Merge "Update language to comply with Android's inclusive language guidance"
diff --git a/camera/ICameraClient.cpp b/camera/ICameraClient.cpp
index c02c81b..bef2ea0 100644
--- a/camera/ICameraClient.cpp
+++ b/camera/ICameraClient.cpp
@@ -142,7 +142,8 @@
camera_frame_metadata_t metadata;
if (data.dataAvail() > 0) {
metadata.number_of_faces = data.readInt32();
- if (metadata.number_of_faces <= 0 ||
+ // Zero faces is a valid case, to notify clients that no faces are now visible
+ if (metadata.number_of_faces < 0 ||
metadata.number_of_faces > (int32_t)(INT32_MAX / sizeof(camera_face_t))) {
ALOGE("%s: Too large face count: %d", __FUNCTION__, metadata.number_of_faces);
return BAD_VALUE;
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index e8a2e0e..b31a58b 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -273,14 +273,11 @@
SurfaceComposerClient::Transaction& t,
const sp<IBinder>& dpy,
const ui::DisplayState& displayState) {
- const ui::Size& viewport = displayState.viewport;
-
- // Set the region of the layer stack we're interested in, which in our
- // case is "all of it".
- Rect layerStackRect(viewport);
+ // Set the region of the layer stack we're interested in, which in our case is "all of it".
+ Rect layerStackRect(displayState.layerStackSpaceRect);
// We need to preserve the aspect ratio of the display.
- float displayAspect = viewport.getHeight() / static_cast<float>(viewport.getWidth());
+ float displayAspect = layerStackRect.getHeight() / static_cast<float>(layerStackRect.getWidth());
// Set the way we map the output onto the display surface (which will
@@ -699,20 +696,21 @@
return err;
}
- const ui::Size& viewport = displayState.viewport;
+ const ui::Size& layerStackSpaceRect = displayState.layerStackSpaceRect;
if (gVerbose) {
printf("Display is %dx%d @%.2ffps (orientation=%s), layerStack=%u\n",
- viewport.getWidth(), viewport.getHeight(), displayConfig.refreshRate,
- toCString(displayState.orientation), displayState.layerStack);
+ layerStackSpaceRect.getWidth(), layerStackSpaceRect.getHeight(),
+ displayConfig.refreshRate, toCString(displayState.orientation),
+ displayState.layerStack);
fflush(stdout);
}
// Encoder can't take odd number as config
if (gVideoWidth == 0) {
- gVideoWidth = floorToEven(viewport.getWidth());
+ gVideoWidth = floorToEven(layerStackSpaceRect.getWidth());
}
if (gVideoHeight == 0) {
- gVideoHeight = floorToEven(viewport.getHeight());
+ gVideoHeight = floorToEven(layerStackSpaceRect.getHeight());
}
// Configure and start the encoder.
diff --git a/drm/common/include/DrmEngineBase.h b/drm/common/include/DrmEngineBase.h
index 73f11a4..c0a5e3b 100644
--- a/drm/common/include/DrmEngineBase.h
+++ b/drm/common/include/DrmEngineBase.h
@@ -309,7 +309,7 @@
/**
* Removes all the rights information of each plug-in associated with
- * DRM framework. Will be used in master reset
+ * DRM framework.
*
* @param[in] uniqueId Unique identifier for a session
* @return status_t
diff --git a/drm/common/include/IDrmEngine.h b/drm/common/include/IDrmEngine.h
index 1837a11..a545941 100644
--- a/drm/common/include/IDrmEngine.h
+++ b/drm/common/include/IDrmEngine.h
@@ -250,7 +250,7 @@
/**
* Removes all the rights information of each plug-in associated with
- * DRM framework. Will be used in master reset
+ * DRM framework.
*
* @param[in] uniqueId Unique identifier for a session
* @return status_t
diff --git a/drm/libdrmframework/include/DrmManagerClientImpl.h b/drm/libdrmframework/include/DrmManagerClientImpl.h
index 3858675..8c8783b 100644
--- a/drm/libdrmframework/include/DrmManagerClientImpl.h
+++ b/drm/libdrmframework/include/DrmManagerClientImpl.h
@@ -230,7 +230,7 @@
/**
* Removes all the rights information of each plug-in associated with
- * DRM framework. Will be used in master reset
+ * DRM framework.
*
* @param[in] uniqueId Unique identifier for a session
* @return status_t
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h
index b62ddb9..eb5b0f6 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/include/FwdLockEngine.h
@@ -252,8 +252,7 @@
/**
* Removes all the rights information of each plug-in associated with
- * DRM framework. Will be used in master reset but does nothing for
- * Forward Lock Engine.
+ * DRM framework. Does nothing for Forward Lock Engine.
*
* @param uniqueId Unique identifier for a session
* @return status_t
diff --git a/drm/libdrmframework/plugins/forward-lock/internal-format/doc/FwdLock.html b/drm/libdrmframework/plugins/forward-lock/internal-format/doc/FwdLock.html
index 8f95cd2..c1d5b3d 100644
--- a/drm/libdrmframework/plugins/forward-lock/internal-format/doc/FwdLock.html
+++ b/drm/libdrmframework/plugins/forward-lock/internal-format/doc/FwdLock.html
@@ -488,7 +488,7 @@
<p class=MsoBodyText><b>Note:</b> The key-encryption key must be unique to each
device; this is what makes the files forward lockÂprotected. Ideally, it should
be derived from secret hardware parameters, but at the very least it should be
-persistent from one master reset to the next.</p>
+persistent from one factory reset to the next.</p>
<div style='margin-bottom:24.0pt;border:solid windowtext 1.0pt;padding:1.0pt 4.0pt 1.0pt 4.0pt;
background:#F2F2F2'>
diff --git a/include/drm/DrmManagerClient.h b/include/drm/DrmManagerClient.h
index 866edac..a38aa9b 100644
--- a/include/drm/DrmManagerClient.h
+++ b/include/drm/DrmManagerClient.h
@@ -318,7 +318,7 @@
/**
* Removes all the rights information of each plug-in associated with
- * DRM framework. Will be used in master reset
+ * DRM framework.
*
* @return status_t
* Returns DRM_NO_ERROR for success, DRM_ERROR_UNKNOWN for failure
diff --git a/media/codec2/components/avc/C2SoftAvcDec.cpp b/media/codec2/components/avc/C2SoftAvcDec.cpp
index d7b9e12..3afd670 100644
--- a/media/codec2/components/avc/C2SoftAvcDec.cpp
+++ b/media/codec2/components/avc/C2SoftAvcDec.cpp
@@ -34,7 +34,11 @@
constexpr size_t kMinInputBufferSize = 2 * 1024 * 1024;
constexpr char COMPONENT_NAME[] = "c2.android.avc.decoder";
constexpr uint32_t kDefaultOutputDelay = 8;
-constexpr uint32_t kMaxOutputDelay = 16;
+/* avc specification allows for a maximum delay of 16 frames.
+ As soft avc decoder supports interlaced, this delay would be 32 fields.
+ And avc decoder implementation has an additional delay of 2 decode calls.
+ So total maximum output delay is 34 */
+constexpr uint32_t kMaxOutputDelay = 34;
constexpr uint32_t kMinInputBytes = 4;
} // namespace
diff --git a/media/codec2/components/gav1/C2SoftGav1Dec.cpp b/media/codec2/components/gav1/C2SoftGav1Dec.cpp
index dfed76c..a1929e7 100644
--- a/media/codec2/components/gav1/C2SoftGav1Dec.cpp
+++ b/media/codec2/components/gav1/C2SoftGav1Dec.cpp
@@ -678,7 +678,7 @@
if (format == HAL_PIXEL_FORMAT_RGBA_1010102) {
convertYUV420Planar16ToY410(
(uint32_t *)dstY, srcY, srcU, srcV, srcYStride / 2, srcUStride / 2,
- srcVStride / 2, align(mWidth, 16), mWidth, mHeight);
+ srcVStride / 2, dstYStride / sizeof(uint32_t), mWidth, mHeight);
} else {
convertYUV420Planar16ToYUV420Planar(dstY, dstU, dstV,
srcY, srcU, srcV,
diff --git a/media/codec2/sfplugin/CCodecBuffers.cpp b/media/codec2/sfplugin/CCodecBuffers.cpp
index e58a1e4..c49a16c 100644
--- a/media/codec2/sfplugin/CCodecBuffers.cpp
+++ b/media/codec2/sfplugin/CCodecBuffers.cpp
@@ -272,8 +272,6 @@
// The output format can be processed without a registered slot.
if (outputFormat) {
- ALOGD("[%s] popFromStashAndRegister: output format changed to %s",
- mName, outputFormat->debugString().c_str());
updateSkipCutBuffer(outputFormat, entry.notify);
}
@@ -301,6 +299,10 @@
}
if (!entry.notify) {
+ if (outputFormat) {
+ ALOGD("[%s] popFromStashAndRegister: output format changed to %s",
+ mName, outputFormat->debugString().c_str());
+ }
mPending.pop_front();
return DISCARD;
}
@@ -317,6 +319,10 @@
// Append information from the front stash entry to outBuffer.
(*outBuffer)->meta()->setInt64("timeUs", entry.timestamp);
(*outBuffer)->meta()->setInt32("flags", entry.flags);
+ if (outputFormat) {
+ ALOGD("[%s] popFromStashAndRegister: output format changed to %s",
+ mName, outputFormat->debugString().c_str());
+ }
ALOGV("[%s] popFromStashAndRegister: "
"out buffer index = %zu [%p] => %p + %zu (%lld)",
mName, *index, outBuffer->get(),
diff --git a/media/extractors/tests/AndroidTest.xml b/media/extractors/tests/AndroidTest.xml
index 1f17d42..fc8152c 100644
--- a/media/extractors/tests/AndroidTest.xml
+++ b/media/extractors/tests/AndroidTest.xml
@@ -19,7 +19,7 @@
<option name="cleanup" value="true" />
<option name="push" value="ExtractorUnitTest->/data/local/tmp/ExtractorUnitTest" />
<option name="push-file"
- key="https://storage.googleapis.com/android_media/frameworks/av/media/extractors/tests/extractor-1.3.zip?unzip=true"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/extractors/tests/extractor-1.4.zip?unzip=true"
value="/data/local/tmp/ExtractorUnitTestRes/" />
</target_preparer>
diff --git a/media/extractors/tests/ExtractorUnitTest.cpp b/media/extractors/tests/ExtractorUnitTest.cpp
index b7c6c59..d91fffa 100644
--- a/media/extractors/tests/ExtractorUnitTest.cpp
+++ b/media/extractors/tests/ExtractorUnitTest.cpp
@@ -138,10 +138,23 @@
mDisableTest = false;
static const std::map<std::string, standardExtractors> mapExtractor = {
- {"aac", AAC}, {"amr", AMR}, {"mp3", MP3}, {"ogg", OGG},
- {"wav", WAV}, {"mkv", MKV}, {"flac", FLAC}, {"midi", MIDI},
- {"mpeg4", MPEG4}, {"mpeg2ts", MPEG2TS}, {"mpeg2ps", MPEG2PS}, {"mp4", MPEG4},
- {"webm", MKV}, {"ts", MPEG2TS}, {"mpeg", MPEG2PS}};
+ {"aac", AAC},
+ {"amr", AMR},
+ {"flac", FLAC},
+ {"mid", MIDI},
+ {"midi", MIDI},
+ {"mkv", MKV},
+ {"mp3", MP3},
+ {"mp4", MPEG4},
+ {"mpeg2ps", MPEG2PS},
+ {"mpeg2ts", MPEG2TS},
+ {"mpeg4", MPEG4},
+ {"mpg", MPEG2PS},
+ {"ogg", OGG},
+ {"opus", OGG},
+ {"ts", MPEG2TS},
+ {"wav", WAV},
+ {"webm", MKV}};
// Find the component type
if (mapExtractor.find(writerFormat) != mapExtractor.end()) {
mExtractorName = mapExtractor.at(writerFormat);
@@ -940,36 +953,55 @@
}
}
- virtual void SetUp() override {
- string input0 = GetParam().first;
- string input1 = GetParam().second;
-
- // Allocate memory to hold extracted data for both extractors
- struct stat buf;
- int32_t status = stat((gEnv->getRes() + input0).c_str(), &buf);
- ASSERT_EQ(status, 0) << "Unable to get file properties";
-
- // allocating the buffer size as 2x since some
- // extractors like flac, midi and wav decodes the file.
- mExtractorOutput[0] = (int8_t *)calloc(1, buf.st_size * 2);
- ASSERT_NE(mExtractorOutput[0], nullptr)
- << "Unable to allocate memory for writing extractor's output";
- mExtractorOuputSize[0] = buf.st_size * 2;
-
- status = stat((gEnv->getRes() + input1).c_str(), &buf);
- ASSERT_EQ(status, 0) << "Unable to get file properties";
-
- // allocate buffer for extractor output, 2x input file size.
- mExtractorOutput[1] = (int8_t *)calloc(1, buf.st_size * 2);
- ASSERT_NE(mExtractorOutput[1], nullptr)
- << "Unable to allocate memory for writing extractor's output";
- mExtractorOuputSize[1] = buf.st_size * 2;
- }
-
int8_t *mExtractorOutput[2]{};
size_t mExtractorOuputSize[2]{};
};
+size_t allocateOutputBuffers(string inputFileName, AMediaFormat *extractorFormat) {
+ size_t bufferSize = 0u;
+ // allocating the buffer size as sampleRate * channelCount * clipDuration since
+ // some extractors like flac, midi and wav decodes the file. These extractors
+ // advertise the mime type as raw.
+ const char *mime;
+ AMediaFormat_getString(extractorFormat, AMEDIAFORMAT_KEY_MIME, &mime);
+ if (!strcmp(mime, MEDIA_MIMETYPE_AUDIO_RAW)) {
+ int64_t clipDurationUs = -1;
+ int32_t channelCount = -1;
+ int32_t sampleRate = -1;
+ int32_t bitsPerSampple = -1;
+ if (!AMediaFormat_getInt32(extractorFormat, AMEDIAFORMAT_KEY_CHANNEL_COUNT,
+ &channelCount) || channelCount <= 0) {
+ ALOGE("Invalid channelCount for input file : %s", inputFileName.c_str());
+ return 0;
+ }
+ if (!AMediaFormat_getInt32(extractorFormat, AMEDIAFORMAT_KEY_SAMPLE_RATE, &sampleRate) ||
+ sampleRate <= 0) {
+ ALOGE("Invalid sampleRate for input file : %s", inputFileName.c_str());
+ return 0;
+ }
+ if (!AMediaFormat_getInt64(extractorFormat, AMEDIAFORMAT_KEY_DURATION, &clipDurationUs) ||
+ clipDurationUs <= 0) {
+ ALOGE("Invalid clip duration for input file : %s", inputFileName.c_str());
+ return 0;
+ }
+ if (!AMediaFormat_getInt32(extractorFormat, AMEDIAFORMAT_KEY_PCM_ENCODING,
+ &bitsPerSampple) || bitsPerSampple <= 0) {
+ ALOGE("Invalid bits per sample for input file : %s", inputFileName.c_str());
+ return 0;
+ }
+ bufferSize = bitsPerSampple * channelCount * sampleRate * (clipDurationUs / 1000000 + 1);
+ } else {
+ struct stat buf;
+ int32_t status = stat(inputFileName.c_str(), &buf);
+ if (status != 0) {
+ ALOGE("Unable to get file properties for: %s", inputFileName.c_str());
+ return 0;
+ }
+ bufferSize = buf.st_size;
+ }
+ return bufferSize;
+}
+
// Compare output of two extractors for identical content
TEST_P(ExtractorComparison, ExtractorComparisonTest) {
vector<string> inputFileNames = {GetParam().first, GetParam().second};
@@ -1011,6 +1043,13 @@
CMediaTrack *cTrack = wrap(track);
ASSERT_NE(cTrack, nullptr) << "Failed to get track wrapper for index " << trackIdx;
+ mExtractorOuputSize[idx] = allocateOutputBuffers(inputFileName, extractorFormat[idx]);
+ ASSERT_GT(mExtractorOuputSize[idx], 0u) << " Invalid size for output buffers";
+
+ mExtractorOutput[idx] = (int8_t *)calloc(1, mExtractorOuputSize[idx]);
+ ASSERT_NE(mExtractorOutput[idx], nullptr)
+ << "Unable to allocate memory for writing extractor's output";
+
MediaBufferGroup *bufferGroup = new MediaBufferGroup();
status = cTrack->start(track, bufferGroup->wrap());
ASSERT_EQ(OK, (media_status_t)status) << "Failed to start the track";
@@ -1087,14 +1126,44 @@
<< inputFileNames[1] << " extractors";
}
-INSTANTIATE_TEST_SUITE_P(ExtractorComparisonAll, ExtractorComparison,
- ::testing::Values(make_pair("swirl_144x136_vp9.mp4",
- "swirl_144x136_vp9.webm"),
- make_pair("video_480x360_mp4_vp9_333kbps_25fps.mp4",
- "video_480x360_webm_vp9_333kbps_25fps.webm"),
- make_pair("video_1280x720_av1_hdr_static_3mbps.mp4",
- "video_1280x720_av1_hdr_static_3mbps.webm"),
- make_pair("loudsoftaac.aac", "loudsoftaac.mkv")));
+INSTANTIATE_TEST_SUITE_P(
+ ExtractorComparisonAll, ExtractorComparison,
+ ::testing::Values(make_pair("swirl_144x136_vp9.mp4", "swirl_144x136_vp9.webm"),
+ make_pair("video_480x360_mp4_vp9_333kbps_25fps.mp4",
+ "video_480x360_webm_vp9_333kbps_25fps.webm"),
+ make_pair("video_1280x720_av1_hdr_static_3mbps.mp4",
+ "video_1280x720_av1_hdr_static_3mbps.webm"),
+ make_pair("swirl_132x130_mpeg4.3gp", "swirl_132x130_mpeg4.mkv"),
+ make_pair("swirl_144x136_avc.mkv", "swirl_144x136_avc.mp4"),
+ make_pair("swirl_132x130_mpeg4.mp4", "swirl_132x130_mpeg4.mkv"),
+ make_pair("crowd_508x240_25fps_hevc.mp4","crowd_508x240_25fps_hevc.mkv"),
+ make_pair("bbb_cif_768kbps_30fps_mpeg2.mp4",
+ "bbb_cif_768kbps_30fps_mpeg2.ts"),
+
+ make_pair("loudsoftaac.aac", "loudsoftaac.mkv"),
+ make_pair("sinesweepflacmkv.mkv", "sinesweepflacmp4.mp4"),
+ make_pair("sinesweepmp3lame.mp3", "sinesweepmp3lame.mkv"),
+ make_pair("sinesweepoggmp4.mp4", "sinesweepogg.ogg"),
+ make_pair("sinesweepvorbis.mp4", "sinesweepvorbis.ogg"),
+ make_pair("sinesweepvorbis.mkv", "sinesweepvorbis.ogg"),
+ make_pair("testopus.mkv", "testopus.mp4"),
+ make_pair("testopus.mp4", "testopus.opus"),
+
+ make_pair("loudsoftaac.aac", "loudsoftaac.aac"),
+ make_pair("testamr.amr", "testamr.amr"),
+ make_pair("sinesweepflac.flac", "sinesweepflac.flac"),
+ make_pair("midi_a.mid", "midi_a.mid"),
+ make_pair("sinesweepvorbis.mkv", "sinesweepvorbis.mkv"),
+ make_pair("sinesweepmp3lame.mp3", "sinesweepmp3lame.mp3"),
+ make_pair("sinesweepoggmp4.mp4", "sinesweepoggmp4.mp4"),
+ make_pair("testopus.opus", "testopus.opus"),
+ make_pair("john_cage.ogg", "john_cage.ogg"),
+ make_pair("monotestgsm.wav", "monotestgsm.wav"),
+
+ make_pair("swirl_144x136_mpeg2.mpg", "swirl_144x136_mpeg2.mpg"),
+ make_pair("swirl_132x130_mpeg4.mp4", "swirl_132x130_mpeg4.mp4"),
+ make_pair("swirl_144x136_vp9.webm", "swirl_144x136_vp9.webm"),
+ make_pair("swirl_144x136_vp8.webm", "swirl_144x136_vp8.webm")));
INSTANTIATE_TEST_SUITE_P(ConfigParamTestAll, ConfigParamTest,
::testing::Values(make_pair("aac", AAC_1),
diff --git a/media/extractors/tests/README.md b/media/extractors/tests/README.md
index 69538b6..cff09ca 100644
--- a/media/extractors/tests/README.md
+++ b/media/extractors/tests/README.md
@@ -22,7 +22,7 @@
adb push ${OUT}/data/nativetest/ExtractorUnitTest/ExtractorUnitTest /data/local/tmp/
```
-The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/extractors/tests/extractor.zip). Download, unzip and push these files into device for testing.
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/extractors/tests/extractor-1.4.zip). Download, unzip and push these files into device for testing.
```
adb push extractor /data/local/tmp/
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 7aaf908..6666788 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -689,7 +689,7 @@
aaudio_performance_mode_t mode) __INTRODUCED_IN(26);
/**
- * Set the intended use case for the stream.
+ * Set the intended use case for the output stream.
*
* The AAudio system will use this information to optimize the
* behavior of the stream.
@@ -706,7 +706,7 @@
aaudio_usage_t usage) __INTRODUCED_IN(28);
/**
- * Set the type of audio data that the stream will carry.
+ * Set the type of audio data that the output stream will carry.
*
* The AAudio system will use this information to optimize the
* behavior of the stream.
@@ -1037,6 +1037,11 @@
* but still allow queries to the stream to occur from other threads. This often
* happens if you are monitoring stream progress from a UI thread.
*
+ * NOTE: This function is only fully implemented for MMAP streams,
+ * which are low latency streams supported by some devices.
+ * On other "Legacy" streams some audio resources will still be in use
+ * and some callbacks may still be in process after this call.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index d014608..55fc986 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -14,8 +14,6 @@
* limitations under the License.
*/
-#define LOG_TAG (mInService ? "AudioStreamInternalCapture_Service" \
- : "AudioStreamInternalCapture_Client")
//#define LOG_NDEBUG 0
#include <utils/Log.h>
@@ -29,6 +27,14 @@
#define ATRACE_TAG ATRACE_TAG_AUDIO
#include <utils/Trace.h>
+// We do this after the #includes because if a header uses ALOG.
+// it would fail on the reference to mInService.
+#undef LOG_TAG
+// This file is used in both client and server processes.
+// This is needed to make sense of the logs more easily.
+#define LOG_TAG (mInService ? "AudioStreamInternalCapture_Service" \
+ : "AudioStreamInternalCapture_Client")
+
using android::WrappingBuffer;
using namespace aaudio;
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 6337b53..b47b472 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -14,8 +14,6 @@
* limitations under the License.
*/
-#define LOG_TAG (mInService ? "AudioStreamInternalPlay_Service" \
- : "AudioStreamInternalPlay_Client")
//#define LOG_NDEBUG 0
#include <utils/Log.h>
@@ -26,6 +24,14 @@
#include "client/AudioStreamInternalPlay.h"
#include "utility/AudioClock.h"
+// We do this after the #includes because if a header uses ALOG.
+// it would fail on the reference to mInService.
+#undef LOG_TAG
+// This file is used in both client and server processes.
+// This is needed to make sense of the logs more easily.
+#define LOG_TAG (mInService ? "AudioStreamInternalPlay_Service" \
+ : "AudioStreamInternalPlay_Client")
+
using android::WrappingBuffer;
using namespace aaudio;
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 8965875..cfa7221 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -255,17 +255,16 @@
if (audioStream != nullptr) {
aaudio_stream_id_t id = audioStream->getId();
ALOGD("%s(s#%u) called ---------------", __func__, id);
- result = audioStream->safeRelease();
- // safeRelease will only fail if called illegally, for example, from a callback.
+ result = audioStream->safeReleaseClose();
+ // safeReleaseClose will only fail if called illegally, for example, from a callback.
// That would result in deleting an active stream, which would cause a crash.
if (result != AAUDIO_OK) {
ALOGW("%s(s#%u) failed. Close it from another thread.",
__func__, id);
} else {
audioStream->unregisterPlayerBase();
- // Mark CLOSED to keep destructors from asserting.
- audioStream->closeFinal();
- delete audioStream;
+ // Allow the stream to be deleted.
+ AudioStreamBuilder::stopUsingStream(audioStream);
}
ALOGD("%s(s#%u) returned %d ---------", __func__, id, result);
}
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 983887b..ac2da57 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -39,7 +39,7 @@
}
AudioStream::AudioStream()
- : mPlayerBase(new MyPlayerBase(this))
+ : mPlayerBase(new MyPlayerBase())
, mStreamId(AAudio_getNextStreamId())
{
// mThread is a pthread_t of unknown size so we need memset.
@@ -48,6 +48,10 @@
}
AudioStream::~AudioStream() {
+ // Please preserve this log because there have been several bugs related to
+ // AudioStream deletion and late callbacks.
+ ALOGD("%s(s#%u) mPlayerBase strongCount = %d",
+ __func__, getId(), mPlayerBase->getStrongCount());
// If the stream is deleted when OPEN or in use then audio resources will leak.
// This would indicate an internal error. So we want to find this ASAP.
LOG_ALWAYS_FATAL_IF(!(getState() == AAUDIO_STREAM_STATE_CLOSED
@@ -55,8 +59,6 @@
|| getState() == AAUDIO_STREAM_STATE_DISCONNECTED),
"~AudioStream() - still in use, state = %s",
AudioGlobal_convertStreamStateToText(getState()));
-
- mPlayerBase->clearParentReference(); // remove reference to this AudioStream
}
aaudio_result_t AudioStream::open(const AudioStreamBuilder& builder)
@@ -301,18 +303,29 @@
}
aaudio_result_t AudioStream::safeRelease() {
- // This get temporarily unlocked in the release() when joining callback threads.
+ // This get temporarily unlocked in the MMAP release() when joining callback threads.
std::lock_guard<std::mutex> lock(mStreamLock);
if (collidesWithCallback()) {
ALOGE("%s cannot be called from a callback!", __func__);
return AAUDIO_ERROR_INVALID_STATE;
}
- if (getState() == AAUDIO_STREAM_STATE_CLOSING) {
+ if (getState() == AAUDIO_STREAM_STATE_CLOSING) { // already released?
return AAUDIO_OK;
}
return release_l();
}
+aaudio_result_t AudioStream::safeReleaseClose() {
+ // This get temporarily unlocked in the MMAP release() when joining callback threads.
+ std::lock_guard<std::mutex> lock(mStreamLock);
+ if (collidesWithCallback()) {
+ ALOGE("%s cannot be called from a callback!", __func__);
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+ releaseCloseFinal();
+ return AAUDIO_OK;
+}
+
void AudioStream::setState(aaudio_stream_state_t state) {
ALOGD("%s(s#%d) from %d to %d", __func__, getId(), mState, state);
if (state == mState) {
@@ -523,11 +536,18 @@
}
#if AAUDIO_USE_VOLUME_SHAPER
-android::media::VolumeShaper::Status AudioStream::applyVolumeShaper(
- const android::media::VolumeShaper::Configuration& configuration __unused,
- const android::media::VolumeShaper::Operation& operation __unused) {
- ALOGW("applyVolumeShaper() is not supported");
- return android::media::VolumeShaper::Status::ok();
+::android::binder::Status AudioStream::MyPlayerBase::applyVolumeShaper(
+ const ::android::media::VolumeShaper::Configuration& configuration,
+ const ::android::media::VolumeShaper::Operation& operation) {
+ android::sp<AudioStream> audioStream;
+ {
+ std::lock_guard<std::mutex> lock(mParentLock);
+ audioStream = mParent.promote();
+ }
+ if (audioStream) {
+ return audioStream->applyVolumeShaper(configuration, operation);
+ }
+ return android::NO_ERROR;
}
#endif
@@ -537,26 +557,36 @@
doSetVolume(); // apply this change
}
-AudioStream::MyPlayerBase::MyPlayerBase(AudioStream *parent) : mParent(parent) {
-}
-
-AudioStream::MyPlayerBase::~MyPlayerBase() {
-}
-
-void AudioStream::MyPlayerBase::registerWithAudioManager() {
+void AudioStream::MyPlayerBase::registerWithAudioManager(const android::sp<AudioStream>& parent) {
+ std::lock_guard<std::mutex> lock(mParentLock);
+ mParent = parent;
if (!mRegistered) {
- init(android::PLAYER_TYPE_AAUDIO, AAudioConvert_usageToInternal(mParent->getUsage()));
+ init(android::PLAYER_TYPE_AAUDIO, AAudioConvert_usageToInternal(parent->getUsage()));
mRegistered = true;
}
}
void AudioStream::MyPlayerBase::unregisterWithAudioManager() {
+ std::lock_guard<std::mutex> lock(mParentLock);
if (mRegistered) {
baseDestroy();
mRegistered = false;
}
}
+android::status_t AudioStream::MyPlayerBase::playerSetVolume() {
+ android::sp<AudioStream> audioStream;
+ {
+ std::lock_guard<std::mutex> lock(mParentLock);
+ audioStream = mParent.promote();
+ }
+ if (audioStream) {
+ // No pan and only left volume is taken into account from IPLayer interface
+ audioStream->setDuckAndMuteVolume(mVolumeMultiplierL /* * mPanMultiplierL */);
+ }
+ return android::NO_ERROR;
+}
+
void AudioStream::MyPlayerBase::destroy() {
unregisterWithAudioManager();
}
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index fb71c36..e0bd9d8 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -25,8 +25,10 @@
#include <binder/Status.h>
#include <utils/StrongPointer.h>
-#include "media/VolumeShaper.h"
-#include "media/PlayerBase.h"
+#include <media/AudioSystem.h>
+#include <media/PlayerBase.h>
+#include <media/VolumeShaper.h>
+
#include "utility/AAudioUtilities.h"
#include "utility/MonotonicCounter.h"
@@ -45,7 +47,8 @@
/**
* AAudio audio stream.
*/
-class AudioStream {
+// By extending AudioDeviceCallback, we also inherit from RefBase.
+class AudioStream : public android::AudioSystem::AudioDeviceCallback {
public:
AudioStream();
@@ -117,6 +120,17 @@
virtual void logOpen();
void logReleaseBufferState();
+ /* Note about naming for "release" and "close" related methods.
+ *
+ * These names are intended to match the public AAudio API.
+ * The original AAudio API had an AAudioStream_close() function that
+ * released the hardware and deleted the stream. That made it difficult
+ * because apps want to release the HW ASAP but are not in a rush to delete
+ * the stream object. So in R we added an AAudioStream_release() function
+ * that just released the hardware.
+ * The AAudioStream_close() method releases if needed and then closes.
+ */
+
/**
* Free any hardware or system resources from the open() call.
* It is safe to call release_l() multiple times.
@@ -126,22 +140,27 @@
return AAUDIO_OK;
}
- aaudio_result_t closeFinal() {
+ /**
+ * Free any resources not already freed by release_l().
+ * Assume release_l() already called.
+ */
+ virtual void close_l() {
+ // Releasing the stream will set the state to CLOSING.
+ assert(getState() == AAUDIO_STREAM_STATE_CLOSING);
+ // setState() prevents a transition from CLOSING to any state other than CLOSED.
// State is checked by destructor.
setState(AAUDIO_STREAM_STATE_CLOSED);
- return AAUDIO_OK;
}
/**
* Release then close the stream.
- * @return AAUDIO_OK or negative error.
*/
- aaudio_result_t releaseCloseFinal() {
- aaudio_result_t result = release_l(); // TODO review locking
- if (result == AAUDIO_OK) {
- result = closeFinal();
+ void releaseCloseFinal() {
+ if (getState() != AAUDIO_STREAM_STATE_CLOSING) { // not already released?
+ // Ignore result and keep closing.
+ (void) release_l();
}
- return result;
+ close_l();
}
// This is only used to identify a stream in the logs without
@@ -328,6 +347,10 @@
*/
bool collidesWithCallback() const;
+ // Implement AudioDeviceCallback
+ void onAudioDeviceUpdate(audio_io_handle_t audioIo,
+ audio_port_handle_t deviceId) override {};
+
// ============== I/O ===========================
// A Stream will only implement read() or write() depending on its direction.
virtual aaudio_result_t write(const void *buffer __unused,
@@ -366,7 +389,7 @@
*/
void registerPlayerBase() {
if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
- mPlayerBase->registerWithAudioManager();
+ mPlayerBase->registerWithAudioManager(this);
}
}
@@ -395,21 +418,33 @@
*/
aaudio_result_t systemStopFromCallback();
+ /**
+ * Safely RELEASE a stream after taking mStreamLock and checking
+ * to make sure we are not being called from a callback.
+ * @return AAUDIO_OK or a negative error
+ */
aaudio_result_t safeRelease();
+ /**
+ * Safely RELEASE and CLOSE a stream after taking mStreamLock and checking
+ * to make sure we are not being called from a callback.
+ * @return AAUDIO_OK or a negative error
+ */
+ aaudio_result_t safeReleaseClose();
+
protected:
// PlayerBase allows the system to control the stream volume.
class MyPlayerBase : public android::PlayerBase {
public:
- explicit MyPlayerBase(AudioStream *parent);
+ MyPlayerBase() {};
- virtual ~MyPlayerBase();
+ virtual ~MyPlayerBase() = default;
/**
* Register for volume changes and remote control.
*/
- void registerWithAudioManager();
+ void registerWithAudioManager(const android::sp<AudioStream>& parent);
/**
* UnRegister.
@@ -421,8 +456,6 @@
*/
void destroy() override;
- void clearParentReference() { mParent = nullptr; }
-
// Just a stub. The ability to start audio through PlayerBase is being deprecated.
android::status_t playerStart() override {
return android::NO_ERROR;
@@ -438,18 +471,10 @@
return android::NO_ERROR;
}
- android::status_t playerSetVolume() override {
- // No pan and only left volume is taken into account from IPLayer interface
- mParent->setDuckAndMuteVolume(mVolumeMultiplierL /* * mPanMultiplierL */);
- return android::NO_ERROR;
- }
+ android::status_t playerSetVolume() override;
#if AAUDIO_USE_VOLUME_SHAPER
- ::android::binder::Status applyVolumeShaper(
- const ::android::media::VolumeShaper::Configuration& configuration,
- const ::android::media::VolumeShaper::Operation& operation) {
- return mParent->applyVolumeShaper(configuration, operation);
- }
+ ::android::binder::Status applyVolumeShaper();
#endif
aaudio_result_t getResult() {
@@ -457,9 +482,12 @@
}
private:
- AudioStream *mParent;
- aaudio_result_t mResult = AAUDIO_OK;
- bool mRegistered = false;
+ // Use a weak pointer so the AudioStream can be deleted.
+
+ std::mutex mParentLock;
+ android::wp<AudioStream> mParent;
+ aaudio_result_t mResult = AAUDIO_OK;
+ bool mRegistered = false;
};
/**
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 60dad84..630b289 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -63,27 +63,26 @@
static aaudio_result_t builder_createStream(aaudio_direction_t direction,
aaudio_sharing_mode_t sharingMode,
bool tryMMap,
- AudioStream **audioStreamPtr) {
- *audioStreamPtr = nullptr;
+ android::sp<AudioStream> &stream) {
aaudio_result_t result = AAUDIO_OK;
switch (direction) {
case AAUDIO_DIRECTION_INPUT:
if (tryMMap) {
- *audioStreamPtr = new AudioStreamInternalCapture(AAudioBinderClient::getInstance(),
+ stream = new AudioStreamInternalCapture(AAudioBinderClient::getInstance(),
false);
} else {
- *audioStreamPtr = new AudioStreamRecord();
+ stream = new AudioStreamRecord();
}
break;
case AAUDIO_DIRECTION_OUTPUT:
if (tryMMap) {
- *audioStreamPtr = new AudioStreamInternalPlay(AAudioBinderClient::getInstance(),
+ stream = new AudioStreamInternalPlay(AAudioBinderClient::getInstance(),
false);
} else {
- *audioStreamPtr = new AudioStreamTrack();
+ stream = new AudioStreamTrack();
}
break;
@@ -98,7 +97,7 @@
// Fall back to Legacy path if MMAP not available.
// Exact behavior is controlled by MMapPolicy.
aaudio_result_t AudioStreamBuilder::build(AudioStream** streamPtr) {
- AudioStream *audioStream = nullptr;
+
if (streamPtr == nullptr) {
ALOGE("%s() streamPtr is null", __func__);
return AAUDIO_ERROR_NULL;
@@ -171,41 +170,48 @@
setPrivacySensitive(true);
}
- result = builder_createStream(getDirection(), sharingMode, allowMMap, &audioStream);
+ android::sp<AudioStream> audioStream;
+ result = builder_createStream(getDirection(), sharingMode, allowMMap, audioStream);
if (result == AAUDIO_OK) {
// Open the stream using the parameters from the builder.
result = audioStream->open(*this);
- if (result == AAUDIO_OK) {
- *streamPtr = audioStream;
- } else {
+ if (result != AAUDIO_OK) {
bool isMMap = audioStream->isMMap();
- delete audioStream;
- audioStream = nullptr;
-
if (isMMap && allowLegacy) {
ALOGV("%s() MMAP stream did not open so try Legacy path", __func__);
// If MMAP stream failed to open then TRY using a legacy stream.
result = builder_createStream(getDirection(), sharingMode,
- false, &audioStream);
+ false, audioStream);
if (result == AAUDIO_OK) {
result = audioStream->open(*this);
- if (result == AAUDIO_OK) {
- *streamPtr = audioStream;
- } else {
- delete audioStream;
- audioStream = nullptr;
- }
}
}
}
- if (audioStream != nullptr) {
+ if (result == AAUDIO_OK) {
audioStream->logOpen();
- }
+ *streamPtr = startUsingStream(audioStream);
+ } // else audioStream will go out of scope and be deleted
}
return result;
}
+AudioStream *AudioStreamBuilder::startUsingStream(android::sp<AudioStream> &audioStream) {
+ // Increment the smart pointer so it will not get deleted when
+ // we pass it to the C caller and it goes out of scope.
+ // The C code cannot hold a smart pointer so we increment the reference
+ // count to indicate that the C app owns a reference.
+ audioStream->incStrong(nullptr);
+ return audioStream.get();
+}
+
+void AudioStreamBuilder::stopUsingStream(AudioStream *stream) {
+ // Undo the effect of startUsingStream()
+ android::sp<AudioStream> spAudioStream(stream);
+ ALOGV("%s() strongCount = %d", __func__, spAudioStream->getStrongCount());
+ spAudioStream->decStrong(nullptr);
+}
+
aaudio_result_t AudioStreamBuilder::validate() const {
// Check for values that are ridiculously out of range to prevent math overflow exploits.
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.h b/media/libaaudio/src/core/AudioStreamBuilder.h
index d5fb80d..9f93341 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.h
+++ b/media/libaaudio/src/core/AudioStreamBuilder.h
@@ -108,9 +108,16 @@
virtual aaudio_result_t validate() const override;
+
void logParameters() const;
+ // Mark the stream so it can be deleted.
+ static void stopUsingStream(AudioStream *stream);
+
private:
+ // Extract a raw pointer that we can pass to a 'C' app.
+ static AudioStream *startUsingStream(android::sp<AudioStream> &spAudioStream);
+
bool mSharingModeMatchRequired = false; // must match sharing mode requested
aaudio_performance_mode_t mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index c062882..33c1bf5 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -34,8 +34,7 @@
using namespace aaudio;
AudioStreamLegacy::AudioStreamLegacy()
- : AudioStream()
- , mDeviceCallback(new StreamDeviceCallback(this)) {
+ : AudioStream() {
}
AudioStreamLegacy::~AudioStreamLegacy() {
@@ -163,7 +162,11 @@
}
void AudioStreamLegacy::forceDisconnect(bool errorCallbackEnabled) {
- if (getState() != AAUDIO_STREAM_STATE_DISCONNECTED) {
+ // There is no need to disconnect if already in these states.
+ if (getState() != AAUDIO_STREAM_STATE_DISCONNECTED
+ && getState() != AAUDIO_STREAM_STATE_CLOSING
+ && getState() != AAUDIO_STREAM_STATE_CLOSED
+ ) {
setState(AAUDIO_STREAM_STATE_DISCONNECTED);
if (errorCallbackEnabled) {
maybeCallErrorCallback(AAUDIO_ERROR_DISCONNECTED);
@@ -205,24 +208,30 @@
return AAudioConvert_androidToAAudioResult(status);
}
-void AudioStreamLegacy::onAudioDeviceUpdate(audio_port_handle_t deviceId)
-{
+void AudioStreamLegacy::onAudioDeviceUpdate(audio_io_handle_t /* audioIo */,
+ audio_port_handle_t deviceId) {
// Device routing is a common source of errors and DISCONNECTS.
- // Please leave this log in place.
- ALOGD("%s() devId %d => %d", __func__, (int) getDeviceId(), (int)deviceId);
- if (getDeviceId() != AAUDIO_UNSPECIFIED && getDeviceId() != deviceId &&
- getState() != AAUDIO_STREAM_STATE_DISCONNECTED) {
+ // Please leave this log in place. If there is a bug then this might
+ // get called after the stream has been deleted so log before we
+ // touch the stream object.
+ ALOGD("%s(deviceId = %d)", __func__, (int)deviceId);
+ if (getDeviceId() != AAUDIO_UNSPECIFIED
+ && getDeviceId() != deviceId
+ && getState() != AAUDIO_STREAM_STATE_DISCONNECTED
+ ) {
// Note that isDataCallbackActive() is affected by state so call it before DISCONNECTING.
// If we have a data callback and the stream is active, then ask the data callback
// to DISCONNECT and call the error callback.
if (isDataCallbackActive()) {
- ALOGD("onAudioDeviceUpdate() request DISCONNECT in data callback due to device change");
+ ALOGD("%s() request DISCONNECT in data callback, device %d => %d",
+ __func__, (int) getDeviceId(), (int) deviceId);
// If the stream is stopped before the data callback has a chance to handle the
// request then the requestStop() and requestPause() methods will handle it after
// the callback has stopped.
mRequestDisconnect.request();
} else {
- ALOGD("onAudioDeviceUpdate() DISCONNECT the stream now");
+ ALOGD("%s() DISCONNECT the stream now, device %d => %d",
+ __func__, (int) getDeviceId(), (int) deviceId);
forceDisconnect();
}
}
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index 9c24b2b..fefe6e0 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -87,29 +87,13 @@
protected:
- class StreamDeviceCallback : public android::AudioSystem::AudioDeviceCallback
- {
- public:
-
- StreamDeviceCallback(AudioStreamLegacy *parent) : mParent(parent) {}
- virtual ~StreamDeviceCallback() {}
-
- virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo __unused,
- audio_port_handle_t deviceId) {
- if (mParent != nullptr) {
- mParent->onAudioDeviceUpdate(deviceId);
- }
- }
-
- AudioStreamLegacy *mParent;
- };
-
aaudio_result_t getBestTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds,
android::ExtendedTimestamp *extendedTimestamp);
- void onAudioDeviceUpdate(audio_port_handle_t deviceId);
+ void onAudioDeviceUpdate(audio_io_handle_t audioIo,
+ audio_port_handle_t deviceId) override;
/*
* Check to see whether a callback thread has requested a disconnected.
@@ -140,7 +124,6 @@
int32_t mBlockAdapterBytesPerFrame = 0;
aaudio_wrapping_frames_t mPositionWhenStarting = 0;
int32_t mCallbackBufferSize = 0;
- const android::sp<StreamDeviceCallback> mDeviceCallback;
AtomicRequestor mRequestDisconnect;
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index 3bfa2b7..43b63d6 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -279,7 +279,7 @@
: (aaudio_session_id_t) mAudioRecord->getSessionId();
setSessionId(actualSessionId);
- mAudioRecord->addAudioDeviceCallback(mDeviceCallback);
+ mAudioRecord->addAudioDeviceCallback(this);
return AAUDIO_OK;
}
@@ -288,16 +288,24 @@
// TODO add close() or release() to AudioFlinger's AudioRecord API.
// Then call it from here
if (getState() != AAUDIO_STREAM_STATE_CLOSING) {
- mAudioRecord->removeAudioDeviceCallback(mDeviceCallback);
+ mAudioRecord->removeAudioDeviceCallback(this);
logReleaseBufferState();
- mAudioRecord.clear();
- mFixedBlockWriter.close();
+ // Data callbacks may still be running!
return AudioStream::release_l();
} else {
return AAUDIO_OK; // already released
}
}
+void AudioStreamRecord::close_l() {
+ mAudioRecord.clear();
+ // Do not close mFixedBlockWriter because a data callback
+ // thread might still be running if someone else has a reference
+ // to mAudioRecord.
+ // It has a unique_ptr to its buffer so it will clean up by itself.
+ AudioStream::close_l();
+}
+
const void * AudioStreamRecord::maybeConvertDeviceData(const void *audioData, int32_t numFrames) {
if (mFormatConversionBufferFloat.get() != nullptr) {
LOG_ALWAYS_FATAL_IF(numFrames > mFormatConversionBufferSizeInFrames,
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index c5944c7..e4ef1c0 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -39,6 +39,7 @@
aaudio_result_t open(const AudioStreamBuilder & builder) override;
aaudio_result_t release_l() override;
+ void close_l() override;
aaudio_result_t requestStart() override;
aaudio_result_t requestStop() override;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 0427220..9e826bd 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -221,7 +221,7 @@
mInitialBufferCapacity = getBufferCapacity();
mInitialFramesPerBurst = getFramesPerBurst();
- mAudioTrack->addAudioDeviceCallback(mDeviceCallback);
+ mAudioTrack->addAudioDeviceCallback(this);
// Update performance mode based on the actual stream flags.
// For example, if the sample rate is not allowed then you won't get a FAST track.
@@ -250,19 +250,26 @@
aaudio_result_t AudioStreamTrack::release_l() {
if (getState() != AAUDIO_STREAM_STATE_CLOSING) {
- mAudioTrack->removeAudioDeviceCallback(mDeviceCallback);
+ status_t err = mAudioTrack->removeAudioDeviceCallback(this);
+ ALOGE_IF(err, "%s() removeAudioDeviceCallback returned %d", __func__, err);
logReleaseBufferState();
- // TODO Investigate why clear() causes a hang in test_various.cpp
- // if I call close() from a data callback.
- // But the same thing in AudioRecord is OK!
- // mAudioTrack.clear();
- mFixedBlockReader.close();
+ // Data callbacks may still be running!
return AudioStream::release_l();
} else {
return AAUDIO_OK; // already released
}
}
+void AudioStreamTrack::close_l() {
+ // Stop callbacks before deleting mFixedBlockReader memory.
+ mAudioTrack.clear();
+ // Do not close mFixedBlockReader because a data callback
+ // thread might still be running if someone else has a reference
+ // to mAudioRecord.
+ // It has a unique_ptr to its buffer so it will clean up by itself.
+ AudioStream::close_l();
+}
+
void AudioStreamTrack::processCallback(int event, void *info) {
switch (event) {
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index 93a1ff4..6334f66 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h
@@ -42,6 +42,7 @@
aaudio_result_t open(const AudioStreamBuilder & builder) override;
aaudio_result_t release_l() override;
+ void close_l() override;
aaudio_result_t requestStart() override;
aaudio_result_t requestPause() override;
diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
index a20c799..cbf863f 100644
--- a/media/libaaudio/tests/test_various.cpp
+++ b/media/libaaudio/tests/test_various.cpp
@@ -33,6 +33,11 @@
void *audioData,
int32_t numFrames
) {
+ aaudio_direction_t direction = AAudioStream_getDirection(stream);
+ if (direction == AAUDIO_DIRECTION_INPUT) {
+ return AAUDIO_CALLBACK_RESULT_CONTINUE;
+ }
+ // Check to make sure the buffer is initialized to all zeros.
int channels = AAudioStream_getChannelCount(stream);
int numSamples = channels * numFrames;
bool allZeros = true;
@@ -48,7 +53,8 @@
constexpr int64_t NANOS_PER_MILLISECOND = 1000 * 1000;
void checkReleaseThenClose(aaudio_performance_mode_t perfMode,
- aaudio_sharing_mode_t sharingMode) {
+ aaudio_sharing_mode_t sharingMode,
+ aaudio_direction_t direction = AAUDIO_DIRECTION_OUTPUT) {
AAudioStreamBuilder* aaudioBuilder = nullptr;
AAudioStream* aaudioStream = nullptr;
@@ -61,6 +67,7 @@
nullptr);
AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
AAudioStreamBuilder_setSharingMode(aaudioBuilder, sharingMode);
+ AAudioStreamBuilder_setDirection(aaudioBuilder, direction);
AAudioStreamBuilder_setFormat(aaudioBuilder, AAUDIO_FORMAT_PCM_FLOAT);
// Create an AAudioStream using the Builder.
@@ -88,14 +95,28 @@
// We should NOT be able to start or change a stream after it has been released.
EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestStart(aaudioStream));
EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
- EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestPause(aaudioStream));
+ // Pause is only implemented for OUTPUT.
+ if (direction == AAUDIO_DIRECTION_OUTPUT) {
+ EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE,
+ AAudioStream_requestPause(aaudioStream));
+ }
EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestStop(aaudioStream));
EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
// Does this crash?
- EXPECT_LT(0, AAudioStream_getFramesRead(aaudioStream));
- EXPECT_LT(0, AAudioStream_getFramesWritten(aaudioStream));
+ EXPECT_GT(AAudioStream_getFramesRead(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getFramesWritten(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getFramesPerBurst(aaudioStream), 0);
+ EXPECT_GE(AAudioStream_getXRunCount(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getBufferCapacityInFrames(aaudioStream), 0);
+ EXPECT_GT(AAudioStream_getBufferSizeInFrames(aaudioStream), 0);
+
+ int64_t timestampFrames = 0;
+ int64_t timestampNanos = 0;
+ aaudio_result_t result = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
+ ×tampFrames, ×tampNanos);
+ EXPECT_TRUE(result == AAUDIO_ERROR_INVALID_STATE || result == AAUDIO_ERROR_UNIMPLEMENTED);
// Verify Closing State. Does this crash?
aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
@@ -107,20 +128,42 @@
EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
}
-TEST(test_various, aaudio_release_close_none) {
+TEST(test_various, aaudio_release_close_none_output) {
checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_NONE,
- AAUDIO_SHARING_MODE_SHARED);
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_OUTPUT);
// No EXCLUSIVE streams with MODE_NONE.
}
-TEST(test_various, aaudio_release_close_low_shared) {
- checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
- AAUDIO_SHARING_MODE_SHARED);
+TEST(test_various, aaudio_release_close_none_input) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_NONE,
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_INPUT);
+ // No EXCLUSIVE streams with MODE_NONE.
}
-TEST(test_various, aaudio_release_close_low_exclusive) {
+TEST(test_various, aaudio_release_close_low_shared_output) {
checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
- AAUDIO_SHARING_MODE_EXCLUSIVE);
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_OUTPUT);
+}
+
+TEST(test_various, aaudio_release_close_low_shared_input) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_SHARING_MODE_SHARED,
+ AAUDIO_DIRECTION_INPUT);
+}
+
+TEST(test_various, aaudio_release_close_low_exclusive_output) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_SHARING_MODE_EXCLUSIVE,
+ AAUDIO_DIRECTION_OUTPUT);
+}
+
+TEST(test_various, aaudio_release_close_low_exclusive_input) {
+ checkReleaseThenClose(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
+ AAUDIO_SHARING_MODE_EXCLUSIVE,
+ AAUDIO_DIRECTION_INPUT);
}
enum FunctionToCall {
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index e8e1a09..2a1e56c 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -77,8 +77,6 @@
"IAudioPolicyService.cpp",
"IAudioPolicyServiceClient.cpp",
"IAudioTrack.cpp",
- "IEffect.cpp",
- "IEffectClient.cpp",
"ToneGenerator.cpp",
"PlayerBase.cpp",
"RecordingActivityTracker.cpp",
@@ -99,6 +97,7 @@
"libmediautils",
"libnblog",
"libprocessgroup",
+ "libshmemcompat",
"libutils",
"libvibrator",
],
@@ -108,7 +107,8 @@
"frameworks/av/media/libnbaio/include_mono/",
],
local_include_dirs: [
- "include/media", "aidl"
+ "include/media",
+ "aidl",
],
header_libs: [
"libaudioclient_headers",
@@ -116,10 +116,16 @@
"libmedia_headers",
],
export_header_lib_headers: ["libaudioclient_headers"],
+ export_static_lib_headers: [
+ "effect-aidl-cpp",
+ "shared-file-region-aidl-cpp",
+ ],
- // for memory heap analysis
static_libs: [
+ "effect-aidl-cpp",
+ // for memory heap analysis
"libc_malloc_debug_backtrace",
+ "shared-file-region-aidl-cpp",
],
cflags: [
"-Wall",
@@ -127,7 +133,7 @@
"-Wno-error=deprecated-declarations",
],
sanitize: {
- misc_undefined : [
+ misc_undefined: [
"unsigned-integer-overflow",
"signed-integer-overflow",
],
@@ -170,3 +176,16 @@
"aidl/android/media/ICaptureStateListener.aidl",
],
}
+
+aidl_interface {
+ name: "effect-aidl",
+ unstable: true,
+ local_include_dir: "aidl",
+ srcs: [
+ "aidl/android/media/IEffect.aidl",
+ "aidl/android/media/IEffectClient.aidl",
+ ],
+ imports: [
+ "shared-file-region-aidl",
+ ],
+}
diff --git a/media/libaudioclient/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index 73b96ab..1282474 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -23,16 +23,28 @@
#include <sys/types.h>
#include <limits.h>
-#include <private/media/AudioEffectShared.h>
-#include <media/AudioEffect.h>
-
-#include <utils/Log.h>
#include <binder/IPCThreadState.h>
-
-
+#include <media/AudioEffect.h>
+#include <media/ShmemCompat.h>
+#include <private/media/AudioEffectShared.h>
+#include <utils/Log.h>
namespace android {
+using binder::Status;
+
+namespace {
+
+// Copy from a raw pointer + size into a vector of bytes.
+void appendToBuffer(const void* data,
+ size_t size,
+ std::vector<uint8_t>* buffer) {
+ const uint8_t* p = reinterpret_cast<const uint8_t*>(data);
+ buffer->insert(buffer->end(), p, p + size);
+}
+
+} // namespace
+
// ---------------------------------------------------------------------------
AudioEffect::AudioEffect(const String16& opPackageName)
@@ -50,7 +62,7 @@
const AudioDeviceTypeAddr& device,
bool probe)
{
- sp<IEffect> iEffect;
+ sp<media::IEffect> iEffect;
sp<IMemory> cblk;
int enabled;
@@ -112,8 +124,10 @@
mEnabled = (volatile int32_t)enabled;
- cblk = iEffect->getCblk();
- if (cblk == 0) {
+ if (media::SharedFileRegion shmem;
+ !iEffect->getCblk(&shmem).isOk()
+ || !convertSharedFileRegionToIMemory(shmem, &cblk)
+ || cblk == 0) {
mStatus = NO_INIT;
ALOGE("Could not get control block");
return mStatus;
@@ -216,15 +230,19 @@
}
status_t status = NO_ERROR;
-
AutoMutex lock(mLock);
if (enabled != mEnabled) {
+ Status bs;
+
if (enabled) {
ALOGV("enable %p", this);
- status = mIEffect->enable();
+ bs = mIEffect->enable(&status);
} else {
ALOGV("disable %p", this);
- status = mIEffect->disable();
+ bs = mIEffect->disable(&status);
+ }
+ if (!bs.isOk()) {
+ status = bs.transactionError();
}
if (status == NO_ERROR) {
mEnabled = enabled;
@@ -257,7 +275,20 @@
mLock.lock();
}
- status_t status = mIEffect->command(cmdCode, cmdSize, cmdData, replySize, replyData);
+ std::vector<uint8_t> data;
+ appendToBuffer(cmdData, cmdSize, &data);
+
+ status_t status;
+ std::vector<uint8_t> response;
+
+ Status bs = mIEffect->command(cmdCode, data, *replySize, &response, &status);
+ if (!bs.isOk()) {
+ status = bs.transactionError();
+ }
+ if (status == NO_ERROR) {
+ memcpy(replyData, response.data(), response.size());
+ *replySize = response.size();
+ }
if (cmdCode == EFFECT_CMD_ENABLE || cmdCode == EFFECT_CMD_DISABLE) {
if (status == NO_ERROR) {
@@ -272,7 +303,6 @@
return status;
}
-
status_t AudioEffect::setParameter(effect_param_t *param)
{
if (mProbe) {
@@ -286,14 +316,27 @@
return BAD_VALUE;
}
- uint32_t size = sizeof(int);
uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize;
ALOGV("setParameter: param: %d, param2: %d", *(int *)param->data,
(param->psize == 8) ? *((int *)param->data + 1): -1);
- return mIEffect->command(EFFECT_CMD_SET_PARAM, sizeof (effect_param_t) + psize, param, &size,
- ¶m->status);
+ std::vector<uint8_t> cmd;
+ appendToBuffer(param, sizeof(effect_param_t) + psize, &cmd);
+ std::vector<uint8_t> response;
+ status_t status;
+ Status bs = mIEffect->command(EFFECT_CMD_SET_PARAM,
+ cmd,
+ sizeof(int),
+ &response,
+ &status);
+ if (!bs.isOk()) {
+ status = bs.transactionError();
+ return status;
+ }
+ assert(response.size() == sizeof(int));
+ memcpy(¶m->status, response.data(), response.size());
+ return status;
}
status_t AudioEffect::setParameterDeferred(effect_param_t *param)
@@ -338,8 +381,18 @@
if (mCblk->clientIndex == 0) {
return INVALID_OPERATION;
}
- uint32_t size = 0;
- return mIEffect->command(EFFECT_CMD_SET_PARAM_COMMIT, 0, NULL, &size, NULL);
+ std::vector<uint8_t> cmd;
+ std::vector<uint8_t> response;
+ status_t status;
+ Status bs = mIEffect->command(EFFECT_CMD_SET_PARAM_COMMIT,
+ cmd,
+ 0,
+ &response,
+ &status);
+ if (!bs.isOk()) {
+ status = bs.transactionError();
+ }
+ return status;
}
status_t AudioEffect::getParameter(effect_param_t *param)
@@ -361,8 +414,18 @@
uint32_t psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
param->vsize;
- return mIEffect->command(EFFECT_CMD_GET_PARAM, sizeof(effect_param_t) + param->psize, param,
- &psize, param);
+ status_t status;
+ std::vector<uint8_t> cmd;
+ std::vector<uint8_t> response;
+ appendToBuffer(param, sizeof(effect_param_t) + param->psize, &cmd);
+
+ Status bs = mIEffect->command(EFFECT_CMD_GET_PARAM, cmd, psize, &response, &status);
+ if (!bs.isOk()) {
+ status = bs.transactionError();
+ return status;
+ }
+ memcpy(param, response.data(), response.size());
+ return status;
}
@@ -410,19 +473,18 @@
}
}
-void AudioEffect::commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize __unused,
- void *cmdData,
- uint32_t replySize __unused,
- void *replyData)
+void AudioEffect::commandExecuted(int32_t cmdCode,
+ const std::vector<uint8_t>& cmdData,
+ const std::vector<uint8_t>& replyData)
{
- if (cmdData == NULL || replyData == NULL) {
+ if (cmdData.empty() || replyData.empty()) {
return;
}
if (mCbf != NULL && cmdCode == EFFECT_CMD_SET_PARAM) {
- effect_param_t *cmd = (effect_param_t *)cmdData;
- cmd->status = *(int32_t *)replyData;
+ std::vector<uint8_t> cmdDataCopy(cmdData);
+ effect_param_t* cmd = reinterpret_cast<effect_param_t *>(cmdDataCopy.data());
+ cmd->status = *reinterpret_cast<const int32_t *>(replyData.data());
mCbf(EVENT_PARAMETER_CHANGED, mUserData, cmd);
}
}
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index f621aa5..49c4bc0 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -1362,7 +1362,7 @@
return aps->registerPolicyMixes(mixes, registration);
}
-status_t AudioSystem::setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices)
+status_t AudioSystem::setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
@@ -1376,7 +1376,7 @@
}
status_t AudioSystem::setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices)
+ const AudioDeviceTypeAddrVector& devices)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
@@ -1603,33 +1603,35 @@
return aps->isCallScreenModeSupported();
}
-status_t AudioSystem::setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device)
+status_t AudioSystem::setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role,
+ const AudioDeviceTypeAddrVector &devices)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) {
return PERMISSION_DENIED;
}
- return aps->setPreferredDeviceForStrategy(strategy, device);
+ return aps->setDevicesRoleForStrategy(strategy, role, devices);
}
-status_t AudioSystem::removePreferredDeviceForStrategy(product_strategy_t strategy)
+status_t AudioSystem::removeDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) {
return PERMISSION_DENIED;
}
- return aps->removePreferredDeviceForStrategy(strategy);
+ return aps->removeDevicesRoleForStrategy(strategy, role);
}
-status_t AudioSystem::getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device)
+status_t AudioSystem::getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role,
+ AudioDeviceTypeAddrVector &devices)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) {
return PERMISSION_DENIED;
}
- return aps->getPreferredDeviceForStrategy(strategy, device);
+ return aps->getDevicesForRoleAndStrategy(strategy, role, devices);
}
class CaptureStateListenerImpl : public media::BnCaptureStateListener,
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 6d79aba..225713a 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -653,9 +653,9 @@
return NO_ERROR;
}
- virtual sp<IEffect> createEffect(
+ virtual sp<media::IEffect> createEffect(
effect_descriptor_t *pDesc,
- const sp<IEffectClient>& client,
+ const sp<media::IEffectClient>& client,
int32_t priority,
audio_io_handle_t output,
audio_session_t sessionId,
@@ -668,7 +668,7 @@
int *enabled)
{
Parcel data, reply;
- sp<IEffect> effect;
+ sp<media::IEffect> effect;
if (pDesc == NULL) {
if (status != NULL) {
*status = BAD_VALUE;
@@ -705,7 +705,7 @@
if (enabled != NULL) {
*enabled = tmp;
}
- effect = interface_cast<IEffect>(reply.readStrongBinder());
+ effect = interface_cast<media::IEffect>(reply.readStrongBinder());
reply.read(pDesc, sizeof(effect_descriptor_t));
}
if (status != NULL) {
@@ -1386,7 +1386,8 @@
if (data.read(&desc, sizeof(effect_descriptor_t)) != NO_ERROR) {
ALOGE("b/23905951");
}
- sp<IEffectClient> client = interface_cast<IEffectClient>(data.readStrongBinder());
+ sp<media::IEffectClient> client =
+ interface_cast<media::IEffectClient>(data.readStrongBinder());
int32_t priority = data.readInt32();
audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
audio_session_t sessionId = (audio_session_t) data.readInt32();
@@ -1402,8 +1403,8 @@
int id = 0;
int enabled = 0;
- sp<IEffect> effect = createEffect(&desc, client, priority, output, sessionId, device,
- opPackageName, pid, probe, &status, &id, &enabled);
+ sp<media::IEffect> effect = createEffect(&desc, client, priority, output, sessionId,
+ device, opPackageName, pid, probe, &status, &id, &enabled);
reply->writeInt32(status);
reply->writeInt32(id);
reply->writeInt32(enabled);
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 60af84b..1491afe 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -112,9 +112,9 @@
MOVE_EFFECTS_TO_IO,
SET_RTT_ENABLED,
IS_CALL_SCREEN_MODE_SUPPORTED,
- SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
- REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
- GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ SET_DEVICES_ROLE_FOR_PRODUCT_STRATEGY,
+ REMOVE_DEVICES_ROLE_FOR_PRODUCT_STRATEGY,
+ GET_DEVICES_FOR_ROLE_AND_PRODUCT_STRATEGY,
GET_DEVICES_FOR_ATTRIBUTES,
AUDIO_MODULES_UPDATED, // oneway
SET_CURRENT_IME_UID,
@@ -1173,31 +1173,18 @@
return reply.readBool();
}
- virtual status_t setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices)
+ virtual status_t setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32((int32_t) uid);
- size_t size = devices.size();
- size_t sizePosition = data.dataPosition();
- data.writeInt32((int32_t) size);
- size_t finalSize = size;
- for (size_t i = 0; i < size; i++) {
- size_t position = data.dataPosition();
- if (devices[i].writeToParcel(&data) != NO_ERROR) {
- data.setDataPosition(position);
- finalSize--;
- }
- }
- if (size != finalSize) {
- size_t position = data.dataPosition();
- data.setDataPosition(sizePosition);
- data.writeInt32(finalSize);
- data.setDataPosition(position);
+ status_t status = data.writeParcelableVector(devices);
+ if (status != NO_ERROR) {
+ return status;
}
- status_t status = remote()->transact(SET_UID_DEVICE_AFFINITY, data, &reply);
+ status = remote()->transact(SET_UID_DEVICE_AFFINITY, data, &reply);
if (status == NO_ERROR) {
status = (status_t)reply.readInt32();
}
@@ -1218,51 +1205,37 @@
return status;
}
- virtual status_t setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ virtual status_t setUserIdDeviceAffinities(int userId, const AudioDeviceTypeAddrVector& devices)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32((int32_t) userId);
- size_t size = devices.size();
- size_t sizePosition = data.dataPosition();
- data.writeInt32((int32_t) size);
- size_t finalSize = size;
- for (size_t i = 0; i < size; i++) {
- size_t position = data.dataPosition();
- if (devices[i].writeToParcel(&data) != NO_ERROR) {
- data.setDataPosition(position);
- finalSize--;
- }
- }
- if (size != finalSize) {
- size_t position = data.dataPosition();
- data.setDataPosition(sizePosition);
- data.writeInt32(finalSize);
- data.setDataPosition(position);
- }
-
- status_t status = remote()->transact(SET_USERID_DEVICE_AFFINITY, data, &reply);
- if (status == NO_ERROR) {
- status = (status_t)reply.readInt32();
- }
+ data.writeInt32((int32_t) userId);
+ status_t status = data.writeParcelableVector(devices);
+ if (status != NO_ERROR) {
return status;
}
- virtual status_t removeUserIdDeviceAffinities(int userId) {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-
- data.writeInt32((int32_t) userId);
-
- status_t status =
- remote()->transact(REMOVE_USERID_DEVICE_AFFINITY, data, &reply);
- if (status == NO_ERROR) {
- status = (status_t) reply.readInt32();
- }
- return status;
+ status = remote()->transact(SET_USERID_DEVICE_AFFINITY, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
}
+ return status;
+ }
+
+ virtual status_t removeUserIdDeviceAffinities(int userId) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+
+ data.writeInt32((int32_t) userId);
+
+ status_t status =
+ remote()->transact(REMOVE_USERID_DEVICE_AFFINITY, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t) reply.readInt32();
+ }
+ return status;
+ }
virtual status_t listAudioProductStrategies(AudioProductStrategyVector &strategies)
{
@@ -1384,17 +1357,31 @@
return reply.readBool();
}
- virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device)
+ virtual status_t setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role, const AudioDeviceTypeAddrVector &devices)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeUint32(static_cast<uint32_t>(strategy));
- status_t status = device.writeToParcel(&data);
+ data.writeUint32(static_cast<uint32_t>(role));
+ status_t status = data.writeParcelableVector(devices);
if (status != NO_ERROR) {
return BAD_VALUE;
}
- status = remote()->transact(SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ status = remote()->transact(SET_DEVICES_ROLE_FOR_PRODUCT_STRATEGY, data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ return static_cast<status_t>(reply.readInt32());
+ }
+
+ virtual status_t removeDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ data.writeUint32(static_cast<uint32_t>(strategy));
+ data.writeUint32(static_cast<uint32_t>(role));
+ status_t status = remote()->transact(REMOVE_DEVICES_ROLE_FOR_PRODUCT_STRATEGY,
data, &reply);
if (status != NO_ERROR) {
return status;
@@ -1402,31 +1389,19 @@
return static_cast<status_t>(reply.readInt32());
}
- virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy)
+ virtual status_t getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role, AudioDeviceTypeAddrVector &devices)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeUint32(static_cast<uint32_t>(strategy));
- status_t status = remote()->transact(REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
- data, &reply);
- if (status != NO_ERROR) {
- return status;
- }
- return static_cast<status_t>(reply.readInt32());
- }
-
- virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeUint32(static_cast<uint32_t>(strategy));
- status_t status = remote()->transact(GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ data.writeUint32(static_cast<uint32_t>(role));
+ status_t status = remote()->transact(GET_DEVICES_FOR_ROLE_AND_PRODUCT_STRATEGY,
data, &reply);
if (status != NO_ERROR) {
return status;
}
- status = device.readFromParcel(&reply);
+ status = reply.readParcelableVector(&devices);
if (status != NO_ERROR) {
return status;
}
@@ -1561,10 +1536,10 @@
case RELEASE_SOUNDTRIGGER_SESSION:
case SET_RTT_ENABLED:
case IS_CALL_SCREEN_MODE_SUPPORTED:
- case SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY:
+ case SET_DEVICES_ROLE_FOR_PRODUCT_STRATEGY:
case SET_SUPPORTED_SYSTEM_USAGES:
- case REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY:
- case GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY:
+ case REMOVE_DEVICES_ROLE_FOR_PRODUCT_STRATEGY:
+ case GET_DEVICES_FOR_ROLE_AND_PRODUCT_STRATEGY:
case GET_DEVICES_FOR_ATTRIBUTES:
case SET_ALLOWED_CAPTURE_POLICY:
case AUDIO_MODULES_UPDATED:
@@ -2460,15 +2435,12 @@
case SET_UID_DEVICE_AFFINITY: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
const uid_t uid = (uid_t) data.readInt32();
- Vector<AudioDeviceTypeAddr> devices;
- size_t size = (size_t)data.readInt32();
- for (size_t i = 0; i < size; i++) {
- AudioDeviceTypeAddr device;
- if (device.readFromParcel((Parcel*)&data) == NO_ERROR) {
- devices.add(device);
- }
+ AudioDeviceTypeAddrVector devices;
+ status_t status = data.readParcelableVector(&devices);
+ if (status != NO_ERROR) {
+ return status;
}
- status_t status = setUidDeviceAffinities(uid, devices);
+ status = setUidDeviceAffinities(uid, devices);
reply->writeInt32(status);
return NO_ERROR;
}
@@ -2484,15 +2456,12 @@
case SET_USERID_DEVICE_AFFINITY: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
const int userId = (int) data.readInt32();
- Vector<AudioDeviceTypeAddr> devices;
- size_t size = (size_t)data.readInt32();
- for (size_t i = 0; i < size; i++) {
- AudioDeviceTypeAddr device;
- if (device.readFromParcel((Parcel*)&data) == NO_ERROR) {
- devices.add(device);
- }
+ AudioDeviceTypeAddrVector devices;
+ status_t status = data.readParcelableVector(&devices);
+ if (status != NO_ERROR) {
+ return status;
}
- status_t status = setUserIdDeviceAffinities(userId, devices);
+ status = setUserIdDeviceAffinities(userId, devices);
reply->writeInt32(status);
return NO_ERROR;
}
@@ -2649,33 +2618,36 @@
return NO_ERROR;
}
- case SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY: {
+ case SET_DEVICES_ROLE_FOR_PRODUCT_STRATEGY: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
product_strategy_t strategy = (product_strategy_t) data.readUint32();
- AudioDeviceTypeAddr device;
- status_t status = device.readFromParcel((Parcel*)&data);
+ device_role_t role = (device_role_t) data.readUint32();
+ AudioDeviceTypeAddrVector devices;
+ status_t status = data.readParcelableVector(&devices);
if (status != NO_ERROR) {
return status;
}
- status = setPreferredDeviceForStrategy(strategy, device);
+ status = setDevicesRoleForStrategy(strategy, role, devices);
reply->writeInt32(status);
return NO_ERROR;
}
- case REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY: {
+ case REMOVE_DEVICES_ROLE_FOR_PRODUCT_STRATEGY: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
product_strategy_t strategy = (product_strategy_t) data.readUint32();
- status_t status = removePreferredDeviceForStrategy(strategy);
+ device_role_t role = (device_role_t) data.readUint32();
+ status_t status = removeDevicesRoleForStrategy(strategy, role);
reply->writeInt32(status);
return NO_ERROR;
}
- case GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY: {
+ case GET_DEVICES_FOR_ROLE_AND_PRODUCT_STRATEGY: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
product_strategy_t strategy = (product_strategy_t) data.readUint32();
- AudioDeviceTypeAddr device;
- status_t status = getPreferredDeviceForStrategy(strategy, device);
- status_t marshall_status = device.writeToParcel(reply);
+ device_role_t role = (device_role_t) data.readUint32();
+ AudioDeviceTypeAddrVector devices;
+ status_t status = getDevicesForRoleAndStrategy(strategy, role, devices);
+ status_t marshall_status = reply->writeParcelableVector(devices);
if (marshall_status != NO_ERROR) {
return marshall_status;
}
diff --git a/media/libaudioclient/IEffect.cpp b/media/libaudioclient/IEffect.cpp
deleted file mode 100644
index 5d47dff..0000000
--- a/media/libaudioclient/IEffect.cpp
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
-**
-** Copyright 2010, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "IEffect"
-#include <utils/Log.h>
-#include <stdint.h>
-#include <sys/types.h>
-#include <binder/Parcel.h>
-#include <media/IEffect.h>
-
-namespace android {
-
-// Maximum command/reply size expected
-#define EFFECT_PARAM_SIZE_MAX 65536
-
-enum {
- ENABLE = IBinder::FIRST_CALL_TRANSACTION,
- DISABLE,
- COMMAND,
- DISCONNECT,
- GET_CBLK
-};
-
-class BpEffect: public BpInterface<IEffect>
-{
-public:
- explicit BpEffect(const sp<IBinder>& impl)
- : BpInterface<IEffect>(impl)
- {
- }
-
- status_t enable()
- {
- ALOGV("enable");
- Parcel data, reply;
- data.writeInterfaceToken(IEffect::getInterfaceDescriptor());
- remote()->transact(ENABLE, data, &reply);
- return reply.readInt32();
- }
-
- status_t disable()
- {
- ALOGV("disable");
- Parcel data, reply;
- data.writeInterfaceToken(IEffect::getInterfaceDescriptor());
- remote()->transact(DISABLE, data, &reply);
- return reply.readInt32();
- }
-
- status_t command(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *pReplySize,
- void *pReplyData)
- {
- ALOGV("command");
- Parcel data, reply;
- data.writeInterfaceToken(IEffect::getInterfaceDescriptor());
- data.writeInt32(cmdCode);
- int size = cmdSize;
- if (pCmdData == NULL) {
- size = 0;
- }
- data.writeInt32(size);
- if (size) {
- data.write(pCmdData, size);
- }
- if (pReplySize == NULL) {
- size = 0;
- } else {
- size = *pReplySize;
- }
- data.writeInt32(size);
-
- status_t status = remote()->transact(COMMAND, data, &reply);
- if (status == NO_ERROR) {
- status = reply.readInt32();
- }
- if (status != NO_ERROR) {
- if (pReplySize != NULL)
- *pReplySize = 0;
- return status;
- }
-
- size = reply.readInt32();
- if (size != 0 && pReplyData != NULL && pReplySize != NULL) {
- reply.read(pReplyData, size);
- *pReplySize = size;
- }
- return status;
- }
-
- void disconnect()
- {
- ALOGV("disconnect");
- Parcel data, reply;
- data.writeInterfaceToken(IEffect::getInterfaceDescriptor());
- remote()->transact(DISCONNECT, data, &reply);
- return;
- }
-
- virtual sp<IMemory> getCblk() const
- {
- Parcel data, reply;
- sp<IMemory> cblk;
- data.writeInterfaceToken(IEffect::getInterfaceDescriptor());
- status_t status = remote()->transact(GET_CBLK, data, &reply);
- if (status == NO_ERROR) {
- cblk = interface_cast<IMemory>(reply.readStrongBinder());
- if (cblk != 0 && cblk->unsecurePointer() == NULL) {
- cblk.clear();
- }
- }
- return cblk;
- }
- };
-
-IMPLEMENT_META_INTERFACE(Effect, "android.media.IEffect");
-
-// ----------------------------------------------------------------------
-
-status_t BnEffect::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- switch (code) {
- case ENABLE: {
- ALOGV("ENABLE");
- CHECK_INTERFACE(IEffect, data, reply);
- reply->writeInt32(enable());
- return NO_ERROR;
- } break;
-
- case DISABLE: {
- ALOGV("DISABLE");
- CHECK_INTERFACE(IEffect, data, reply);
- reply->writeInt32(disable());
- return NO_ERROR;
- } break;
-
- case COMMAND: {
- ALOGV("COMMAND");
- CHECK_INTERFACE(IEffect, data, reply);
- uint32_t cmdCode = data.readInt32();
- uint32_t cmdSize = data.readInt32();
- char *cmd = NULL;
- if (cmdSize) {
- if (cmdSize > EFFECT_PARAM_SIZE_MAX) {
- reply->writeInt32(NO_MEMORY);
- return NO_ERROR;
- }
- cmd = (char *)calloc(cmdSize, 1);
- if (cmd == NULL) {
- reply->writeInt32(NO_MEMORY);
- return NO_ERROR;
- }
- data.read(cmd, cmdSize);
- }
- uint32_t replySize = data.readInt32();
- uint32_t replySz = replySize;
- char *resp = NULL;
- if (replySize) {
- if (replySize > EFFECT_PARAM_SIZE_MAX) {
- free(cmd);
- reply->writeInt32(NO_MEMORY);
- return NO_ERROR;
- }
- resp = (char *)calloc(replySize, 1);
- if (resp == NULL) {
- free(cmd);
- reply->writeInt32(NO_MEMORY);
- return NO_ERROR;
- }
- }
- status_t status = command(cmdCode, cmdSize, cmd, &replySz, resp);
- reply->writeInt32(status);
- if (status == NO_ERROR) {
- if (replySz < replySize) {
- replySize = replySz;
- }
- reply->writeInt32(replySize);
- if (replySize) {
- reply->write(resp, replySize);
- }
- }
- if (cmd) {
- free(cmd);
- }
- if (resp) {
- free(resp);
- }
- return NO_ERROR;
- } break;
-
- case DISCONNECT: {
- ALOGV("DISCONNECT");
- CHECK_INTERFACE(IEffect, data, reply);
- disconnect();
- return NO_ERROR;
- } break;
-
- case GET_CBLK: {
- CHECK_INTERFACE(IEffect, data, reply);
- reply->writeStrongBinder(IInterface::asBinder(getCblk()));
- return NO_ERROR;
- } break;
-
- default:
- return BBinder::onTransact(code, data, reply, flags);
- }
-}
-
-// ----------------------------------------------------------------------------
-
-} // namespace android
diff --git a/media/libaudioclient/IEffectClient.cpp b/media/libaudioclient/IEffectClient.cpp
deleted file mode 100644
index 3f2c67d..0000000
--- a/media/libaudioclient/IEffectClient.cpp
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
-**
-** Copyright 2010, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "IEffectClient"
-#include <utils/Log.h>
-#include <stdint.h>
-#include <sys/types.h>
-#include <media/IEffectClient.h>
-
-namespace android {
-
-enum {
- CONTROL_STATUS_CHANGED = IBinder::FIRST_CALL_TRANSACTION,
- ENABLE_STATUS_CHANGED,
- COMMAND_EXECUTED
-};
-
-class BpEffectClient: public BpInterface<IEffectClient>
-{
-public:
- explicit BpEffectClient(const sp<IBinder>& impl)
- : BpInterface<IEffectClient>(impl)
- {
- }
-
- void controlStatusChanged(bool controlGranted)
- {
- ALOGV("controlStatusChanged");
- Parcel data, reply;
- data.writeInterfaceToken(IEffectClient::getInterfaceDescriptor());
- data.writeInt32((uint32_t)controlGranted);
- remote()->transact(CONTROL_STATUS_CHANGED, data, &reply, IBinder::FLAG_ONEWAY);
- }
-
- void enableStatusChanged(bool enabled)
- {
- ALOGV("enableStatusChanged");
- Parcel data, reply;
- data.writeInterfaceToken(IEffectClient::getInterfaceDescriptor());
- data.writeInt32((uint32_t)enabled);
- remote()->transact(ENABLE_STATUS_CHANGED, data, &reply, IBinder::FLAG_ONEWAY);
- }
-
- void commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t replySize,
- void *pReplyData)
- {
- ALOGV("commandExecuted");
- Parcel data, reply;
- data.writeInterfaceToken(IEffectClient::getInterfaceDescriptor());
- data.writeInt32(cmdCode);
- int size = cmdSize;
- if (pCmdData == NULL) {
- size = 0;
- }
- data.writeInt32(size);
- if (size) {
- data.write(pCmdData, size);
- }
- size = replySize;
- if (pReplyData == NULL) {
- size = 0;
- }
- data.writeInt32(size);
- if (size) {
- data.write(pReplyData, size);
- }
- remote()->transact(COMMAND_EXECUTED, data, &reply, IBinder::FLAG_ONEWAY);
- }
-
-};
-
-IMPLEMENT_META_INTERFACE(EffectClient, "android.media.IEffectClient");
-
-// ----------------------------------------------------------------------
-
-status_t BnEffectClient::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- switch (code) {
- case CONTROL_STATUS_CHANGED: {
- ALOGV("CONTROL_STATUS_CHANGED");
- CHECK_INTERFACE(IEffectClient, data, reply);
- bool hasControl = (bool)data.readInt32();
- controlStatusChanged(hasControl);
- return NO_ERROR;
- } break;
- case ENABLE_STATUS_CHANGED: {
- ALOGV("ENABLE_STATUS_CHANGED");
- CHECK_INTERFACE(IEffectClient, data, reply);
- bool enabled = (bool)data.readInt32();
- enableStatusChanged(enabled);
- return NO_ERROR;
- } break;
- case COMMAND_EXECUTED: {
- ALOGV("COMMAND_EXECUTED");
- CHECK_INTERFACE(IEffectClient, data, reply);
- uint32_t cmdCode = data.readInt32();
- uint32_t cmdSize = data.readInt32();
- char *cmd = NULL;
- if (cmdSize) {
- cmd = (char *)malloc(cmdSize);
- data.read(cmd, cmdSize);
- }
- uint32_t replySize = data.readInt32();
- char *resp = NULL;
- if (replySize) {
- resp = (char *)malloc(replySize);
- data.read(resp, replySize);
- }
- commandExecuted(cmdCode, cmdSize, cmd, replySize, resp);
- if (cmd) {
- free(cmd);
- }
- if (resp) {
- free(resp);
- }
- return NO_ERROR;
- } break;
- default:
- return BBinder::onTransact(code, data, reply, flags);
- }
-}
-
-// ----------------------------------------------------------------------------
-
-} // namespace android
diff --git a/media/libaudioclient/aidl/android/media/IEffect.aidl b/media/libaudioclient/aidl/android/media/IEffect.aidl
new file mode 100644
index 0000000..9548e46
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/IEffect.aidl
@@ -0,0 +1,65 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.SharedFileRegion;
+
+/**
+ * The IEffect interface enables control of the effect module activity and parameters.
+ *
+ * @hide
+ */
+interface IEffect {
+ /**
+ * Activates the effect module by connecting it to the audio path.
+ * @return a status_t code.
+ */
+ int enable();
+
+ /**
+ * Deactivates the effect module by disconnecting it from the audio path.
+ * @return a status_t code.
+ */
+ int disable();
+
+ /**
+ * Sends control, reads or writes parameters. Same behavior as the command() method in the
+ * effect control interface.
+ * Refer to system/audio_effect.h for a description of the valid command codes and their
+ * associated parameter and return messages. The cmdData and response parameters are expected to
+ * contain the respective types in a standard C memory layout.
+ *
+ * TODO(ytai): replace opaque byte arrays with strongly typed parameters.
+ */
+ int command(int cmdCode, in byte[] cmdData, int maxResponseSize, out byte[] response);
+
+ /**
+ * Disconnects the IEffect interface from the effect module.
+ * This will also delete the effect module and release the effect engine in the library if this
+ * is the last client disconnected. To release control of the effect module, the application can
+ * disconnect or delete the IEffect interface.
+ */
+ void disconnect();
+
+ /**
+ * returns a pointer to a shared memory area used to pass multiple parameters to the effect
+ * module without multiplying the binder calls.
+ *
+ * TODO(ytai): Explain how this should be used exactly.
+ */
+ SharedFileRegion getCblk();
+}
diff --git a/media/libaudioclient/aidl/android/media/IEffectClient.aidl b/media/libaudioclient/aidl/android/media/IEffectClient.aidl
new file mode 100644
index 0000000..d1e331c
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/IEffectClient.aidl
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * A callback interface for getting effect-related notifications.
+ *
+ * @hide
+ */
+interface IEffectClient {
+ /**
+ * Called whenever the status of granting control over the effect to the application
+ * has changed.
+ * @param controlGranted true iff the application has the control of the effect module.
+ */
+ oneway void controlStatusChanged(boolean controlGranted);
+
+ /**
+ * Called whenever the effect has been enabled or disabled. Received only if the client is not
+ * currently controlling the effect.
+ * @param enabled true if the effect module has been activated, false if deactivated.
+ */
+ oneway void enableStatusChanged(boolean enabled);
+
+ /**
+ * A command has been send to the effect engine. Received only if the client is not currently
+ * controlling the effect. See IEffect.command() for a description of buffer contents.
+ *
+ * TODO(ytai): replace opaque byte arrays with strongly typed parameters.
+ */
+ oneway void commandExecuted(int cmdCode, in byte[] cmdData, in byte[] replyData);
+}
diff --git a/media/libaudioclient/include/media/AudioEffect.h b/media/libaudioclient/include/media/AudioEffect.h
index 3d4bb4e..8371711 100644
--- a/media/libaudioclient/include/media/AudioEffect.h
+++ b/media/libaudioclient/include/media/AudioEffect.h
@@ -22,8 +22,6 @@
#include <media/IAudioFlinger.h>
#include <media/IAudioPolicyService.h>
-#include <media/IEffect.h>
-#include <media/IEffectClient.h>
#include <media/AudioSystem.h>
#include <system/audio_effect.h>
@@ -31,6 +29,9 @@
#include <utils/Errors.h>
#include <binder/IInterface.h>
+#include "android/media/IEffect.h"
+#include "android/media/BnEffectClient.h"
+
namespace android {
@@ -549,45 +550,43 @@
// IEffectClient
virtual void controlStatusChanged(bool controlGranted);
virtual void enableStatusChanged(bool enabled);
- virtual void commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t replySize,
- void *pReplyData);
+ virtual void commandExecuted(int32_t cmdCode,
+ const std::vector<uint8_t>& cmdData,
+ const std::vector<uint8_t>& replyData);
private:
// Implements the IEffectClient interface
class EffectClient :
- public android::BnEffectClient, public android::IBinder::DeathRecipient
+ public media::BnEffectClient, public android::IBinder::DeathRecipient
{
public:
EffectClient(AudioEffect *effect) : mEffect(effect){}
// IEffectClient
- virtual void controlStatusChanged(bool controlGranted) {
+ binder::Status controlStatusChanged(bool controlGranted) override {
sp<AudioEffect> effect = mEffect.promote();
if (effect != 0) {
effect->controlStatusChanged(controlGranted);
}
+ return binder::Status::ok();
}
- virtual void enableStatusChanged(bool enabled) {
+ binder::Status enableStatusChanged(bool enabled) override {
sp<AudioEffect> effect = mEffect.promote();
if (effect != 0) {
effect->enableStatusChanged(enabled);
}
+ return binder::Status::ok();
}
- virtual void commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t replySize,
- void *pReplyData) {
+ binder::Status commandExecuted(int32_t cmdCode,
+ const std::vector<uint8_t>& cmdData,
+ const std::vector<uint8_t>& replyData) override {
sp<AudioEffect> effect = mEffect.promote();
if (effect != 0) {
- effect->commandExecuted(
- cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+ effect->commandExecuted(cmdCode, cmdData, replyData);
}
+ return binder::Status::ok();
}
// IBinder::DeathRecipient
@@ -604,7 +603,7 @@
void binderDied();
- sp<IEffect> mIEffect; // IEffect binder interface
+ sp<media::IEffect> mIEffect; // IEffect binder interface
sp<EffectClient> mIEffectClient; // IEffectClient implementation
sp<IMemory> mCblkMemory; // shared memory for deferred parameter setting
effect_param_cblk_t* mCblk = nullptr; // control block for deferred parameter setting
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 19c2cbd..09025d1 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -361,11 +361,11 @@
static status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration);
- static status_t setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices);
+ static status_t setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices);
static status_t removeUidDeviceAffinities(uid_t uid);
- static status_t setUserIdDeviceAffinities(int userId, const Vector<AudioDeviceTypeAddr>& devices);
+ static status_t setUserIdDeviceAffinities(int userId, const AudioDeviceTypeAddrVector& devices);
static status_t removeUserIdDeviceAffinities(int userId);
@@ -425,13 +425,13 @@
*/
static status_t setAudioHalPids(const std::vector<pid_t>& pids);
- static status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device);
+ static status_t setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role, const AudioDeviceTypeAddrVector &devices);
- static status_t removePreferredDeviceForStrategy(product_strategy_t strategy);
+ static status_t removeDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role);
- static status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device);
+ static status_t getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role, AudioDeviceTypeAddrVector &devices);
static status_t getDeviceForStrategy(product_strategy_t strategy,
AudioDeviceTypeAddr &device);
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 612ce7a..b950d0f 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -33,14 +33,14 @@
#include <system/audio.h>
#include <system/audio_effect.h>
#include <system/audio_policy.h>
-#include <media/IEffect.h>
-#include <media/IEffectClient.h>
#include <utils/String8.h>
#include <media/MicrophoneInfo.h>
#include <vector>
#include "android/media/IAudioRecord.h"
#include "android/media/IAudioTrackCallback.h"
+#include "android/media/IEffect.h"
+#include "android/media/IEffectClient.h"
namespace android {
@@ -463,9 +463,9 @@
uint32_t preferredTypeFlag,
effect_descriptor_t *pDescriptor) const = 0;
- virtual sp<IEffect> createEffect(
+ virtual sp<media::IEffect> createEffect(
effect_descriptor_t *pDesc,
- const sp<IEffectClient>& client,
+ const sp<media::IEffectClient>& client,
int32_t priority,
// AudioFlinger doesn't take over handle reference from client
audio_io_handle_t output,
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index bb1c07f..afb0fda 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -196,13 +196,13 @@
virtual status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration) = 0;
- virtual status_t setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices)
+ virtual status_t setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices)
= 0;
virtual status_t removeUidDeviceAffinities(uid_t uid) = 0;
virtual status_t setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices) = 0;
+ const AudioDeviceTypeAddrVector& devices) = 0;
virtual status_t removeUserIdDeviceAffinities(int userId) = 0;
@@ -241,13 +241,16 @@
virtual bool isCallScreenModeSupported() = 0;
- virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device) = 0;
+ virtual status_t setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role,
+ const AudioDeviceTypeAddrVector &devices) = 0;
- virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy) = 0;
+ virtual status_t removeDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role) = 0;
- virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device) = 0;
+ virtual status_t getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role,
+ AudioDeviceTypeAddrVector &devices) = 0;
// The return code here is only intended to represent transport errors. The
// actual server implementation should always return NO_ERROR.
diff --git a/media/libaudioclient/include/media/IEffect.h b/media/libaudioclient/include/media/IEffect.h
deleted file mode 100644
index ff04869..0000000
--- a/media/libaudioclient/include/media/IEffect.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_IEFFECT_H
-#define ANDROID_IEFFECT_H
-
-#include <utils/RefBase.h>
-#include <binder/IInterface.h>
-#include <binder/Parcel.h>
-#include <binder/IMemory.h>
-
-namespace android {
-
-class IEffect: public IInterface
-{
-public:
- DECLARE_META_INTERFACE(Effect);
-
- virtual status_t enable() = 0;
-
- virtual status_t disable() = 0;
-
- virtual status_t command(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *pReplySize,
- void *pReplyData) = 0;
-
- virtual void disconnect() = 0;
-
- virtual sp<IMemory> getCblk() const = 0;
-};
-
-// ----------------------------------------------------------------------------
-
-class BnEffect: public BnInterface<IEffect>
-{
-public:
- virtual status_t onTransact( uint32_t code,
- const Parcel& data,
- Parcel* reply,
- uint32_t flags = 0);
-};
-
-}; // namespace android
-
-#endif // ANDROID_IEFFECT_H
diff --git a/media/libaudioclient/include/media/IEffectClient.h b/media/libaudioclient/include/media/IEffectClient.h
deleted file mode 100644
index 2f78c98..0000000
--- a/media/libaudioclient/include/media/IEffectClient.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_IEFFECTCLIENT_H
-#define ANDROID_IEFFECTCLIENT_H
-
-#include <utils/RefBase.h>
-#include <binder/IInterface.h>
-#include <binder/Parcel.h>
-#include <binder/IMemory.h>
-
-namespace android {
-
-class IEffectClient: public IInterface
-{
-public:
- DECLARE_META_INTERFACE(EffectClient);
-
- virtual void controlStatusChanged(bool controlGranted) = 0;
- virtual void enableStatusChanged(bool enabled) = 0;
- virtual void commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t replySize,
- void *pReplyData) = 0;
-};
-
-// ----------------------------------------------------------------------------
-
-class BnEffectClient: public BnInterface<IEffectClient>
-{
-public:
- virtual status_t onTransact( uint32_t code,
- const Parcel& data,
- Parcel* reply,
- uint32_t flags = 0);
-};
-
-}; // namespace android
-
-#endif // ANDROID_IEFFECTCLIENT_H
diff --git a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
index b44043a..da2e109 100644
--- a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
+++ b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
@@ -16,12 +16,56 @@
#include <media/AudioDeviceTypeAddr.h>
+#include <arpa/inet.h>
+#include <iostream>
+#include <regex>
+#include <sstream>
+
namespace android {
+namespace {
+
+static const std::string SUPPRESSED = "SUPPRESSED";
+static const std::regex MAC_ADDRESS_REGEX("([0-9a-fA-F]{2}:){5}[0-9a-fA-F]{2}");
+
+bool isSenstiveAddress(const std::string &address) {
+ if (std::regex_match(address, MAC_ADDRESS_REGEX)) {
+ return true;
+ }
+
+ sockaddr_storage ss4;
+ if (inet_pton(AF_INET, address.c_str(), &ss4) > 0) {
+ return true;
+ }
+
+ sockaddr_storage ss6;
+ if (inet_pton(AF_INET6, address.c_str(), &ss6) > 0) {
+ return true;
+ }
+
+ return false;
+}
+
+} // namespace
+
+AudioDeviceTypeAddr::AudioDeviceTypeAddr(audio_devices_t type, const std::string &address) :
+ mType(type), mAddress(address) {
+ mIsAddressSensitive = isSenstiveAddress(mAddress);
+}
+
const char* AudioDeviceTypeAddr::getAddress() const {
return mAddress.c_str();
}
+const std::string& AudioDeviceTypeAddr::address() const {
+ return mAddress;
+}
+
+void AudioDeviceTypeAddr::setAddress(const std::string& address) {
+ mAddress = address;
+ mIsAddressSensitive = isSenstiveAddress(mAddress);
+}
+
bool AudioDeviceTypeAddr::equals(const AudioDeviceTypeAddr& other) const {
return mType == other.mType && mAddress == other.mAddress;
}
@@ -38,7 +82,17 @@
void AudioDeviceTypeAddr::reset() {
mType = AUDIO_DEVICE_NONE;
- mAddress = "";
+ setAddress("");
+}
+
+std::string AudioDeviceTypeAddr::toString(bool includeSensitiveInfo) const {
+ std::stringstream sstream;
+ sstream << "type:0x" << std::hex << mType;
+ // IP and MAC address are sensitive information. The sensitive information will be suppressed
+ // is `includeSensitiveInfo` is false.
+ sstream << ",@:"
+ << (!includeSensitiveInfo && mIsAddressSensitive ? SUPPRESSED : mAddress);
+ return sstream.str();
}
status_t AudioDeviceTypeAddr::readFromParcel(const Parcel *parcel) {
@@ -64,4 +118,16 @@
return deviceTypes;
}
-}
\ No newline at end of file
+std::string dumpAudioDeviceTypeAddrVector(const AudioDeviceTypeAddrVector& deviceTypeAddrs,
+ bool includeSensitiveInfo) {
+ std::stringstream stream;
+ for (auto it = deviceTypeAddrs.begin(); it != deviceTypeAddrs.end(); ++it) {
+ if (it != deviceTypeAddrs.begin()) {
+ stream << " ";
+ }
+ stream << it->toString(includeSensitiveInfo);
+ }
+ return stream.str();
+}
+
+} // namespace android
\ No newline at end of file
diff --git a/media/libaudiofoundation/DeviceDescriptorBase.cpp b/media/libaudiofoundation/DeviceDescriptorBase.cpp
index e9b589d..16cf71a 100644
--- a/media/libaudiofoundation/DeviceDescriptorBase.cpp
+++ b/media/libaudiofoundation/DeviceDescriptorBase.cpp
@@ -22,9 +22,6 @@
#include <media/DeviceDescriptorBase.h>
#include <media/TypeConverter.h>
-#include <arpa/inet.h>
-#include <regex>
-
namespace android {
DeviceDescriptorBase::DeviceDescriptorBase(audio_devices_t type) :
@@ -37,46 +34,19 @@
{
}
-namespace {
-
-static const std::string SUPPRESSED = "SUPPRESSED";
-static const std::regex MAC_ADDRESS_REGEX("([0-9a-fA-F]{2}:){5}[0-9a-fA-F]{2}");
-
-bool isAddressSensitive(const std::string &address) {
- if (std::regex_match(address, MAC_ADDRESS_REGEX)) {
- return true;
- }
-
- sockaddr_storage ss4;
- if (inet_pton(AF_INET, address.c_str(), &ss4) > 0) {
- return true;
- }
-
- sockaddr_storage ss6;
- if (inet_pton(AF_INET6, address.c_str(), &ss6) > 0) {
- return true;
- }
-
- return false;
-}
-
-} // namespace
-
DeviceDescriptorBase::DeviceDescriptorBase(const AudioDeviceTypeAddr &deviceTypeAddr) :
AudioPort("", AUDIO_PORT_TYPE_DEVICE,
audio_is_output_device(deviceTypeAddr.mType) ? AUDIO_PORT_ROLE_SINK :
AUDIO_PORT_ROLE_SOURCE),
mDeviceTypeAddr(deviceTypeAddr)
{
- if (mDeviceTypeAddr.mAddress.empty() && audio_is_remote_submix_device(mDeviceTypeAddr.mType)) {
- mDeviceTypeAddr.mAddress = "0";
+ if (mDeviceTypeAddr.address().empty() && audio_is_remote_submix_device(mDeviceTypeAddr.mType)) {
+ mDeviceTypeAddr.setAddress("0");
}
- mIsAddressSensitive = isAddressSensitive(mDeviceTypeAddr.mAddress);
}
void DeviceDescriptorBase::setAddress(const std::string &address) {
- mDeviceTypeAddr.mAddress = address;
- mIsAddressSensitive = isAddressSensitive(address);
+ mDeviceTypeAddr.setAddress(address);
}
void DeviceDescriptorBase::toAudioPortConfig(struct audio_port_config *dstConfig,
@@ -157,7 +127,7 @@
"%*s- supported encapsulation metadata types: %u",
spaces, "", mEncapsulationMetadataTypes));
- if (mDeviceTypeAddr.mAddress.size() != 0) {
+ if (mDeviceTypeAddr.address().size() != 0) {
dst->append(base::StringPrintf(
"%*s- address: %-32s\n", spaces, "", mDeviceTypeAddr.getAddress()));
}
@@ -166,14 +136,7 @@
std::string DeviceDescriptorBase::toString(bool includeSensitiveInfo) const
{
- std::stringstream sstream;
- sstream << "type:0x" << std::hex << type();
- // IP and MAC address are sensitive information. The sensitive information will be suppressed
- // is `includeSensitiveInfo` is false.
- sstream << ",@:"
- << (!includeSensitiveInfo && mIsAddressSensitive ? SUPPRESSED
- : mDeviceTypeAddr.mAddress);
- return sstream.str();
+ return mDeviceTypeAddr.toString(includeSensitiveInfo);
}
void DeviceDescriptorBase::log() const
diff --git a/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
index 60ea78e..3e03df7 100644
--- a/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
+++ b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
@@ -27,14 +27,20 @@
namespace android {
-struct AudioDeviceTypeAddr : public Parcelable {
+class AudioDeviceTypeAddr : public Parcelable {
+public:
AudioDeviceTypeAddr() = default;
- AudioDeviceTypeAddr(audio_devices_t type, const std::string& address) :
- mType(type), mAddress(address) {}
+ AudioDeviceTypeAddr(audio_devices_t type, const std::string& address);
const char* getAddress() const;
+ const std::string& address() const;
+
+ void setAddress(const std::string& address);
+
+ bool isAddressSensitive();
+
bool equals(const AudioDeviceTypeAddr& other) const;
AudioDeviceTypeAddr& operator= (const AudioDeviceTypeAddr&) = default;
@@ -43,12 +49,17 @@
void reset();
+ std::string toString(bool includeSensitiveInfo=false) const;
+
status_t readFromParcel(const Parcel *parcel) override;
status_t writeToParcel(Parcel *parcel) const override;
audio_devices_t mType = AUDIO_DEVICE_NONE;
+
+private:
std::string mAddress;
+ bool mIsAddressSensitive;
};
using AudioDeviceTypeAddrVector = std::vector<AudioDeviceTypeAddr>;
@@ -58,4 +69,7 @@
*/
DeviceTypeSet getAudioDeviceTypes(const AudioDeviceTypeAddrVector& deviceTypeAddrs);
-}
+std::string dumpAudioDeviceTypeAddrVector(const AudioDeviceTypeAddrVector& deviceTypeAddrs,
+ bool includeSensitiveInfo=false);
+
+} // namespace android
diff --git a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
index c143c7e..0cbd1de 100644
--- a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
+++ b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
@@ -41,7 +41,7 @@
virtual ~DeviceDescriptorBase() {}
audio_devices_t type() const { return mDeviceTypeAddr.mType; }
- std::string address() const { return mDeviceTypeAddr.mAddress; }
+ const std::string& address() const { return mDeviceTypeAddr.address(); }
void setAddress(const std::string &address);
const AudioDeviceTypeAddr& getDeviceTypeAddr() const { return mDeviceTypeAddr; }
@@ -77,7 +77,6 @@
protected:
AudioDeviceTypeAddr mDeviceTypeAddr;
- bool mIsAddressSensitive;
uint32_t mEncapsulationModes = 0;
uint32_t mEncapsulationMetadataTypes = 0;
};
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index 1709d1e..fab0fea 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -18,6 +18,7 @@
"libaudiohal@4.0",
"libaudiohal@5.0",
"libaudiohal@6.0",
+// "libaudiohal@7.0",
],
shared_libs: [
diff --git a/media/libaudiohal/FactoryHalHidl.cpp b/media/libaudiohal/FactoryHalHidl.cpp
index 5985ef0..7228b22 100644
--- a/media/libaudiohal/FactoryHalHidl.cpp
+++ b/media/libaudiohal/FactoryHalHidl.cpp
@@ -31,6 +31,7 @@
/** Supported HAL versions, in order of preference.
*/
const char* sAudioHALVersions[] = {
+ "7.0",
"6.0",
"5.0",
"4.0",
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index 967fba1..df006b5 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -116,3 +116,20 @@
]
}
+cc_library_shared {
+ enabled: false,
+ name: "libaudiohal@7.0",
+ defaults: ["libaudiohal_default"],
+ shared_libs: [
+ "android.hardware.audio.common@7.0",
+ "android.hardware.audio.common@7.0-util",
+ "android.hardware.audio.effect@7.0",
+ "android.hardware.audio@7.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=7",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
+
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 9192a31..80e2b87 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -37,7 +37,7 @@
EffectsFactoryHalHidl::EffectsFactoryHalHidl(sp<IEffectsFactory> effectsFactory)
: ConversionHelperHidl("EffectsFactory") {
- ALOG_ASSERT(effectsFactory != nullptr, "Provided IDevicesFactory service is NULL");
+ ALOG_ASSERT(effectsFactory != nullptr, "Provided IEffectsFactory service is NULL");
mEffectsFactory = effectsFactory;
}
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index dece1bb..5fa85e7 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -54,6 +54,8 @@
virtual status_t dumpEffects(int fd);
+ virtual float getHalVersion() { return MAJOR_VERSION + (float)MINOR_VERSION / 10; }
+
status_t allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) override;
status_t mirrorBuffer(void* external, size_t size,
sp<EffectBufferHalInterface>* buffer) override;
diff --git a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
index 3a76f9f..9fb56ae 100644
--- a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
@@ -46,6 +46,8 @@
virtual status_t dumpEffects(int fd) = 0;
+ virtual float getHalVersion() = 0;
+
static sp<EffectsFactoryHalInterface> create();
virtual status_t allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) = 0;
diff --git a/media/libaudioprocessing/AudioResamplerDyn.cpp b/media/libaudioprocessing/AudioResamplerDyn.cpp
index 96d6104..1aacfd1 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.cpp
+++ b/media/libaudioprocessing/AudioResamplerDyn.cpp
@@ -25,7 +25,6 @@
#include <cutils/compiler.h>
#include <cutils/properties.h>
-#include <utils/Debug.h>
#include <utils/Log.h>
#include <audio_utils/primitives.h>
diff --git a/media/libeffects/lvm/tests/Android.bp b/media/libeffects/lvm/tests/Android.bp
index 674c246..aea7703 100644
--- a/media/libeffects/lvm/tests/Android.bp
+++ b/media/libeffects/lvm/tests/Android.bp
@@ -44,6 +44,36 @@
}
cc_test {
+ name: "reverb_test",
+ host_supported: false,
+ proprietary: true,
+
+ include_dirs: [
+ "frameworks/av/media/libeffects/lvm/wrapper/Reverb"
+ ],
+
+ header_libs: [
+ "libaudioeffects",
+ ],
+
+ shared_libs: [
+ "libaudioutils",
+ "liblog",
+ "libreverbwrapper",
+ ],
+
+ srcs: [
+ "reverb_test.cpp",
+ ],
+
+ cflags: [
+ "-Wall",
+ "-Werror",
+ "-Wextra",
+ ],
+}
+
+cc_test {
name: "snr",
host_supported: false,
diff --git a/media/libeffects/lvm/tests/build_and_run_all_unit_tests_reverb.sh b/media/libeffects/lvm/tests/build_and_run_all_unit_tests_reverb.sh
new file mode 100755
index 0000000..5a972db
--- /dev/null
+++ b/media/libeffects/lvm/tests/build_and_run_all_unit_tests_reverb.sh
@@ -0,0 +1,87 @@
+#!/bin/bash
+#
+# reverb test
+#
+
+if [ -z "$ANDROID_BUILD_TOP" ]; then
+ echo "Android build environment not set"
+ exit -1
+fi
+
+# ensure we have mm
+. $ANDROID_BUILD_TOP/build/envsetup.sh
+
+mm -j
+
+echo "waiting for device"
+
+adb root && adb wait-for-device remount
+
+# location of test files
+testdir="/data/local/tmp/revTest"
+
+echo "========================================"
+echo "testing reverb"
+adb shell mkdir -p $testdir
+adb push $ANDROID_BUILD_TOP/cts/tests/tests/media/res/raw/sinesweepraw.raw $testdir
+
+E_VAL=1
+cmds="adb push $OUT/testcases/reverb_test/arm/reverb_test $testdir"
+
+fs_arr=(
+ 8000
+ 16000
+ 22050
+ 32000
+ 44100
+ 48000
+ 88200
+ 96000
+ 176400
+ 192000
+)
+
+# run reverb at different configs, saving only the stereo channel
+# pair.
+error_count=0
+for cmd in "${cmds[@]}"
+do
+ $cmd
+ for preset_val in {0..6}
+ do
+ for fs in ${fs_arr[*]}
+ do
+ for chMask in {1..22}
+ do
+ adb shell LD_LIBRARY_PATH=/system/vendor/lib/soundfx $testdir/reverb_test \
+ --input $testdir/sinesweepraw.raw \
+ --output $testdir/sinesweep_$((chMask))_$((fs)).raw \
+ --chMask $chMask --fs $fs --preset $preset_val
+
+ shell_ret=$?
+ if [ $shell_ret -ne 0 ]; then
+ echo "error: $shell_ret"
+ ((++error_count))
+ fi
+
+ # two channel files should be identical to higher channel
+ # computation (first 2 channels).
+ if [[ "$chMask" -gt 1 ]]
+ then
+ adb shell cmp $testdir/sinesweep_1_$((fs)).raw \
+ $testdir/sinesweep_$((chMask))_$((fs)).raw
+ fi
+ # cmp returns EXIT_FAILURE on mismatch.
+ shell_ret=$?
+ if [ $shell_ret -ne 0 ]; then
+ echo "error: $shell_ret"
+ ((++error_count))
+ fi
+ done
+ done
+ done
+done
+
+adb shell rm -r $testdir
+echo "$error_count errors"
+exit $error_count
diff --git a/media/libeffects/lvm/tests/reverb_test.cpp b/media/libeffects/lvm/tests/reverb_test.cpp
new file mode 100644
index 0000000..a9cf348
--- /dev/null
+++ b/media/libeffects/lvm/tests/reverb_test.cpp
@@ -0,0 +1,395 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <assert.h>
+#include <getopt.h>
+#include <inttypes.h>
+#include <iterator>
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <vector>
+
+#include <audio_utils/channels.h>
+#include <audio_utils/primitives.h>
+#include <log/log.h>
+#include <system/audio.h>
+
+#include "EffectReverb.h"
+
+// This is the only symbol that needs to be exported
+extern audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM;
+
+// Global Variables
+enum ReverbParams {
+ ARG_HELP = 1,
+ ARG_INPUT,
+ ARG_OUTPUT,
+ ARG_FS,
+ ARG_CH_MASK,
+ ARG_PRESET,
+ ARG_AUX,
+ ARG_MONO_MODE,
+ ARG_FILE_CH,
+};
+
+const effect_uuid_t kReverbUuids[] = {
+ {0x172cdf00,
+ 0xa3bc,
+ 0x11df,
+ 0xa72f,
+ {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // preset-insert mode
+ {0xf29a1400,
+ 0xa3bb,
+ 0x11df,
+ 0x8ddc,
+ {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // preset-aux mode
+};
+
+// structures
+struct reverbConfigParams_t {
+ int fChannels = 2;
+ int monoMode = false;
+ int frameLength = 256;
+ int preset = 0;
+ int nrChannels = 2;
+ int sampleRate = 48000;
+ int auxiliary = 0;
+ audio_channel_mask_t chMask = AUDIO_CHANNEL_OUT_STEREO;
+};
+
+constexpr audio_channel_mask_t kReverbConfigChMask[] = {
+ AUDIO_CHANNEL_OUT_MONO,
+ AUDIO_CHANNEL_OUT_STEREO,
+ AUDIO_CHANNEL_OUT_2POINT1,
+ AUDIO_CHANNEL_OUT_2POINT0POINT2,
+ AUDIO_CHANNEL_OUT_QUAD,
+ AUDIO_CHANNEL_OUT_QUAD_BACK,
+ AUDIO_CHANNEL_OUT_QUAD_SIDE,
+ AUDIO_CHANNEL_OUT_SURROUND,
+ (1 << 4) - 1,
+ AUDIO_CHANNEL_OUT_2POINT1POINT2,
+ AUDIO_CHANNEL_OUT_3POINT0POINT2,
+ AUDIO_CHANNEL_OUT_PENTA,
+ (1 << 5) - 1,
+ AUDIO_CHANNEL_OUT_3POINT1POINT2,
+ AUDIO_CHANNEL_OUT_5POINT1,
+ AUDIO_CHANNEL_OUT_5POINT1_BACK,
+ AUDIO_CHANNEL_OUT_5POINT1_SIDE,
+ (1 << 6) - 1,
+ AUDIO_CHANNEL_OUT_6POINT1,
+ (1 << 7) - 1,
+ AUDIO_CHANNEL_OUT_5POINT1POINT2,
+ AUDIO_CHANNEL_OUT_7POINT1,
+ (1 << 8) - 1,
+};
+
+constexpr int kReverbConfigChMaskCount = std::size(kReverbConfigChMask);
+
+int reverbCreateEffect(effect_handle_t *pEffectHandle, effect_config_t *pConfig, int sessionId,
+ int ioId, int auxFlag) {
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(&kReverbUuids[auxFlag], sessionId,
+ ioId, pEffectHandle);
+ status != 0) {
+ ALOGE("Reverb create returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ (**pEffectHandle)
+ ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG, sizeof(effect_config_t), pConfig,
+ &replySize, &reply);
+ return reply;
+}
+
+int reverbSetConfigParam(uint32_t paramType, uint32_t paramValue, effect_handle_t effectHandle) {
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ uint32_t paramData[2] = {paramType, paramValue};
+ effect_param_t *effectParam =
+ (effect_param_t *)malloc(sizeof(*effectParam) + sizeof(paramData));
+ memcpy(&effectParam->data[0], ¶mData[0], sizeof(paramData));
+ effectParam->psize = sizeof(paramData[0]);
+ effectParam->vsize = sizeof(paramData[1]);
+ int status =
+ (*effectHandle)
+ ->command(effectHandle, EFFECT_CMD_SET_PARAM,
+ sizeof(effect_param_t) + sizeof(paramData), effectParam, &replySize, &reply);
+ free(effectParam);
+ if (status != 0) {
+ ALOGE("Reverb set config returned an error = %d\n", status);
+ return status;
+ }
+ return reply;
+}
+
+void printUsage() {
+ printf("\nUsage: ");
+ printf("\n <executable> [options]\n");
+ printf("\nwhere options are, ");
+ printf("\n --input <inputfile>");
+ printf("\n path to the input file");
+ printf("\n --output <outputfile>");
+ printf("\n path to the output file");
+ printf("\n --help");
+ printf("\n prints this usage information");
+ printf("\n --chMask <channel_mask>\n");
+ printf("\n 0 - AUDIO_CHANNEL_OUT_MONO");
+ printf("\n 1 - AUDIO_CHANNEL_OUT_STEREO");
+ printf("\n 2 - AUDIO_CHANNEL_OUT_2POINT1");
+ printf("\n 3 - AUDIO_CHANNEL_OUT_2POINT0POINT2");
+ printf("\n 4 - AUDIO_CHANNEL_OUT_QUAD");
+ printf("\n 5 - AUDIO_CHANNEL_OUT_QUAD_BACK");
+ printf("\n 6 - AUDIO_CHANNEL_OUT_QUAD_SIDE");
+ printf("\n 7 - AUDIO_CHANNEL_OUT_SURROUND");
+ printf("\n 8 - canonical channel index mask for 4 ch: (1 << 4) - 1");
+ printf("\n 9 - AUDIO_CHANNEL_OUT_2POINT1POINT2");
+ printf("\n 10 - AUDIO_CHANNEL_OUT_3POINT0POINT2");
+ printf("\n 11 - AUDIO_CHANNEL_OUT_PENTA");
+ printf("\n 12 - canonical channel index mask for 5 ch: (1 << 5) - 1");
+ printf("\n 13 - AUDIO_CHANNEL_OUT_3POINT1POINT2");
+ printf("\n 14 - AUDIO_CHANNEL_OUT_5POINT1");
+ printf("\n 15 - AUDIO_CHANNEL_OUT_5POINT1_BACK");
+ printf("\n 16 - AUDIO_CHANNEL_OUT_5POINT1_SIDE");
+ printf("\n 17 - canonical channel index mask for 6 ch: (1 << 6) - 1");
+ printf("\n 18 - AUDIO_CHANNEL_OUT_6POINT1");
+ printf("\n 19 - canonical channel index mask for 7 ch: (1 << 7) - 1");
+ printf("\n 20 - AUDIO_CHANNEL_OUT_5POINT1POINT2");
+ printf("\n 21 - AUDIO_CHANNEL_OUT_7POINT1");
+ printf("\n 22 - canonical channel index mask for 8 ch: (1 << 8) - 1");
+ printf("\n default 0");
+ printf("\n --fs <sampling_freq>");
+ printf("\n Sampling frequency in Hz, default 48000.");
+ printf("\n --preset <preset_value>");
+ printf("\n 0 - None");
+ printf("\n 1 - Small Room");
+ printf("\n 2 - Medium Room");
+ printf("\n 3 - Large Room");
+ printf("\n 4 - Medium Hall");
+ printf("\n 5 - Large Hall");
+ printf("\n 6 - Plate");
+ printf("\n default 0");
+ printf("\n --fch <file_channels>");
+ printf("\n number of channels in input file (1 through 8), default 1");
+ printf("\n --M");
+ printf("\n Mono mode (force all input audio channels to be identical)");
+ printf("\n --aux <auxiliary_flag> ");
+ printf("\n 0 - Insert Mode on");
+ printf("\n 1 - auxiliary Mode on");
+ printf("\n default 0");
+ printf("\n");
+}
+
+int main(int argc, const char *argv[]) {
+ if (argc == 1) {
+ printUsage();
+ return EXIT_FAILURE;
+ }
+
+ reverbConfigParams_t revConfigParams{}; // default initialize
+ const char *inputFile = nullptr;
+ const char *outputFile = nullptr;
+
+ const option long_opts[] = {
+ {"help", no_argument, nullptr, ARG_HELP},
+ {"input", required_argument, nullptr, ARG_INPUT},
+ {"output", required_argument, nullptr, ARG_OUTPUT},
+ {"fs", required_argument, nullptr, ARG_FS},
+ {"chMask", required_argument, nullptr, ARG_CH_MASK},
+ {"preset", required_argument, nullptr, ARG_PRESET},
+ {"aux", required_argument, nullptr, ARG_AUX},
+ {"M", no_argument, &revConfigParams.monoMode, true},
+ {"fch", required_argument, nullptr, ARG_FILE_CH},
+ {nullptr, 0, nullptr, 0},
+ };
+
+ while (true) {
+ const int opt = getopt_long(argc, (char *const *)argv, "i:o:", long_opts, nullptr);
+ if (opt == -1) {
+ break;
+ }
+ switch (opt) {
+ case ARG_HELP:
+ printUsage();
+ return EXIT_SUCCESS;
+ case ARG_INPUT: {
+ inputFile = (char *)optarg;
+ break;
+ }
+ case ARG_OUTPUT: {
+ outputFile = (char *)optarg;
+ break;
+ }
+ case ARG_FS: {
+ revConfigParams.sampleRate = atoi(optarg);
+ break;
+ }
+ case ARG_CH_MASK: {
+ int chMaskIdx = atoi(optarg);
+ if (chMaskIdx < 0 or chMaskIdx > kReverbConfigChMaskCount) {
+ ALOGE("Channel Mask index not in correct range\n");
+ printUsage();
+ return EXIT_FAILURE;
+ }
+ revConfigParams.chMask = kReverbConfigChMask[chMaskIdx];
+ break;
+ }
+ case ARG_PRESET: {
+ revConfigParams.preset = atoi(optarg);
+ break;
+ }
+ case ARG_AUX: {
+ revConfigParams.auxiliary = atoi(optarg);
+ break;
+ }
+ case ARG_MONO_MODE: {
+ break;
+ }
+ case ARG_FILE_CH: {
+ revConfigParams.fChannels = atoi(optarg);
+ break;
+ }
+ default:
+ break;
+ }
+ }
+
+ if (inputFile == nullptr) {
+ ALOGE("Error: missing input files\n");
+ printUsage();
+ return EXIT_FAILURE;
+ }
+ std::unique_ptr<FILE, decltype(&fclose)> inputFp(fopen(inputFile, "rb"), &fclose);
+
+ if (inputFp == nullptr) {
+ ALOGE("Cannot open input file %s\n", inputFile);
+ return EXIT_FAILURE;
+ }
+
+ if (outputFile == nullptr) {
+ ALOGE("Error: missing output files\n");
+ printUsage();
+ return EXIT_FAILURE;
+ }
+ std::unique_ptr<FILE, decltype(&fclose)> outputFp(fopen(outputFile, "wb"), &fclose);
+
+ if (outputFp == nullptr) {
+ ALOGE("Cannot open output file %s\n", outputFile);
+ return EXIT_FAILURE;
+ }
+
+ int32_t sessionId = 1;
+ int32_t ioId = 1;
+ effect_handle_t effectHandle = nullptr;
+ effect_config_t config;
+ config.inputCfg.samplingRate = config.outputCfg.samplingRate = revConfigParams.sampleRate;
+ config.inputCfg.channels = config.outputCfg.channels = revConfigParams.chMask;
+ config.inputCfg.format = config.outputCfg.format = AUDIO_FORMAT_PCM_FLOAT;
+ if (int status =
+ reverbCreateEffect(&effectHandle, &config, sessionId, ioId, revConfigParams.auxiliary);
+ status != 0) {
+ ALOGE("Create effect call returned error %i", status);
+ return EXIT_FAILURE;
+ }
+
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ (*effectHandle)->command(effectHandle, EFFECT_CMD_ENABLE, 0, nullptr, &replySize, &reply);
+ if (reply != 0) {
+ ALOGE("Command enable call returned error %d\n", reply);
+ return EXIT_FAILURE;
+ }
+
+ if (int status = reverbSetConfigParam(REVERB_PARAM_PRESET, (uint32_t)revConfigParams.preset,
+ effectHandle);
+ status != 0) {
+ ALOGE("Invalid reverb preset. Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+
+ revConfigParams.nrChannels = audio_channel_count_from_out_mask(revConfigParams.chMask);
+ const int channelCount = revConfigParams.nrChannels;
+ const int frameLength = revConfigParams.frameLength;
+#ifdef BYPASS_EXEC
+ const int frameSize = (int)channelCount * sizeof(float);
+#endif
+ const int ioChannelCount = revConfigParams.fChannels;
+ const int ioFrameSize = ioChannelCount * sizeof(short);
+ const int maxChannelCount = std::max(channelCount, ioChannelCount);
+ /*
+ * Mono input will be converted to 2 channels internally in the process call
+ * by copying the same data into the second channel.
+ * Hence when channelCount is 1, output buffer should be allocated for
+ * 2 channels. The memAllocChCount takes care of allocation of sufficient
+ * memory for the output buffer.
+ */
+ const int memAllocChCount = (channelCount == 1 ? 2 : channelCount);
+
+ std::vector<short> in(frameLength * maxChannelCount);
+ std::vector<short> out(frameLength * maxChannelCount);
+ std::vector<float> floatIn(frameLength * channelCount);
+ std::vector<float> floatOut(frameLength * memAllocChCount);
+
+ int frameCounter = 0;
+
+ while (fread(in.data(), ioFrameSize, frameLength, inputFp.get()) == (size_t)frameLength) {
+ if (ioChannelCount != channelCount) {
+ adjust_channels(in.data(), ioChannelCount, in.data(), channelCount, sizeof(short),
+ frameLength * ioFrameSize);
+ }
+ memcpy_to_float_from_i16(floatIn.data(), in.data(), frameLength * channelCount);
+
+ // Mono mode will replicate the first channel to all other channels.
+ // This ensures all audio channels are identical. This is useful for testing
+ // Bass Boost, which extracts a mono signal for processing.
+ if (revConfigParams.monoMode && channelCount > 1) {
+ for (int i = 0; i < frameLength; ++i) {
+ auto *fp = &floatIn[i * channelCount];
+ std::fill(fp + 1, fp + channelCount, *fp); // replicate ch 0
+ }
+ }
+
+ audio_buffer_t inputBuffer, outputBuffer;
+ inputBuffer.frameCount = outputBuffer.frameCount = frameLength;
+ inputBuffer.f32 = floatIn.data();
+ outputBuffer.f32 = floatOut.data();
+#ifndef BYPASS_EXEC
+ if (int status = (*effectHandle)->process(effectHandle, &inputBuffer, &outputBuffer);
+ status != 0) {
+ ALOGE("\nError: Process returned with error %d\n", status);
+ return EXIT_FAILURE;
+ }
+#else
+ memcpy(floatOut.data(), floatIn.data(), frameLength * frameSize);
+#endif
+ memcpy_to_i16_from_float(out.data(), floatOut.data(), frameLength * channelCount);
+
+ if (ioChannelCount != channelCount) {
+ adjust_channels(out.data(), channelCount, out.data(), ioChannelCount, sizeof(short),
+ frameLength * channelCount * sizeof(short));
+ }
+ (void)fwrite(out.data(), ioFrameSize, frameLength, outputFp.get());
+ frameCounter += frameLength;
+ }
+
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle);
+ status != 0) {
+ ALOGE("Audio Preprocessing release returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ printf("frameCounter: [%d]\n", frameCounter);
+
+ return EXIT_SUCCESS;
+}
diff --git a/media/libeffects/preprocessing/tests/Android.bp b/media/libeffects/preprocessing/tests/Android.bp
new file mode 100644
index 0000000..71f6e8f
--- /dev/null
+++ b/media/libeffects/preprocessing/tests/Android.bp
@@ -0,0 +1,30 @@
+// audio preprocessing unit test
+cc_test {
+ name: "AudioPreProcessingTest",
+
+ vendor: true,
+
+ relative_install_path: "soundfx",
+
+ srcs: ["PreProcessingTest.cpp"],
+
+ shared_libs: [
+ "libaudiopreprocessing",
+ "libaudioutils",
+ "liblog",
+ "libutils",
+ "libwebrtc_audio_preprocessing",
+ ],
+
+ cflags: [
+ "-DWEBRTC_POSIX",
+ "-fvisibility=default",
+ "-Wall",
+ "-Werror",
+ ],
+
+ header_libs: [
+ "libaudioeffects",
+ "libhardware_headers",
+ ],
+}
diff --git a/media/libeffects/preprocessing/tests/PreProcessingTest.cpp b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
new file mode 100644
index 0000000..5c81d78
--- /dev/null
+++ b/media/libeffects/preprocessing/tests/PreProcessingTest.cpp
@@ -0,0 +1,429 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <audio_effects/effect_aec.h>
+#include <audio_effects/effect_agc.h>
+#include <audio_effects/effect_ns.h>
+#include <audio_processing.h>
+#include <getopt.h>
+#include <hardware/audio_effect.h>
+#include <module_common_types.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/stat.h>
+#include <utils/Log.h>
+#include <utils/Timers.h>
+
+#include <audio_utils/channels.h>
+#include <audio_utils/primitives.h>
+#include <log/log.h>
+#include <system/audio.h>
+
+// This is the only symbol that needs to be imported
+extern audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM;
+
+//------------------------------------------------------------------------------
+// local definitions
+//------------------------------------------------------------------------------
+
+// types of pre processing modules
+enum PreProcId {
+ PREPROC_AGC, // Automatic Gain Control
+ PREPROC_AEC, // Acoustic Echo Canceler
+ PREPROC_NS, // Noise Suppressor
+ PREPROC_NUM_EFFECTS
+};
+
+enum PreProcParams {
+ ARG_HELP = 1,
+ ARG_INPUT,
+ ARG_OUTPUT,
+ ARG_FAR,
+ ARG_FS,
+ ARG_CH_MASK,
+ ARG_AGC_TGT_LVL,
+ ARG_AGC_COMP_LVL,
+ ARG_AEC_DELAY,
+ ARG_NS_LVL,
+};
+
+struct preProcConfigParams_t {
+ int samplingFreq = 16000;
+ audio_channel_mask_t chMask = AUDIO_CHANNEL_IN_MONO;
+ int nsLevel = 0; // a value between 0-3
+ int agcTargetLevel = 3; // in dB
+ int agcCompLevel = 9; // in dB
+ int aecDelay = 0; // in ms
+};
+
+const effect_uuid_t kPreProcUuids[PREPROC_NUM_EFFECTS] = {
+ {0xaa8130e0, 0x66fc, 0x11e0, 0xbad0, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // agc uuid
+ {0xbb392ec0, 0x8d4d, 0x11e0, 0xa896, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // aec uuid
+ {0xc06c8400, 0x8e06, 0x11e0, 0x9cb6, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, // ns uuid
+};
+
+constexpr audio_channel_mask_t kPreProcConfigChMask[] = {
+ AUDIO_CHANNEL_IN_MONO,
+ AUDIO_CHANNEL_IN_STEREO,
+ AUDIO_CHANNEL_IN_FRONT_BACK,
+ AUDIO_CHANNEL_IN_6,
+ AUDIO_CHANNEL_IN_2POINT0POINT2,
+ AUDIO_CHANNEL_IN_2POINT1POINT2,
+ AUDIO_CHANNEL_IN_3POINT0POINT2,
+ AUDIO_CHANNEL_IN_3POINT1POINT2,
+ AUDIO_CHANNEL_IN_5POINT1,
+ AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO,
+ AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO,
+ AUDIO_CHANNEL_IN_VOICE_CALL_MONO,
+};
+
+constexpr int kPreProcConfigChMaskCount = std::size(kPreProcConfigChMask);
+
+void printUsage() {
+ printf("\nUsage: ");
+ printf("\n <executable> [options]\n");
+ printf("\nwhere options are, ");
+ printf("\n --input <inputfile>");
+ printf("\n path to the input file");
+ printf("\n --output <outputfile>");
+ printf("\n path to the output file");
+ printf("\n --help");
+ printf("\n Prints this usage information");
+ printf("\n --fs <sampling_freq>");
+ printf("\n Sampling frequency in Hz, default 16000.");
+ printf("\n -ch_mask <channel_mask>\n");
+ printf("\n 0 - AUDIO_CHANNEL_IN_MONO");
+ printf("\n 1 - AUDIO_CHANNEL_IN_STEREO");
+ printf("\n 2 - AUDIO_CHANNEL_IN_FRONT_BACK");
+ printf("\n 3 - AUDIO_CHANNEL_IN_6");
+ printf("\n 4 - AUDIO_CHANNEL_IN_2POINT0POINT2");
+ printf("\n 5 - AUDIO_CHANNEL_IN_2POINT1POINT2");
+ printf("\n 6 - AUDIO_CHANNEL_IN_3POINT0POINT2");
+ printf("\n 7 - AUDIO_CHANNEL_IN_3POINT1POINT2");
+ printf("\n 8 - AUDIO_CHANNEL_IN_5POINT1");
+ printf("\n 9 - AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO");
+ printf("\n 10 - AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO ");
+ printf("\n 11 - AUDIO_CHANNEL_IN_VOICE_CALL_MONO ");
+ printf("\n default 0");
+ printf("\n --far <farend_file>");
+ printf("\n Path to far-end file needed for echo cancellation");
+ printf("\n --aec");
+ printf("\n Enable Echo Cancellation, default disabled");
+ printf("\n --ns");
+ printf("\n Enable Noise Suppression, default disabled");
+ printf("\n --agc");
+ printf("\n Enable Gain Control, default disabled");
+ printf("\n --ns_lvl <ns_level>");
+ printf("\n Noise Suppression level in dB, default value 0dB");
+ printf("\n --agc_tgt_lvl <target_level>");
+ printf("\n AGC Target Level in dB, default value 3dB");
+ printf("\n --agc_comp_lvl <comp_level>");
+ printf("\n AGC Comp Level in dB, default value 9dB");
+ printf("\n --aec_delay <delay>");
+ printf("\n AEC delay value in ms, default value 0ms");
+ printf("\n");
+}
+
+constexpr float kTenMilliSecVal = 0.01;
+
+int preProcCreateEffect(effect_handle_t *pEffectHandle, uint32_t effectType,
+ effect_config_t *pConfig, int sessionId, int ioId) {
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.create_effect(&kPreProcUuids[effectType],
+ sessionId, ioId, pEffectHandle);
+ status != 0) {
+ ALOGE("Audio Preprocessing create returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ if (effectType == PREPROC_AEC) {
+ (**pEffectHandle)
+ ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG_REVERSE, sizeof(effect_config_t), pConfig,
+ &replySize, &reply);
+ }
+ (**pEffectHandle)
+ ->command(*pEffectHandle, EFFECT_CMD_SET_CONFIG, sizeof(effect_config_t), pConfig,
+ &replySize, &reply);
+ return reply;
+}
+
+int preProcSetConfigParam(uint32_t paramType, uint32_t paramValue, effect_handle_t effectHandle) {
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ uint32_t paramData[2] = {paramType, paramValue};
+ effect_param_t *effectParam =
+ (effect_param_t *)malloc(sizeof(*effectParam) + sizeof(paramData));
+ memcpy(&effectParam->data[0], ¶mData[0], sizeof(paramData));
+ effectParam->psize = sizeof(paramData[0]);
+ (*effectHandle)
+ ->command(effectHandle, EFFECT_CMD_SET_PARAM, sizeof(effect_param_t), effectParam,
+ &replySize, &reply);
+ free(effectParam);
+ return reply;
+}
+
+int main(int argc, const char *argv[]) {
+ if (argc == 1) {
+ printUsage();
+ return EXIT_FAILURE;
+ }
+ const char *inputFile = nullptr;
+ const char *outputFile = nullptr;
+ const char *farFile = nullptr;
+ int effectEn[PREPROC_NUM_EFFECTS] = {0};
+
+ const option long_opts[] = {
+ {"help", no_argument, nullptr, ARG_HELP},
+ {"input", required_argument, nullptr, ARG_INPUT},
+ {"output", required_argument, nullptr, ARG_OUTPUT},
+ {"far", required_argument, nullptr, ARG_FAR},
+ {"fs", required_argument, nullptr, ARG_FS},
+ {"ch_mask", required_argument, nullptr, ARG_CH_MASK},
+ {"agc_tgt_lvl", required_argument, nullptr, ARG_AGC_TGT_LVL},
+ {"agc_comp_lvl", required_argument, nullptr, ARG_AGC_COMP_LVL},
+ {"aec_delay", required_argument, nullptr, ARG_AEC_DELAY},
+ {"ns_lvl", required_argument, nullptr, ARG_NS_LVL},
+ {"aec", no_argument, &effectEn[PREPROC_AEC], 1},
+ {"agc", no_argument, &effectEn[PREPROC_AGC], 1},
+ {"ns", no_argument, &effectEn[PREPROC_NS], 1},
+ {nullptr, 0, nullptr, 0},
+ };
+ struct preProcConfigParams_t preProcCfgParams {};
+
+ while (true) {
+ const int opt = getopt_long(argc, (char *const *)argv, "i:o:", long_opts, nullptr);
+ if (opt == -1) {
+ break;
+ }
+ switch (opt) {
+ case ARG_HELP:
+ printUsage();
+ return 0;
+ case ARG_INPUT: {
+ inputFile = (char *)optarg;
+ break;
+ }
+ case ARG_OUTPUT: {
+ outputFile = (char *)optarg;
+ break;
+ }
+ case ARG_FAR: {
+ farFile = (char *)optarg;
+ break;
+ }
+ case ARG_FS: {
+ preProcCfgParams.samplingFreq = atoi(optarg);
+ break;
+ }
+ case ARG_CH_MASK: {
+ int chMaskIdx = atoi(optarg);
+ if (chMaskIdx < 0 or chMaskIdx > kPreProcConfigChMaskCount) {
+ ALOGE("Channel Mask index not in correct range\n");
+ printUsage();
+ return EXIT_FAILURE;
+ }
+ preProcCfgParams.chMask = kPreProcConfigChMask[chMaskIdx];
+ break;
+ }
+ case ARG_AGC_TGT_LVL: {
+ preProcCfgParams.agcTargetLevel = atoi(optarg);
+ break;
+ }
+ case ARG_AGC_COMP_LVL: {
+ preProcCfgParams.agcCompLevel = atoi(optarg);
+ break;
+ }
+ case ARG_AEC_DELAY: {
+ preProcCfgParams.aecDelay = atoi(optarg);
+ break;
+ }
+ case ARG_NS_LVL: {
+ preProcCfgParams.nsLevel = atoi(optarg);
+ break;
+ }
+ default:
+ break;
+ }
+ }
+
+ if (inputFile == nullptr) {
+ ALOGE("Error: missing input file\n");
+ printUsage();
+ return EXIT_FAILURE;
+ }
+
+ std::unique_ptr<FILE, decltype(&fclose)> inputFp(fopen(inputFile, "rb"), &fclose);
+ if (inputFp == nullptr) {
+ ALOGE("Cannot open input file %s\n", inputFile);
+ return EXIT_FAILURE;
+ }
+
+ std::unique_ptr<FILE, decltype(&fclose)> farFp(fopen(farFile, "rb"), &fclose);
+ std::unique_ptr<FILE, decltype(&fclose)> outputFp(fopen(outputFile, "wb"), &fclose);
+ if (effectEn[PREPROC_AEC]) {
+ if (farFile == nullptr) {
+ ALOGE("Far end signal file required for echo cancellation \n");
+ return EXIT_FAILURE;
+ }
+ if (farFp == nullptr) {
+ ALOGE("Cannot open far end stream file %s\n", farFile);
+ return EXIT_FAILURE;
+ }
+ struct stat statInput, statFar;
+ (void)fstat(fileno(inputFp.get()), &statInput);
+ (void)fstat(fileno(farFp.get()), &statFar);
+ if (statInput.st_size != statFar.st_size) {
+ ALOGE("Near and far end signals are of different sizes");
+ return EXIT_FAILURE;
+ }
+ }
+ if (outputFile != nullptr && outputFp == nullptr) {
+ ALOGE("Cannot open output file %s\n", outputFile);
+ return EXIT_FAILURE;
+ }
+
+ int32_t sessionId = 1;
+ int32_t ioId = 1;
+ effect_handle_t effectHandle[PREPROC_NUM_EFFECTS] = {nullptr};
+ effect_config_t config;
+ config.inputCfg.samplingRate = config.outputCfg.samplingRate = preProcCfgParams.samplingFreq;
+ config.inputCfg.channels = config.outputCfg.channels = preProcCfgParams.chMask;
+ config.inputCfg.format = config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+
+ // Create all the effect handles
+ for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
+ if (int status = preProcCreateEffect(&effectHandle[i], i, &config, sessionId, ioId);
+ status != 0) {
+ ALOGE("Create effect call returned error %i", status);
+ return EXIT_FAILURE;
+ }
+ }
+
+ for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
+ if (effectEn[i] == 1) {
+ int reply = 0;
+ uint32_t replySize = sizeof(reply);
+ (*effectHandle[i])
+ ->command(effectHandle[i], EFFECT_CMD_ENABLE, 0, nullptr, &replySize, &reply);
+ if (reply != 0) {
+ ALOGE("Command enable call returned error %d\n", reply);
+ return EXIT_FAILURE;
+ }
+ }
+ }
+
+ // Set Config Params of the effects
+ if (effectEn[PREPROC_AGC]) {
+ if (int status = preProcSetConfigParam(AGC_PARAM_TARGET_LEVEL,
+ (uint32_t)preProcCfgParams.agcTargetLevel,
+ effectHandle[PREPROC_AGC]);
+ status != 0) {
+ ALOGE("Invalid AGC Target Level. Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+ if (int status =
+ preProcSetConfigParam(AGC_PARAM_COMP_GAIN, (uint32_t)preProcCfgParams.agcCompLevel,
+ effectHandle[PREPROC_AGC]);
+ status != 0) {
+ ALOGE("Invalid AGC Comp Gain. Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+ }
+ if (effectEn[PREPROC_NS]) {
+ if (int status = preProcSetConfigParam(NS_PARAM_LEVEL, (uint32_t)preProcCfgParams.nsLevel,
+ effectHandle[PREPROC_NS]);
+ status != 0) {
+ ALOGE("Invalid Noise Suppression level Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+ }
+
+ // Process Call
+ const int frameLength = (int)(preProcCfgParams.samplingFreq * kTenMilliSecVal);
+ const int ioChannelCount = audio_channel_count_from_in_mask(preProcCfgParams.chMask);
+ const int ioFrameSize = ioChannelCount * sizeof(short);
+ int frameCounter = 0;
+ while (true) {
+ std::vector<short> in(frameLength * ioChannelCount);
+ std::vector<short> out(frameLength * ioChannelCount);
+ std::vector<short> farIn(frameLength * ioChannelCount);
+ size_t samplesRead = fread(in.data(), ioFrameSize, frameLength, inputFp.get());
+ if (samplesRead == 0) {
+ break;
+ }
+ audio_buffer_t inputBuffer, outputBuffer;
+ audio_buffer_t farInBuffer{};
+ inputBuffer.frameCount = samplesRead;
+ outputBuffer.frameCount = samplesRead;
+ inputBuffer.s16 = in.data();
+ outputBuffer.s16 = out.data();
+
+ if (farFp != nullptr) {
+ samplesRead = fread(farIn.data(), ioFrameSize, frameLength, farFp.get());
+ if (samplesRead == 0) {
+ break;
+ }
+ farInBuffer.frameCount = samplesRead;
+ farInBuffer.s16 = farIn.data();
+ }
+
+ for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
+ if (effectEn[i] == 1) {
+ if (i == PREPROC_AEC) {
+ if (int status =
+ preProcSetConfigParam(AEC_PARAM_ECHO_DELAY, (uint32_t)preProcCfgParams.aecDelay,
+ effectHandle[PREPROC_AEC]);
+ status != 0) {
+ ALOGE("preProcSetConfigParam returned Error %d\n", status);
+ return EXIT_FAILURE;
+ }
+ }
+ if (int status =
+ (*effectHandle[i])->process(effectHandle[i], &inputBuffer, &outputBuffer);
+ status != 0) {
+ ALOGE("\nError: Process i = %d returned with error %d\n", i, status);
+ return EXIT_FAILURE;
+ }
+ if (i == PREPROC_AEC) {
+ if (int status = (*effectHandle[i])
+ ->process_reverse(effectHandle[i], &farInBuffer, &outputBuffer);
+ status != 0) {
+ ALOGE("\nError: Process reverse i = %d returned with error %d\n", i, status);
+ return EXIT_FAILURE;
+ }
+ }
+ }
+ }
+ if (outputFp != nullptr) {
+ size_t samplesWritten =
+ fwrite(out.data(), ioFrameSize, outputBuffer.frameCount, outputFp.get());
+ if (samplesWritten != outputBuffer.frameCount) {
+ ALOGE("\nError: Output file writing failed");
+ break;
+ }
+ }
+ frameCounter += frameLength;
+ }
+ // Release all the effect handles created
+ for (int i = 0; i < PREPROC_NUM_EFFECTS; i++) {
+ if (int status = AUDIO_EFFECT_LIBRARY_INFO_SYM.release_effect(effectHandle[i]);
+ status != 0) {
+ ALOGE("Audio Preprocessing release returned an error = %d\n", status);
+ return EXIT_FAILURE;
+ }
+ }
+ return EXIT_SUCCESS;
+}
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 637322f..1be82d8 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -63,7 +63,7 @@
searchDirs[1] + fileName,
searchDirs[2] + fileName,
searchDirs[3] + fileName,
- "system/etc/media_profiles_V1_0.xml" // System fallback
+ "system/etc/media_profiles.xml" // System fallback
};
}();
static std::array<char const*, 5> const cPaths = {
diff --git a/media/libmedia/include/media/mediaplayer.h b/media/libmedia/include/media/mediaplayer.h
index d0a8e38..0073375 100644
--- a/media/libmedia/include/media/mediaplayer.h
+++ b/media/libmedia/include/media/mediaplayer.h
@@ -178,7 +178,10 @@
KEY_PARAMETER_PLAYBACK_RATE_PERMILLE = 1300, // set only
// Set a Parcel containing the value of a parcelled Java AudioAttribute instance
- KEY_PARAMETER_AUDIO_ATTRIBUTES = 1400 // set only
+ KEY_PARAMETER_AUDIO_ATTRIBUTES = 1400, // set only
+
+ // Set a Parcel containing the values of RTP attribute
+ KEY_PARAMETER_RTP_ATTRIBUTES = 2000 // set only
};
// Keep INVOKE_ID_* in sync with MediaPlayer.java.
diff --git a/media/libmediahelper/TypeConverter.cpp b/media/libmediahelper/TypeConverter.cpp
index 6382ce4..705959a 100644
--- a/media/libmediahelper/TypeConverter.cpp
+++ b/media/libmediahelper/TypeConverter.cpp
@@ -57,6 +57,8 @@
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_USB_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_HEARING_AID),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_ECHO_CANCELLER),
+ MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLE_HEADSET),
+ MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_BLE_SPEAKER),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_DEFAULT),
// STUB must be after DEFAULT, so the latter is picked up by toString first.
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_OUT_STUB),
@@ -96,6 +98,7 @@
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_USB_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_BLE),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_ECHO_REFERENCE),
+ MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_BLE_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_DEFAULT),
// STUB must be after DEFAULT, so the latter is picked up by toString first.
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_STUB),
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 7e5fe56..02ae456 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -17,6 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "StagefrightRecorder"
#include <inttypes.h>
+// TODO/workaround: including base logging now as it conflicts with ADebug.h
+// and it must be included first.
+#include <android-base/logging.h>
#include <utils/Log.h>
#include "WebmWriter.h"
@@ -575,12 +578,14 @@
mVideoBitRate = bitRate;
// A new bitrate(TMMBR) should be applied on runtime as well if OutputFormat is RTP_AVP
- if (mOutputFormat == OUTPUT_FORMAT_RTP_AVP && mStarted && mPauseStartTimeUs == 0) {
+ if (mOutputFormat == OUTPUT_FORMAT_RTP_AVP) {
// Regular I frames may overload the network so we reduce the bitrate to allow
// margins for the I frame overruns.
// Still send requested bitrate (TMMBR) in the reply (TMMBN).
const float coefficient = 0.8f;
mVideoBitRate = (bitRate * coefficient) / 1000 * 1000;
+ }
+ if (mOutputFormat == OUTPUT_FORMAT_RTP_AVP && mStarted && mPauseStartTimeUs == 0) {
mVideoEncoderSource->setEncodingBitrate(mVideoBitRate);
ARTPWriter* rtpWriter = static_cast<ARTPWriter*>(mWriter.get());
rtpWriter->setTMMBNInfo(mOpponentID, bitRate);
@@ -1967,10 +1972,6 @@
format->setInt32("stride", stride);
format->setInt32("slice-height", sliceHeight);
format->setInt32("color-format", colorFormat);
- if (mOutputFormat == OUTPUT_FORMAT_RTP_AVP) {
- // This indicates that a raw image provided to encoder needs to be rotated.
- format->setInt32("rotation-degrees", mRotationDegrees);
- }
} else {
format->setInt32("width", mVideoWidth);
format->setInt32("height", mVideoHeight);
@@ -1988,6 +1989,11 @@
}
}
+ if (mOutputFormat == OUTPUT_FORMAT_RTP_AVP) {
+ // This indicates that a raw image provided to encoder needs to be rotated.
+ format->setInt32("rotation-degrees", mRotationDegrees);
+ }
+
format->setInt32("bitrate", mVideoBitRate);
format->setInt32("bitrate-mode", mVideoBitRateMode);
format->setInt32("frame-rate", mFrameRate);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 4e7daa5..47362ef 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1702,6 +1702,12 @@
updateRebufferingTimer(false /* stopping */, false /* exiting */);
}
+void NuPlayer::setTargetBitrate(int bitrate) {
+ if (mSource != NULL) {
+ mSource->setTargetBitrate(bitrate);
+ }
+}
+
void NuPlayer::onPause() {
updatePlaybackTimer(true /* stopping */, "onPause");
@@ -2868,6 +2874,27 @@
}
break;
}
+ case NuPlayer::RTPSource::RTP_QUALITY:
+ {
+ int32_t feedbackType, bitrate;
+ int32_t highestSeqNum, baseSeqNum, prevExpected;
+ int32_t numBufRecv, prevNumBufRecv;
+ CHECK(msg->findInt32("feedback-type", &feedbackType));
+ CHECK(msg->findInt32("bit-rate", &bitrate));
+ CHECK(msg->findInt32("highest-seq-num", &highestSeqNum));
+ CHECK(msg->findInt32("base-seq-num", &baseSeqNum));
+ CHECK(msg->findInt32("prev-expected", &prevExpected));
+ CHECK(msg->findInt32("num-buf-recv", &numBufRecv));
+ CHECK(msg->findInt32("prev-num-buf-recv", &prevNumBufRecv));
+ in.writeInt32(feedbackType);
+ in.writeInt32(bitrate);
+ in.writeInt32(highestSeqNum);
+ in.writeInt32(baseSeqNum);
+ in.writeInt32(prevExpected);
+ in.writeInt32(numBufRecv);
+ in.writeInt32(prevNumBufRecv);
+ break;
+ }
case NuPlayer::RTPSource::RTP_CVO:
{
int32_t cvo;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 0105248..adb7075 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -102,6 +102,8 @@
void updateInternalTimers();
+ void setTargetBitrate(int bitrate /* bps */);
+
protected:
virtual ~NuPlayer();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 2d82944..2a50fc2 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -817,7 +817,11 @@
}
status_t NuPlayerDriver::setParameter(
- int /* key */, const Parcel & /* request */) {
+ int key, const Parcel &request ) {
+ if (key == KEY_PARAMETER_RTP_ATTRIBUTES) {
+ mPlayer->setTargetBitrate(request.readInt32());
+ return OK;
+ }
return INVALID_OPERATION;
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
index eb39870..bf6b539 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
@@ -132,6 +132,8 @@
virtual void setOffloadAudio(bool /* offload */) {}
+ virtual void setTargetBitrate(int32_t) {}
+
// Modular DRM
virtual status_t prepareDrm(
const uint8_t /*uuid*/[16], const Vector<uint8_t> &/*drmSessionId*/,
diff --git a/media/libmediaplayerservice/nuplayer/RTPSource.cpp b/media/libmediaplayerservice/nuplayer/RTPSource.cpp
index a6601cd..b1901e8 100644
--- a/media/libmediaplayerservice/nuplayer/RTPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTPSource.cpp
@@ -114,7 +114,8 @@
// index(i) should be started from 1. 0 is reserved for [root]
mRTPConn->addStream(sockRtp, sockRtcp, desc, i + 1, notify, false);
mRTPConn->setSelfID(info->mSelfID);
- mRTPConn->setMinMaxBitrate(kMinVideoBitrate, info->mAS * 1000 /* kbps */);
+ mRTPConn->setJbTime(
+ (info->mJbTimeMs <= 3000 && info->mJbTimeMs >= 40) ? info->mJbTimeMs : 300);
info->mRTPSocket = sockRtp;
info->mRTCPSocket = sockRtcp;
@@ -135,11 +136,16 @@
if (info->mIsAudio) {
mAudioTrack = source;
+ info->mTimeScale = 16000;
} else {
mVideoTrack = source;
+ info->mTimeScale = 90000;
}
info->mSource = source;
+ info->mRTPTime = 0;
+ info->mNormalPlaytimeUs = 0;
+ info->mNPTMappingValid = false;
}
if (mInPreparationPhase) {
@@ -280,20 +286,19 @@
}
int32_t cvo;
- if ((*accessUnit) != NULL && (*accessUnit)->meta()->findInt32("cvo", &cvo)) {
- if (cvo != mLastCVOUpdated) {
- sp<AMessage> msg = new AMessage();
- msg->setInt32("payload-type", NuPlayer::RTPSource::RTP_CVO);
- msg->setInt32("cvo", cvo);
+ if ((*accessUnit) != NULL && (*accessUnit)->meta()->findInt32("cvo", &cvo) &&
+ cvo != mLastCVOUpdated) {
+ sp<AMessage> msg = new AMessage();
+ msg->setInt32("payload-type", NuPlayer::RTPSource::RTP_CVO);
+ msg->setInt32("cvo", cvo);
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatIMSRxNotice);
- notify->setMessage("message", msg);
- notify->post();
+ sp<AMessage> notify = dupNotify();
+ notify->setInt32("what", kWhatIMSRxNotice);
+ notify->setMessage("message", msg);
+ notify->post();
- ALOGV("notify cvo updated (%d)->(%d) to upper layer", mLastCVOUpdated, cvo);
- mLastCVOUpdated = cvo;
- }
+ ALOGV("notify cvo updated (%d)->(%d) to upper layer", mLastCVOUpdated, cvo);
+ mLastCVOUpdated = cvo;
}
return finalResult;
@@ -347,6 +352,11 @@
schedulePollBuffering();
}
+bool NuPlayer::RTPSource::isRealTime() const {
+ ALOGD("RTPSource::isRealTime=%d", true);
+ return true;
+}
+
void NuPlayer::RTPSource::onMessageReceived(const sp<AMessage> &msg) {
ALOGV("onMessageReceived =%d", msg->what());
@@ -429,7 +439,6 @@
source->queueAccessUnit(accessUnit);
break;
}
- */
int64_t nptUs =
((double)rtpTime - (double)info->mRTPTime)
@@ -437,7 +446,8 @@
* 1000000ll
+ info->mNormalPlaytimeUs;
- accessUnit->meta()->setInt64("timeUs", nptUs);
+ */
+ accessUnit->meta()->setInt64("timeUs", ALooper::GetNowUs());
source->queueAccessUnit(accessUnit);
}
@@ -490,6 +500,10 @@
}
}
+void NuPlayer::RTPSource::setTargetBitrate(int32_t bitrate) {
+ mRTPConn->setTargetBitrate(bitrate);
+}
+
void NuPlayer::RTPSource::onTimeUpdate(int32_t trackIndex, uint32_t rtpTime, uint64_t ntpTime) {
ALOGV("onTimeUpdate track %d, rtpTime = 0x%08x, ntpTime = %#016llx",
trackIndex, rtpTime, (long long)ntpTime);
@@ -656,6 +670,7 @@
newTrackInfo.mIsAudio = isAudioKey;
mTracks.push(newTrackInfo);
info = &mTracks.editTop();
+ info->mJbTimeMs = 300;
}
if (key == "rtp-param-mime-type") {
@@ -698,6 +713,8 @@
} else if (key == "rtp-param-set-socket-network") {
int64_t networkHandle = atoll(value);
setSocketNetwork(networkHandle);
+ } else if (key == "rtp-param-jitter-buffer-time") {
+ info->mJbTimeMs = atoi(value);
}
return OK;
diff --git a/media/libmediaplayerservice/nuplayer/RTPSource.h b/media/libmediaplayerservice/nuplayer/RTPSource.h
index 5085a7e..fb2d3b9 100644
--- a/media/libmediaplayerservice/nuplayer/RTPSource.h
+++ b/media/libmediaplayerservice/nuplayer/RTPSource.h
@@ -52,6 +52,9 @@
const String8& rtpParams);
enum {
+ RTP_FIRST_PACKET = 100,
+ RTCP_FIRST_PACKET = 101,
+ RTP_QUALITY = 102,
RTCP_TSFB = 205,
RTCP_PSFB = 206,
RTP_CVO = 300,
@@ -77,8 +80,12 @@
int64_t seekTimeUs,
MediaPlayerSeekMode mode = MediaPlayerSeekMode::SEEK_PREVIOUS_SYNC) override;
+ virtual bool isRealTime() const;
+
void onMessageReceived(const sp<AMessage> &msg);
+ virtual void setTargetBitrate(int32_t bitrate) override;
+
protected:
virtual ~RTPSource();
@@ -95,7 +102,6 @@
};
const int64_t kBufferingPollIntervalUs = 1000000ll;
- const int32_t kMinVideoBitrate = 192000; /* bps */
enum State {
DISCONNECTED,
@@ -123,6 +129,8 @@
int32_t mTimeScale;
int32_t mAS;
+ /* RTP jitter buffer time in milliseconds */
+ uint32_t mJbTimeMs;
/* Unique ID indicates itself */
uint32_t mSelfID;
/* extmap:<value> for CVO will be set to here */
diff --git a/media/libmediatranscoding/transcoder/benchmark/Android.bp b/media/libmediatranscoding/transcoder/benchmark/Android.bp
index 507c943..b755206 100644
--- a/media/libmediatranscoding/transcoder/benchmark/Android.bp
+++ b/media/libmediatranscoding/transcoder/benchmark/Android.bp
@@ -4,3 +4,11 @@
shared_libs: ["libmediatranscoder", "libmediandk"],
static_libs: ["libgoogle-benchmark"],
}
+
+cc_test {
+ name: "MediaSampleReaderBenchmark",
+ srcs: ["MediaSampleReaderBenchmark.cpp"],
+ shared_libs: ["libmediatranscoder", "libmediandk", "libbase"],
+ static_libs: ["libgoogle-benchmark"],
+}
+
diff --git a/media/libmediatranscoding/transcoder/benchmark/MediaSampleReaderBenchmark.cpp b/media/libmediatranscoding/transcoder/benchmark/MediaSampleReaderBenchmark.cpp
new file mode 100644
index 0000000..a651fa2
--- /dev/null
+++ b/media/libmediatranscoding/transcoder/benchmark/MediaSampleReaderBenchmark.cpp
@@ -0,0 +1,138 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/**
+ * MediaSampleReader benchmark tests.
+ *
+ * How to run the benchmark:
+ *
+ * 1. Download the media assets from http://go/transcodingbenchmark and push the directory
+ * ("TranscodingBenchmark") to /data/local/tmp.
+ *
+ * 2. Compile the benchmark and sync to device:
+ * $ mm -j72 && adb sync
+ *
+ * 3. Run:
+ * $ adb shell /data/nativetest64/MediaSampleReaderBenchmark/MediaSampleReaderBenchmark
+ */
+
+#define LOG_TAG "MediaSampleReaderBenchmark"
+
+#include <android-base/logging.h>
+#include <benchmark/benchmark.h>
+#include <fcntl.h>
+#include <media/MediaSampleReaderNDK.h>
+#include <unistd.h>
+
+#include <thread>
+
+using namespace android;
+
+static void ReadMediaSamples(benchmark::State& state, const std::string& srcFileName,
+ bool readAudio) {
+ // Asset directory
+ static const std::string kAssetDirectory = "/data/local/tmp/TranscodingBenchmark/";
+
+ int srcFd = 0;
+ std::string srcPath = kAssetDirectory + srcFileName;
+
+ if ((srcFd = open(srcPath.c_str(), O_RDONLY)) < 0) {
+ state.SkipWithError("Unable to open source file");
+ return;
+ }
+
+ const size_t fileSize = lseek(srcFd, 0, SEEK_END);
+ lseek(srcFd, 0, SEEK_SET);
+
+ for (auto _ : state) {
+ auto sampleReader = MediaSampleReaderNDK::createFromFd(srcFd, 0, fileSize);
+
+ std::vector<std::thread> trackThreads;
+
+ for (int trackIndex = 0; trackIndex < sampleReader->getTrackCount(); ++trackIndex) {
+ const char* mime = nullptr;
+
+ AMediaFormat* trackFormat = sampleReader->getTrackFormat(trackIndex);
+ AMediaFormat_getString(trackFormat, AMEDIAFORMAT_KEY_MIME, &mime);
+
+ if (strncmp(mime, "video/", 6) == 0) {
+ int32_t frameCount;
+ if (AMediaFormat_getInt32(trackFormat, AMEDIAFORMAT_KEY_FRAME_COUNT, &frameCount)) {
+ state.counters["VideoFrameRate"] =
+ benchmark::Counter(frameCount, benchmark::Counter::kIsRate);
+ }
+ } else if (!readAudio && strncmp(mime, "audio/", 6) == 0) {
+ continue;
+ }
+
+ trackThreads.emplace_back([trackIndex, sampleReader, &state] {
+ LOG(INFO) << "Track " << trackIndex << " started";
+ MediaSampleInfo info;
+
+ size_t bufferSize = 0;
+ std::unique_ptr<uint8_t[]> buffer;
+
+ while (true) {
+ media_status_t status = sampleReader->getSampleInfoForTrack(trackIndex, &info);
+ if (status == AMEDIA_ERROR_END_OF_STREAM) {
+ break;
+ }
+
+ if (info.size > bufferSize) {
+ bufferSize = info.size;
+ buffer.reset(new uint8_t[bufferSize]);
+ }
+
+ status = sampleReader->readSampleDataForTrack(trackIndex, buffer.get(),
+ bufferSize);
+ if (status != AMEDIA_OK) {
+ state.SkipWithError("Error reading sample data");
+ break;
+ }
+
+ sampleReader->advanceTrack(trackIndex);
+ }
+
+ LOG(INFO) << "Track " << trackIndex << " finished";
+ });
+ }
+
+ for (auto& thread : trackThreads) {
+ thread.join();
+ }
+ }
+
+ close(srcFd);
+}
+
+// Benchmark registration wrapper for transcoding.
+#define TRANSCODER_BENCHMARK(func) \
+ BENCHMARK(func)->UseRealTime()->MeasureProcessCPUTime()->Unit(benchmark::kMillisecond)
+
+static void BM_MediaSampleReader_AudioVideo(benchmark::State& state) {
+ ReadMediaSamples(state, "video_1920x1080_3648frame_h264_22Mbps_30fps_aac.mp4",
+ true /* readAudio */);
+}
+
+static void BM_MediaSampleReader_Video(benchmark::State& state) {
+ ReadMediaSamples(state, "video_1920x1080_3648frame_h264_22Mbps_30fps_aac.mp4",
+ false /* readAudio */);
+}
+
+TRANSCODER_BENCHMARK(BM_MediaSampleReader_AudioVideo);
+TRANSCODER_BENCHMARK(BM_MediaSampleReader_Video);
+
+BENCHMARK_MAIN();
diff --git a/media/libshmem/Android.bp b/media/libshmem/Android.bp
new file mode 100644
index 0000000..ee33f9e
--- /dev/null
+++ b/media/libshmem/Android.bp
@@ -0,0 +1,50 @@
+aidl_interface {
+ name: "shared-file-region-aidl",
+ unstable: true,
+ local_include_dir: "aidl",
+ srcs: [
+ "aidl/android/media/SharedFileRegion.aidl",
+ ],
+}
+
+cc_library {
+ name: "libshmemcompat",
+ export_include_dirs: ["include"],
+ srcs: ["ShmemCompat.cpp"],
+ shared_libs: [
+ "libbinder",
+ "libshmemutil",
+ "libutils",
+ "shared-file-region-aidl-cpp",
+ ],
+ export_shared_lib_headers: [
+ "libbinder",
+ "libutils",
+ "shared-file-region-aidl-cpp",
+ ],
+}
+
+cc_library {
+ name: "libshmemutil",
+ export_include_dirs: ["include"],
+ srcs: ["ShmemUtil.cpp"],
+ shared_libs: [
+ "shared-file-region-aidl-cpp",
+ ],
+ export_shared_lib_headers: [
+ "shared-file-region-aidl-cpp",
+ ],
+}
+
+cc_test {
+ name: "shmemTest",
+ srcs: ["ShmemTest.cpp"],
+ shared_libs: [
+ "libbinder",
+ "libshmemcompat",
+ "libshmemutil",
+ "libutils",
+ "shared-file-region-aidl-cpp",
+ ],
+ test_suites: ["device-tests"],
+}
diff --git a/media/libshmem/OWNERS b/media/libshmem/OWNERS
new file mode 100644
index 0000000..29fa2f5
--- /dev/null
+++ b/media/libshmem/OWNERS
@@ -0,0 +1,3 @@
+ytai@google.com
+mnaganov@google.com
+elaurent@google.com
diff --git a/media/libshmem/README.md b/media/libshmem/README.md
new file mode 100644
index 0000000..c25fa7f
--- /dev/null
+++ b/media/libshmem/README.md
@@ -0,0 +1,6 @@
+# libshmem
+
+This library provides facilities for sharing memory across processes over (stable) AIDL. The main
+feature is the definition of the `android.media.SharedMemory` AIDL type, which represents a block of
+memory that can be shared between processes. In addition, a few utilities are provided to facilitate
+the use of shared memory and to integrate with legacy code that uses older facilities.
\ No newline at end of file
diff --git a/media/libshmem/ShmemCompat.cpp b/media/libshmem/ShmemCompat.cpp
new file mode 100644
index 0000000..5dd83f4
--- /dev/null
+++ b/media/libshmem/ShmemCompat.cpp
@@ -0,0 +1,98 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include "media/ShmemCompat.h"
+
+#include "binder/MemoryBase.h"
+#include "binder/MemoryHeapBase.h"
+#include "media/ShmemUtil.h"
+
+namespace android {
+namespace media {
+
+bool convertSharedFileRegionToIMemory(const SharedFileRegion& shmem,
+ sp<IMemory>* result) {
+ if (!validateSharedFileRegion(shmem)) {
+ return false;
+ }
+
+ if (shmem.fd.get() < 0) {
+ *result = nullptr;
+ return true;
+ }
+
+ // Heap offset and size must be page aligned.
+ const size_t pageSize = getpagesize();
+ const size_t pageMask = ~(pageSize - 1);
+
+ // OK if this wraps.
+ const uint64_t endOffset = static_cast<uint64_t>(shmem.offset) +
+ static_cast<uint64_t>(shmem.size);
+
+ // Round down to page boundary.
+ const uint64_t heapStartOffset = shmem.offset & pageMask;
+ // Round up to page boundary.
+ const uint64_t heapEndOffset = (endOffset + pageSize - 1) & pageMask;
+ const uint64_t heapSize = heapEndOffset - heapStartOffset;
+
+ if (heapStartOffset > std::numeric_limits<size_t>::max() ||
+ heapSize > std::numeric_limits<size_t>::max()) {
+ return false;
+ }
+
+ const sp<MemoryHeapBase> heap =
+ new MemoryHeapBase(shmem.fd.get(), heapSize, 0, heapStartOffset);
+ *result = sp<MemoryBase>::make(heap,
+ shmem.offset - heapStartOffset,
+ shmem.size);
+ return true;
+}
+
+bool convertIMemoryToSharedFileRegion(const sp<IMemory>& mem,
+ SharedFileRegion* result) {
+ *result = SharedFileRegion();
+ if (mem == nullptr) {
+ return true;
+ }
+
+ ssize_t offset;
+ size_t size;
+
+ sp<IMemoryHeap> heap = mem->getMemory(&offset, &size);
+ if (heap != nullptr) {
+ // Make sure the offset and size do not overflow from int64 boundaries.
+ if (size > std::numeric_limits<int64_t>::max() ||
+ offset > std::numeric_limits<int64_t>::max() ||
+ heap->getOffset() > std::numeric_limits<int64_t>::max() ||
+ static_cast<uint64_t>(heap->getOffset()) +
+ static_cast<uint64_t>(offset)
+ > std::numeric_limits<int64_t>::max()) {
+ return false;
+ }
+
+ const int fd = fcntl(heap->getHeapID(), F_DUPFD_CLOEXEC, 0);
+ if (fd < 0) {
+ return false;
+ }
+ result->fd.reset(base::unique_fd(fd));
+ result->size = size;
+ result->offset = heap->getOffset() + offset;
+ }
+
+ return true;
+}
+
+} // namespace media
+} // namespace android
diff --git a/media/libshmem/ShmemTest.cpp b/media/libshmem/ShmemTest.cpp
new file mode 100644
index 0000000..4f11b51
--- /dev/null
+++ b/media/libshmem/ShmemTest.cpp
@@ -0,0 +1,87 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <gtest/gtest.h>
+
+#include "binder/MemoryBase.h"
+#include "binder/MemoryHeapBase.h"
+#include "media/ShmemCompat.h"
+#include "media/ShmemUtil.h"
+
+namespace android {
+namespace media {
+namespace {
+
+// Creates a SharedFileRegion instance with a null FD.
+SharedFileRegion makeSharedFileRegion(int64_t offset, int64_t size) {
+ SharedFileRegion shmem;
+ shmem.offset = offset;
+ shmem.size = size;
+ return shmem;
+}
+
+sp<IMemory> makeIMemory(const std::vector<uint8_t>& content) {
+ constexpr size_t kOffset = 19;
+
+ sp<MemoryHeapBase> heap = new MemoryHeapBase(content.size());
+ sp<IMemory> result = sp<MemoryBase>::make(heap, kOffset, content.size());
+ memcpy(result->unsecurePointer(), content.data(), content.size());
+ return result;
+}
+
+TEST(ShmemTest, Validate) {
+ EXPECT_TRUE(validateSharedFileRegion(makeSharedFileRegion(0, 0)));
+ EXPECT_TRUE(validateSharedFileRegion(makeSharedFileRegion(1, 2)));
+ EXPECT_FALSE(validateSharedFileRegion(makeSharedFileRegion(-1, 2)));
+ EXPECT_FALSE(validateSharedFileRegion(makeSharedFileRegion(2, -1)));
+ EXPECT_TRUE(validateSharedFileRegion(makeSharedFileRegion(
+ std::numeric_limits<int64_t>::max(),
+ std::numeric_limits<int64_t>::max())));
+}
+
+TEST(ShmemTest, Conversion) {
+ sp<IMemory> reconstructed;
+ {
+ SharedFileRegion shmem;
+ sp<IMemory> imem = makeIMemory({6, 5, 3});
+ ASSERT_TRUE(convertIMemoryToSharedFileRegion(imem, &shmem));
+ ASSERT_EQ(3, shmem.size);
+ ASSERT_GE(shmem.fd.get(), 0);
+ ASSERT_TRUE(convertSharedFileRegionToIMemory(shmem, &reconstructed));
+ }
+ ASSERT_EQ(3, reconstructed->size());
+ const uint8_t* p =
+ reinterpret_cast<const uint8_t*>(reconstructed->unsecurePointer());
+ EXPECT_EQ(6, p[0]);
+ EXPECT_EQ(5, p[1]);
+ EXPECT_EQ(3, p[2]);
+}
+
+TEST(ShmemTest, NullConversion) {
+ sp<IMemory> reconstructed;
+ {
+ SharedFileRegion shmem;
+ sp<IMemory> imem;
+ ASSERT_TRUE(convertIMemoryToSharedFileRegion(imem, &shmem));
+ ASSERT_EQ(0, shmem.size);
+ ASSERT_LT(shmem.fd.get(), 0);
+ ASSERT_TRUE(convertSharedFileRegionToIMemory(shmem, &reconstructed));
+ }
+ ASSERT_EQ(nullptr, reconstructed);
+}
+
+} // namespace
+} // namespace media
+} // namespace android
diff --git a/media/libshmem/ShmemUtil.cpp b/media/libshmem/ShmemUtil.cpp
new file mode 100644
index 0000000..a6d047f
--- /dev/null
+++ b/media/libshmem/ShmemUtil.cpp
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include "media/ShmemUtil.h"
+
+namespace android {
+namespace media {
+
+bool validateSharedFileRegion(const SharedFileRegion& shmem) {
+ // Size and offset must be non-negative.
+ if (shmem.size < 0 || shmem.offset < 0) {
+ return false;
+ }
+
+ uint64_t size = shmem.size;
+ uint64_t offset = shmem.offset;
+
+ // Must not wrap.
+ if (offset > offset + size) {
+ return false;
+ }
+
+ return true;
+}
+
+} // namespace media
+} // namespace android
diff --git a/media/libshmem/aidl/android/media/SharedFileRegion.aidl b/media/libshmem/aidl/android/media/SharedFileRegion.aidl
new file mode 100644
index 0000000..c99ad95
--- /dev/null
+++ b/media/libshmem/aidl/android/media/SharedFileRegion.aidl
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * A shared file region.
+ *
+ * This type contains the required information to share a region of a file between processes over
+ * AIDL. An invalid (null) region may be represented using a negative file descriptor.
+ * Primarily, this is intended for shared memory blocks.
+ *
+ * @hide
+ */
+parcelable SharedFileRegion {
+ /** File descriptor of the region. */
+ ParcelFileDescriptor fd;
+ /** Offset, in bytes within the file of the start of the region. Must be non-negative. */
+ long offset;
+ /** Size, in bytes of the memory region. Must be non-negative. */
+ long size;
+}
diff --git a/media/libshmem/include/media/ShmemCompat.h b/media/libshmem/include/media/ShmemCompat.h
new file mode 100644
index 0000000..3bf7f67
--- /dev/null
+++ b/media/libshmem/include/media/ShmemCompat.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+// This module contains utilities for interfacing between legacy code that is using IMemory and new
+// code that is using android.os.SharedFileRegion.
+
+#include "android/media/SharedFileRegion.h"
+#include "binder/IMemory.h"
+#include "utils/StrongPointer.h"
+
+namespace android {
+namespace media {
+
+/**
+ * Converts a SharedFileRegion parcelable to an IMemory instance.
+ * @param shmem The SharedFileRegion instance.
+ * @param result The resulting IMemory instance, or null of the SharedFileRegion is null (has a
+ * negative FD).
+ * @return true if the conversion is successful (should always succeed under normal circumstances,
+ * failure usually means corrupt data).
+ */
+bool convertSharedFileRegionToIMemory(const SharedFileRegion& shmem,
+ sp<IMemory>* result);
+
+/**
+ * Converts an IMemory instance to SharedFileRegion.
+ * @param mem The IMemory instance. May be null.
+ * @param result The resulting SharedFileRegion instance.
+ * @return true if the conversion is successful (should always succeed under normal circumstances,
+ * failure usually means corrupt data).
+ */
+bool convertIMemoryToSharedFileRegion(const sp<IMemory>& mem,
+ SharedFileRegion* result);
+
+} // namespace media
+} // namespace android
diff --git a/media/libshmem/include/media/ShmemUtil.h b/media/libshmem/include/media/ShmemUtil.h
new file mode 100644
index 0000000..563cb71
--- /dev/null
+++ b/media/libshmem/include/media/ShmemUtil.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+// This module contains utilities for working with android.os.SharedFileRegion.
+
+#include "android/media/SharedFileRegion.h"
+
+namespace android {
+namespace media {
+
+/**
+ * Checks whether a SharedFileRegion instance is valid (all the fields have sane values).
+ * A null SharedFileRegion (having a negative FD) is considered valid.
+ */
+bool validateSharedFileRegion(const SharedFileRegion& shmem);
+
+} // namespace media
+} // namespace android
diff --git a/media/libstagefright/FrameCaptureProcessor.cpp b/media/libstagefright/FrameCaptureProcessor.cpp
index ee642d4..8cd7f82 100644
--- a/media/libstagefright/FrameCaptureProcessor.cpp
+++ b/media/libstagefright/FrameCaptureProcessor.cpp
@@ -171,13 +171,7 @@
err = OK;
}
}
- mRE->cleanupPostRender();
- // Unbind the buffer now to remove it from the RenderEngine's image cache.
- // The buffer was put into the image cache during the drawLayers() call above.
- const sp<GraphicBuffer> &gbuf = layerSettings.source.buffer.buffer;
- if (gbuf != nullptr) {
- mRE->unbindExternalTextureBuffer(gbuf->getId());
- }
+ mRE->cleanupPostRender(renderengine::RenderEngine::CleanupMode::CLEAN_ALL);
return err;
}
diff --git a/media/libstagefright/HevcUtils.cpp b/media/libstagefright/HevcUtils.cpp
index aac656a..5f9c20e 100644
--- a/media/libstagefright/HevcUtils.cpp
+++ b/media/libstagefright/HevcUtils.cpp
@@ -380,6 +380,54 @@
return reader.overRead() ? ERROR_MALFORMED : OK;
}
+void HevcParameterSets::FindHEVCDimensions(const sp<ABuffer> &SpsBuffer, int32_t *width, int32_t *height)
+{
+ ALOGD("FindHEVCDimensions");
+ // See Rec. ITU-T H.265 v3 (04/2015) Chapter 7.3.2.2 for reference
+ ABitReader reader(SpsBuffer->data() + 1, SpsBuffer->size() - 1);
+ // Skip sps_video_parameter_set_id
+ reader.skipBits(4);
+ uint8_t maxSubLayersMinus1 = reader.getBitsWithFallback(3, 0);
+ // Skip sps_temporal_id_nesting_flag;
+ reader.skipBits(1);
+ // Skip general profile
+ reader.skipBits(96);
+ if (maxSubLayersMinus1 > 0) {
+ bool subLayerProfilePresentFlag[8];
+ bool subLayerLevelPresentFlag[8];
+ for (int i = 0; i < maxSubLayersMinus1; ++i) {
+ subLayerProfilePresentFlag[i] = reader.getBitsWithFallback(1, 0);
+ subLayerLevelPresentFlag[i] = reader.getBitsWithFallback(1, 0);
+ }
+ // Skip reserved
+ reader.skipBits(2 * (8 - maxSubLayersMinus1));
+ for (int i = 0; i < maxSubLayersMinus1; ++i) {
+ if (subLayerProfilePresentFlag[i]) {
+ // Skip profile
+ reader.skipBits(88);
+ }
+ if (subLayerLevelPresentFlag[i]) {
+ // Skip sub_layer_level_idc[i]
+ reader.skipBits(8);
+ }
+ }
+ }
+ // Skip sps_seq_parameter_set_id
+ skipUE(&reader);
+ uint8_t chromaFormatIdc = parseUEWithFallback(&reader, 0);
+ if (chromaFormatIdc == 3) {
+ // Skip separate_colour_plane_flag
+ reader.skipBits(1);
+ }
+ skipUE(&reader);
+ skipUE(&reader);
+
+ // pic_width_in_luma_samples
+ *width = parseUEWithFallback(&reader, 0);
+ // pic_height_in_luma_samples
+ *height = parseUEWithFallback(&reader, 0);
+}
+
status_t HevcParameterSets::parsePps(
const uint8_t* data UNUSED_PARAM, size_t size UNUSED_PARAM) {
return OK;
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index e975ee6..5015787 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -2044,20 +2044,25 @@
} else if (mFlags & kFlagOutputBuffersChanged) {
PostReplyWithError(replyID, INFO_OUTPUT_BUFFERS_CHANGED);
mFlags &= ~kFlagOutputBuffersChanged;
- } else if (mFlags & kFlagOutputFormatChanged) {
- PostReplyWithError(replyID, INFO_FORMAT_CHANGED);
- mFlags &= ~kFlagOutputFormatChanged;
} else {
sp<AMessage> response = new AMessage;
- ssize_t index = dequeuePortBuffer(kPortIndexOutput);
-
- if (index < 0) {
- CHECK_EQ(index, -EAGAIN);
+ BufferInfo *info = peekNextPortBuffer(kPortIndexOutput);
+ if (!info) {
return false;
}
- const sp<MediaCodecBuffer> &buffer =
- mPortBuffers[kPortIndexOutput][index].mData;
+ // In synchronous mode, output format change should be handled
+ // at dequeue to put the event at the correct order.
+
+ const sp<MediaCodecBuffer> &buffer = info->mData;
+ handleOutputFormatChangeIfNeeded(buffer);
+ if (mFlags & kFlagOutputFormatChanged) {
+ PostReplyWithError(replyID, INFO_FORMAT_CHANGED);
+ mFlags &= ~kFlagOutputFormatChanged;
+ return true;
+ }
+
+ ssize_t index = dequeuePortBuffer(kPortIndexOutput);
response->setSize("index", index);
response->setSize("offset", buffer->offset());
@@ -2094,8 +2099,8 @@
CHECK(msg->findInt32("err", &err));
CHECK(msg->findInt32("actionCode", &actionCode));
- ALOGE("Codec reported err %#x, actionCode %d, while in state %d",
- err, actionCode, mState);
+ ALOGE("Codec reported err %#x, actionCode %d, while in state %d/%s",
+ err, actionCode, mState, stateString(mState).c_str());
if (err == DEAD_OBJECT) {
mFlags |= kFlagSawMediaServerDie;
mFlags &= ~kFlagIsComponentAllocated;
@@ -2253,8 +2258,8 @@
if (mState == RELEASING || mState == UNINITIALIZED) {
// In case a kWhatError or kWhatRelease message came in and replied,
// we log a warning and ignore.
- ALOGW("allocate interrupted by error or release, current state %d",
- mState);
+ ALOGW("allocate interrupted by error or release, current state %d/%s",
+ mState, stateString(mState).c_str());
break;
}
CHECK_EQ(mState, INITIALIZING);
@@ -2300,8 +2305,8 @@
if (mState == RELEASING || mState == UNINITIALIZED || mState == INITIALIZED) {
// In case a kWhatError or kWhatRelease message came in and replied,
// we log a warning and ignore.
- ALOGW("configure interrupted by error or release, current state %d",
- mState);
+ ALOGW("configure interrupted by error or release, current state %d/%s",
+ mState, stateString(mState).c_str());
break;
}
CHECK_EQ(mState, CONFIGURING);
@@ -2430,7 +2435,8 @@
if (mState == RELEASING || mState == UNINITIALIZED) {
// In case a kWhatRelease message came in and replied,
// we log a warning and ignore.
- ALOGW("start interrupted by release, current state %d", mState);
+ ALOGW("start interrupted by release, current state %d/%s",
+ mState, stateString(mState).c_str());
break;
}
@@ -2541,107 +2547,13 @@
break;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<MediaCodecBuffer> buffer = static_cast<MediaCodecBuffer *>(obj.get());
-
- if (mOutputFormat != buffer->format()) {
- if (mFlags & kFlagUseBlockModel) {
- sp<AMessage> diff1 = mOutputFormat->changesFrom(buffer->format());
- sp<AMessage> diff2 = buffer->format()->changesFrom(mOutputFormat);
- std::set<std::string> keys;
- size_t numEntries = diff1->countEntries();
- AMessage::Type type;
- for (size_t i = 0; i < numEntries; ++i) {
- keys.emplace(diff1->getEntryNameAt(i, &type));
- }
- numEntries = diff2->countEntries();
- for (size_t i = 0; i < numEntries; ++i) {
- keys.emplace(diff2->getEntryNameAt(i, &type));
- }
- sp<WrapperObject<std::set<std::string>>> changedKeys{
- new WrapperObject<std::set<std::string>>{std::move(keys)}};
- buffer->meta()->setObject("changedKeys", changedKeys);
- }
- mOutputFormat = buffer->format();
- ALOGV("[%s] output format changed to: %s",
- mComponentName.c_str(), mOutputFormat->debugString(4).c_str());
-
- if (mSoftRenderer == NULL &&
- mSurface != NULL &&
- (mFlags & kFlagUsesSoftwareRenderer)) {
- AString mime;
- CHECK(mOutputFormat->findString("mime", &mime));
-
- // TODO: propagate color aspects to software renderer to allow better
- // color conversion to RGB. For now, just mark dataspace for YUV
- // rendering.
- int32_t dataSpace;
- if (mOutputFormat->findInt32("android._dataspace", &dataSpace)) {
- ALOGD("[%s] setting dataspace on output surface to #%x",
- mComponentName.c_str(), dataSpace);
- int err = native_window_set_buffers_data_space(
- mSurface.get(), (android_dataspace)dataSpace);
- ALOGW_IF(err != 0, "failed to set dataspace on surface (%d)", err);
- }
- if (mOutputFormat->contains("hdr-static-info")) {
- HDRStaticInfo info;
- if (ColorUtils::getHDRStaticInfoFromFormat(mOutputFormat, &info)) {
- setNativeWindowHdrMetadata(mSurface.get(), &info);
- }
- }
-
- sp<ABuffer> hdr10PlusInfo;
- if (mOutputFormat->findBuffer("hdr10-plus-info", &hdr10PlusInfo)
- && hdr10PlusInfo != nullptr && hdr10PlusInfo->size() > 0) {
- native_window_set_buffers_hdr10_plus_metadata(mSurface.get(),
- hdr10PlusInfo->size(), hdr10PlusInfo->data());
- }
-
- if (mime.startsWithIgnoreCase("video/")) {
- mSurface->setDequeueTimeout(-1);
- mSoftRenderer = new SoftwareRenderer(mSurface, mRotationDegrees);
- }
- }
-
- requestCpuBoostIfNeeded();
-
- if (mFlags & kFlagIsEncoder) {
- // Before we announce the format change we should
- // collect codec specific data and amend the output
- // format as necessary.
- int32_t flags = 0;
- (void) buffer->meta()->findInt32("flags", &flags);
- if ((flags & BUFFER_FLAG_CODECCONFIG) && !(mFlags & kFlagIsSecure)) {
- status_t err =
- amendOutputFormatWithCodecSpecificData(buffer);
-
- if (err != OK) {
- ALOGE("Codec spit out malformed codec "
- "specific data!");
- }
- }
- }
- if (mFlags & kFlagIsAsync) {
- onOutputFormatChanged();
- } else {
- mFlags |= kFlagOutputFormatChanged;
- postActivityNotificationIfPossible();
- }
-
- // Notify mCrypto of video resolution changes
- if (mCrypto != NULL) {
- int32_t left, top, right, bottom, width, height;
- if (mOutputFormat->findRect("crop", &left, &top, &right, &bottom)) {
- mCrypto->notifyResolution(right - left + 1, bottom - top + 1);
- } else if (mOutputFormat->findInt32("width", &width)
- && mOutputFormat->findInt32("height", &height)) {
- mCrypto->notifyResolution(width, height);
- }
- }
- }
-
if (mFlags & kFlagIsAsync) {
+ sp<RefBase> obj;
+ CHECK(msg->findObject("buffer", &obj));
+ sp<MediaCodecBuffer> buffer = static_cast<MediaCodecBuffer *>(obj.get());
+
+ // In asynchronous mode, output format change is processed immediately.
+ handleOutputFormatChangeIfNeeded(buffer);
onOutputBufferAvailable();
} else if (mFlags & kFlagDequeueOutputPending) {
CHECK(handleDequeueOutputBuffer(mDequeueOutputReplyID));
@@ -2666,7 +2578,8 @@
case kWhatStopCompleted:
{
if (mState != STOPPING) {
- ALOGW("Received kWhatStopCompleted in state %d", mState);
+ ALOGW("Received kWhatStopCompleted in state %d/%s",
+ mState, stateString(mState).c_str());
break;
}
setState(INITIALIZED);
@@ -2677,7 +2590,8 @@
case kWhatReleaseCompleted:
{
if (mState != RELEASING) {
- ALOGW("Received kWhatReleaseCompleted in state %d", mState);
+ ALOGW("Received kWhatReleaseCompleted in state %d/%s",
+ mState, stateString(mState).c_str());
break;
}
setState(UNINITIALIZED);
@@ -2707,8 +2621,8 @@
case kWhatFlushCompleted:
{
if (mState != FLUSHING) {
- ALOGW("received FlushCompleted message in state %d",
- mState);
+ ALOGW("received FlushCompleted message in state %d/%s",
+ mState, stateString(mState).c_str());
break;
}
@@ -3470,6 +3384,106 @@
}
}
+void MediaCodec::handleOutputFormatChangeIfNeeded(const sp<MediaCodecBuffer> &buffer) {
+ sp<AMessage> format = buffer->format();
+ if (mOutputFormat == format) {
+ return;
+ }
+ if (mFlags & kFlagUseBlockModel) {
+ sp<AMessage> diff1 = mOutputFormat->changesFrom(format);
+ sp<AMessage> diff2 = format->changesFrom(mOutputFormat);
+ std::set<std::string> keys;
+ size_t numEntries = diff1->countEntries();
+ AMessage::Type type;
+ for (size_t i = 0; i < numEntries; ++i) {
+ keys.emplace(diff1->getEntryNameAt(i, &type));
+ }
+ numEntries = diff2->countEntries();
+ for (size_t i = 0; i < numEntries; ++i) {
+ keys.emplace(diff2->getEntryNameAt(i, &type));
+ }
+ sp<WrapperObject<std::set<std::string>>> changedKeys{
+ new WrapperObject<std::set<std::string>>{std::move(keys)}};
+ buffer->meta()->setObject("changedKeys", changedKeys);
+ }
+ mOutputFormat = format;
+ ALOGV("[%s] output format changed to: %s",
+ mComponentName.c_str(), mOutputFormat->debugString(4).c_str());
+
+ if (mSoftRenderer == NULL &&
+ mSurface != NULL &&
+ (mFlags & kFlagUsesSoftwareRenderer)) {
+ AString mime;
+ CHECK(mOutputFormat->findString("mime", &mime));
+
+ // TODO: propagate color aspects to software renderer to allow better
+ // color conversion to RGB. For now, just mark dataspace for YUV
+ // rendering.
+ int32_t dataSpace;
+ if (mOutputFormat->findInt32("android._dataspace", &dataSpace)) {
+ ALOGD("[%s] setting dataspace on output surface to #%x",
+ mComponentName.c_str(), dataSpace);
+ int err = native_window_set_buffers_data_space(
+ mSurface.get(), (android_dataspace)dataSpace);
+ ALOGW_IF(err != 0, "failed to set dataspace on surface (%d)", err);
+ }
+ if (mOutputFormat->contains("hdr-static-info")) {
+ HDRStaticInfo info;
+ if (ColorUtils::getHDRStaticInfoFromFormat(mOutputFormat, &info)) {
+ setNativeWindowHdrMetadata(mSurface.get(), &info);
+ }
+ }
+
+ sp<ABuffer> hdr10PlusInfo;
+ if (mOutputFormat->findBuffer("hdr10-plus-info", &hdr10PlusInfo)
+ && hdr10PlusInfo != nullptr && hdr10PlusInfo->size() > 0) {
+ native_window_set_buffers_hdr10_plus_metadata(mSurface.get(),
+ hdr10PlusInfo->size(), hdr10PlusInfo->data());
+ }
+
+ if (mime.startsWithIgnoreCase("video/")) {
+ mSurface->setDequeueTimeout(-1);
+ mSoftRenderer = new SoftwareRenderer(mSurface, mRotationDegrees);
+ }
+ }
+
+ requestCpuBoostIfNeeded();
+
+ if (mFlags & kFlagIsEncoder) {
+ // Before we announce the format change we should
+ // collect codec specific data and amend the output
+ // format as necessary.
+ int32_t flags = 0;
+ (void) buffer->meta()->findInt32("flags", &flags);
+ if ((flags & BUFFER_FLAG_CODECCONFIG) && !(mFlags & kFlagIsSecure)) {
+ status_t err =
+ amendOutputFormatWithCodecSpecificData(buffer);
+
+ if (err != OK) {
+ ALOGE("Codec spit out malformed codec "
+ "specific data!");
+ }
+ }
+ }
+ if (mFlags & kFlagIsAsync) {
+ onOutputFormatChanged();
+ } else {
+ mFlags |= kFlagOutputFormatChanged;
+ postActivityNotificationIfPossible();
+ }
+
+ // Notify mCrypto of video resolution changes
+ if (mCrypto != NULL) {
+ int32_t left, top, right, bottom, width, height;
+ if (mOutputFormat->findRect("crop", &left, &top, &right, &bottom)) {
+ mCrypto->notifyResolution(right - left + 1, bottom - top + 1);
+ } else if (mOutputFormat->findInt32("width", &width)
+ && mOutputFormat->findInt32("height", &height)) {
+ mCrypto->notifyResolution(width, height);
+ }
+ }
+}
+
void MediaCodec::extractCSD(const sp<AMessage> &format) {
mCSD.clear();
@@ -3935,19 +3949,31 @@
return OK;
}
-ssize_t MediaCodec::dequeuePortBuffer(int32_t portIndex) {
+MediaCodec::BufferInfo *MediaCodec::peekNextPortBuffer(int32_t portIndex) {
CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
List<size_t> *availBuffers = &mAvailPortBuffers[portIndex];
if (availBuffers->empty()) {
+ return nullptr;
+ }
+
+ return &mPortBuffers[portIndex][*availBuffers->begin()];
+}
+
+ssize_t MediaCodec::dequeuePortBuffer(int32_t portIndex) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+
+ BufferInfo *info = peekNextPortBuffer(portIndex);
+ if (!info) {
return -EAGAIN;
}
+ List<size_t> *availBuffers = &mAvailPortBuffers[portIndex];
size_t index = *availBuffers->begin();
+ CHECK_EQ(info, &mPortBuffers[portIndex][index]);
availBuffers->erase(availBuffers->begin());
- BufferInfo *info = &mPortBuffers[portIndex][index];
CHECK(!info->mOwnedByClient);
{
Mutex::Autolock al(mBufferLock);
diff --git a/media/libstagefright/OWNERS b/media/libstagefright/OWNERS
new file mode 100644
index 0000000..819389d
--- /dev/null
+++ b/media/libstagefright/OWNERS
@@ -0,0 +1,7 @@
+set noparent
+chz@google.com
+essick@google.com
+lajos@google.com
+marcone@google.com
+taklee@google.com
+wonsik@google.com
\ No newline at end of file
diff --git a/media/libstagefright/codecs/m4v_h263/dec/src/block_idct.cpp b/media/libstagefright/codecs/m4v_h263/dec/src/block_idct.cpp
index 3d10086..bc708e2 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/src/block_idct.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/src/block_idct.cpp
@@ -506,6 +506,7 @@
/*----------------------------------------------------------------------------
; Function Code FOR idctrow
----------------------------------------------------------------------------*/
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctrow(
int16 *blk, uint8 *pred, uint8 *dst, int width
)
@@ -828,6 +829,7 @@
/*----------------------------------------------------------------------------
; Function Code FOR idctcol
----------------------------------------------------------------------------*/
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctcol(
int16 *blk
)
diff --git a/media/libstagefright/codecs/m4v_h263/dec/src/idct_vca.cpp b/media/libstagefright/codecs/m4v_h263/dec/src/idct_vca.cpp
index f35ce4f..0ba4944 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/src/idct_vca.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/src/idct_vca.cpp
@@ -94,6 +94,7 @@
return;
}
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctrow2(int16 *blk, uint8 *pred, uint8 *dst, int width)
{
int32 x0, x1, x2, x4, x5;
@@ -182,6 +183,7 @@
return ;
}
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctrow3(int16 *blk, uint8 *pred, uint8 *dst, int width)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -291,6 +293,7 @@
}
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctrow4(int16 *blk, uint8 *pred, uint8 *dst, int width)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -368,6 +371,7 @@
return ;
}
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctcol4(int16 *blk)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8;
@@ -445,6 +449,7 @@
return;
}
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctrow2_intra(int16 *blk, PIXEL *comp, int width)
{
int32 x0, x1, x2, x4, x5, temp;
@@ -502,6 +507,7 @@
return ;
}
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctrow3_intra(int16 *blk, PIXEL *comp, int width)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8, temp;
@@ -575,6 +581,7 @@
return ;
}
+__attribute__((no_sanitize("signed-integer-overflow")))
void idctrow4_intra(int16 *blk, PIXEL *comp, int width)
{
int32 x0, x1, x2, x3, x4, x5, x6, x7, x8, temp;
diff --git a/media/libstagefright/include/HevcUtils.h b/media/libstagefright/include/HevcUtils.h
index d2a86eb..6a4a168 100644
--- a/media/libstagefright/include/HevcUtils.h
+++ b/media/libstagefright/include/HevcUtils.h
@@ -94,6 +94,8 @@
// Note that this method does not write the start code.
bool write(size_t index, uint8_t* dest, size_t size);
status_t makeHvcc(uint8_t *hvcc, size_t *hvccSize, size_t nalSizeLength);
+ void FindHEVCDimensions(
+ const sp<ABuffer> &SpsBuffer, int32_t *width, int32_t *height);
Info getInfo() const { return mInfo; }
static bool IsHevcIDR(const uint8_t *data, size_t size);
diff --git a/media/libstagefright/include/media/stagefright/MediaCodec.h b/media/libstagefright/include/media/stagefright/MediaCodec.h
index 1f8e780..c4026ec 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodec.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodec.h
@@ -452,6 +452,7 @@
size_t updateBuffers(int32_t portIndex, const sp<AMessage> &msg);
status_t onQueueInputBuffer(const sp<AMessage> &msg);
status_t onReleaseOutputBuffer(const sp<AMessage> &msg);
+ BufferInfo *peekNextPortBuffer(int32_t portIndex);
ssize_t dequeuePortBuffer(int32_t portIndex);
status_t getBufferAndFormat(
@@ -483,6 +484,7 @@
status_t onSetParameters(const sp<AMessage> ¶ms);
status_t amendOutputFormatWithCodecSpecificData(const sp<MediaCodecBuffer> &buffer);
+ void handleOutputFormatChangeIfNeeded(const sp<MediaCodecBuffer> &buffer);
bool isExecuting() const;
uint64_t getGraphicBufferSize();
diff --git a/media/libstagefright/include/media/stagefright/MetaDataBase.h b/media/libstagefright/include/media/stagefright/MetaDataBase.h
index 173b701..2f34094 100644
--- a/media/libstagefright/include/media/stagefright/MetaDataBase.h
+++ b/media/libstagefright/include/media/stagefright/MetaDataBase.h
@@ -248,8 +248,9 @@
// Treat empty track as malformed for MediaRecorder.
kKeyEmptyTrackMalFormed = 'nemt', // bool (int32_t)
- kKeySps = 'sSps', // int32_t, indicates that a buffer is sps (value ignored).
- kKeyPps = 'sPps', // int32_t, indicates that a buffer is pps (value ignored).
+ kKeyVps = 'sVps', // int32_t, indicates that a buffer has vps.
+ kKeySps = 'sSps', // int32_t, indicates that a buffer has sps.
+ kKeyPps = 'sPps', // int32_t, indicates that a buffer has pps.
kKeySelfID = 'sfid', // int32_t, source ID to identify itself on RTP protocol.
kKeyPayloadType = 'pTyp', // int32_t, SDP negotiated payload type.
kKeyRtpExtMap = 'extm', // int32_t, rtp extension ID for cvo on RTP protocol.
diff --git a/media/libstagefright/omx/1.0/OmxStore.cpp b/media/libstagefright/omx/1.0/OmxStore.cpp
index 67f478e..b5c1166 100644
--- a/media/libstagefright/omx/1.0/OmxStore.cpp
+++ b/media/libstagefright/omx/1.0/OmxStore.cpp
@@ -54,6 +54,24 @@
});
}
+ if (!nodes.empty()) {
+ auto anyNode = nodes.cbegin();
+ std::string::const_iterator first = anyNode->cbegin();
+ std::string::const_iterator last = anyNode->cend();
+ for (const std::string &name : nodes) {
+ std::string::const_iterator it1 = first;
+ for (std::string::const_iterator it2 = name.cbegin();
+ it1 != last && it2 != name.cend() && tolower(*it1) == tolower(*it2);
+ ++it1, ++it2) {
+ }
+ last = it1;
+ }
+ mPrefix = std::string(first, last);
+ LOG(INFO) << "omx common prefix: '" << mPrefix.c_str() << "'";
+ } else {
+ LOG(INFO) << "omx common prefix: no nodes";
+ }
+
MediaCodecsXmlParser parser;
parser.parseXmlFilesInSearchDirs(xmlNames, searchDirs);
if (profilingResultsXmlPath != nullptr) {
@@ -112,8 +130,6 @@
mRoleList[i] = std::move(role);
++i;
}
-
- mPrefix = parser.getCommonPrefix();
}
OmxStore::~OmxStore() {
diff --git a/media/libstagefright/rtsp/AAVCAssembler.cpp b/media/libstagefright/rtsp/AAVCAssembler.cpp
index 0164040..a0b66a7 100644
--- a/media/libstagefright/rtsp/AAVCAssembler.cpp
+++ b/media/libstagefright/rtsp/AAVCAssembler.cpp
@@ -37,12 +37,73 @@
mAccessUnitRTPTime(0),
mNextExpectedSeqNoValid(false),
mNextExpectedSeqNo(0),
- mAccessUnitDamaged(false) {
+ mAccessUnitDamaged(false),
+ mFirstIFrameProvided(false),
+ mLastIFrameProvidedAtMs(0) {
}
AAVCAssembler::~AAVCAssembler() {
}
+int32_t AAVCAssembler::addNack(
+ const sp<ARTPSource> &source) {
+ List<sp<ABuffer>> *queue = source->queue();
+ int32_t nackCount = 0;
+
+ List<sp<ABuffer> >::iterator it = queue->begin();
+
+ if (it == queue->end()) {
+ return nackCount /* 0 */;
+ }
+
+ uint16_t queueHeadSeqNum = (*it)->int32Data();
+
+ // move to the packet after which RTCP:NACK was sent.
+ for (; it != queue->end(); ++it) {
+ int32_t seqNum = (*it)->int32Data();
+ if (seqNum >= source->mHighestNackNumber) {
+ break;
+ }
+ }
+
+ int32_t nackStartAt = -1;
+
+ while (it != queue->end()) {
+ int32_t seqBeforeLast = (*it)->int32Data();
+ // increase iterator.
+ if ((++it) == queue->end()) {
+ break;
+ }
+ int32_t seqLast = (*it)->int32Data();
+
+ if ((seqLast - seqBeforeLast) < 0) {
+ ALOGD("addNack: found end of seqNum from(%d) to(%d)", seqBeforeLast, seqLast);
+ source->mHighestNackNumber = 0;
+ }
+
+ // missed packet found
+ if (seqLast > (seqBeforeLast + 1) &&
+ // we didn't send RTCP:NACK for this packet yet.
+ (seqLast - 1) > source->mHighestNackNumber) {
+ source->mHighestNackNumber = seqLast - 1;
+ nackStartAt = seqBeforeLast + 1;
+ break;
+ }
+
+ }
+
+ if (nackStartAt != -1) {
+ nackCount = source->mHighestNackNumber - nackStartAt + 1;
+ ALOGD("addNack: nackCount=%d, nackFrom=%d, nackTo=%d", nackCount,
+ nackStartAt, source->mHighestNackNumber);
+
+ uint16_t mask = (uint16_t)(0xffff) >> (16 - nackCount + 1);
+ source->setSeqNumToNACK(nackStartAt, mask, queueHeadSeqNum);
+ }
+
+ return nackCount;
+}
+
ARTPAssembler::AssemblyStatus AAVCAssembler::addNALUnit(
const sp<ARTPSource> &source) {
List<sp<ABuffer> > *queue = source->queue();
@@ -52,78 +113,54 @@
}
sp<ABuffer> buffer = *queue->begin();
- int32_t rtpTime;
- CHECK(buffer->meta()->findInt32("rtp-time", &rtpTime));
+ uint32_t rtpTime;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
int64_t startTime = source->mFirstSysTime / 1000;
int64_t nowTime = ALooper::GetNowUs() / 1000;
int64_t playedTime = nowTime - startTime;
- int32_t playedTimeRtp = source->mFirstRtpTime +
- (((uint32_t)playedTime) * (source->mClockRate / 1000));
- const int32_t jitterTime = source->mClockRate / 5; // 200ms
- int32_t expiredTimeInJb = rtpTime + jitterTime;
+ int64_t playedTimeRtp =
+ source->mFirstRtpTime + (((uint32_t)playedTime) * (source->mClockRate / 1000));
+ const uint32_t jitterTime =
+ (uint32_t)(source->mClockRate / ((float)1000 / (source->mJbTimeMs)));
+ uint32_t expiredTimeInJb = rtpTime + jitterTime;
bool isExpired = expiredTimeInJb <= (playedTimeRtp);
bool isTooLate200 = expiredTimeInJb < (playedTimeRtp - jitterTime);
bool isTooLate300 = expiredTimeInJb < (playedTimeRtp - (jitterTime * 3 / 2));
if (mShowQueue && mShowQueueCnt < 20) {
showCurrentQueue(queue);
- ALOGD("start=%lld, now=%lld, played=%lld", (long long)startTime,
- (long long)nowTime, (long long)playedTime);
- ALOGD("rtp-time(JB)=%d, played-rtp-time(JB)=%d, expired-rtp-time(JB)=%d isExpired=%d",
- rtpTime, playedTimeRtp, expiredTimeInJb, isExpired);
+ printNowTimeUs(startTime, nowTime, playedTime);
+ printRTPTime(rtpTime, playedTimeRtp, expiredTimeInJb, isExpired);
mShowQueueCnt++;
}
- ALOGV("start=%lld, now=%lld, played=%lld", (long long)startTime,
- (long long)nowTime, (long long)playedTime);
- ALOGV("rtp-time(JB)=%d, played-rtp-time(JB)=%d, expired-rtp-time(JB)=%d isExpired=%d",
- rtpTime, playedTimeRtp, expiredTimeInJb, isExpired);
+ AAVCAssembler::addNack(source);
if (!isExpired) {
ALOGV("buffering in jitter buffer.");
return NOT_ENOUGH_DATA;
}
- if (isTooLate200)
+ if (isTooLate200) {
ALOGW("=== WARNING === buffer arrived 200ms late. === WARNING === ");
+ }
if (isTooLate300) {
- ALOGW("buffer arrived too late. 300ms..");
- ALOGW("start=%lld, now=%lld, played=%lld", (long long)startTime,
- (long long)nowTime, (long long)playedTime);
- ALOGW("rtp-time(JB)=%d, plyed-rtp-time(JB)=%d, exp-rtp-time(JB)=%d diff=%lld isExpired=%d",
- rtpTime, playedTimeRtp, expiredTimeInJb,
- ((long long)playedTimeRtp) - expiredTimeInJb, isExpired);
- ALOGW("expected Seq. NO =%d", buffer->int32Data());
+ ALOGW("buffer arrived after 300ms ... \t Diff in Jb=%lld \t Seq# %d",
+ ((long long)playedTimeRtp) - expiredTimeInJb, buffer->int32Data());
+ printNowTimeUs(startTime, nowTime, playedTime);
+ printRTPTime(rtpTime, playedTimeRtp, expiredTimeInJb, isExpired);
- List<sp<ABuffer> >::iterator it = queue->begin();
- while (it != queue->end()) {
- CHECK((*it)->meta()->findInt32("rtp-time", &rtpTime));
- if (rtpTime + jitterTime >= playedTimeRtp) {
- mNextExpectedSeqNo = (*it)->int32Data();
- break;
- }
- it++;
- }
- source->noticeAbandonBuffer();
+ mNextExpectedSeqNo = pickProperSeq(queue, jitterTime, playedTimeRtp);
}
if (mNextExpectedSeqNoValid) {
int32_t size = queue->size();
- int32_t cnt = 0;
- List<sp<ABuffer> >::iterator it = queue->begin();
- while (it != queue->end()) {
- if ((uint32_t)(*it)->int32Data() >= mNextExpectedSeqNo) {
- break;
- }
+ int32_t cntRemove = deleteUnitUnderSeq(queue, mNextExpectedSeqNo);
- it = queue->erase(it);
- cnt++;
- }
-
- if (cnt > 0) {
- source->noticeAbandonBuffer(cnt);
- ALOGW("delete %d of %d buffers", cnt, size);
+ if (cntRemove > 0) {
+ source->noticeAbandonBuffer(cntRemove);
+ ALOGW("delete %d of %d buffers", cntRemove, size);
}
if (queue->empty()) {
return NOT_ENOUGH_DATA;
@@ -187,12 +224,30 @@
}
}
+void AAVCAssembler::checkIFrameProvided(const sp<ABuffer> &buffer) {
+ if (buffer->size() == 0) {
+ return;
+ }
+ const uint8_t *data = buffer->data();
+ unsigned nalType = data[0] & 0x1f;
+ if (nalType == 0x5) {
+ mFirstIFrameProvided = true;
+ mLastIFrameProvidedAtMs = ALooper::GetNowUs() / 1000;
+
+ uint32_t rtpTime;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+ ALOGD("got First I-frame to be decoded. rtpTime=%u, size=%zu", rtpTime, buffer->size());
+ }
+}
+
void AAVCAssembler::addSingleNALUnit(const sp<ABuffer> &buffer) {
ALOGV("addSingleNALUnit of size %zu", buffer->size());
#if !LOG_NDEBUG
hexdump(buffer->data(), buffer->size());
#endif
+ checkIFrameProvided(buffer);
+
uint32_t rtpTime;
CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
@@ -280,6 +335,11 @@
size_t totalCount = 1;
bool complete = false;
+ uint32_t rtpTimeStartAt;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTimeStartAt));
+ uint32_t startSeqNo = buffer->int32Data();
+ bool pFrame = nalType == 0x1;
+
if (data[1] & 0x40) {
// Huh? End bit also set on the first buffer.
@@ -288,6 +348,8 @@
complete = true;
} else {
List<sp<ABuffer> >::iterator it = ++queue->begin();
+ int32_t connected = 1;
+ bool snapped = false;
while (it != queue->end()) {
ALOGV("sequence length %zu", totalCount);
@@ -297,26 +359,32 @@
size_t size = buffer->size();
if ((uint32_t)buffer->int32Data() != expectedSeqNo) {
- ALOGV("sequence not complete, expected seqNo %d, got %d",
- expectedSeqNo, (uint32_t)buffer->int32Data());
+ ALOGD("sequence not complete, expected seqNo %u, got %u, nalType %u",
+ expectedSeqNo, (unsigned)buffer->int32Data(), nalType);
+ snapped = true;
- return WRONG_SEQUENCE_NUMBER;
+ if (!pFrame) {
+ return WRONG_SEQUENCE_NUMBER;
+ }
}
+ if (!snapped) {
+ connected++;
+ }
+
+ uint32_t rtpTime;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
if (size < 2
|| data[0] != indicator
|| (data[1] & 0x1f) != nalType
- || (data[1] & 0x80)) {
+ || (data[1] & 0x80)
+ || rtpTime != rtpTimeStartAt) {
ALOGV("Ignoring malformed FU buffer.");
// Delete the whole start of the FU.
- it = queue->begin();
- for (size_t i = 0; i <= totalCount; ++i) {
- it = queue->erase(it);
- }
-
mNextExpectedSeqNo = expectedSeqNo + 1;
+ deleteUnitUnderSeq(queue, mNextExpectedSeqNo);
return MALFORMED_PACKET;
}
@@ -324,9 +392,17 @@
totalSize += size - 2;
++totalCount;
- expectedSeqNo = expectedSeqNo + 1;
+ expectedSeqNo = (uint32_t)buffer->int32Data() + 1;
if (data[1] & 0x40) {
+ if (pFrame && !recycleUnit(startSeqNo, expectedSeqNo,
+ connected, totalCount, 0.5f)) {
+ mNextExpectedSeqNo = expectedSeqNo;
+ deleteUnitUnderSeq(queue, mNextExpectedSeqNo);
+
+ return MALFORMED_PACKET;
+ }
+
// This is the last fragment.
complete = true;
break;
@@ -433,22 +509,78 @@
msg->post();
}
+int32_t AAVCAssembler::pickProperSeq(const Queue *queue, uint32_t jit, int64_t play) {
+ sp<ABuffer> buffer = *(queue->begin());
+ uint32_t rtpTime;
+ int32_t nextSeqNo = buffer->int32Data();
+
+ Queue::const_iterator it = queue->begin();
+ while (it != queue->end()) {
+ CHECK((*it)->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+ // if pkt in time exists, that should be the next pivot
+ if (rtpTime + jit >= play) {
+ nextSeqNo = (*it)->int32Data();
+ break;
+ }
+ it++;
+ }
+ return nextSeqNo;
+}
+
+bool AAVCAssembler::recycleUnit(uint32_t start, uint32_t end, uint32_t connected,
+ size_t avail, float goodRatio) {
+ float total = end - start;
+ float valid = connected;
+ float exist = avail;
+ bool isRecycle = (valid / total) >= goodRatio;
+
+ ALOGV("checking p-frame losses.. recvBufs %f valid %f diff %f recycle? %d",
+ exist, valid, total, isRecycle);
+
+ return isRecycle;
+}
+
+int32_t AAVCAssembler::deleteUnitUnderSeq(Queue *queue, uint32_t seq) {
+ int32_t initSize = queue->size();
+ Queue::iterator it = queue->begin();
+ while (it != queue->end()) {
+ if ((uint32_t)(*it)->int32Data() >= seq) {
+ break;
+ }
+ it++;
+ }
+ queue->erase(queue->begin(), it);
+ return initSize - queue->size();
+}
+
+inline void AAVCAssembler::printNowTimeUs(int64_t start, int64_t now, int64_t play) {
+ ALOGD("start=%lld, now=%lld, played=%lld",
+ (long long)start, (long long)now, (long long)play);
+}
+
+inline void AAVCAssembler::printRTPTime(uint32_t rtp, int64_t play, uint32_t exp, bool isExp) {
+ ALOGD("rtp-time(JB)=%u, played-rtp-time(JB)=%lld, expired-rtp-time(JB)=%u isExpired=%d",
+ rtp, (long long)play, exp, isExp);
+}
+
ARTPAssembler::AssemblyStatus AAVCAssembler::assembleMore(
const sp<ARTPSource> &source) {
AssemblyStatus status = addNALUnit(source);
if (status == MALFORMED_PACKET) {
- mAccessUnitDamaged = true;
+ uint64_t msecsSinceLastIFrame = (ALooper::GetNowUs() / 1000) - mLastIFrameProvidedAtMs;
+ if (msecsSinceLastIFrame > 1000) {
+ ALOGV("request FIR to get a new I-Frame, time since "
+ "last I-Frame %llu ms", (unsigned long long)msecsSinceLastIFrame);
+ source->onIssueFIRByAssembler();
+ }
}
return status;
}
void AAVCAssembler::packetLost() {
CHECK(mNextExpectedSeqNoValid);
- ALOGV("packetLost (expected %d)", mNextExpectedSeqNo);
-
+ ALOGD("packetLost (expected %u)", mNextExpectedSeqNo);
++mNextExpectedSeqNo;
-
- mAccessUnitDamaged = true;
}
void AAVCAssembler::onByeReceived() {
diff --git a/media/libstagefright/rtsp/AAVCAssembler.h b/media/libstagefright/rtsp/AAVCAssembler.h
index e19480c..913a868 100644
--- a/media/libstagefright/rtsp/AAVCAssembler.h
+++ b/media/libstagefright/rtsp/AAVCAssembler.h
@@ -31,6 +31,7 @@
struct AAVCAssembler : public ARTPAssembler {
explicit AAVCAssembler(const sp<AMessage> ¬ify);
+ typedef List<sp<ABuffer> > Queue;
protected:
virtual ~AAVCAssembler();
@@ -45,8 +46,12 @@
bool mNextExpectedSeqNoValid;
uint32_t mNextExpectedSeqNo;
bool mAccessUnitDamaged;
+ bool mFirstIFrameProvided;
+ uint64_t mLastIFrameProvidedAtMs;
List<sp<ABuffer> > mNALUnits;
+ int32_t addNack(const sp<ARTPSource> &source);
+ void checkIFrameProvided(const sp<ABuffer> &buffer);
AssemblyStatus addNALUnit(const sp<ARTPSource> &source);
void addSingleNALUnit(const sp<ABuffer> &buffer);
AssemblyStatus addFragmentedNALUnit(List<sp<ABuffer> > *queue);
@@ -54,6 +59,13 @@
void submitAccessUnit();
+ int32_t pickProperSeq(const Queue *q, uint32_t jit, int64_t play);
+ bool recycleUnit(uint32_t start, uint32_t end, uint32_t connected,
+ size_t avail, float goodRatio);
+ int32_t deleteUnitUnderSeq(Queue *q, uint32_t seq);
+ void printNowTimeUs(int64_t start, int64_t now, int64_t play);
+ void printRTPTime(uint32_t rtp, int64_t play, uint32_t exp, bool isExp);
+
DISALLOW_EVIL_CONSTRUCTORS(AAVCAssembler);
};
diff --git a/media/libstagefright/rtsp/AHEVCAssembler.cpp b/media/libstagefright/rtsp/AHEVCAssembler.cpp
index 93869fb..148a0ba 100644
--- a/media/libstagefright/rtsp/AHEVCAssembler.cpp
+++ b/media/libstagefright/rtsp/AHEVCAssembler.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_NDEBUG 0
+//#define LOG_NDEBUG 0
#define LOG_TAG "AHEVCAssembler"
#include <utils/Log.h>
@@ -25,6 +25,7 @@
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
+#include <include/HevcUtils.h>
#include <media/stagefright/foundation/hexdump.h>
#include <stdint.h>
@@ -46,7 +47,11 @@
mAccessUnitRTPTime(0),
mNextExpectedSeqNoValid(false),
mNextExpectedSeqNo(0),
- mAccessUnitDamaged(false) {
+ mAccessUnitDamaged(false),
+ mFirstIFrameProvided(false),
+ mLastIFrameProvidedAtMs(0),
+ mWidth(0),
+ mHeight(0) {
ALOGV("Constructor");
}
@@ -54,6 +59,66 @@
AHEVCAssembler::~AHEVCAssembler() {
}
+int32_t AHEVCAssembler::addNack(
+ const sp<ARTPSource> &source) {
+ List<sp<ABuffer>> *queue = source->queue();
+ int32_t nackCount = 0;
+
+ List<sp<ABuffer> >::iterator it = queue->begin();
+
+ if (it == queue->end()) {
+ return nackCount /* 0 */;
+ }
+
+ uint16_t queueHeadSeqNum = (*it)->int32Data();
+
+ // move to the packet after which RTCP:NACK was sent.
+ for (; it != queue->end(); ++it) {
+ int32_t seqNum = (*it)->int32Data();
+ if (seqNum >= source->mHighestNackNumber) {
+ break;
+ }
+ }
+
+ int32_t nackStartAt = -1;
+
+ while (it != queue->end()) {
+ int32_t seqBeforeLast = (*it)->int32Data();
+ // increase iterator.
+ if ((++it) == queue->end()) {
+ break;
+ }
+
+ int32_t seqLast = (*it)->int32Data();
+
+ if ((seqLast - seqBeforeLast) < 0) {
+ ALOGD("addNack: found end of seqNum from(%d) to(%d)", seqBeforeLast, seqLast);
+ source->mHighestNackNumber = 0;
+ }
+
+ // missed packet found
+ if (seqLast > (seqBeforeLast + 1) &&
+ // we didn't send RTCP:NACK for this packet yet.
+ (seqLast - 1) > source->mHighestNackNumber) {
+ source->mHighestNackNumber = seqLast -1;
+ nackStartAt = seqBeforeLast + 1;
+ break;
+ }
+
+ }
+
+ if (nackStartAt != -1) {
+ nackCount = source->mHighestNackNumber - nackStartAt + 1;
+ ALOGD("addNack: nackCount=%d, nackFrom=%d, nackTo=%d", nackCount,
+ nackStartAt, source->mHighestNackNumber);
+
+ uint16_t mask = (uint16_t)(0xffff) >> (16 - nackCount + 1);
+ source->setSeqNumToNACK(nackStartAt, mask, queueHeadSeqNum);
+ }
+
+ return nackCount;
+}
+
ARTPAssembler::AssemblyStatus AHEVCAssembler::addNALUnit(
const sp<ARTPSource> &source) {
List<sp<ABuffer> > *queue = source->queue();
@@ -63,33 +128,54 @@
}
sp<ABuffer> buffer = *queue->begin();
- int32_t rtpTime;
- CHECK(buffer->meta()->findInt32("rtp-time", &rtpTime));
+ buffer->meta()->setObject("source", source);
+ uint32_t rtpTime;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
int64_t startTime = source->mFirstSysTime / 1000;
int64_t nowTime = ALooper::GetNowUs() / 1000;
int64_t playedTime = nowTime - startTime;
- int32_t playedTimeRtp = source->mFirstRtpTime +
+ int64_t playedTimeRtp = source->mFirstRtpTime +
(((uint32_t)playedTime) * (source->mClockRate / 1000));
- int32_t expiredTimeInJb = rtpTime + (source->mClockRate / 5);
+ const uint32_t jitterTime = (uint32_t)(source->mClockRate / ((float)1000 / (source->mJbTimeMs)));
+ uint32_t expiredTimeInJb = rtpTime + jitterTime;
bool isExpired = expiredTimeInJb <= (playedTimeRtp);
- ALOGV("start=%lld, now=%lld, played=%lld", (long long)startTime,
- (long long)nowTime, (long long)playedTime);
- ALOGV("rtp-time(JB)=%d, played-rtp-time(JB)=%d, expired-rtp-time(JB)=%d isExpired=%d",
- rtpTime, playedTimeRtp, expiredTimeInJb, isExpired);
+ bool isTooLate200 = expiredTimeInJb < (playedTimeRtp - jitterTime);
+ bool isTooLate300 = expiredTimeInJb < (playedTimeRtp - (jitterTime * 3 / 2));
+
+ if (mShowQueueCnt < 20) {
+ showCurrentQueue(queue);
+ printNowTimeUs(startTime, nowTime, playedTime);
+ printRTPTime(rtpTime, playedTimeRtp, expiredTimeInJb, isExpired);
+ mShowQueueCnt++;
+ }
+
+ AHEVCAssembler::addNack(source);
if (!isExpired) {
ALOGV("buffering in jitter buffer.");
return NOT_ENOUGH_DATA;
}
- if (mNextExpectedSeqNoValid) {
- List<sp<ABuffer> >::iterator it = queue->begin();
- while (it != queue->end()) {
- if ((uint32_t)(*it)->int32Data() >= mNextExpectedSeqNo) {
- break;
- }
+ if (isTooLate200) {
+ ALOGW("=== WARNING === buffer arrived 200ms late. === WARNING === ");
+ }
- it = queue->erase(it);
+ if (isTooLate300) {
+ ALOGW("buffer arrived after 300ms ... \t Diff in Jb=%lld \t Seq# %d",
+ ((long long)playedTimeRtp) - expiredTimeInJb, buffer->int32Data());
+ printNowTimeUs(startTime, nowTime, playedTime);
+ printRTPTime(rtpTime, playedTimeRtp, expiredTimeInJb, isExpired);
+
+ mNextExpectedSeqNo = pickProperSeq(queue, jitterTime, playedTimeRtp);
+ }
+
+ if (mNextExpectedSeqNoValid) {
+ int32_t size = queue->size();
+ int32_t cntRemove = deleteUnitUnderSeq(queue, mNextExpectedSeqNo);
+
+ if (cntRemove > 0) {
+ source->noticeAbandonBuffer(cntRemove);
+ ALOGW("delete %d of %d buffers", cntRemove, size);
}
if (queue->empty()) {
@@ -154,15 +240,74 @@
}
}
+void AHEVCAssembler::checkSpsUpdated(const sp<ABuffer> &buffer) {
+ if (buffer->size() == 0) {
+ return;
+ }
+ const uint8_t *data = buffer->data();
+ HevcParameterSets paramSets;
+ unsigned nalType = (data[0] >> 1) & H265_NALU_MASK;
+ if (nalType == H265_NALU_SPS) {
+ int32_t width = 0, height = 0;
+ paramSets.FindHEVCDimensions(buffer, &width, &height);
+ ALOGV("existing resolution (%u x %u)", mWidth, mHeight);
+ if (width != mWidth || height != mHeight) {
+ mFirstIFrameProvided = false;
+ mWidth = width;
+ mHeight = height;
+ ALOGD("found a new resolution (%u x %u)", mWidth, mHeight);
+ }
+ }
+}
+
+void AHEVCAssembler::checkIFrameProvided(const sp<ABuffer> &buffer) {
+ if (buffer->size() == 0) {
+ return;
+ }
+ const uint8_t *data = buffer->data();
+ unsigned nalType = (data[0] >> 1) & H265_NALU_MASK;
+ if (nalType > 0x0F && nalType < 0x18) {
+ mLastIFrameProvidedAtMs = ALooper::GetNowUs() / 1000;
+ if (!mFirstIFrameProvided) {
+ mFirstIFrameProvided = true;
+ uint32_t rtpTime;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+ ALOGD("got First I-frame to be decoded. rtpTime=%d, size=%zu", rtpTime, buffer->size());
+ }
+ }
+}
+
+bool AHEVCAssembler::dropFramesUntilIframe(const sp<ABuffer> &buffer) {
+ if (buffer->size() == 0) {
+ return false;
+ }
+ const uint8_t *data = buffer->data();
+ unsigned nalType = (data[0] >> 1) & H265_NALU_MASK;
+ return !mFirstIFrameProvided && nalType < 0x10;
+}
+
void AHEVCAssembler::addSingleNALUnit(const sp<ABuffer> &buffer) {
ALOGV("addSingleNALUnit of size %zu", buffer->size());
#if !LOG_NDEBUG
hexdump(buffer->data(), buffer->size());
#endif
+ checkSpsUpdated(buffer);
+ checkIFrameProvided(buffer);
uint32_t rtpTime;
CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+ if (dropFramesUntilIframe(buffer)) {
+ sp<ARTPSource> source = nullptr;
+ buffer->meta()->findObject("source", (sp<android::RefBase>*)&source);
+ if (source != nullptr) {
+ ALOGD("Issued FIR to get the I-frame");
+ source->onIssueFIRByAssembler();
+ }
+ ALOGD("drop P-frames till an I-frame provided. rtpTime %u", rtpTime);
+ return;
+ }
+
if (!mNALUnits.empty() && rtpTime != mAccessUnitRTPTime) {
submitAccessUnit();
}
@@ -260,6 +405,11 @@
size_t totalCount = 1;
bool complete = false;
+ uint32_t rtpTimeStartAt;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTimeStartAt));
+ uint32_t startSeqNo = buffer->int32Data();
+ bool pFrame = (nalType < 0x10);
+
if (data[2] & 0x40) {
// Huh? End bit also set on the first buffer.
@@ -268,6 +418,8 @@
complete = true;
} else {
List<sp<ABuffer> >::iterator it = ++queue->begin();
+ int32_t connected = 1;
+ bool snapped = false;
while (it != queue->end()) {
ALOGV("sequence length %zu", totalCount);
@@ -277,26 +429,32 @@
size_t size = buffer->size();
if ((uint32_t)buffer->int32Data() != expectedSeqNo) {
- ALOGV("sequence not complete, expected seqNo %d, got %d",
- expectedSeqNo, (uint32_t)buffer->int32Data());
+ ALOGV("sequence not complete, expected seqNo %u, got %u, nalType %u",
+ expectedSeqNo, (uint32_t)buffer->int32Data(), nalType);
+ snapped = true;
- return WRONG_SEQUENCE_NUMBER;
+ if (!pFrame) {
+ return WRONG_SEQUENCE_NUMBER;
+ }
}
+ if (!snapped) {
+ connected++;
+ }
+
+ uint32_t rtpTime;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
if (size < 3
|| ((data[0] >> 1) & H265_NALU_MASK) != indicator
|| (data[2] & H265_NALU_MASK) != nalType
- || (data[2] & 0x80)) {
+ || (data[2] & 0x80)
+ || rtpTime != rtpTimeStartAt) {
ALOGV("Ignoring malformed FU buffer.");
// Delete the whole start of the FU.
- it = queue->begin();
- for (size_t i = 0; i <= totalCount; ++i) {
- it = queue->erase(it);
- }
-
mNextExpectedSeqNo = expectedSeqNo + 1;
+ deleteUnitUnderSeq(queue, mNextExpectedSeqNo);
return MALFORMED_PACKET;
}
@@ -304,9 +462,16 @@
totalSize += size - 3;
++totalCount;
- expectedSeqNo = expectedSeqNo + 1;
+ expectedSeqNo = (uint32_t)buffer->int32Data() + 1;
if (data[2] & 0x40) {
+ if (pFrame && !recycleUnit(startSeqNo, expectedSeqNo,
+ connected, totalCount, 0.5f)) {
+ mNextExpectedSeqNo = expectedSeqNo;
+ deleteUnitUnderSeq(queue, mNextExpectedSeqNo);
+
+ return MALFORMED_PACKET;
+ }
// This is the last fragment.
complete = true;
break;
@@ -335,6 +500,7 @@
unit->data()[1] = tid;
size_t offset = 2;
+ int32_t cvo = -1;
List<sp<ABuffer> >::iterator it = queue->begin();
for (size_t i = 0; i < totalCount; ++i) {
const sp<ABuffer> &buffer = *it;
@@ -345,6 +511,7 @@
#endif
memcpy(unit->data() + offset, buffer->data() + 3, buffer->size() - 3);
+ buffer->meta()->findInt32("cvo", &cvo);
offset += buffer->size() - 3;
it = queue->erase(it);
@@ -352,6 +519,10 @@
unit->setRange(0, totalSize);
+ if (cvo >= 0) {
+ unit->meta()->setInt32("cvo", cvo);
+ }
+
addSingleNALUnit(unit);
ALOGV("successfully assembled a NAL unit from fragments.");
@@ -372,6 +543,7 @@
sp<ABuffer> accessUnit = new ABuffer(totalSize);
size_t offset = 0;
+ int32_t cvo = -1;
for (List<sp<ABuffer> >::iterator it = mNALUnits.begin();
it != mNALUnits.end(); ++it) {
memcpy(accessUnit->data() + offset, "\x00\x00\x00\x01", 4);
@@ -380,6 +552,7 @@
sp<ABuffer> nal = *it;
memcpy(accessUnit->data() + offset, nal->data(), nal->size());
offset += nal->size();
+ nal->meta()->findInt32("cvo", &cvo);
}
CopyTimes(accessUnit, *mNALUnits.begin());
@@ -388,6 +561,9 @@
printf(mAccessUnitDamaged ? "X" : ".");
fflush(stdout);
#endif
+ if (cvo >= 0) {
+ accessUnit->meta()->setInt32("cvo", cvo);
+ }
if (mAccessUnitDamaged) {
accessUnit->meta()->setInt32("damaged", true);
@@ -401,22 +577,80 @@
msg->post();
}
+int32_t AHEVCAssembler::pickProperSeq(const Queue *queue, uint32_t jit, int64_t play) {
+ sp<ABuffer> buffer = *(queue->begin());
+ uint32_t rtpTime;
+ int32_t nextSeqNo = buffer->int32Data();
+
+ Queue::const_iterator it = queue->begin();
+ while (it != queue->end()) {
+ CHECK((*it)->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+ // if pkt in time exists, that should be the next pivot
+ if (rtpTime + jit >= play) {
+ nextSeqNo = (*it)->int32Data();
+ break;
+ }
+ it++;
+ }
+ return nextSeqNo;
+}
+
+bool AHEVCAssembler::recycleUnit(uint32_t start, uint32_t end, uint32_t connected,
+ size_t avail, float goodRatio) {
+ float total = end - start;
+ float valid = connected;
+ float exist = avail;
+ bool isRecycle = (valid / total) >= goodRatio;
+
+ ALOGV("checking p-frame losses.. recvBufs %f valid %f diff %f recycle? %d",
+ exist, valid, total, isRecycle);
+
+ return isRecycle;
+}
+
+int32_t AHEVCAssembler::deleteUnitUnderSeq(Queue *queue, uint32_t seq) {
+ int32_t initSize = queue->size();
+ Queue::iterator it = queue->begin();
+ while (it != queue->end()) {
+ if ((uint32_t)(*it)->int32Data() >= seq) {
+ break;
+ }
+ it++;
+ }
+ queue->erase(queue->begin(), it);
+ return initSize - queue->size();
+}
+
+inline void AHEVCAssembler::printNowTimeUs(int64_t start, int64_t now, int64_t play) {
+ ALOGD("start=%lld, now=%lld, played=%lld",
+ (long long)start, (long long)now, (long long)play);
+}
+
+inline void AHEVCAssembler::printRTPTime(uint32_t rtp, int64_t play, uint32_t exp, bool isExp) {
+ ALOGD("rtp-time(JB)=%u, played-rtp-time(JB)=%lld, expired-rtp-time(JB)=%u isExpired=%d",
+ rtp, (long long)play, exp, isExp);
+}
+
+
ARTPAssembler::AssemblyStatus AHEVCAssembler::assembleMore(
const sp<ARTPSource> &source) {
AssemblyStatus status = addNALUnit(source);
if (status == MALFORMED_PACKET) {
- mAccessUnitDamaged = true;
+ uint64_t msecsSinceLastIFrame = (ALooper::GetNowUs() / 1000) - mLastIFrameProvidedAtMs;
+ if (msecsSinceLastIFrame > 1000) {
+ ALOGV("request FIR to get a new I-Frame, time after "
+ "last I-Frame in %llu ms", (unsigned long long)msecsSinceLastIFrame);
+ source->onIssueFIRByAssembler();
+ }
}
return status;
}
void AHEVCAssembler::packetLost() {
CHECK(mNextExpectedSeqNoValid);
- ALOGV("packetLost (expected %d)", mNextExpectedSeqNo);
+ ALOGD("packetLost (expected %u)", mNextExpectedSeqNo);
++mNextExpectedSeqNo;
-
- mAccessUnitDamaged = true;
}
void AHEVCAssembler::onByeReceived() {
diff --git a/media/libstagefright/rtsp/AHEVCAssembler.h b/media/libstagefright/rtsp/AHEVCAssembler.h
index cc20622..16fc1c8 100644
--- a/media/libstagefright/rtsp/AHEVCAssembler.h
+++ b/media/libstagefright/rtsp/AHEVCAssembler.h
@@ -31,6 +31,8 @@
struct AHEVCAssembler : public ARTPAssembler {
AHEVCAssembler(const sp<AMessage> ¬ify);
+ typedef List<sp<ABuffer> > Queue;
+
protected:
virtual ~AHEVCAssembler();
@@ -45,8 +47,16 @@
bool mNextExpectedSeqNoValid;
uint32_t mNextExpectedSeqNo;
bool mAccessUnitDamaged;
+ bool mFirstIFrameProvided;
+ uint64_t mLastIFrameProvidedAtMs;
+ int32_t mWidth;
+ int32_t mHeight;
List<sp<ABuffer> > mNALUnits;
+ int32_t addNack(const sp<ARTPSource> &source);
+ void checkSpsUpdated(const sp<ABuffer> &buffer);
+ void checkIFrameProvided(const sp<ABuffer> &buffer);
+ bool dropFramesUntilIframe(const sp<ABuffer> &buffer);
AssemblyStatus addNALUnit(const sp<ARTPSource> &source);
void addSingleNALUnit(const sp<ABuffer> &buffer);
AssemblyStatus addFragmentedNALUnit(List<sp<ABuffer> > *queue);
@@ -54,6 +64,13 @@
void submitAccessUnit();
+ int32_t pickProperSeq(const Queue *queue, uint32_t jit, int64_t play);
+ bool recycleUnit(uint32_t start, uint32_t end, uint32_t conneceted,
+ size_t avail, float goodRatio);
+ int32_t deleteUnitUnderSeq(Queue *queue, uint32_t seq);
+ void printNowTimeUs(int64_t start, int64_t now, int64_t play);
+ void printRTPTime(uint32_t rtp, int64_t play, uint32_t exp, bool isExp);
+
DISALLOW_EVIL_CONSTRUCTORS(AHEVCAssembler);
};
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 1346c9a..f57077c 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -78,7 +78,9 @@
: mFlags(flags),
mPollEventPending(false),
mLastReceiverReportTimeUs(-1),
- mLastBitrateReportTimeUs(-1) {
+ mLastBitrateReportTimeUs(-1),
+ mTargetBitrate(-1),
+ mJbTimeMs(300) {
}
ARTPConnection::~ARTPConnection() {
@@ -439,6 +441,24 @@
continue;
}
+ // add NACK and FIR that needs to be sent immediately.
+ sp<ABuffer> buffer = new ABuffer(kMaxUDPSize);
+ for (size_t i = 0; i < it->mSources.size(); ++i) {
+ buffer->setRange(0, 0);
+ int cnt = it->mSources.valueAt(i)->addNACK(buffer);
+ if (cnt > 0) {
+ ALOGV("Send NACK for lost %d Packets", cnt);
+ send(&*it, buffer);
+ }
+
+ buffer->setRange(0, 0);
+ it->mSources.valueAt(i)->addFIR(buffer);
+ if (buffer->size() > 0) {
+ ALOGD("Send FIR immediately for lost Packets");
+ send(&*it, buffer);
+ }
+ }
+
++it;
}
}
@@ -524,8 +544,9 @@
(!receiveRTP && s->mNumRTCPPacketsReceived == 0)
? sizeSockSt : 0;
- if (mFlags & kViLTEConnection)
+ if (mFlags & kViLTEConnection) {
remoteAddrLen = 0;
+ }
ssize_t nbytes;
do {
@@ -1012,8 +1033,12 @@
source = new ARTPSource(
srcId, info->mSessionDesc, info->mIndex, info->mNotifyMsg);
+ if (mFlags & kViLTEConnection) {
+ source->setPeriodicFIR(false);
+ }
+
source->setSelfID(mSelfID);
- source->setMinMaxBitrate(mMinBitrate, mMaxBitrate);
+ source->setJbTime(mJbTimeMs > 0 ? mJbTimeMs : 300);
info->mSources.add(srcId, source);
} else {
source = info->mSources.valueAt(index);
@@ -1033,9 +1058,12 @@
mSelfID = selfID;
}
-void ARTPConnection::setMinMaxBitrate(int32_t min, int32_t max) {
- mMinBitrate = min;
- mMaxBitrate = max;
+void ARTPConnection::setJbTime(const uint32_t jbTimeMs) {
+ mJbTimeMs = jbTimeMs;
+}
+
+void ARTPConnection::setTargetBitrate(int32_t targetBitrate) {
+ mTargetBitrate = targetBitrate;
}
void ARTPConnection::checkRxBitrate(int64_t nowUs) {
@@ -1068,17 +1096,8 @@
for (size_t i = 0; i < s->mSources.size(); ++i) {
sp<ARTPSource> source = s->mSources.valueAt(i);
- source->setBitrateData(bitrate, nowUs);
- source->setTargetBitrate();
- source->addTMMBR(buffer);
- if (source->isNeedToDowngrade()) {
- sp<AMessage> notify = s->mNotifyMsg->dup();
- notify->setInt32("rtcp-event", 1);
- notify->setInt32("payload-type", 400);
- notify->setInt32("feedback-type", 1);
- notify->setInt32("sender", source->getSelfID());
- notify->post();
- }
+ source->notifyPktInfo(bitrate, nowUs);
+ source->addTMMBR(buffer, mTargetBitrate);
}
if (buffer->size() > 0) {
ALOGV("Sending TMMBR...");
diff --git a/media/libstagefright/rtsp/ARTPConnection.h b/media/libstagefright/rtsp/ARTPConnection.h
index 712eec5..7c8218f 100644
--- a/media/libstagefright/rtsp/ARTPConnection.h
+++ b/media/libstagefright/rtsp/ARTPConnection.h
@@ -46,7 +46,8 @@
void injectPacket(int index, const sp<ABuffer> &buffer);
void setSelfID(const uint32_t selfID);
- void setMinMaxBitrate(int32_t min, int32_t max);
+ void setJbTime(const uint32_t jbTimeMs);
+ void setTargetBitrate(int32_t targetBitrate);
// Creates a pair of UDP datagram sockets bound to adjacent ports
// (the rtpSocket is bound to an even port, the rtcpSocket to the
@@ -85,9 +86,10 @@
int64_t mLastBitrateReportTimeUs;
int32_t mSelfID;
+ int32_t mTargetBitrate;
- int32_t mMinBitrate;
- int32_t mMaxBitrate;
+ uint32_t mJbTimeMs;
+
int32_t mCumulativeBytes;
void onAddStream(const sp<AMessage> &msg);
diff --git a/media/libstagefright/rtsp/ARTPSource.cpp b/media/libstagefright/rtsp/ARTPSource.cpp
index bbe9d94..6303fc4 100644
--- a/media/libstagefright/rtsp/ARTPSource.cpp
+++ b/media/libstagefright/rtsp/ARTPSource.cpp
@@ -46,15 +46,21 @@
mFirstRtpTime(0),
mFirstSysTime(0),
mClockRate(0),
+ mJbTimeMs(300), // default jitter buffer time is 300ms.
+ mFirstSsrc(0),
+ mHighestNackNumber(0),
mID(id),
mHighestSeqNumber(0),
mPrevExpected(0),
mBaseSeqNumber(0),
mNumBuffersReceived(0),
mPrevNumBuffersReceived(0),
+ mPrevExpectedForRR(0),
+ mPrevNumBuffersReceivedForRR(0),
mLastNTPTime(0),
mLastNTPTimeUpdateUs(0),
mIssueFIRRequests(false),
+ mIssueFIRByAssembler(false),
mLastFIRRequestUs(-1),
mNextFIRSeqNo((rand() * 256.0) / RAND_MAX),
mNotify(notify) {
@@ -120,20 +126,29 @@
bool ARTPSource::queuePacket(const sp<ABuffer> &buffer) {
uint32_t seqNum = (uint32_t)buffer->int32Data();
+ int32_t ssrc = 0;
+ buffer->meta()->findInt32("ssrc", &ssrc);
+
if (mNumBuffersReceived++ == 0 && mFirstSysTime == 0) {
- int32_t firstRtpTime;
- CHECK(buffer->meta()->findInt32("rtp-time", &firstRtpTime));
+ uint32_t firstRtpTime;
+ CHECK(buffer->meta()->findInt32("rtp-time", (int32_t *)&firstRtpTime));
mFirstSysTime = ALooper::GetNowUs();
mHighestSeqNumber = seqNum;
mBaseSeqNumber = seqNum;
mFirstRtpTime = firstRtpTime;
- ALOGV("first-rtp arrived: first-rtp-time=%d, sys-time=%lld, seq-num=%u",
- mFirstRtpTime, (long long)mFirstSysTime, mHighestSeqNumber);
+ mFirstSsrc = ssrc;
+ ALOGD("first-rtp arrived: first-rtp-time=%d, sys-time=%lld, seq-num=%u, ssrc=%d",
+ mFirstRtpTime, (long long)mFirstSysTime, mHighestSeqNumber, mFirstSsrc);
mClockRate = 90000;
mQueue.push_back(buffer);
return true;
}
+ if (mFirstSsrc != ssrc) {
+ ALOGW("Discarding a buffer due to unexpected ssrc");
+ return false;
+ }
+
// Only the lower 16-bit of the sequence numbers are transmitted,
// derive the high-order bits by choosing the candidate closest
// to the highest sequence number (extended to 32 bits) received so far.
@@ -196,20 +211,34 @@
}
void ARTPSource::addFIR(const sp<ABuffer> &buffer) {
- if (!mIssueFIRRequests) {
+ if (!mIssueFIRRequests && !mIssueFIRByAssembler) {
return;
}
+ bool send = false;
int64_t nowUs = ALooper::GetNowUs();
- if (mLastFIRRequestUs >= 0 && mLastFIRRequestUs + 5000000LL > nowUs) {
- // Send FIR requests at most every 5 secs.
+ int64_t usecsSinceLastFIR = nowUs - mLastFIRRequestUs;
+ if (mLastFIRRequestUs < 0) {
+ // A first FIR, just send it.
+ send = true;
+ } else if (mIssueFIRByAssembler && (usecsSinceLastFIR > 1000000)) {
+ // A FIR issued by Assembler.
+ // Send it if last FIR is not sent within a sec.
+ send = true;
+ } else if (mIssueFIRRequests && (usecsSinceLastFIR > 5000000)) {
+ // A FIR issued periodically reagardless packet loss.
+ // Send it if last FIR is not sent within 5 secs.
+ send = true;
+ }
+
+ if (!send) {
return;
}
mLastFIRRequestUs = nowUs;
if (buffer->size() + 20 > buffer->capacity()) {
- ALOGW("RTCP buffer too small to accomodate FIR.");
+ ALOGW("RTCP buffer too small to accommodate FIR.");
return;
}
@@ -218,7 +247,7 @@
data[0] = 0x80 | 4;
data[1] = 206; // PSFB
data[2] = 0;
- data[3] = 4;
+ data[3] = 4; // total (4+1) * sizeof(int32_t) = 20 bytes
data[4] = kSourceID >> 24;
data[5] = (kSourceID >> 16) & 0xff;
data[6] = (kSourceID >> 8) & 0xff;
@@ -240,14 +269,16 @@
data[18] = 0x00;
data[19] = 0x00;
- buffer->setRange(buffer->offset(), buffer->size() + 20);
+ buffer->setRange(buffer->offset(), buffer->size() + (data[3] + 1) * sizeof(int32_t));
+
+ mIssueFIRByAssembler = false;
ALOGV("Added FIR request.");
}
void ARTPSource::addReceiverReport(const sp<ABuffer> &buffer) {
if (buffer->size() + 32 > buffer->capacity()) {
- ALOGW("RTCP buffer too small to accomodate RR.");
+ ALOGW("RTCP buffer too small to accommodate RR.");
return;
}
@@ -255,16 +286,16 @@
// According to appendix A.3 in RFC 3550
uint32_t expected = mHighestSeqNumber - mBaseSeqNumber + 1;
- int64_t intervalExpected = expected - mPrevExpected;
- int64_t intervalReceived = mNumBuffersReceived - mPrevNumBuffersReceived;
+ int64_t intervalExpected = expected - mPrevExpectedForRR;
+ int64_t intervalReceived = mNumBuffersReceived - mPrevNumBuffersReceivedForRR;
int64_t intervalPacketLost = intervalExpected - intervalReceived;
if (intervalExpected > 0 && intervalPacketLost > 0) {
fraction = (intervalPacketLost << 8) / intervalExpected;
}
- mPrevExpected = expected;
- mPrevNumBuffersReceived = mNumBuffersReceived;
+ mPrevExpectedForRR = expected;
+ mPrevNumBuffersReceivedForRR = mNumBuffersReceived;
int32_t cumulativePacketLost = (int32_t)expected - mNumBuffersReceived;
uint8_t *data = buffer->data() + buffer->size();
@@ -272,7 +303,7 @@
data[0] = 0x80 | 1;
data[1] = 201; // RR
data[2] = 0;
- data[3] = 7;
+ data[3] = 7; // total (7+1) * sizeof(int32_t) = 32 bytes
data[4] = kSourceID >> 24;
data[5] = (kSourceID >> 16) & 0xff;
data[6] = (kSourceID >> 8) & 0xff;
@@ -318,18 +349,18 @@
data[30] = (DLSR >> 8) & 0xff;
data[31] = DLSR & 0xff;
- buffer->setRange(buffer->offset(), buffer->size() + 32);
+ buffer->setRange(buffer->offset(), buffer->size() + (data[3] + 1) * sizeof(int32_t));
}
-void ARTPSource::addTMMBR(const sp<ABuffer> &buffer) {
+void ARTPSource::addTMMBR(const sp<ABuffer> &buffer, int32_t targetBitrate) {
if (buffer->size() + 20 > buffer->capacity()) {
ALOGW("RTCP buffer too small to accommodate RR.");
return;
}
- int32_t targetBitrate = mQualManager.getTargetBitrate();
- if (targetBitrate <= 0)
+ if (targetBitrate <= 0) {
return;
+ }
uint8_t *data = buffer->data() + buffer->size();
@@ -363,52 +394,145 @@
data[18] = (mantissa & 0x0007f) << 1;
data[19] = 40; // 40 bytes overhead;
- buffer->setRange(buffer->offset(), buffer->size() + 20);
+ buffer->setRange(buffer->offset(), buffer->size() + (data[3] + 1) * sizeof(int32_t));
+}
+
+int ARTPSource::addNACK(const sp<ABuffer> &buffer) {
+ constexpr size_t kMaxFCIs = 10; // max number of FCIs
+ if (buffer->size() + (3 + kMaxFCIs) * sizeof(int32_t) > buffer->capacity()) {
+ ALOGW("RTCP buffer too small to accommodate NACK.");
+ return -1;
+ }
+
+ uint8_t *data = buffer->data() + buffer->size();
+
+ data[0] = 0x80 | 1; // Generic NACK
+ data[1] = 205; // TSFB
+ data[2] = 0;
+ data[3] = 0; // will be decided later
+ data[4] = kSourceID >> 24;
+ data[5] = (kSourceID >> 16) & 0xff;
+ data[6] = (kSourceID >> 8) & 0xff;
+ data[7] = kSourceID & 0xff;
+
+ data[8] = mID >> 24;
+ data[9] = (mID >> 16) & 0xff;
+ data[10] = (mID >> 8) & 0xff;
+ data[11] = mID & 0xff;
+
+ List<int> list;
+ List<int>::iterator it;
+ getSeqNumToNACK(list, kMaxFCIs);
+ size_t cnt = 0;
+
+ int *FCI = (int *)(data + 12);
+ for (it = list.begin(); it != list.end() && cnt < kMaxFCIs; it++) {
+ *(FCI + cnt) = *it;
+ cnt++;
+ }
+
+ data[3] = (3 + cnt) - 1; // total (3 + #ofFCI) * sizeof(int32_t) byte
+
+ buffer->setRange(buffer->offset(), buffer->size() + (data[3] + 1) * sizeof(int32_t));
+
+ return cnt;
+}
+
+int ARTPSource::getSeqNumToNACK(List<int>& list, int size) {
+ AutoMutex _l(mMapLock);
+ int cnt = 0;
+
+ std::map<uint16_t, infoNACK>::iterator it;
+ for(it = mNACKMap.begin(); it != mNACKMap.end() && cnt < size; it++) {
+ infoNACK &info_it = it->second;
+ if (info_it.needToNACK) {
+ info_it.needToNACK = false;
+ // switch LSB to MSB for sending N/W
+ uint32_t FCI;
+ uint8_t *temp = (uint8_t *)&FCI;
+ temp[0] = (info_it.seqNum >> 8) & 0xff;
+ temp[1] = (info_it.seqNum) & 0xff;
+ temp[2] = (info_it.mask >> 8) & 0xff;
+ temp[3] = (info_it.mask) & 0xff;
+
+ list.push_back(FCI);
+ cnt++;
+ }
+ }
+
+ return cnt;
+}
+
+void ARTPSource::setSeqNumToNACK(uint16_t seqNum, uint16_t mask, uint16_t nowJitterHeadSeqNum) {
+ AutoMutex _l(mMapLock);
+ infoNACK info = {seqNum, mask, nowJitterHeadSeqNum, true};
+ std::map<uint16_t, infoNACK>::iterator it;
+
+ it = mNACKMap.find(seqNum);
+ if (it != mNACKMap.end()) {
+ infoNACK &info_it = it->second;
+ // renew if (mask or head seq) is changed
+ if ((info_it.mask != mask) || (info_it.nowJitterHeadSeqNum != nowJitterHeadSeqNum)) {
+ info_it = info;
+ }
+ } else {
+ mNACKMap[seqNum] = info;
+ }
+
+ // delete all NACK far from current Jitter's first sequence number
+ it = mNACKMap.begin();
+ while (it != mNACKMap.end()) {
+ infoNACK &info_it = it->second;
+
+ int diff = nowJitterHeadSeqNum - info_it.nowJitterHeadSeqNum;
+ if (diff > 100) {
+ ALOGV("Delete %d pkt from NACK map ", info_it.seqNum);
+ it = mNACKMap.erase(it);
+ } else {
+ it++;
+ }
+ }
+
}
uint32_t ARTPSource::getSelfID() {
return kSourceID;
}
+
void ARTPSource::setSelfID(const uint32_t selfID) {
kSourceID = selfID;
}
-void ARTPSource::setMinMaxBitrate(int32_t min, int32_t max) {
- mQualManager.setMinMaxBitrate(min, max);
+void ARTPSource::setJbTime(const uint32_t jbTimeMs) {
+ mJbTimeMs = jbTimeMs;
}
-void ARTPSource::setBitrateData(int32_t bitrate, int64_t time) {
- mQualManager.setBitrateData(bitrate, time);
+void ARTPSource::setPeriodicFIR(bool enable) {
+ ALOGD("setPeriodicFIR %d", enable);
+ mIssueFIRRequests = enable;
}
-void ARTPSource::setTargetBitrate() {
- uint8_t fraction = 0;
+void ARTPSource::notifyPktInfo(int32_t bitrate, int64_t /*time*/) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("rtcp-event", 1);
+ notify->setInt32("payload-type", 102);
+ notify->setInt32("feedback-type", 0);
+ // sending target bitrate up to application to share rtp quality.
+ notify->setInt32("bit-rate", bitrate);
+ notify->setInt32("highest-seq-num", mHighestSeqNumber);
+ notify->setInt32("base-seq-num", mBaseSeqNumber);
+ notify->setInt32("prev-expected", mPrevExpected);
+ notify->setInt32("num-buf-recv", mNumBuffersReceived);
+ notify->setInt32("prev-num-buf-recv", mPrevNumBuffersReceived);
+ notify->post();
- // According to appendix A.3 in RFC 3550
uint32_t expected = mHighestSeqNumber - mBaseSeqNumber + 1;
- int64_t intervalExpected = expected - mPrevExpected;
- int64_t intervalReceived = mNumBuffersReceived - mPrevNumBuffersReceived;
- int64_t intervalPacketLost = intervalExpected - intervalReceived;
-
- ALOGI("UID %p expectedPkts %lld lostPkts %lld", this, (long long)intervalExpected, (long long)intervalPacketLost);
-
- if (intervalPacketLost < 0 || intervalExpected == 0)
- fraction = 0;
- else if (intervalExpected <= intervalPacketLost)
- fraction = 255;
- else
- fraction = (intervalPacketLost << 8) / intervalExpected;
-
- mQualManager.setTargetBitrate(fraction, ALooper::GetNowUs(), intervalExpected < 5);
+ mPrevExpected = expected;
+ mPrevNumBuffersReceived = mNumBuffersReceived;
}
-bool ARTPSource::isNeedToReport() {
- int64_t intervalReceived = mNumBuffersReceived - mPrevNumBuffersReceived;
- return (intervalReceived > 0) ? true : false;
-}
-
-bool ARTPSource::isNeedToDowngrade() {
- return mQualManager.isNeedToDowngrade();
+void ARTPSource::onIssueFIRByAssembler() {
+ mIssueFIRByAssembler = true;
}
void ARTPSource::noticeAbandonBuffer(int cnt) {
diff --git a/media/libstagefright/rtsp/ARTPSource.h b/media/libstagefright/rtsp/ARTPSource.h
index 652e753..ea683a0 100644
--- a/media/libstagefright/rtsp/ARTPSource.h
+++ b/media/libstagefright/rtsp/ARTPSource.h
@@ -23,7 +23,9 @@
#include <media/stagefright/foundation/ABase.h>
#include <utils/List.h>
#include <utils/RefBase.h>
-#include <QualManager.h>
+#include <utils/Thread.h>
+
+#include <map>
namespace android {
@@ -46,23 +48,28 @@
void addReceiverReport(const sp<ABuffer> &buffer);
void addFIR(const sp<ABuffer> &buffer);
- void addTMMBR(const sp<ABuffer> &buffer);
+ void addTMMBR(const sp<ABuffer> &buffer, int32_t targetBitrate);
+ int addNACK(const sp<ABuffer> &buffer);
+ void setSeqNumToNACK(uint16_t seqNum, uint16_t mask, uint16_t nowJitterHeadSeqNum);
uint32_t getSelfID();
void setSelfID(const uint32_t selfID);
- void setMinMaxBitrate(int32_t min, int32_t max);
- void setBitrateData(int32_t bitrate, int64_t time);
- void setTargetBitrate();
-
- bool isNeedToReport();
- bool isNeedToDowngrade();
+ void setJbTime(const uint32_t jbTimeMs);
+ void setPeriodicFIR(bool enable);
+ void notifyPktInfo(int32_t bitrate, int64_t time);
+ // FIR needs to be sent by missing packet or broken video image.
+ void onIssueFIRByAssembler();
void noticeAbandonBuffer(int cnt=1);
int32_t mFirstSeqNumber;
- int32_t mFirstRtpTime;
+ uint32_t mFirstRtpTime;
int64_t mFirstSysTime;
int32_t mClockRate;
+ uint32_t mJbTimeMs;
+ int32_t mFirstSsrc;
+ int32_t mHighestNackNumber;
+
private:
uint32_t mID;
@@ -71,21 +78,33 @@
uint32_t mBaseSeqNumber;
int32_t mNumBuffersReceived;
int32_t mPrevNumBuffersReceived;
+ uint32_t mPrevExpectedForRR;
+ int32_t mPrevNumBuffersReceivedForRR;
List<sp<ABuffer> > mQueue;
sp<ARTPAssembler> mAssembler;
+ typedef struct infoNACK {
+ uint16_t seqNum;
+ uint16_t mask;
+ uint16_t nowJitterHeadSeqNum;
+ bool needToNACK;
+ } infoNACK;
+
+ Mutex mMapLock;
+ std::map<uint16_t, infoNACK> mNACKMap;
+ int getSeqNumToNACK(List<int>& list, int size);
+
uint64_t mLastNTPTime;
int64_t mLastNTPTimeUpdateUs;
bool mIssueFIRRequests;
+ bool mIssueFIRByAssembler;
int64_t mLastFIRRequestUs;
uint8_t mNextFIRSeqNo;
sp<AMessage> mNotify;
- QualManager mQualManager;
-
bool queuePacket(const sp<ABuffer> &buffer);
DISALLOW_EVIL_CONSTRUCTORS(ARTPSource);
diff --git a/media/libstagefright/rtsp/ARTPWriter.cpp b/media/libstagefright/rtsp/ARTPWriter.cpp
index 70d34de..76afb04 100644
--- a/media/libstagefright/rtsp/ARTPWriter.cpp
+++ b/media/libstagefright/rtsp/ARTPWriter.cpp
@@ -42,21 +42,24 @@
#define H264_NALU_PFRAME 0x1
#define H265_NALU_MASK 0x3F
-#define H265_NALU_VPS 0x40
-#define H265_NALU_SPS 0x42
-#define H265_NALU_PPS 0x44
+#define H265_NALU_VPS 0x20
+#define H265_NALU_SPS 0x21
+#define H265_NALU_PPS 0x22
+#define LINK_HEADER_SIZE 14
+#define IP_HEADER_SIZE 20
#define UDP_HEADER_SIZE 8
+#define TCPIP_HEADER_SIZE (LINK_HEADER_SIZE + IP_HEADER_SIZE + UDP_HEADER_SIZE)
#define RTP_HEADER_SIZE 12
-#define RTP_HEADER_EXT_SIZE 1
+#define RTP_HEADER_EXT_SIZE 8
#define RTP_FU_HEADER_SIZE 2
-#define RTP_PAYLOAD_ROOM_SIZE 140
+#define RTP_PAYLOAD_ROOM_SIZE 100 // ROOM size for IPv6 header, ESP and etc.
namespace android {
// static const size_t kMaxPacketSize = 65507; // maximum payload in UDP over IP
-static const size_t kMaxPacketSize = 1500;
+static const size_t kMaxPacketSize = 1280;
static char kCNAME[255] = "someone@somewhere";
static int UniformRand(int limit) {
@@ -67,7 +70,8 @@
: mFlags(0),
mFd(dup(fd)),
mLooper(new ALooper),
- mReflector(new AHandlerReflector<ARTPWriter>(this)) {
+ mReflector(new AHandlerReflector<ARTPWriter>(this)),
+ mTrafficRec(new TrafficRecorder<uint32_t, size_t>(128)) {
CHECK_GE(fd, 0);
mIsIPv6 = false;
@@ -117,7 +121,8 @@
: mFlags(0),
mFd(dup(fd)),
mLooper(new ALooper),
- mReflector(new AHandlerReflector<ARTPWriter>(this)) {
+ mReflector(new AHandlerReflector<ARTPWriter>(this)),
+ mTrafficRec(new TrafficRecorder<uint32_t, size_t>(128)) {
CHECK_GE(fd, 0);
mIsIPv6 = false;
@@ -126,6 +131,7 @@
mLooper->start();
makeSocketPairAndBind(localIp, localPort, remoteIp , remotePort);
+ mVPSBuf = NULL;
mSPSBuf = NULL;
mPPSBuf = NULL;
@@ -147,6 +153,11 @@
}
ARTPWriter::~ARTPWriter() {
+ if (mVPSBuf != NULL) {
+ mVPSBuf->release();
+ mVPSBuf = NULL;
+ }
+
if (mSPSBuf != NULL) {
mSPSBuf->release();
mSPSBuf = NULL;
@@ -277,12 +288,9 @@
return OK;
}
-// return size of SPS if there is more NAL unit found following to SPS.
-static uint32_t StripStartcode(MediaBufferBase *buffer) {
- uint32_t nalSize = 0;
-
+static void StripStartcode(MediaBufferBase *buffer) {
if (buffer->range_length() < 4) {
- return 0;
+ return;
}
const uint8_t *ptr =
@@ -292,55 +300,129 @@
buffer->set_range(
buffer->range_offset() + 4, buffer->range_length() - 4);
}
-
- ptr = (const uint8_t *)buffer->data() + buffer->range_offset();
-
- if (buffer->range_length() > 0 && (*ptr & H264_NALU_MASK) == H264_NALU_SPS) {
- for (uint32_t i = 1; i + 4 <= buffer->range_length(); i++) {
-
- if (!memcmp(ptr + i, "\x00\x00\x00\x01", 4)) {
- // Now, we found one more NAL unit in the media buffer.
- // Mostly, it will be a PPS.
- nalSize = i;
- ALOGV("SPS found. size=%d", nalSize);
- }
- }
- }
-
- return nalSize;
}
-static void SpsPpsParser(MediaBufferBase *mediaBuffer,
- MediaBufferBase **spsBuffer, MediaBufferBase **ppsBuffer, uint32_t spsSize) {
+static const uint8_t SPCSize = 4; // Start Prefix Code Size
+static const uint8_t startPrefixCode[SPCSize] = {0, 0, 0, 1};
+static const uint8_t spcKMPidx[SPCSize] = {0, 0, 2, 0};
+static void SpsPpsParser(MediaBufferBase *buffer,
+ MediaBufferBase **spsBuffer, MediaBufferBase **ppsBuffer) {
- if (mediaBuffer == NULL || mediaBuffer->range_length() < 4)
- return;
+ while (buffer->range_length() > 0) {
+ const uint8_t *NALPtr = (const uint8_t *)buffer->data() + buffer->range_offset();
- if ((*spsBuffer) != NULL) {
- (*spsBuffer)->release();
- (*spsBuffer) = NULL;
+ MediaBufferBase **targetPtr = NULL;
+ if ((*NALPtr & H264_NALU_MASK) == H264_NALU_SPS) {
+ targetPtr = spsBuffer;
+ } else if ((*NALPtr & H264_NALU_MASK) == H264_NALU_PPS) {
+ targetPtr = ppsBuffer;
+ } else {
+ return;
+ }
+ ALOGV("SPS(7) or PPS(8) found. Type %d", *NALPtr & H264_NALU_MASK);
+
+ uint32_t bufferSize = buffer->range_length();
+ MediaBufferBase *&target = *targetPtr;
+ uint32_t i = 0, j = 0;
+ bool isBoundFound = false;
+ for (i = 0; i < bufferSize; i++) {
+ while (j > 0 && NALPtr[i] != startPrefixCode[j]) {
+ j = spcKMPidx[j - 1];
+ }
+ if (NALPtr[i] == startPrefixCode[j]) {
+ j++;
+ if (j == SPCSize) {
+ isBoundFound = true;
+ break;
+ }
+ }
+ }
+
+ uint32_t targetSize;
+ if (target != NULL) {
+ target->release();
+ }
+ // note that targetSize is never 0 as the first byte is never part
+ // of a start prefix
+ if (isBoundFound) {
+ targetSize = i - SPCSize + 1;
+ target = MediaBufferBase::Create(targetSize);
+ memcpy(target->data(),
+ (const uint8_t *)buffer->data() + buffer->range_offset(),
+ targetSize);
+ buffer->set_range(buffer->range_offset() + targetSize + SPCSize,
+ buffer->range_length() - targetSize - SPCSize);
+ } else {
+ targetSize = bufferSize;
+ target = MediaBufferBase::Create(targetSize);
+ memcpy(target->data(),
+ (const uint8_t *)buffer->data() + buffer->range_offset(),
+ targetSize);
+ buffer->set_range(buffer->range_offset() + bufferSize, 0);
+ return;
+ }
}
+}
- if ((*ppsBuffer) != NULL) {
- (*ppsBuffer)->release();
- (*ppsBuffer) = NULL;
- }
+static void VpsSpsPpsParser(MediaBufferBase *buffer,
+ MediaBufferBase **vpsBuffer, MediaBufferBase **spsBuffer, MediaBufferBase **ppsBuffer) {
- // we got sps/pps but startcode of sps is striped.
- (*spsBuffer) = MediaBufferBase::Create(spsSize);
- memcpy((*spsBuffer)->data(),
- (const uint8_t *)mediaBuffer->data() + mediaBuffer->range_offset(),
- spsSize);
+ while (buffer->range_length() > 0) {
+ const uint8_t *NALPtr = (const uint8_t *)buffer->data() + buffer->range_offset();
+ uint8_t nalType = ((*NALPtr) >> 1) & H265_NALU_MASK;
- int32_t ppsSize = mediaBuffer->range_length() - spsSize - 4 /*startcode*/;
- if (ppsSize > 0) {
- (*ppsBuffer) = MediaBufferBase::Create(ppsSize);
- ALOGV("PPS found. size=%d", (int)ppsSize);
- mediaBuffer->set_range(mediaBuffer->range_offset() + spsSize + 4 /*startcode*/,
- mediaBuffer->range_length() - spsSize - 4 /*startcode*/);
- memcpy((*ppsBuffer)->data(),
- (const uint8_t *)mediaBuffer->data() + mediaBuffer->range_offset(),
- ppsSize);
+ MediaBufferBase **targetPtr = NULL;
+ if (nalType == H265_NALU_VPS) {
+ targetPtr = vpsBuffer;
+ } else if (nalType == H265_NALU_SPS) {
+ targetPtr = spsBuffer;
+ } else if (nalType == H265_NALU_PPS) {
+ targetPtr = ppsBuffer;
+ } else {
+ return;
+ }
+ ALOGV("VPS(32) SPS(33) or PPS(34) found. Type %d", nalType);
+
+ uint32_t bufferSize = buffer->range_length();
+ MediaBufferBase *&target = *targetPtr;
+ uint32_t i = 0, j = 0;
+ bool isBoundFound = false;
+ for (i = 0; i < bufferSize; i++) {
+ while (j > 0 && NALPtr[i] != startPrefixCode[j]) {
+ j = spcKMPidx[j - 1];
+ }
+ if (NALPtr[i] == startPrefixCode[j]) {
+ j++;
+ if (j == SPCSize) {
+ isBoundFound = true;
+ break;
+ }
+ }
+ }
+
+ if (target != NULL) {
+ target->release();
+ }
+ uint32_t targetSize;
+ // note that targetSize is never 0 as the first byte is never part
+ // of a start prefix
+ if (isBoundFound) {
+ targetSize = i - SPCSize + 1;
+ target = MediaBufferBase::Create(j);
+ memcpy(target->data(),
+ (const uint8_t *)buffer->data() + buffer->range_offset(),
+ j);
+ buffer->set_range(buffer->range_offset() + targetSize + SPCSize,
+ buffer->range_length() - targetSize - SPCSize);
+ } else {
+ targetSize = bufferSize;
+ target = MediaBufferBase::Create(targetSize);
+ memcpy(target->data(),
+ (const uint8_t *)buffer->data() + buffer->range_offset(),
+ targetSize);
+ buffer->set_range(buffer->range_offset() + bufferSize, 0);
+ return;
+ }
}
}
@@ -451,15 +533,17 @@
ALOGV("read buffer of size %zu", mediaBuf->range_length());
if (mMode == H264) {
- uint32_t spsSize = 0;
- if ((spsSize = StripStartcode(mediaBuf)) > 0) {
- SpsPpsParser(mediaBuf, &mSPSBuf, &mPPSBuf, spsSize);
- } else {
+ StripStartcode(mediaBuf);
+ SpsPpsParser(mediaBuf, &mSPSBuf, &mPPSBuf);
+ if (mediaBuf->range_length() > 0) {
sendAVCData(mediaBuf);
}
} else if (mMode == H265) {
StripStartcode(mediaBuf);
- sendHEVCData(mediaBuf);
+ VpsSpsPpsParser(mediaBuf, &mVPSBuf, &mSPSBuf, &mPPSBuf);
+ if (mediaBuf->range_length() > 0) {
+ sendHEVCData(mediaBuf);
+ }
} else if (mMode == H263) {
sendH263Data(mediaBuf);
} else if (mMode == AMR_NB || mMode == AMR_WB) {
@@ -504,11 +588,20 @@
remAddr = (struct sockaddr *)&mRTPAddr;
}
+ // Unseal code if moderator is needed (prevent overflow of instant bandwidth)
+ // Set limit bits per period through the moderator.
+ // ex) 6KByte/10ms = 48KBit/10ms = 4.8MBit/s instant limit
+ // ModerateInstantTraffic(10, 6 * 1024);
+
ssize_t n = sendto(isRTCP ? mRTCPSocket : mRTPSocket,
buffer->data(), buffer->size(), 0, remAddr, sizeSockSt);
if (n != (ssize_t)buffer->size()) {
ALOGW("packets can not be sent. ret=%d, buf=%d", (int)n, (int)buffer->size());
+ } else {
+ // Record current traffic & Print bits while last 1sec (1000ms)
+ mTrafficRec->writeBytes(buffer->size());
+ mTrafficRec->printAccuBitsForLastPeriod(1000, 1000);
}
#if LOG_TO_FILES
@@ -807,12 +900,13 @@
}
void ARTPWriter::sendSPSPPSIfIFrame(MediaBufferBase *mediaBuf, int64_t timeUs) {
+ CHECK(mediaBuf->range_length() > 0);
const uint8_t *mediaData =
(const uint8_t *)mediaBuf->data() + mediaBuf->range_offset();
- if (mediaBuf->range_length() == 0
- || (mediaData[0] & H264_NALU_MASK) != H264_NALU_IFRAME)
+ if ((mediaData[0] & H264_NALU_MASK) != H264_NALU_IFRAME) {
return;
+ }
if (mSPSBuf != NULL) {
mSPSBuf->meta_data().setInt64(kKeyTime, timeUs);
@@ -827,6 +921,35 @@
}
}
+void ARTPWriter::sendVPSSPSPPSIfIFrame(MediaBufferBase *mediaBuf, int64_t timeUs) {
+ CHECK(mediaBuf->range_length() > 0);
+ const uint8_t *mediaData =
+ (const uint8_t *)mediaBuf->data() + mediaBuf->range_offset();
+
+ int nalType = ((mediaData[0] >> 1) & H265_NALU_MASK);
+ if (!(nalType >= 16 && nalType <= 21) /*H265_NALU_IFRAME*/) {
+ return;
+ }
+
+ if (mVPSBuf != NULL) {
+ mVPSBuf->meta_data().setInt64(kKeyTime, timeUs);
+ mVPSBuf->meta_data().setInt32(kKeyVps, 1);
+ sendHEVCData(mVPSBuf);
+ }
+
+ if (mSPSBuf != NULL) {
+ mSPSBuf->meta_data().setInt64(kKeyTime, timeUs);
+ mSPSBuf->meta_data().setInt32(kKeySps, 1);
+ sendHEVCData(mSPSBuf);
+ }
+
+ if (mPPSBuf != NULL) {
+ mPPSBuf->meta_data().setInt64(kKeyTime, timeUs);
+ mPPSBuf->meta_data().setInt32(kKeyPps, 1);
+ sendHEVCData(mPPSBuf);
+ }
+}
+
void ARTPWriter::sendHEVCData(MediaBufferBase *mediaBuf) {
// 12 bytes RTP header + 2 bytes for the FU-indicator and FU-header.
CHECK_GE(kMaxPacketSize, 12u + 2u);
@@ -834,21 +957,33 @@
int64_t timeUs;
CHECK(mediaBuf->meta_data().findInt64(kKeyTime, &timeUs));
- sendSPSPPSIfIFrame(mediaBuf, timeUs);
+ sendVPSSPSPPSIfIFrame(mediaBuf, timeUs);
uint32_t rtpTime = mRTPTimeBase + (timeUs * 9 / 100ll);
+ CHECK(mediaBuf->range_length() > 0);
const uint8_t *mediaData =
(const uint8_t *)mediaBuf->data() + mediaBuf->range_offset();
+ int32_t isNonVCL = 0;
+ if (mediaBuf->meta_data().findInt32(kKeyVps, &isNonVCL) ||
+ mediaBuf->meta_data().findInt32(kKeySps, &isNonVCL) ||
+ mediaBuf->meta_data().findInt32(kKeyPps, &isNonVCL)) {
+ isNonVCL = 1;
+ }
+
sp<ABuffer> buffer = new ABuffer(kMaxPacketSize);
- if (mediaBuf->range_length() + UDP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_PAYLOAD_ROOM_SIZE
- <= buffer->capacity()) {
+ if (mediaBuf->range_length() + TCPIP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_HEADER_EXT_SIZE
+ + RTP_PAYLOAD_ROOM_SIZE <= buffer->capacity()) {
// The data fits into a single packet
uint8_t *data = buffer->data();
data[0] = 0x80;
- data[1] = (1 << 7) | mPayloadType; // M-bit
+ if (isNonVCL) {
+ data[1] = mPayloadType; // Marker bit should not be set in case of Non-VCL
+ } else {
+ data[1] = (1 << 7) | mPayloadType; // M-bit
+ }
data[2] = (mSeqNo >> 8) & 0xff;
data[3] = mSeqNo & 0xff;
data[4] = rtpTime >> 24;
@@ -881,11 +1016,11 @@
while (offset < mediaBuf->range_length()) {
size_t size = mediaBuf->range_length() - offset;
bool lastPacket = true;
- if (size + UDP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_FU_HEADER_SIZE +
- RTP_PAYLOAD_ROOM_SIZE > buffer->capacity()) {
+ if (size + TCPIP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_HEADER_EXT_SIZE +
+ RTP_FU_HEADER_SIZE + RTP_PAYLOAD_ROOM_SIZE > buffer->capacity()) {
lastPacket = false;
- size = buffer->capacity() - UDP_HEADER_SIZE - RTP_HEADER_SIZE -
- RTP_FU_HEADER_SIZE - RTP_PAYLOAD_ROOM_SIZE;
+ size = buffer->capacity() - TCPIP_HEADER_SIZE - RTP_HEADER_SIZE -
+ RTP_HEADER_EXT_SIZE - RTP_FU_HEADER_SIZE - RTP_PAYLOAD_ROOM_SIZE;
}
uint8_t *data = buffer->data();
@@ -963,6 +1098,7 @@
uint32_t rtpTime = mRTPTimeBase + (timeUs * 9 / 100LL);
+ CHECK(mediaBuf->range_length() > 0);
const uint8_t *mediaData =
(const uint8_t *)mediaBuf->data() + mediaBuf->range_offset();
@@ -973,9 +1109,10 @@
isSpsPps = true;
}
+ mTrafficRec->updateClock(ALooper::GetNowUs() / 1000);
sp<ABuffer> buffer = new ABuffer(kMaxPacketSize);
- if (mediaBuf->range_length() + UDP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_PAYLOAD_ROOM_SIZE
- <= buffer->capacity()) {
+ if (mediaBuf->range_length() + TCPIP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_HEADER_EXT_SIZE
+ + RTP_PAYLOAD_ROOM_SIZE <= buffer->capacity()) {
// The data fits into a single packet
uint8_t *data = buffer->data();
data[0] = 0x80;
@@ -1051,11 +1188,11 @@
while (offset < mediaBuf->range_length()) {
size_t size = mediaBuf->range_length() - offset;
bool lastPacket = true;
- if (size + UDP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_FU_HEADER_SIZE +
- RTP_PAYLOAD_ROOM_SIZE > buffer->capacity()) {
+ if (size + TCPIP_HEADER_SIZE + RTP_HEADER_SIZE + RTP_HEADER_EXT_SIZE +
+ RTP_FU_HEADER_SIZE + RTP_PAYLOAD_ROOM_SIZE > buffer->capacity()) {
lastPacket = false;
- size = buffer->capacity() - UDP_HEADER_SIZE - RTP_HEADER_SIZE -
- RTP_FU_HEADER_SIZE - RTP_PAYLOAD_ROOM_SIZE;
+ size = buffer->capacity() - TCPIP_HEADER_SIZE - RTP_HEADER_SIZE -
+ RTP_HEADER_EXT_SIZE - RTP_FU_HEADER_SIZE - RTP_PAYLOAD_ROOM_SIZE;
}
uint8_t *data = buffer->data();
@@ -1408,4 +1545,15 @@
}
}
+// TODO : Develop more advanced moderator based on AS & TMMBR value
+void ARTPWriter::ModerateInstantTraffic(uint32_t samplePeriod, uint32_t limitBytes) {
+ unsigned int bytes = mTrafficRec->readBytesForLastPeriod(samplePeriod);
+ if (bytes > limitBytes) {
+ ALOGI("Nuclear moderator. #seq = %d \t\t %d bits / 10ms",
+ mSeqNo, bytes * 8);
+ usleep(4000);
+ mTrafficRec->updateClock(ALooper::GetNowUs() / 1000);
+ }
+}
+
} // namespace android
diff --git a/media/libstagefright/rtsp/ARTPWriter.h b/media/libstagefright/rtsp/ARTPWriter.h
index f7e2204..6f25a66 100644
--- a/media/libstagefright/rtsp/ARTPWriter.h
+++ b/media/libstagefright/rtsp/ARTPWriter.h
@@ -28,6 +28,7 @@
#include <sys/socket.h>
#include <android/multinetwork.h>
+#include "TrafficRecorder.h"
#define LOG_TO_FILES 0
@@ -102,6 +103,7 @@
AString mSeqParamSet;
AString mPicParamSet;
+ MediaBufferBase *mVPSBuf;
MediaBufferBase *mSPSBuf;
MediaBufferBase *mPPSBuf;
@@ -116,6 +118,7 @@
uint32_t mOpponentID;
uint32_t mBitrate;
+ sp<TrafficRecorder<uint32_t, size_t> > mTrafficRec;
int32_t mNumSRsSent;
int32_t mRTPCVOExtMap;
@@ -143,6 +146,7 @@
void dumpSessionDesc();
void sendBye();
+ void sendVPSSPSPPSIfIFrame(MediaBufferBase *mediaBuf, int64_t timeUs);
void sendSPSPPSIfIFrame(MediaBufferBase *mediaBuf, int64_t timeUs);
void sendHEVCData(MediaBufferBase *mediaBuf);
void sendAVCData(MediaBufferBase *mediaBuf);
@@ -152,6 +156,7 @@
void send(const sp<ABuffer> &buffer, bool isRTCP);
void makeSocketPairAndBind(String8& localIp, int localPort, String8& remoteIp, int remotePort);
+ void ModerateInstantTraffic(uint32_t samplePeriod, uint32_t limitBytes);
DISALLOW_EVIL_CONSTRUCTORS(ARTPWriter);
};
diff --git a/media/libstagefright/rtsp/Android.bp b/media/libstagefright/rtsp/Android.bp
index 6179142..f990ecf 100644
--- a/media/libstagefright/rtsp/Android.bp
+++ b/media/libstagefright/rtsp/Android.bp
@@ -18,7 +18,6 @@
"ARTSPConnection.cpp",
"ASessionDescription.cpp",
"SDPLoader.cpp",
- "QualManager.cpp",
],
shared_libs: [
diff --git a/media/libstagefright/rtsp/TrafficRecorder.h b/media/libstagefright/rtsp/TrafficRecorder.h
new file mode 100644
index 0000000..f8e7c03
--- /dev/null
+++ b/media/libstagefright/rtsp/TrafficRecorder.h
@@ -0,0 +1,151 @@
+/*
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef A_TRAFFIC_RECORDER_H_
+
+#define A_TRAFFIC_RECORDER_H_
+
+#include <android-base/logging.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+// Circular array to save recent amount of bytes
+template <class Time, class Bytes>
+class TrafficRecorder : public RefBase {
+private:
+ size_t mSize;
+ size_t mSizeMask;
+ Time *mTimeArray = NULL;
+ Bytes *mBytesArray = NULL;
+ size_t mHeadIdx = 0;
+ size_t mTailIdx = 0;
+
+ Time mClock = 0;
+ Time mLastTimeOfPrint = 0;
+ Bytes mAccuBytesOfPrint = 0;
+public:
+ TrafficRecorder();
+ TrafficRecorder(size_t size);
+ virtual ~TrafficRecorder();
+
+ void init();
+
+ void updateClock(Time now);
+
+ Bytes readBytesForLastPeriod(Time period);
+ void writeBytes(Bytes bytes);
+
+ void printAccuBitsForLastPeriod(Time period, Time unit);
+};
+
+template <class Time, class Bytes>
+TrafficRecorder<Time, Bytes>::TrafficRecorder() {
+ TrafficRecorder(128);
+}
+
+template <class Time, class Bytes>
+TrafficRecorder<Time, Bytes>::TrafficRecorder(size_t size) {
+ size_t exp;
+ for (exp = 0; exp < 32; exp++) {
+ if (size <= (1ul << exp)) {
+ break;
+ }
+ }
+ mSize = (1ul << exp); // size = 2^exp
+ mSizeMask = mSize - 1;
+
+ LOG(VERBOSE) << "TrafficRecorder Init size " << mSize;
+ mTimeArray = new Time[mSize];
+ mBytesArray = new Bytes[mSize];
+
+ init();
+}
+
+template <class Time, class Bytes>
+TrafficRecorder<Time, Bytes>::~TrafficRecorder() {
+ delete[] mTimeArray;
+ delete[] mBytesArray;
+}
+
+template <class Time, class Bytes>
+void TrafficRecorder<Time, Bytes>::init() {
+ mHeadIdx = 0;
+ mTailIdx = 0;
+ mTimeArray[0] = 0;
+ mBytesArray[0] = 0;
+}
+
+template <class Time, class Bytes>
+void TrafficRecorder<Time, Bytes>::updateClock(Time now) {
+ mClock = now;
+}
+
+template <class Time, class Bytes>
+Bytes TrafficRecorder<Time, Bytes>::readBytesForLastPeriod(Time period) {
+ Bytes bytes = 0;
+
+ size_t i = mTailIdx;
+ while (i != mHeadIdx) {
+ LOG(VERBOSE) << "READ " << i << " time " << mTimeArray[i] << " \t EndOfPeriod " << mClock - period;
+ if (mTimeArray[i] < mClock - period) {
+ break;
+ }
+ bytes += mBytesArray[i];
+ i = (i + mSize - 1) & mSizeMask;
+ }
+ mHeadIdx = i;
+ return bytes;
+}
+
+template <class Time, class Bytes>
+void TrafficRecorder<Time, Bytes>::writeBytes(Bytes bytes) {
+ size_t writeIdx;
+ if (mClock == mTimeArray[mTailIdx]) {
+ writeIdx = mTailIdx;
+ mBytesArray[writeIdx] += bytes;
+ } else {
+ writeIdx = (mTailIdx + 1) % mSize;
+ mTimeArray[writeIdx] = mClock;
+ mBytesArray[writeIdx] = bytes;
+ }
+
+ LOG(VERBOSE) << "WRITE " << writeIdx << " time " << mClock;
+ if (writeIdx == mHeadIdx) {
+ LOG(WARNING) << "Traffic recorder size exceeded at " << mHeadIdx;
+ mHeadIdx = (mHeadIdx + 1) & mSizeMask;
+ }
+
+ mTailIdx = writeIdx;
+ mAccuBytesOfPrint += bytes;
+}
+
+template <class Time, class Bytes>
+void TrafficRecorder<Time, Bytes>::printAccuBitsForLastPeriod(Time period, Time unit) {
+ Time duration = mClock - mLastTimeOfPrint;
+ float numOfUnit = (float)duration / unit;
+ if (duration > period) {
+ ALOGD("Actual Tx period %.0f ms \t %.0f Bits/Unit",
+ numOfUnit * 1000.f, mAccuBytesOfPrint * 8.f / numOfUnit);
+ mLastTimeOfPrint = mClock;
+ mAccuBytesOfPrint = 0;
+ init();
+ }
+}
+
+} // namespace android
+
+#endif // A_TRAFFIC_RECORDER_H_
diff --git a/media/libstagefright/tests/writer/WriterTest.cpp b/media/libstagefright/tests/writer/WriterTest.cpp
index 4ca02b0..d170e7c 100644
--- a/media/libstagefright/tests/writer/WriterTest.cpp
+++ b/media/libstagefright/tests/writer/WriterTest.cpp
@@ -726,6 +726,176 @@
close(fd);
}
+class WriterValidityTest
+ : public WriterTest,
+ public ::testing::TestWithParam<
+ tuple<string /* writerFormat*/, inputId /* inputId0*/, bool /* addSourceFail*/>> {
+ public:
+ virtual void SetUp() override { setupWriterType(get<0>(GetParam())); }
+};
+
+TEST_P(WriterValidityTest, InvalidInputTest) {
+ if (mDisableTest) return;
+ ALOGV("Validates writer's behavior for invalid inputs");
+
+ string writerFormat = get<0>(GetParam());
+ inputId inpId = get<1>(GetParam());
+ bool addSourceFailExpected = get<2>(GetParam());
+
+ // Test writers for invalid FD value
+ int32_t fd = -1;
+ int32_t status = createWriter(fd);
+ if (status != OK) {
+ ALOGV("createWriter failed for invalid FD, this is expected behavior");
+ return;
+ }
+
+ // If writer was created for invalid fd, test it further.
+ string inputFile = gEnv->getRes();
+ string inputInfo = gEnv->getRes();
+ configFormat param;
+ bool isAudio;
+ ASSERT_NE(inpId, UNUSED_ID) << "Test expects first inputId to be a valid id";
+
+ getFileDetails(inputFile, inputInfo, param, isAudio, inpId);
+ ASSERT_NE(inputFile.compare(gEnv->getRes()), 0) << "No input file specified";
+
+ ASSERT_NO_FATAL_FAILURE(getInputBufferInfo(inputFile, inputInfo));
+ status = addWriterSource(isAudio, param);
+ if (status != OK) {
+ ASSERT_TRUE(addSourceFailExpected)
+ << "Failed to add source for " << writerFormat << " writer";
+ ALOGV("addWriterSource failed for invalid FD, this is expected behavior");
+ return;
+ }
+
+ // start the writer with valid argument but invalid FD
+ status = mWriter->start(mFileMeta.get());
+ ASSERT_NE((status_t)OK, status) << "Writer did not fail for invalid FD";
+
+ status = sendBuffersToWriter(mInputStream[0], mBufferInfo[0], mInputFrameId[0],
+ mCurrentTrack[0], 0, mBufferInfo[0].size());
+ ASSERT_NE((status_t)OK, status) << "Writer did not report error for invalid FD";
+
+ status = mCurrentTrack[0]->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop the track";
+
+ status = mWriter->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop " << writerFormat << " writer";
+}
+
+TEST_P(WriterValidityTest, MalFormedDataTest) {
+ if (mDisableTest) return;
+ // Enable test for Ogg writer
+ ASSERT_NE(mWriterName, OGG) << "TODO(b/160105646)";
+ ALOGV("Test writer for malformed inputs");
+
+ string writerFormat = get<0>(GetParam());
+ inputId inpId = get<1>(GetParam());
+ bool addSourceFailExpected = get<2>(GetParam());
+ int32_t fd =
+ open(OUTPUT_FILE_NAME, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ ASSERT_GE(fd, 0) << "Failed to open output file to dump writer's data";
+
+ int32_t status = createWriter(fd);
+ ASSERT_EQ(status, (status_t)OK)
+ << "Failed to create writer for " << writerFormat << " output format";
+
+ string inputFile = gEnv->getRes();
+ string inputInfo = gEnv->getRes();
+ configFormat param;
+ bool isAudio;
+ ASSERT_NE(inpId, UNUSED_ID) << "Test expects first inputId to be a valid id";
+
+ getFileDetails(inputFile, inputInfo, param, isAudio, inpId);
+ ASSERT_NE(inputFile.compare(gEnv->getRes()), 0) << "No input file specified";
+
+ ASSERT_NO_FATAL_FAILURE(getInputBufferInfo(inputFile, inputInfo));
+ // Remove CSD data from input
+ mNumCsds[0] = 0;
+ status = addWriterSource(isAudio, param);
+ if (status != OK) {
+ ASSERT_TRUE(addSourceFailExpected)
+ << "Failed to add source for " << writerFormat << " writer";
+ ALOGV("%s writer failed to addSource after removing CSD from input", writerFormat.c_str());
+ return;
+ }
+
+ status = mWriter->start(mFileMeta.get());
+ ASSERT_EQ((status_t)OK, status) << "Could not start " << writerFormat << "writer";
+
+ // Skip first few frames. These may contain sync frames also.
+ int32_t frameID = mInputFrameId[0] + mBufferInfo[0].size() / 4;
+ status = sendBuffersToWriter(mInputStream[0], mBufferInfo[0], frameID, mCurrentTrack[0], 0,
+ mBufferInfo[0].size());
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+
+ status = mCurrentTrack[0]->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop the track";
+
+ Vector<String16> args;
+ status = mWriter->dump(fd, args);
+ ASSERT_EQ((status_t)OK, status) << "Failed to dump statistics from writer";
+
+ status = mWriter->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop " << writerFormat << " writer";
+ close(fd);
+}
+
+// This test is specific to MPEG4Writer to test more APIs
+TEST_P(WriteFunctionalityTest, Mpeg4WriterTest) {
+ if (mDisableTest) return;
+ if (mWriterName != standardWriters::MPEG4) return;
+ ALOGV("Test MPEG4 writer specific APIs");
+
+ inputId inpId = get<1>(GetParam());
+ int32_t fd =
+ open(OUTPUT_FILE_NAME, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ ASSERT_GE(fd, 0) << "Failed to open output file to dump writer's data";
+
+ int32_t status = createWriter(fd);
+ ASSERT_EQ(status, (status_t)OK) << "Failed to create writer for mpeg4 output format";
+
+ string inputFile = gEnv->getRes();
+ string inputInfo = gEnv->getRes();
+ configFormat param;
+ bool isAudio;
+ ASSERT_NE(inpId, UNUSED_ID) << "Test expects first inputId to be a valid id";
+
+ getFileDetails(inputFile, inputInfo, param, isAudio, inpId);
+ ASSERT_NE(inputFile.compare(gEnv->getRes()), 0) << "No input file specified";
+
+ ASSERT_NO_FATAL_FAILURE(getInputBufferInfo(inputFile, inputInfo));
+ status = addWriterSource(isAudio, param);
+ ASSERT_EQ((status_t)OK, status) << "Failed to add source for mpeg4 Writer";
+
+ // signal meta data for the writer
+ sp<MPEG4Writer> mp4writer = static_cast<MPEG4Writer *>(mWriter.get());
+ status = mp4writer->setInterleaveDuration(kDefaultInterleaveDuration);
+ ASSERT_EQ((status_t)OK, status) << "setInterleaveDuration failed";
+
+ status = mp4writer->setGeoData(kDefaultLatitudex10000, kDefaultLongitudex10000);
+ ASSERT_EQ((status_t)OK, status) << "setGeoData failed";
+
+ status = mp4writer->setCaptureRate(kDefaultFPS);
+ ASSERT_EQ((status_t)OK, status) << "setCaptureRate failed";
+
+ status = mWriter->start(mFileMeta.get());
+ ASSERT_EQ((status_t)OK, status) << "Could not start the writer";
+
+ status = sendBuffersToWriter(mInputStream[0], mBufferInfo[0], mInputFrameId[0],
+ mCurrentTrack[0], 0, mBufferInfo[0].size());
+ ASSERT_EQ((status_t)OK, status) << "mpeg4 writer failed";
+
+ status = mCurrentTrack[0]->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop the track";
+
+ status = mWriter->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop the writer";
+ mp4writer.clear();
+ close(fd);
+}
+
class ListenerTest
: public WriterTest,
public ::testing::TestWithParam<tuple<
@@ -858,6 +1028,7 @@
make_tuple("amrwb", AMR_WB_1, UNUSED_ID, 0.5, 0.5, 1),
make_tuple("mpeg2Ts", AAC_1, UNUSED_ID, 0.2, 1, 1),
make_tuple("mpeg4", AAC_1, UNUSED_ID, 0.4, 0.3, 0.25),
+ make_tuple("mpeg4", AAC_1, UNUSED_ID, 0.3, 1, 0.5),
make_tuple("ogg", OPUS_1, UNUSED_ID, 0.7, 0.3, 1)));
// TODO: (b/144476164)
@@ -904,6 +1075,29 @@
make_tuple("webm", VP8_1, OPUS_1, 0.50),
make_tuple("webm", VORBIS_1, VP8_1, 0.25)));
+INSTANTIATE_TEST_SUITE_P(
+ WriterValidityTest, WriterValidityTest,
+ ::testing::Values(
+ make_tuple("aac", AAC_1, true),
+
+ make_tuple("amrnb", AMR_NB_1, true),
+ make_tuple("amrwb", AMR_WB_1, true),
+
+ make_tuple("mpeg4", AAC_1, false),
+ make_tuple("mpeg4", AMR_NB_1, false),
+ make_tuple("mpeg4", AVC_1, false),
+ make_tuple("mpeg4", H263_1, false),
+ make_tuple("mpeg4", HEIC_1, false),
+ make_tuple("mpeg4", HEVC_1, false),
+ make_tuple("mpeg4", MPEG4_1, false),
+
+ make_tuple("ogg", OPUS_1, true),
+
+ make_tuple("webm", OPUS_1, false),
+ make_tuple("webm", VORBIS_1, true),
+ make_tuple("webm", VP8_1, false),
+ make_tuple("webm", VP9_1, false)));
+
int main(int argc, char **argv) {
ProcessState::self()->startThreadPool();
gEnv = new WriterTestEnvironment();
diff --git a/media/libstagefright/tests/writer/WriterUtility.h b/media/libstagefright/tests/writer/WriterUtility.h
index 716844a..6b456fb 100644
--- a/media/libstagefright/tests/writer/WriterUtility.h
+++ b/media/libstagefright/tests/writer/WriterUtility.h
@@ -35,6 +35,11 @@
constexpr uint32_t kMaxCSDStrlen = 16;
constexpr uint32_t kMaxCount = 20;
constexpr int32_t kMimeSize = 128;
+constexpr int32_t kDefaultInterleaveDuration = 0;
+// Geodata is set according to ISO-6709 standard.
+constexpr int32_t kDefaultLatitudex10000 = 500000;
+constexpr int32_t kDefaultLongitudex10000 = 1000000;
+constexpr float kDefaultFPS = 30.0f;
struct BufferInfo {
int32_t size;
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 3873600..12f6eba 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -54,6 +54,7 @@
"libmediautils",
"libmemunreachable",
"libmedia_helper",
+ "libshmemcompat",
"libvibrator",
],
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 728c38d..34bdac5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -98,6 +98,8 @@
namespace android {
+using media::IEffectClient;
+
static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
static const char kHardwareLockedString[] = "Hardware lock is taken\n";
static const char kClientLockedString[] = "Client lock is taken\n";
@@ -3438,7 +3440,7 @@
return status;
}
-sp<IEffect> AudioFlinger::createEffect(
+sp<media::IEffect> AudioFlinger::createEffect(
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index d3ad908..2db902d 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -101,6 +101,7 @@
#include <vibrator/ExternalVibrationUtils.h>
#include "android/media/BnAudioRecord.h"
+#include "android/media/BnEffect.h"
namespace android {
@@ -232,9 +233,9 @@
uint32_t preferredTypeFlag,
effect_descriptor_t *descriptor) const;
- virtual sp<IEffect> createEffect(
+ virtual sp<media::IEffect> createEffect(
effect_descriptor_t *pDesc,
- const sp<IEffectClient>& effectClient,
+ const sp<media::IEffectClient>& effectClient,
int32_t priority,
audio_io_handle_t io,
audio_session_t sessionId,
diff --git a/services/audioflinger/DeviceEffectManager.cpp b/services/audioflinger/DeviceEffectManager.cpp
index 5ff7215..cecd52b 100644
--- a/services/audioflinger/DeviceEffectManager.cpp
+++ b/services/audioflinger/DeviceEffectManager.cpp
@@ -30,6 +30,8 @@
namespace android {
+using media::IEffectClient;
+
void AudioFlinger::DeviceEffectManager::createAudioPatch(audio_patch_handle_t handle,
const PatchPanel::Patch& patch) {
ALOGV("%s handle %d mHalHandle %d num sinks %d device sink %08x",
@@ -115,10 +117,19 @@
status_t AudioFlinger::DeviceEffectManager::checkEffectCompatibility(
const effect_descriptor_t *desc) {
+ sp<EffectsFactoryHalInterface> effectsFactory = mAudioFlinger.getEffectsFactory();
+ if (effectsFactory == nullptr) {
+ return BAD_VALUE;
+ }
- if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC
- && (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
- ALOGW("%s() non pre/post processing device effect %s", __func__, desc->name);
+ static const float sMinDeviceEffectHalVersion = 6.0;
+ float halVersion = effectsFactory->getHalVersion();
+
+ if (((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC
+ && (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC)
+ || halVersion < sMinDeviceEffectHalVersion) {
+ ALOGW("%s() non pre/post processing device effect %s or incompatible API version %f",
+ __func__, desc->name, halVersion);
return BAD_VALUE;
}
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
index c6d2110..d187df2 100644
--- a/services/audioflinger/DeviceEffectManager.h
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -33,7 +33,7 @@
sp<EffectHandle> createEffect_l(effect_descriptor_t *descriptor,
const AudioDeviceTypeAddr& device,
const sp<AudioFlinger::Client>& client,
- const sp<IEffectClient>& effectClient,
+ const sp<media::IEffectClient>& effectClient,
const std::map<audio_patch_handle_t, PatchPanel::Patch>& patches,
int *enabled,
status_t *status,
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 9ee47c9..eaad6ef 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -34,6 +34,7 @@
#include <media/AudioContainers.h>
#include <media/AudioEffect.h>
#include <media/AudioDeviceTypeAddr.h>
+#include <media/ShmemCompat.h>
#include <media/audiohal/EffectHalInterface.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include <mediautils/ServiceUtilities.h>
@@ -59,6 +60,27 @@
namespace android {
+using binder::Status;
+
+namespace {
+
+// Append a POD value into a vector of bytes.
+template<typename T>
+void appendToBuffer(const T& value, std::vector<uint8_t>* buffer) {
+ const uint8_t* ar(reinterpret_cast<const uint8_t*>(&value));
+ buffer->insert(buffer->end(), ar, ar + sizeof(T));
+}
+
+// Write a POD value into a vector of bytes (clears the previous buffer
+// content).
+template<typename T>
+void writeToBuffer(const T& value, std::vector<uint8_t>* buffer) {
+ buffer->clear();
+ appendToBuffer(value, buffer);
+}
+
+} // namespace
+
// ----------------------------------------------------------------------------
// EffectBase implementation
// ----------------------------------------------------------------------------
@@ -293,6 +315,9 @@
}
}
+ // Prevent calls to process() and other functions on effect interface from now on.
+ // The effect engine will be released by the destructor when the last strong reference on
+ // this object is released which can happen after next process is called.
if (mHandles.size() == 0 && !mPinned) {
mState = DESTROYED;
}
@@ -566,20 +591,6 @@
}
-ssize_t AudioFlinger::EffectModule::removeHandle_l(EffectHandle *handle)
-{
- ssize_t status = EffectBase::removeHandle_l(handle);
-
- // Prevent calls to process() and other functions on effect interface from now on.
- // The effect engine will be released by the destructor when the last strong reference on
- // this object is released which can happen after next process is called.
- if (status == 0 && !mPinned) {
- mEffectInterface->close();
- }
-
- return status;
-}
-
bool AudioFlinger::EffectModule::updateState() {
Mutex::Autolock _l(mLock);
@@ -1166,11 +1177,10 @@
return remainder == 0 ? 0 : divisor - remainder;
}
-status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *replySize,
- void *pReplyData)
+status_t AudioFlinger::EffectModule::command(int32_t cmdCode,
+ const std::vector<uint8_t>& cmdData,
+ int32_t maxReplySize,
+ std::vector<uint8_t>* reply)
{
Mutex::Autolock _l(mLock);
ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface.get());
@@ -1181,63 +1191,68 @@
if (mStatus != NO_ERROR) {
return mStatus;
}
+ if (maxReplySize < 0 || maxReplySize > EFFECT_PARAM_SIZE_MAX) {
+ return -EINVAL;
+ }
+ size_t cmdSize = cmdData.size();
+ const effect_param_t* param = cmdSize >= sizeof(effect_param_t)
+ ? reinterpret_cast<const effect_param_t*>(cmdData.data())
+ : nullptr;
if (cmdCode == EFFECT_CMD_GET_PARAM &&
- (sizeof(effect_param_t) > cmdSize ||
- ((effect_param_t *)pCmdData)->psize > cmdSize
- - sizeof(effect_param_t))) {
+ (param == nullptr || param->psize > cmdSize - sizeof(effect_param_t))) {
android_errorWriteLog(0x534e4554, "32438594");
android_errorWriteLog(0x534e4554, "33003822");
return -EINVAL;
}
if (cmdCode == EFFECT_CMD_GET_PARAM &&
- (*replySize < sizeof(effect_param_t) ||
- ((effect_param_t *)pCmdData)->psize > *replySize - sizeof(effect_param_t))) {
+ (maxReplySize < sizeof(effect_param_t) ||
+ param->psize > maxReplySize - sizeof(effect_param_t))) {
android_errorWriteLog(0x534e4554, "29251553");
return -EINVAL;
}
if (cmdCode == EFFECT_CMD_GET_PARAM &&
- (sizeof(effect_param_t) > *replySize
- || ((effect_param_t *)pCmdData)->psize > *replySize
- - sizeof(effect_param_t)
- || ((effect_param_t *)pCmdData)->vsize > *replySize
- - sizeof(effect_param_t)
- - ((effect_param_t *)pCmdData)->psize
- || roundUpDelta(((effect_param_t *)pCmdData)->psize, (uint32_t)sizeof(int)) >
- *replySize
- - sizeof(effect_param_t)
- - ((effect_param_t *)pCmdData)->psize
- - ((effect_param_t *)pCmdData)->vsize)) {
+ (sizeof(effect_param_t) > maxReplySize
+ || param->psize > maxReplySize - sizeof(effect_param_t)
+ || param->vsize > maxReplySize - sizeof(effect_param_t)
+ - param->psize
+ || roundUpDelta(param->psize, (uint32_t) sizeof(int)) >
+ maxReplySize
+ - sizeof(effect_param_t)
+ - param->psize
+ - param->vsize)) {
ALOGV("\tLVM_ERROR : EFFECT_CMD_GET_PARAM: reply size inconsistent");
android_errorWriteLog(0x534e4554, "32705438");
return -EINVAL;
}
if ((cmdCode == EFFECT_CMD_SET_PARAM
- || cmdCode == EFFECT_CMD_SET_PARAM_DEFERRED) && // DEFERRED not generally used
- (sizeof(effect_param_t) > cmdSize
- || ((effect_param_t *)pCmdData)->psize > cmdSize
- - sizeof(effect_param_t)
- || ((effect_param_t *)pCmdData)->vsize > cmdSize
- - sizeof(effect_param_t)
- - ((effect_param_t *)pCmdData)->psize
- || roundUpDelta(((effect_param_t *)pCmdData)->psize, (uint32_t)sizeof(int)) >
- cmdSize
- - sizeof(effect_param_t)
- - ((effect_param_t *)pCmdData)->psize
- - ((effect_param_t *)pCmdData)->vsize)) {
+ || cmdCode == EFFECT_CMD_SET_PARAM_DEFERRED)
+ && // DEFERRED not generally used
+ (param == nullptr
+ || param->psize > cmdSize - sizeof(effect_param_t)
+ || param->vsize > cmdSize - sizeof(effect_param_t)
+ - param->psize
+ || roundUpDelta(param->psize,
+ (uint32_t) sizeof(int)) >
+ cmdSize
+ - sizeof(effect_param_t)
+ - param->psize
+ - param->vsize)) {
android_errorWriteLog(0x534e4554, "30204301");
return -EINVAL;
}
+ uint32_t replySize = maxReplySize;
+ reply->resize(replySize);
status_t status = mEffectInterface->command(cmdCode,
cmdSize,
- pCmdData,
- replySize,
- pReplyData);
+ const_cast<uint8_t*>(cmdData.data()),
+ &replySize,
+ reply->data());
+ reply->resize(status == NO_ERROR ? replySize : 0);
if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
- uint32_t size = (replySize == NULL) ? 0 : *replySize;
for (size_t i = 1; i < mHandles.size(); i++) {
EffectHandle *h = mHandles[i];
if (h != NULL && !h->disconnected()) {
- h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
+ h->commandExecuted(cmdCode, cmdData, *reply);
}
}
}
@@ -1547,19 +1562,18 @@
return INVALID_OPERATION;
}
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 3];
- effect_param_t *param = (effect_param_t*) buf32;
+ std::vector<uint8_t> request(sizeof(effect_param_t) + 3 * sizeof(uint32_t));
+ effect_param_t *param = (effect_param_t*) request.data();
param->psize = sizeof(int32_t);
param->vsize = sizeof(int32_t) * 2;
*(int32_t*)param->data = HG_PARAM_HAPTIC_INTENSITY;
*((int32_t*)param->data + 1) = id;
*((int32_t*)param->data + 2) = intensity;
- uint32_t size = sizeof(int32_t);
- status_t status = command(
- EFFECT_CMD_SET_PARAM, sizeof(effect_param_t) + param->psize + param->vsize,
- param, &size, ¶m->status);
+ std::vector<uint8_t> response;
+ status_t status = command(EFFECT_CMD_SET_PARAM, request, sizeof(int32_t), &response);
if (status == NO_ERROR) {
- status = param->status;
+ LOG_ALWAYS_FATAL_IF(response.size() != 4);
+ status = *reinterpret_cast<const status_t*>(response.data());
}
return status;
}
@@ -1642,9 +1656,9 @@
#define LOG_TAG "AudioFlinger::EffectHandle"
AudioFlinger::EffectHandle::EffectHandle(const sp<EffectBase>& effect,
- const sp<AudioFlinger::Client>& client,
- const sp<IEffectClient>& effectClient,
- int32_t priority)
+ const sp<AudioFlinger::Client>& client,
+ const sp<media::IEffectClient>& effectClient,
+ int32_t priority)
: BnEffect(),
mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
mPriority(priority), mHasControl(false), mEnabled(false), mDisconnected(false)
@@ -1678,20 +1692,24 @@
return mClient == 0 || mCblkMemory != 0 ? OK : NO_MEMORY;
}
-status_t AudioFlinger::EffectHandle::enable()
+#define RETURN(code) \
+ *_aidl_return = (code); \
+ return Status::ok();
+
+Status AudioFlinger::EffectHandle::enable(int32_t* _aidl_return)
{
AutoMutex _l(mLock);
ALOGV("enable %p", this);
sp<EffectBase> effect = mEffect.promote();
if (effect == 0 || mDisconnected) {
- return DEAD_OBJECT;
+ RETURN(DEAD_OBJECT);
}
if (!mHasControl) {
- return INVALID_OPERATION;
+ RETURN(INVALID_OPERATION);
}
if (mEnabled) {
- return NO_ERROR;
+ RETURN(NO_ERROR);
}
mEnabled = true;
@@ -1699,54 +1717,55 @@
status_t status = effect->updatePolicyState();
if (status != NO_ERROR) {
mEnabled = false;
- return status;
+ RETURN(status);
}
effect->checkSuspendOnEffectEnabled(true, false /*threadLocked*/);
// checkSuspendOnEffectEnabled() can suspend this same effect when enabled
if (effect->suspended()) {
- return NO_ERROR;
+ RETURN(NO_ERROR);
}
status = effect->setEnabled(true, true /*fromHandle*/);
if (status != NO_ERROR) {
mEnabled = false;
}
- return status;
+ RETURN(status);
}
-status_t AudioFlinger::EffectHandle::disable()
+Status AudioFlinger::EffectHandle::disable(int32_t* _aidl_return)
{
ALOGV("disable %p", this);
AutoMutex _l(mLock);
sp<EffectBase> effect = mEffect.promote();
if (effect == 0 || mDisconnected) {
- return DEAD_OBJECT;
+ RETURN(DEAD_OBJECT);
}
if (!mHasControl) {
- return INVALID_OPERATION;
+ RETURN(INVALID_OPERATION);
}
if (!mEnabled) {
- return NO_ERROR;
+ RETURN(NO_ERROR);
}
mEnabled = false;
effect->updatePolicyState();
if (effect->suspended()) {
- return NO_ERROR;
+ RETURN(NO_ERROR);
}
status_t status = effect->setEnabled(false, true /*fromHandle*/);
- return status;
+ RETURN(status);
}
-void AudioFlinger::EffectHandle::disconnect()
+Status AudioFlinger::EffectHandle::disconnect()
{
ALOGV("%s %p", __FUNCTION__, this);
disconnect(true);
+ return Status::ok();
}
void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
@@ -1783,11 +1802,16 @@
}
}
-status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *replySize,
- void *pReplyData)
+Status AudioFlinger::EffectHandle::getCblk(media::SharedFileRegion* _aidl_return) {
+ LOG_ALWAYS_FATAL_IF(!convertIMemoryToSharedFileRegion(mCblkMemory, _aidl_return));
+ return Status::ok();
+}
+
+Status AudioFlinger::EffectHandle::command(int32_t cmdCode,
+ const std::vector<uint8_t>& cmdData,
+ int32_t maxResponseSize,
+ std::vector<uint8_t>* response,
+ int32_t* _aidl_return)
{
ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
cmdCode, mHasControl, mEffect.unsafe_get());
@@ -1807,49 +1831,46 @@
break;
}
android_errorWriteLog(0x534e4554, "62019992");
- return BAD_VALUE;
+ RETURN(BAD_VALUE);
}
if (cmdCode == EFFECT_CMD_ENABLE) {
- if (*replySize < sizeof(int)) {
+ if (maxResponseSize < sizeof(int)) {
android_errorWriteLog(0x534e4554, "32095713");
- return BAD_VALUE;
+ RETURN(BAD_VALUE);
}
- *(int *)pReplyData = NO_ERROR;
- *replySize = sizeof(int);
- return enable();
+ writeToBuffer(NO_ERROR, response);
+ return enable(_aidl_return);
} else if (cmdCode == EFFECT_CMD_DISABLE) {
- if (*replySize < sizeof(int)) {
+ if (maxResponseSize < sizeof(int)) {
android_errorWriteLog(0x534e4554, "32095713");
- return BAD_VALUE;
+ RETURN(BAD_VALUE);
}
- *(int *)pReplyData = NO_ERROR;
- *replySize = sizeof(int);
- return disable();
+ writeToBuffer(NO_ERROR, response);
+ return disable(_aidl_return);
}
AutoMutex _l(mLock);
sp<EffectBase> effect = mEffect.promote();
if (effect == 0 || mDisconnected) {
- return DEAD_OBJECT;
+ RETURN(DEAD_OBJECT);
}
// only get parameter command is permitted for applications not controlling the effect
if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
- return INVALID_OPERATION;
+ RETURN(INVALID_OPERATION);
}
// handle commands that are not forwarded transparently to effect engine
if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
if (mClient == 0) {
- return INVALID_OPERATION;
+ RETURN(INVALID_OPERATION);
}
- if (*replySize < sizeof(int)) {
+ if (maxResponseSize < sizeof(int)) {
android_errorWriteLog(0x534e4554, "32095713");
- return BAD_VALUE;
+ RETURN(BAD_VALUE);
}
- *(int *)pReplyData = NO_ERROR;
- *replySize = sizeof(int);
+ writeToBuffer(NO_ERROR, response);
// No need to trylock() here as this function is executed in the binder thread serving a
// particular client process: no risk to block the whole media server process or mixer
@@ -1862,10 +1883,10 @@
serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
mCblk->serverIndex = 0;
mCblk->clientIndex = 0;
- return BAD_VALUE;
+ RETURN(BAD_VALUE);
}
status_t status = NO_ERROR;
- effect_param_t *param = NULL;
+ std::vector<uint8_t> param;
for (uint32_t index = serverIndex; index < clientIndex;) {
int *p = (int *)(mBuffer + index);
const int size = *p++;
@@ -1877,23 +1898,16 @@
break;
}
- // copy to local memory in case of client corruption b/32220769
- auto *newParam = (effect_param_t *)realloc(param, size);
- if (newParam == NULL) {
- ALOGW("command(): out of memory");
- status = NO_MEMORY;
- break;
- }
- param = newParam;
- memcpy(param, p, size);
+ std::copy(reinterpret_cast<const uint8_t*>(p),
+ reinterpret_cast<const uint8_t*>(p) + size,
+ std::back_inserter(param));
- int reply = 0;
- uint32_t rsize = sizeof(reply);
+ std::vector<uint8_t> replyBuffer;
status_t ret = effect->command(EFFECT_CMD_SET_PARAM,
- size,
param,
- &rsize,
- &reply);
+ sizeof(int),
+ &replyBuffer);
+ int reply = *reinterpret_cast<const int*>(replyBuffer.data());
// verify shared memory: server index shouldn't change; client index can't go back.
if (serverIndex != mCblk->serverIndex
@@ -1906,21 +1920,24 @@
// stop at first error encountered
if (ret != NO_ERROR) {
status = ret;
- *(int *)pReplyData = reply;
+ writeToBuffer(reply, response);
break;
} else if (reply != NO_ERROR) {
- *(int *)pReplyData = reply;
+ writeToBuffer(reply, response);
break;
}
index += size;
}
- free(param);
mCblk->serverIndex = 0;
mCblk->clientIndex = 0;
- return status;
+ RETURN(status);
}
- return effect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+ status_t status = effect->command(cmdCode,
+ cmdData,
+ maxResponseSize,
+ response);
+ RETURN(status);
}
void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
@@ -1936,13 +1953,11 @@
}
void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t replySize,
- void *pReplyData)
+ const std::vector<uint8_t>& cmdData,
+ const std::vector<uint8_t>& replyData)
{
if (mEffectClient != 0) {
- mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
+ mEffectClient->commandExecuted(cmdCode, cmdData, replyData);
}
}
@@ -1955,13 +1970,6 @@
}
}
-status_t AudioFlinger::EffectHandle::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
-{
- return BnEffect::onTransact(code, data, reply, flags);
-}
-
-
void AudioFlinger::EffectHandle::dumpToBuffer(char* buffer, size_t size)
{
bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
@@ -3012,10 +3020,14 @@
Mutex::Autolock _l(mProxyLock);
if (status == NO_ERROR) {
for (auto& handle : mEffectHandles) {
+ Status bs;
if (enabled) {
- status = handle.second->enable();
+ bs = handle.second->enable(&status);
} else {
- status = handle.second->disable();
+ bs = handle.second->disable(&status);
+ }
+ if (!bs.isOk()) {
+ status = bs.transactionError();
}
}
}
@@ -3074,7 +3086,7 @@
__func__, port->type, port->ext.device.type,
port->ext.device.address, port->id, patch.isSoftware());
if (port->type != AUDIO_PORT_TYPE_DEVICE || port->ext.device.type != mDevice.mType
- || port->ext.device.address != mDevice.mAddress) {
+ || port->ext.device.address != mDevice.address()) {
return NAME_NOT_FOUND;
}
status_t status = NAME_NOT_FOUND;
@@ -3123,10 +3135,14 @@
status = BAD_VALUE;
}
if (status == NO_ERROR || status == ALREADY_EXISTS) {
+ Status bs;
if (isEnabled()) {
- (*handle)->enable();
+ bs = (*handle)->enable(&status);
} else {
- (*handle)->disable();
+ bs = (*handle)->disable(&status);
+ }
+ if (!bs.isOk()) {
+ status = bs.transactionError();
}
}
return status;
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 3cc5a44..03bdc60 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -133,11 +133,10 @@
void setSuspended(bool suspended);
bool suspended() const;
- virtual status_t command(uint32_t cmdCode __unused,
- uint32_t cmdSize __unused,
- void *pCmdData __unused,
- uint32_t *replySize __unused,
- void *pReplyData __unused) { return NO_ERROR; };
+ virtual status_t command(int32_t __unused,
+ const std::vector<uint8_t>& __unused,
+ int32_t __unused,
+ std::vector<uint8_t>* __unused) { return NO_ERROR; };
void setCallback(const sp<EffectCallbackInterface>& callback) { mCallback = callback; }
sp<EffectCallbackInterface>& callback() { return mCallback; }
@@ -145,7 +144,7 @@
status_t addHandle(EffectHandle *handle);
ssize_t disconnectHandle(EffectHandle *handle, bool unpinIfLast);
ssize_t removeHandle(EffectHandle *handle);
- virtual ssize_t removeHandle_l(EffectHandle *handle);
+ ssize_t removeHandle_l(EffectHandle *handle);
EffectHandle* controlHandle_l();
bool purgeHandles();
@@ -214,11 +213,10 @@
void process();
bool updateState();
- status_t command(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *replySize,
- void *pReplyData) override;
+ status_t command(int32_t cmdCode,
+ const std::vector<uint8_t>& cmdData,
+ int32_t maxReplySize,
+ std::vector<uint8_t>* reply) override;
void reset_l();
status_t configure();
@@ -241,8 +239,6 @@
return mOutBuffer != 0 ? reinterpret_cast<int16_t*>(mOutBuffer->ptr()) : NULL;
}
- ssize_t removeHandle_l(EffectHandle *handle) override;
-
status_t setDevices(const AudioDeviceTypeAddrVector &devices);
status_t setInputDevice(const AudioDeviceTypeAddr &device);
status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
@@ -322,32 +318,29 @@
// There is one EffectHandle object for each application controlling (or using)
// an effect module.
// The EffectHandle is obtained by calling AudioFlinger::createEffect().
-class EffectHandle: public android::BnEffect {
+class EffectHandle: public android::media::BnEffect {
public:
EffectHandle(const sp<EffectBase>& effect,
const sp<AudioFlinger::Client>& client,
- const sp<IEffectClient>& effectClient,
+ const sp<media::IEffectClient>& effectClient,
int32_t priority);
virtual ~EffectHandle();
virtual status_t initCheck();
// IEffect
- virtual status_t enable();
- virtual status_t disable();
- virtual status_t command(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *replySize,
- void *pReplyData);
- virtual void disconnect();
-private:
- void disconnect(bool unpinIfLast);
-public:
- virtual sp<IMemory> getCblk() const { return mCblkMemory; }
- virtual status_t onTransact(uint32_t code, const Parcel& data,
- Parcel* reply, uint32_t flags);
+ android::binder::Status enable(int32_t* _aidl_return) override;
+ android::binder::Status disable(int32_t* _aidl_return) override;
+ android::binder::Status command(int32_t cmdCode,
+ const std::vector<uint8_t>& cmdData,
+ int32_t maxResponseSize,
+ std::vector<uint8_t>* response,
+ int32_t* _aidl_return) override;
+ android::binder::Status disconnect() override;
+ android::binder::Status getCblk(media::SharedFileRegion* _aidl_return) override;
+private:
+ void disconnect(bool unpinIfLast);
// Give or take control of effect module
// - hasControl: true if control is given, false if removed
@@ -355,10 +348,8 @@
// - enabled: state of the effect when control is passed
void setControl(bool hasControl, bool signal, bool enabled);
void commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t replySize,
- void *pReplyData);
+ const std::vector<uint8_t>& cmdData,
+ const std::vector<uint8_t>& replyData);
void setEnabled(bool enabled);
bool enabled() const { return mEnabled; }
@@ -381,19 +372,20 @@
friend class AudioFlinger; // for mEffect, mHasControl, mEnabled
DISALLOW_COPY_AND_ASSIGN(EffectHandle);
- Mutex mLock; // protects IEffect method calls
- wp<EffectBase> mEffect; // pointer to controlled EffectModule
- sp<IEffectClient> mEffectClient; // callback interface for client notifications
- /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect()
- sp<IMemory> mCblkMemory; // shared memory for control block
- effect_param_cblk_t* mCblk; // control block for deferred parameter setting via
- // shared memory
- uint8_t* mBuffer; // pointer to parameter area in shared memory
- int mPriority; // client application priority to control the effect
- bool mHasControl; // true if this handle is controlling the effect
- bool mEnabled; // cached enable state: needed when the effect is
- // restored after being suspended
- bool mDisconnected; // Set to true by disconnect()
+ Mutex mLock; // protects IEffect method calls
+ wp<EffectBase> mEffect; // pointer to controlled EffectModule
+ sp<media::IEffectClient> mEffectClient; // callback interface for client notifications
+ /*const*/ sp<Client> mClient; // client for shared memory allocation, see
+ // disconnect()
+ sp<IMemory> mCblkMemory; // shared memory for control block
+ effect_param_cblk_t* mCblk; // control block for deferred parameter setting via
+ // shared memory
+ uint8_t* mBuffer; // pointer to parameter area in shared memory
+ int mPriority; // client application priority to control the effect
+ bool mHasControl; // true if this handle is controlling the effect
+ bool mEnabled; // cached enable state: needed when the effect is
+ // restored after being suspended
+ bool mDisconnected; // Set to true by disconnect()
};
// the EffectChain class represents a group of effects associated to one audio session.
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index bf2e953..cd3c743 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -27,7 +27,6 @@
#include "Configuration.h"
#include <time.h>
-#include <utils/Debug.h>
#include <utils/Log.h>
#include <utils/Trace.h>
#include <system/audio.h>
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
index a42e09c..3f20282 100644
--- a/services/audioflinger/FastMixerDumpState.cpp
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -24,7 +24,6 @@
#include <cpustats/ThreadCpuUsage.h>
#endif
#endif
-#include <utils/Debug.h>
#include <utils/Log.h>
#include "FastMixerDumpState.h"
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index cdf3702..affc09e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -116,6 +116,8 @@
namespace android {
+using media::IEffectClient;
+
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
static const int8_t kMaxTrackRetries = 50;
@@ -1944,7 +1946,7 @@
// here instead of constructor of PlaybackThread so that the onFirstRef
// callback would not be made on an incompletely constructed object.
if (mOutput->stream->setEventCallback(this) != OK) {
- ALOGE("Failed to add event callback");
+ ALOGD("Failed to add event callback");
}
}
run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
@@ -2356,10 +2358,20 @@
}
}
+ // Set DIRECT flag if current thread is DirectOutputThread. This can
+ // happen when the playback is rerouted to direct output thread by
+ // dynamic audio policy.
+ // Do NOT report the flag changes back to client, since the client
+ // doesn't explicitly request a direct flag.
+ audio_output_flags_t trackFlags = *flags;
+ if (mType == DIRECT) {
+ trackFlags = static_cast<audio_output_flags_t>(trackFlags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+
track = new Track(this, client, streamType, attr, sampleRate, format,
channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
- sessionId, creatorPid, uid, *flags, TrackBase::TYPE_DEFAULT, portId);
+ sessionId, creatorPid, uid, trackFlags, TrackBase::TYPE_DEFAULT, portId);
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
@@ -8584,7 +8596,7 @@
// store new device and send to effects
mInDeviceTypeAddr.mType = patch->sources[0].ext.device.type;
- mInDeviceTypeAddr.mAddress = patch->sources[0].ext.device.address;
+ mInDeviceTypeAddr.setAddress(patch->sources[0].ext.device.address);
audio_port_handle_t deviceId = patch->sources[0].id;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setInputDevice_l(inDeviceTypeAddr());
@@ -9225,7 +9237,7 @@
deviceId = patch->sources[0].id;
numDevices = mPatch.num_sources;
sourceDeviceTypeAddr.mType = patch->sources[0].ext.device.type;
- sourceDeviceTypeAddr.mAddress = patch->sources[0].ext.device.address;
+ sourceDeviceTypeAddr.setAddress(patch->sources[0].ext.device.address);
}
for (size_t i = 0; i < mEffectChains.size(); i++) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 2e81ae7..ac41e82 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -349,7 +349,7 @@
sp<EffectHandle> createEffect_l(
const sp<AudioFlinger::Client>& client,
- const sp<IEffectClient>& effectClient,
+ const sp<media::IEffectClient>& effectClient,
int32_t priority,
audio_session_t sessionId,
effect_descriptor_t *desc,
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 8d0e5db..0f3ed14 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -250,12 +250,12 @@
virtual status_t registerPolicyMixes(const Vector<AudioMix>& mixes) = 0;
virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes) = 0;
- virtual status_t setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices)
+ virtual status_t setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices)
= 0;
virtual status_t removeUidDeviceAffinities(uid_t uid) = 0;
virtual status_t setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices) = 0;
+ const AudioDeviceTypeAddrVector& devices) = 0;
virtual status_t removeUserIdDeviceAffinities(int userId) = 0;
virtual status_t startAudioSource(const struct audio_port_config *source,
@@ -295,13 +295,17 @@
virtual bool isCallScreenModeSupported() = 0;
- virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device) = 0;
+ virtual status_t setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role,
+ const AudioDeviceTypeAddrVector &devices) = 0;
- virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy) = 0;
+ virtual status_t removeDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role) = 0;
- virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device) = 0;
+
+ virtual status_t getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role,
+ AudioDeviceTypeAddrVector &devices) = 0;
};
diff --git a/services/audiopolicy/common/include/Volume.h b/services/audiopolicy/common/include/Volume.h
index 7c8ce83..736f8b2 100644
--- a/services/audiopolicy/common/include/Volume.h
+++ b/services/audiopolicy/common/include/Volume.h
@@ -126,6 +126,7 @@
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
case AUDIO_DEVICE_OUT_USB_HEADSET:
+ case AUDIO_DEVICE_OUT_BLE_HEADSET:
return DEVICE_CATEGORY_HEADSET;
case AUDIO_DEVICE_OUT_HEARING_AID:
return DEVICE_CATEGORY_HEARING_AID;
@@ -139,6 +140,7 @@
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
case AUDIO_DEVICE_OUT_USB_ACCESSORY:
case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+ case AUDIO_DEVICE_OUT_BLE_SPEAKER:
default:
return DEVICE_CATEGORY_SPEAKER;
}
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index b82305d..c6bdb04 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -101,7 +101,7 @@
* An example of failure is when there are already rules in place to restrict
* a mix to the given uid (i.e. when a MATCH_UID rule was set for it).
*/
- status_t setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices);
+ status_t setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices);
status_t removeUidDeviceAffinities(uid_t uid);
status_t getDevicesForUid(uid_t uid, Vector<AudioDeviceTypeAddr>& devices) const;
@@ -115,7 +115,7 @@
* An example of failure is when there are already rules in place to restrict
* a mix to the given userId (i.e. when a MATCH_USERID rule was set for it).
*/
- status_t setUserIdDeviceAffinities(int userId, const Vector<AudioDeviceTypeAddr>& devices);
+ status_t setUserIdDeviceAffinities(int userId, const AudioDeviceTypeAddrVector& devices);
status_t removeUserIdDeviceAffinities(int userId);
status_t getDevicesForUserId(int userId, Vector<AudioDeviceTypeAddr>& devices) const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 0f9bcc1..c51d6a9 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -146,6 +146,15 @@
// 4) the combination of all devices is invalid for selection
sp<DeviceDescriptor> getDeviceForOpening() const;
+ // Return the device descriptor that matches the given AudioDeviceTypeAddr
+ sp<DeviceDescriptor> getDeviceFromDeviceTypeAddr(
+ const AudioDeviceTypeAddr& deviceTypeAddr) const;
+
+ // Return the device vector that contains device descriptor whose AudioDeviceTypeAddr appears
+ // in the given AudioDeviceTypeAddrVector
+ DeviceVector getDevicesFromDeviceTypeAddrVec(
+ const AudioDeviceTypeAddrVector& deviceTypeAddrVector) const;
+
// If there are devices with the given type and the devices to add is not empty,
// remove all the devices with the given type and add all the devices to add.
void replaceDevicesByType(audio_devices_t typeToRemove, const DeviceVector &devicesToAdd);
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index b6de4be..fc1d0e2 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -463,7 +463,7 @@
}
status_t AudioPolicyMixCollection::setUidDeviceAffinities(uid_t uid,
- const Vector<AudioDeviceTypeAddr>& devices) {
+ const AudioDeviceTypeAddrVector& devices) {
// verify feasibility: for each player mix: if it already contains a
// "match uid" rule for this uid, return an error
// (adding a uid-device affinity would result in contradictory rules)
@@ -565,7 +565,7 @@
}
status_t AudioPolicyMixCollection::setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices) {
+ const AudioDeviceTypeAddrVector& devices) {
// verify feasibility: for each player mix: if it already contains a
// "match userId" rule for this userId, return an error
// (adding a userId-device affinity would result in contradictory rules)
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index d410ffd..a896157 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -390,6 +390,24 @@
return nullptr;
}
+sp<DeviceDescriptor> DeviceVector::getDeviceFromDeviceTypeAddr(
+ const AudioDeviceTypeAddr& deviceTypeAddr) const {
+ return getDevice(deviceTypeAddr.mType, String8(deviceTypeAddr.getAddress()),
+ AUDIO_FORMAT_DEFAULT);
+}
+
+DeviceVector DeviceVector::getDevicesFromDeviceTypeAddrVec(
+ const AudioDeviceTypeAddrVector& deviceTypeAddrVector) const {
+ DeviceVector devices;
+ for (const auto& deviceTypeAddr : deviceTypeAddrVector) {
+ sp<DeviceDescriptor> device = getDeviceFromDeviceTypeAddr(deviceTypeAddr);
+ if (device != nullptr) {
+ devices.add(device);
+ }
+ }
+ return devices;
+}
+
void DeviceVector::replaceDevicesByType(
audio_devices_t typeToRemove, const DeviceVector &devicesToAdd) {
DeviceVector devicesToRemove = getDevicesFromType(typeToRemove);
diff --git a/services/audiopolicy/config/a2dp_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/a2dp_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..2d323f6
--- /dev/null
+++ b/services/audiopolicy/config/a2dp_audio_policy_configuration_7_0.xml
@@ -0,0 +1,44 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- A2dp Audio HAL Audio Policy Configuration file -->
+<module name="a2dp" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="a2dp output" role="source"/>
+ <mixPort name="a2dp input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="BT A2DP Out"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Headphones"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Speaker"
+ sources="a2dp output"/>
+ <route type="mix" sink="a2dp input"
+ sources="BT A2DP In"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/a2dp_in_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/a2dp_in_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..d59ad70
--- /dev/null
+++ b/services/audiopolicy/config/a2dp_in_audio_policy_configuration_7_0.xml
@@ -0,0 +1,22 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Bluetooth Input Audio HAL Audio Policy Configuration file -->
+<module name="a2dp" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="a2dp input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="BT A2DP In" type="AUDIO_DEVICE_IN_BLUETOOTH_A2DP" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="a2dp input"
+ sources="BT A2DP In"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/audio_policy_configuration_7_0.xml b/services/audiopolicy/config/audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..6087bf2
--- /dev/null
+++ b/services/audiopolicy/config/audio_policy_configuration_7_0.xml
@@ -0,0 +1,211 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” -->
+
+ <!-- Global configuration Decalaration -->
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+
+ <!-- Modules section:
+ There is one section per audio HW module present on the platform.
+ Each module section will contains two mandatory tags for audio HAL “halVersion” and “name”.
+ The module names are the same as in current .conf file:
+ “primary”, “A2DP”, “remote_submix”, “USB”
+ Each module will contain the following sections:
+ “devicePorts”: a list of device descriptors for all input and output devices accessible via this
+ module.
+ This contains both permanently attached devices and removable devices.
+ “mixPorts”: listing all output and input streams exposed by the audio HAL
+ “routes”: list of possible connections between input and output devices or between stream and
+ devices.
+ "route": is defined by an attribute:
+ -"type": <mux|mix> means all sources are mutual exclusive (mux) or can be mixed (mix)
+ -"sink": the sink involved in this route
+ -"sources": all the sources than can be connected to the sink via vis route
+ “attachedDevices”: permanently attached devices.
+ The attachedDevices section is a list of devices names. The names correspond to device names
+ defined in <devicePorts> section.
+ “defaultOutputDevice”: device to be used by default when no policy rule applies
+ -->
+ <modules>
+ <!-- Primary Audio HAL -->
+ <module name="primary" halVersion="3.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ <item>Built-In Back Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="deep_buffer" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DEEP_BUFFER">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="compressed_offload" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+ <profile name="" format="AUDIO_FORMAT_MP3"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AAC"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ <profile name="" format="AUDIO_FORMAT_AAC_LC"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_MONO"/>
+ </mixPort>
+ <mixPort name="voice_tx" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </mixPort>
+ <mixPort name="voice_rx" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
+ <devicePort tagName="Earpiece" type="AUDIO_DEVICE_OUT_EARPIECE" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </devicePort>
+ <devicePort tagName="Speaker" role="sink" type="AUDIO_DEVICE_OUT_SPEAKER" address="">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <gains>
+ <gain name="gain_1" mode="AUDIO_GAIN_MODE_JOINT"
+ minValueMB="-8400"
+ maxValueMB="4000"
+ defaultValueMB="0"
+ stepValueMB="100"/>
+ </gains>
+ </devicePort>
+ <devicePort tagName="Wired Headset" type="AUDIO_DEVICE_OUT_WIRED_HEADSET" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="Wired Headphones" type="AUDIO_DEVICE_OUT_WIRED_HEADPHONE" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </devicePort>
+ <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </devicePort>
+ <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </devicePort>
+ <devicePort tagName="Telephony Tx" type="AUDIO_DEVICE_OUT_TELEPHONY_TX" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+ </devicePort>
+
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </devicePort>
+ <devicePort tagName="Built-In Back Mic" type="AUDIO_DEVICE_IN_BACK_MIC" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </devicePort>
+ <devicePort tagName="Wired Headset Mic" type="AUDIO_DEVICE_IN_WIRED_HEADSET" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO AUDIO_CHANNEL_IN_FRONT_BACK"/>
+ </devicePort>
+ <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </devicePort>
+ <devicePort tagName="Telephony Rx" type="AUDIO_DEVICE_IN_TELEPHONY_RX" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </devicePort>
+ </devicePorts>
+ <!-- route declaration, i.e. list all available sources for a given sink -->
+ <routes>
+ <route type="mix" sink="Earpiece"
+ sources="primary output,deep_buffer,BT SCO Headset Mic"/>
+ <route type="mix" sink="Speaker"
+ sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+ <route type="mix" sink="Wired Headset"
+ sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+ <route type="mix" sink="Wired Headphones"
+ sources="primary output,deep_buffer,compressed_offload,BT SCO Headset Mic,Telephony Rx"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic"/>
+ <route type="mix" sink="Telephony Tx"
+ sources="Built-In Mic,Built-In Back Mic,Wired Headset Mic,BT SCO Headset Mic, voice_tx"/>
+ <route type="mix" sink="voice_rx"
+ sources="Telephony Rx"/>
+ </routes>
+
+ </module>
+
+ <!-- A2dp Input Audio HAL -->
+ <xi:include href="a2dp_in_audio_policy_configuration_7_0.xml"/>
+
+ <!-- Usb Audio HAL -->
+ <xi:include href="usb_audio_policy_configuration.xml"/>
+
+ <!-- Remote Submix Audio HAL -->
+ <xi:include href="r_submix_audio_policy_configuration.xml"/>
+
+ <!-- Bluetooth Audio HAL -->
+ <xi:include href="bluetooth_audio_policy_configuration_7_0.xml"/>
+
+ <!-- MSD Audio HAL (optional) -->
+ <xi:include href="msd_audio_policy_configuration_7_0.xml"/>
+
+ </modules>
+ <!-- End of Modules section -->
+
+ <!-- Volume section:
+ IMPORTANT NOTE: Volume tables have been moved to engine configuration.
+ Keep it here for legacy.
+ Engine will fallback on these files if none are provided by engine.
+ -->
+
+ <xi:include href="audio_policy_volumes.xml"/>
+ <xi:include href="default_volume_tables.xml"/>
+
+ <!-- End of Volume section -->
+
+ <!-- Surround Sound configuration -->
+
+ <xi:include href="surround_sound_configuration_5_0.xml"/>
+
+ <!-- End of Surround Sound configuration -->
+
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/config/bluetooth_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/bluetooth_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..2dffe02
--- /dev/null
+++ b/services/audiopolicy/config/bluetooth_audio_policy_configuration_7_0.xml
@@ -0,0 +1,44 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Bluetooth Audio HAL Audio Policy Configuration file -->
+<module name="bluetooth" halVersion="2.0">
+ <mixPorts>
+ <!-- A2DP Audio Ports -->
+ <mixPort name="a2dp output" role="source"/>
+ <!-- Hearing AIDs Audio Ports -->
+ <mixPort name="hearing aid output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="24000 16000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <!-- A2DP Audio Ports -->
+ <devicePort tagName="BT A2DP Out" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 88200 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Headphones" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 88200 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="BT A2DP Speaker" type="AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100 48000 88200 96000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <!-- Hearing AIDs Audio Ports -->
+ <devicePort tagName="BT Hearing Aid Out" type="AUDIO_DEVICE_OUT_HEARING_AID" role="sink"/>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="BT A2DP Out"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Headphones"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT A2DP Speaker"
+ sources="a2dp output"/>
+ <route type="mix" sink="BT Hearing Aid Out"
+ sources="hearing aid output"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/hearing_aid_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/hearing_aid_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..8c364e4
--- /dev/null
+++ b/services/audiopolicy/config/hearing_aid_audio_policy_configuration_7_0.xml
@@ -0,0 +1,17 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Hearing aid Audio HAL Audio Policy Configuration file -->
+<module name="hearing_aid" halVersion="2.0">
+ <mixPorts>
+ <mixPort name="hearing aid output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="24000 16000"
+ channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="BT Hearing Aid Out" type="AUDIO_DEVICE_OUT_HEARING_AID" role="sink"/>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="BT Hearing Aid Out" sources="hearing aid output"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/msd_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/msd_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..ae0ba80
--- /dev/null
+++ b/services/audiopolicy/config/msd_audio_policy_configuration_7_0.xml
@@ -0,0 +1,78 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright (C) 2017-2018 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<!-- Multi Stream Decoder Audio Policy Configuration file -->
+<module name="msd" halVersion="2.0">
+ <attachedDevices>
+ <item>MS12 Input</item>
+ <item>MS12 Output</item>
+ </attachedDevices>
+ <mixPorts>
+ <mixPort name="ms12 input" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="ms12 compressed input" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_AC4"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ </mixPort>
+ <!-- The HW AV Sync flag is not required, but is recommended -->
+ <mixPort name="ms12 output" role="sink" flags="AUDIO_INPUT_FLAG_HW_AV_SYNC|AUDIO_INPUT_FLAG_DIRECT">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="MS12 Input" type="AUDIO_DEVICE_OUT_BUS" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ <profile name="" format="AUDIO_FORMAT_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_E_AC3_JOC"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ <profile name="" format="AUDIO_FORMAT_AC4"
+ samplingRates="32000 44100 48000"
+ channelMasks="AUDIO_CHANNEL_OUT_MONO AUDIO_CHANNEL_OUT_STEREO AUDIO_CHANNEL_OUT_5POINT1 AUDIO_CHANNEL_OUT_7POINT1"/>
+ </devicePort>
+ <devicePort tagName="MS12 Output" type="AUDIO_DEVICE_IN_BUS" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="MS12 Input" sources="ms12 input,ms12 compressed input"/>
+ <route type="mix" sink="ms12 output" sources="MS12 Output"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/config/primary_audio_policy_configuration_7_0.xml b/services/audiopolicy/config/primary_audio_policy_configuration_7_0.xml
new file mode 100644
index 0000000..68a56b2
--- /dev/null
+++ b/services/audiopolicy/config/primary_audio_policy_configuration_7_0.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Default Primary Audio HAL Module Audio Policy Configuration include file -->
+<module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="44100" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000 16000" channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+ </devicePort>
+
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker"
+ sources="primary output"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/engine/common/include/EngineBase.h b/services/audiopolicy/engine/common/include/EngineBase.h
index 7f339dc..804a802 100755
--- a/services/audiopolicy/engine/common/include/EngineBase.h
+++ b/services/audiopolicy/engine/common/include/EngineBase.h
@@ -93,13 +93,13 @@
void dump(String8 *dst) const override;
- status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device) override;
+ status_t setDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role,
+ const AudioDeviceTypeAddrVector &devices) override;
- status_t removePreferredDeviceForStrategy(product_strategy_t strategy) override;
+ status_t removeDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role) override;
- status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device) const override;
+ status_t getDevicesForRoleAndStrategy(product_strategy_t strategy, device_role_t role,
+ AudioDeviceTypeAddrVector &devices) const override;
engineConfig::ParsingResult loadAudioPolicyEngineConfig();
diff --git a/services/audiopolicy/engine/common/include/ProductStrategy.h b/services/audiopolicy/engine/common/include/ProductStrategy.h
index 3ebe7d1..c505456 100644
--- a/services/audiopolicy/engine/common/include/ProductStrategy.h
+++ b/services/audiopolicy/engine/common/include/ProductStrategy.h
@@ -28,8 +28,11 @@
#include <utils/String8.h>
#include <media/AudioAttributes.h>
#include <media/AudioContainers.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioPolicy.h>
+#include <vector>
+
namespace android {
/**
@@ -164,7 +167,8 @@
product_strategy_t mDefaultStrategy = PRODUCT_STRATEGY_NONE;
};
-class ProductStrategyPreferredRoutingMap : public std::map<product_strategy_t, AudioDeviceTypeAddr>
+class ProductStrategyPreferredRoutingMap : public std::map<product_strategy_t,
+ AudioDeviceTypeAddrVector>
{
public:
void dump(String8 *dst, int spaces = 0) const;
diff --git a/services/audiopolicy/engine/common/src/EngineBase.cpp b/services/audiopolicy/engine/common/src/EngineBase.cpp
index 1bc7fe3..ae4f7f4 100644
--- a/services/audiopolicy/engine/common/src/EngineBase.cpp
+++ b/services/audiopolicy/engine/common/src/EngineBase.cpp
@@ -339,8 +339,8 @@
return NO_ERROR;
}
-status_t EngineBase::setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device)
+status_t EngineBase::setDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role,
+ const AudioDeviceTypeAddrVector &devices)
{
// verify strategy exists
if (mProductStrategies.find(strategy) == mProductStrategies.end()) {
@@ -348,11 +348,24 @@
return BAD_VALUE;
}
- mProductStrategyPreferredDevices[strategy] = device;
+ switch (role) {
+ case DEVICE_ROLE_PREFERRED:
+ mProductStrategyPreferredDevices[strategy] = devices;
+ break;
+ case DEVICE_ROLE_DISABLED:
+ // TODO: support set devices role as disabled for strategy.
+ ALOGI("%s no implemented for role as %d", __func__, role);
+ break;
+ case DEVICE_ROLE_NONE:
+ // Intentionally fall-through as it is no need to set device role as none for a strategy.
+ default:
+ ALOGE("%s invalid role %d", __func__, role);
+ return BAD_VALUE;
+ }
return NO_ERROR;
}
-status_t EngineBase::removePreferredDeviceForStrategy(product_strategy_t strategy)
+status_t EngineBase::removeDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role)
{
// verify strategy exists
if (mProductStrategies.find(strategy) == mProductStrategies.end()) {
@@ -360,29 +373,53 @@
return BAD_VALUE;
}
- if (mProductStrategyPreferredDevices.erase(strategy) == 0) {
- // no preferred device was set
- return NAME_NOT_FOUND;
+ switch (role) {
+ case DEVICE_ROLE_PREFERRED:
+ if (mProductStrategyPreferredDevices.erase(strategy) == 0) {
+ // no preferred device was set
+ return NAME_NOT_FOUND;
+ }
+ break;
+ case DEVICE_ROLE_DISABLED:
+ // TODO: support remove devices role as disabled for strategy.
+ ALOGI("%s no implemented for role as %d", __func__, role);
+ break;
+ case DEVICE_ROLE_NONE:
+ // Intentionally fall-through as it makes no sense to remove devices with
+ // role as DEVICE_ROLE_NONE for a strategy
+ default:
+ ALOGE("%s invalid role %d", __func__, role);
+ return BAD_VALUE;
}
return NO_ERROR;
}
-status_t EngineBase::getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device) const
+status_t EngineBase::getDevicesForRoleAndStrategy(product_strategy_t strategy, device_role_t role,
+ AudioDeviceTypeAddrVector &devices) const
{
// verify strategy exists
if (mProductStrategies.find(strategy) == mProductStrategies.end()) {
ALOGE("%s unknown strategy %u", __func__, strategy);
return BAD_VALUE;
}
- // preferred device for this strategy?
- auto devIt = mProductStrategyPreferredDevices.find(strategy);
- if (devIt == mProductStrategyPreferredDevices.end()) {
- ALOGV("%s no preferred device for strategy %u", __func__, strategy);
- return NAME_NOT_FOUND;
- }
- device = devIt->second;
+ switch (role) {
+ case DEVICE_ROLE_PREFERRED: {
+ // preferred device for this strategy?
+ auto devIt = mProductStrategyPreferredDevices.find(strategy);
+ if (devIt == mProductStrategyPreferredDevices.end()) {
+ ALOGV("%s no preferred device for strategy %u", __func__, strategy);
+ return NAME_NOT_FOUND;
+ }
+
+ devices = devIt->second;
+ } break;
+ case DEVICE_ROLE_NONE:
+ // Intentionally fall-through as the DEVICE_ROLE_NONE is never set
+ default:
+ ALOGE("%s invalid role %d", __func__, role);
+ return BAD_VALUE;
+ }
return NO_ERROR;
}
diff --git a/services/audiopolicy/engine/common/src/ProductStrategy.cpp b/services/audiopolicy/engine/common/src/ProductStrategy.cpp
index 151c7bb..060568a 100644
--- a/services/audiopolicy/engine/common/src/ProductStrategy.cpp
+++ b/services/audiopolicy/engine/common/src/ProductStrategy.cpp
@@ -321,10 +321,11 @@
void ProductStrategyPreferredRoutingMap::dump(android::String8* dst, int spaces) const {
dst->appendFormat("\n%*sPreferred devices per product strategy dump:", spaces, "");
for (const auto& iter : *this) {
- dst->appendFormat("\n%*sStrategy %u dev:%08x addr:%s",
+ dst->appendFormat("\n%*sStrategy %u %s",
spaces + 2, "",
(uint32_t) iter.first,
- iter.second.mType, iter.second.mAddress.c_str());
+ dumpAudioDeviceTypeAddrVector(iter.second, true /*includeSensitiveInfo*/)
+ .c_str());
}
dst->appendFormat("\n");
}
diff --git a/services/audiopolicy/engine/interface/EngineInterface.h b/services/audiopolicy/engine/interface/EngineInterface.h
index dfb20b5..d45e71c 100644
--- a/services/audiopolicy/engine/interface/EngineInterface.h
+++ b/services/audiopolicy/engine/interface/EngineInterface.h
@@ -293,36 +293,44 @@
virtual status_t listAudioVolumeGroups(AudioVolumeGroupVector &groups) const = 0;
/**
- * @brief setPreferredDeviceForStrategy sets the default device to be used for a
- * strategy when available
+ * @brief setDevicesRoleForStrategy sets devices role for a strategy when available. To remove
+ * devices role, removeDevicesRoleForStrategy must be called. When devices role is set
+ * successfully, previously set devices for the same role and strategy will be removed.
* @param strategy the audio strategy whose routing will be affected
- * @param device the audio device to route to when available
- * @return BAD_VALUE if the strategy is invalid,
- * or NO_ERROR if the preferred device was set
+ * @param role the role of the devices for the strategy. All device roles are defined at
+ * system/media/audio/include/system/audio_policy.h. DEVICE_ROLE_NONE is invalid
+ * for setting.
+ * @param devices the audio devices to be set
+ * @return BAD_VALUE if the strategy or role is invalid,
+ * or NO_ERROR if the role of the devices for strategy was set
*/
- virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device) = 0;
+ virtual status_t setDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role,
+ const AudioDeviceTypeAddrVector &devices) = 0;
/**
- * @brief removePreferredDeviceForStrategy removes the preferred device previously set
+ * @brief removeDevicesRoleForStrategy removes the role of device(s) previously set
* for the given strategy
* @param strategy the audio strategy whose routing will be affected
- * @return BAD_VALUE if the strategy is invalid,
- * or NO_ERROR if the preferred device was removed
+ * @param role the role of the devices for strategy
+ * @return BAD_VALUE if the strategy or role is invalid,
+ * or NO_ERROR if the devices for this role was removed
*/
- virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy) = 0;
+ virtual status_t removeDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role) = 0;
/**
- * @brief getPreferredDeviceForStrategy queries which device is set as the
- * preferred device for the given strategy
+ * @brief getDevicesForRoleAndStrategy queries which devices have the specified role for the
+ * specified strategy
* @param strategy the strategy to query
- * @param device returns configured as the preferred device if one was set
- * @return BAD_VALUE if the strategy is invalid,
- * or NAME_NOT_FOUND if no preferred device was set
- * or NO_ERROR if the device parameter was initialized to the preferred device
+ * @param role the role of the devices to query
+ * @param devices returns list of devices with matching role for the specified strategy.
+ * DEVICE_ROLE_NONE is invalid as input.
+ * @return BAD_VALUE if the strategy or role is invalid,
+ * or NAME_NOT_FOUND if no device for the role and strategy was set
+ * or NO_ERROR if the devices parameter contains a list of devices
*/
- virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device) const = 0;
+ virtual status_t getDevicesForRoleAndStrategy(product_strategy_t strategy, device_role_t role,
+ AudioDeviceTypeAddrVector &devices) const = 0;
virtual void dump(String8 *dst) const = 0;
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index b14d2bb..ec50b14 100755
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -452,22 +452,26 @@
devices = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_TELEPHONY_TX);
break;
+ case STRATEGY_NONE:
+ // Happens when internal strategies are processed ("rerouting", "patch"...)
+ break;
+
default:
- ALOGW("getDevicesForStrategy() unknown strategy: %d", strategy);
+ ALOGW("%s unknown strategy: %d", __func__, strategy);
break;
}
if (devices.isEmpty()) {
- ALOGV("getDevicesForStrategy() no device found for strategy %d", strategy);
+ ALOGV("%s no device found for strategy %d", __func__, strategy);
sp<DeviceDescriptor> defaultOutputDevice = getApmObserver()->getDefaultOutputDevice();
if (defaultOutputDevice != nullptr) {
devices.add(defaultOutputDevice);
}
ALOGE_IF(devices.isEmpty(),
- "getDevicesForStrategy() no default device defined");
+ "%s no default device defined", __func__);
}
- ALOGVV("getDevices ForStrategy() strategy %d, device %s",
+ ALOGVV("%s strategy %d, device %s", __func__,
strategy, dumpDeviceTypes(devices.types()).c_str());
return devices;
}
@@ -631,19 +635,17 @@
// check if this strategy has a preferred device that is available,
// if yes, give priority to it
- AudioDeviceTypeAddr preferredStrategyDevice;
- const status_t status = getPreferredDeviceForStrategy(strategy, preferredStrategyDevice);
+ AudioDeviceTypeAddrVector preferredStrategyDevices;
+ const status_t status = getDevicesForRoleAndStrategy(
+ strategy, DEVICE_ROLE_PREFERRED, preferredStrategyDevices);
if (status == NO_ERROR) {
// there is a preferred device, is it available?
- sp<DeviceDescriptor> preferredAvailableDevDescr = availableOutputDevices.getDevice(
- preferredStrategyDevice.mType,
- String8(preferredStrategyDevice.mAddress.c_str()),
- AUDIO_FORMAT_DEFAULT);
- if (preferredAvailableDevDescr != nullptr) {
- ALOGVV("%s using pref device 0x%08x/%s for strategy %u",
- __func__, preferredStrategyDevice.mType,
- preferredStrategyDevice.mAddress.c_str(), strategy);
- return DeviceVector(preferredAvailableDevDescr);
+ DeviceVector preferredAvailableDevVec =
+ availableOutputDevices.getDevicesFromDeviceTypeAddrVec(preferredStrategyDevices);
+ if (preferredAvailableDevVec.size() == preferredAvailableDevVec.size()) {
+ ALOGVV("%s using pref device %s for strategy %u",
+ __func__, preferredAvailableDevVec.toString().c_str(), strategy);
+ return preferredAvailableDevVec;
}
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f4d3d75..84687fd 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -463,7 +463,16 @@
}
}
}
-
+ auto musicStrategy = streamToStrategy(AUDIO_STREAM_MUSIC);
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ // mute media strategies and delay device switch by the largest
+ // This avoid sending the music tail into the earpiece or headset.
+ setStrategyMute(musicStrategy, true, desc);
+ setStrategyMute(musicStrategy, false, desc, MUTE_TIME_MS,
+ mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA),
+ nullptr, true /*fromCache*/).types());
+ }
// Toggle the device state: UNAVAILABLE -> AVAILABLE
// This will force reading again the device configuration
status = setDeviceConnectionState(device,
@@ -1780,7 +1789,8 @@
checkAndSetVolume(curves, client->volumeSource(),
curves.getVolumeIndex(outputDesc->devices().types()),
outputDesc,
- outputDesc->devices().types());
+ outputDesc->devices().types(), 0 /*delay*/,
+ outputDesc->useHwGain() /*force*/);
// update the outputs if starting an output with a stream that can affect notification
// routing
@@ -3098,16 +3108,16 @@
// Returns true if all devices types match the predicate and are supported by one HW module
bool AudioPolicyManager::areAllDevicesSupported(
- const Vector<AudioDeviceTypeAddr>& devices,
+ const AudioDeviceTypeAddrVector& devices,
std::function<bool(audio_devices_t)> predicate,
const char *context) {
for (size_t i = 0; i < devices.size(); i++) {
sp<DeviceDescriptor> devDesc = mHwModules.getDeviceDescriptor(
- devices[i].mType, devices[i].mAddress.c_str(), String8(),
+ devices[i].mType, devices[i].getAddress(), String8(),
AUDIO_FORMAT_DEFAULT, false /*allowToCreate*/, true /*matchAddress*/);
if (devDesc == nullptr || (predicate != nullptr && !predicate(devices[i].mType))) {
ALOGE("%s: device type %#x address %s not supported or not an output device",
- context, devices[i].mType, devices[i].mAddress.c_str());
+ context, devices[i].mType, devices[i].getAddress());
return false;
}
}
@@ -3115,7 +3125,7 @@
}
status_t AudioPolicyManager::setUidDeviceAffinities(uid_t uid,
- const Vector<AudioDeviceTypeAddr>& devices) {
+ const AudioDeviceTypeAddrVector& devices) {
ALOGV("%s() uid=%d num devices %zu", __FUNCTION__, uid, devices.size());
if (!areAllDevicesSupported(devices, audio_is_output_device, __func__)) {
return BAD_VALUE;
@@ -3147,20 +3157,19 @@
return res;
}
-status_t AudioPolicyManager::setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device) {
- ALOGV("%s() strategy=%d device=%08x addr=%s", __FUNCTION__,
- strategy, device.mType, device.mAddress.c_str());
+status_t AudioPolicyManager::setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role,
+ const AudioDeviceTypeAddrVector &devices) {
+ ALOGV("%s() strategy=%d role=%d %s", __func__, strategy, role,
+ dumpAudioDeviceTypeAddrVector(devices).c_str());
- Vector<AudioDeviceTypeAddr> devices;
- devices.add(device);
if (!areAllDevicesSupported(devices, audio_is_output_device, __func__)) {
return BAD_VALUE;
}
- status_t status = mEngine->setPreferredDeviceForStrategy(strategy, device);
+ status_t status = mEngine->setDevicesRoleForStrategy(strategy, role, devices);
if (status != NO_ERROR) {
- ALOGW("Engine could not set preferred device %08x %s for strategy %d",
- device.mType, device.mAddress.c_str(), strategy);
+ ALOGW("Engine could not set preferred devices %s for strategy %d role %d",
+ dumpAudioDeviceTypeAddrVector(devices).c_str(), strategy, role);
return status;
}
@@ -3176,6 +3185,8 @@
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
waitMs = updateCallRouting(newDevices, delayMs);
+ // Only apply special touch sound delay once
+ delayMs = 0;
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
@@ -3185,6 +3196,8 @@
// preventing the force re-routing in case of default dev that distinguishes on address.
// Let's give back to engine full device choice decision however.
waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
+ // Only apply special touch sound delay once
+ delayMs = 0;
}
if (forceVolumeReeval && !newDevices.isEmpty()) {
applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
@@ -3192,11 +3205,12 @@
}
}
-status_t AudioPolicyManager::removePreferredDeviceForStrategy(product_strategy_t strategy)
+status_t AudioPolicyManager::removeDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role)
{
- ALOGI("%s() strategy=%d", __FUNCTION__, strategy);
+ ALOGI("%s() strategy=%d role=%d", __func__, strategy, role);
- status_t status = mEngine->removePreferredDeviceForStrategy(strategy);
+ status_t status = mEngine->removeDevicesRoleForStrategy(strategy, role);
if (status != NO_ERROR) {
ALOGW("Engine could not remove preferred device for strategy %d", strategy);
return status;
@@ -3208,14 +3222,15 @@
return NO_ERROR;
}
-status_t AudioPolicyManager::getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device) {
- return mEngine->getPreferredDeviceForStrategy(strategy, device);
+status_t AudioPolicyManager::getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role,
+ AudioDeviceTypeAddrVector &devices) {
+ return mEngine->getDevicesForRoleAndStrategy(strategy, role, devices);
}
status_t AudioPolicyManager::setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices) {
- ALOGI("%s() userId=%d num devices %zu", __FUNCTION__, userId, devices.size());\
+ const AudioDeviceTypeAddrVector& devices) {
+ ALOGI("%s() userId=%d num devices %zu", __func__, userId, devices.size());
if (!areAllDevicesSupported(devices, audio_is_output_device, __func__)) {
return BAD_VALUE;
}
@@ -6053,7 +6068,8 @@
if (!Intersection(deviceTypes,
{AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,
AUDIO_DEVICE_OUT_WIRED_HEADSET, AUDIO_DEVICE_OUT_WIRED_HEADPHONE,
- AUDIO_DEVICE_OUT_USB_HEADSET, AUDIO_DEVICE_OUT_HEARING_AID}).empty() &&
+ AUDIO_DEVICE_OUT_USB_HEADSET, AUDIO_DEVICE_OUT_HEARING_AID,
+ AUDIO_DEVICE_OUT_BLE_HEADSET}).empty() &&
((volumeSource == alarmVolumeSrc ||
volumeSource == ringVolumeSrc) ||
(volumeSource == toVolumeSource(AUDIO_STREAM_NOTIFICATION)) ||
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 201abc6..11077f1 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -263,17 +263,23 @@
virtual status_t registerPolicyMixes(const Vector<AudioMix>& mixes);
virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
virtual status_t setUidDeviceAffinities(uid_t uid,
- const Vector<AudioDeviceTypeAddr>& devices);
+ const AudioDeviceTypeAddrVector& devices);
virtual status_t removeUidDeviceAffinities(uid_t uid);
virtual status_t setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices);
+ const AudioDeviceTypeAddrVector& devices);
virtual status_t removeUserIdDeviceAffinities(int userId);
- virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device);
- virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy);
- virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device);
+ virtual status_t setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role,
+ const AudioDeviceTypeAddrVector &devices);
+
+ virtual status_t removeDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role);
+
+
+ virtual status_t getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role,
+ AudioDeviceTypeAddrVector &devices);
virtual status_t startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
@@ -939,7 +945,7 @@
sp<AudioPatch> *patchDescPtr);
bool areAllDevicesSupported(
- const Vector<AudioDeviceTypeAddr>& devices,
+ const AudioDeviceTypeAddrVector& devices,
std::function<bool(audio_devices_t)> predicate,
const char* context);
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 34d07b6..7d1ad63 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -1257,7 +1257,7 @@
}
status_t AudioPolicyService::setUidDeviceAffinities(uid_t uid,
- const Vector<AudioDeviceTypeAddr>& devices) {
+ const AudioDeviceTypeAddrVector& devices) {
Mutex::Autolock _l(mLock);
if(!modifyAudioRoutingAllowed()) {
return PERMISSION_DENIED;
@@ -1282,7 +1282,7 @@
}
status_t AudioPolicyService::setUserIdDeviceAffinities(int userId,
- const Vector<AudioDeviceTypeAddr>& devices) {
+ const AudioDeviceTypeAddrVector& devices) {
Mutex::Autolock _l(mLock);
if(!modifyAudioRoutingAllowed()) {
return PERMISSION_DENIED;
@@ -1494,33 +1494,36 @@
return mAudioPolicyManager->isCallScreenModeSupported();
}
-status_t AudioPolicyService::setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device)
+status_t AudioPolicyService::setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role,
+ const AudioDeviceTypeAddrVector &devices)
{
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
Mutex::Autolock _l(mLock);
- return mAudioPolicyManager->setPreferredDeviceForStrategy(strategy, device);
+ return mAudioPolicyManager->setDevicesRoleForStrategy(strategy, role, devices);
}
-status_t AudioPolicyService::removePreferredDeviceForStrategy(product_strategy_t strategy)
+status_t AudioPolicyService::removeDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role)
{
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
Mutex::Autolock _l(mLock);
- return mAudioPolicyManager->removePreferredDeviceForStrategy(strategy);
+ return mAudioPolicyManager->removeDevicesRoleForStrategy(strategy, role);
}
-status_t AudioPolicyService::getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device)
+status_t AudioPolicyService::getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role,
+ AudioDeviceTypeAddrVector &devices)
{
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
Mutex::Autolock _l(mLock);
- return mAudioPolicyManager->getPreferredDeviceForStrategy(strategy, device);
+ return mAudioPolicyManager->getDevicesForRoleAndStrategy(strategy, role, devices);
}
status_t AudioPolicyService::registerSoundTriggerCaptureStateListener(
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 869a963..a851863 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -226,19 +226,22 @@
virtual status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration);
- virtual status_t setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices);
+ virtual status_t setUidDeviceAffinities(uid_t uid, const AudioDeviceTypeAddrVector& devices);
virtual status_t removeUidDeviceAffinities(uid_t uid);
- virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
- const AudioDeviceTypeAddr &device);
+ virtual status_t setDevicesRoleForStrategy(product_strategy_t strategy,
+ device_role_t role,
+ const AudioDeviceTypeAddrVector &devices);
- virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy);
+ virtual status_t removeDevicesRoleForStrategy(product_strategy_t strategy, device_role_t role);
+ virtual status_t getDevicesForRoleAndStrategy(product_strategy_t strategy,
+ device_role_t role,
+ AudioDeviceTypeAddrVector &devices);
- virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
- AudioDeviceTypeAddr &device);
- virtual status_t setUserIdDeviceAffinities(int userId, const Vector<AudioDeviceTypeAddr>& devices);
+ virtual status_t setUserIdDeviceAffinities(int userId,
+ const AudioDeviceTypeAddrVector& devices);
virtual status_t removeUserIdDeviceAffinities(int userId);
diff --git a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
index a63f402..4fe5adf 100644
--- a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
+++ b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
@@ -510,7 +510,8 @@
sp<camera3::StatusTracker> statusTracker = mStatusTracker.promote();
if (statusTracker != nullptr) {
- mStatusId = statusTracker->addComponent();
+ std::string name = std::string("HeicStream ") + std::to_string(getStreamId());
+ mStatusId = statusTracker->addComponent(name);
}
run("HeicCompositeStreamProc");
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 35c389e..d27f11f 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -270,7 +270,7 @@
}
/** Register in-flight map to the status tracker */
- mInFlightStatusId = mStatusTracker->addComponent();
+ mInFlightStatusId = mStatusTracker->addComponent("InflightRequests");
if (mUseHalBufManager) {
res = mRequestBufferSM.initialize(mStatusTracker);
@@ -1768,6 +1768,7 @@
maxExpectedDuration);
status_t res = waitUntilStateThenRelock(/*active*/ false, maxExpectedDuration);
if (res != OK) {
+ mStatusTracker->dumpActiveComponents();
SET_ERR_L("Error waiting for HAL to drain: %s (%d)", strerror(-res),
res);
}
@@ -1833,10 +1834,12 @@
mStatusWaiters++;
+ bool signalPipelineDrain = false;
if (!active && mUseHalBufManager) {
auto streamIds = mOutputStreams.getStreamIds();
if (mStatus == STATUS_ACTIVE) {
mRequestThread->signalPipelineDrain(streamIds);
+ signalPipelineDrain = true;
}
mRequestBufferSM.onWaitUntilIdle();
}
@@ -1866,6 +1869,10 @@
}
} while (!stateSeen);
+ if (signalPipelineDrain) {
+ mRequestThread->resetPipelineDrain();
+ }
+
mStatusWaiters--;
return res;
@@ -2306,6 +2313,15 @@
newRequest->mRotateAndCropAuto = false;
}
+ auto zoomRatioEntry =
+ newRequest->mSettingsList.begin()->metadata.find(ANDROID_CONTROL_ZOOM_RATIO);
+ if (zoomRatioEntry.count > 0 &&
+ zoomRatioEntry.data.f[0] == 1.0f) {
+ newRequest->mZoomRatioIs1x = true;
+ } else {
+ newRequest->mZoomRatioIs1x = false;
+ }
+
return newRequest;
}
@@ -3779,7 +3795,7 @@
mSessionParamKeys(sessionParamKeys),
mLatestSessionParams(sessionParamKeys.size()),
mUseHalBufManager(useHalBufManager) {
- mStatusId = statusTracker->addComponent();
+ mStatusId = statusTracker->addComponent("RequestThread");
}
Camera3Device::RequestThread::~RequestThread() {}
@@ -4426,13 +4442,17 @@
parent->mDistortionMappers.end()) {
continue;
}
- res = parent->mDistortionMappers[it->cameraId].correctCaptureRequest(
- &(it->metadata));
- if (res != OK) {
- SET_ERR("RequestThread: Unable to correct capture requests "
- "for lens distortion for request %d: %s (%d)",
- halRequest->frame_number, strerror(-res), res);
- return INVALID_OPERATION;
+
+ if (!captureRequest->mDistortionCorrectionUpdated) {
+ res = parent->mDistortionMappers[it->cameraId].correctCaptureRequest(
+ &(it->metadata));
+ if (res != OK) {
+ SET_ERR("RequestThread: Unable to correct capture requests "
+ "for lens distortion for request %d: %s (%d)",
+ halRequest->frame_number, strerror(-res), res);
+ return INVALID_OPERATION;
+ }
+ captureRequest->mDistortionCorrectionUpdated = true;
}
}
@@ -4443,21 +4463,24 @@
continue;
}
- camera_metadata_entry_t e = it->metadata.find(ANDROID_CONTROL_ZOOM_RATIO);
- if (e.count > 0 && e.data.f[0] != 1.0f) {
+ if (!captureRequest->mZoomRatioIs1x) {
cameraIdsWithZoom.insert(it->cameraId);
}
- res = parent->mZoomRatioMappers[it->cameraId].updateCaptureRequest(
- &(it->metadata));
- if (res != OK) {
- SET_ERR("RequestThread: Unable to correct capture requests "
- "for zoom ratio for request %d: %s (%d)",
- halRequest->frame_number, strerror(-res), res);
- return INVALID_OPERATION;
+ if (!captureRequest->mZoomRatioUpdated) {
+ res = parent->mZoomRatioMappers[it->cameraId].updateCaptureRequest(
+ &(it->metadata));
+ if (res != OK) {
+ SET_ERR("RequestThread: Unable to correct capture requests "
+ "for zoom ratio for request %d: %s (%d)",
+ halRequest->frame_number, strerror(-res), res);
+ return INVALID_OPERATION;
+ }
+ captureRequest->mZoomRatioUpdated = true;
}
}
- if (captureRequest->mRotateAndCropAuto) {
+ if (captureRequest->mRotateAndCropAuto &&
+ !captureRequest->mRotationAndCropUpdated) {
for (it = captureRequest->mSettingsList.begin();
it != captureRequest->mSettingsList.end(); it++) {
auto mapper = parent->mRotateAndCropMappers.find(it->cameraId);
@@ -4471,6 +4494,7 @@
}
}
}
+ captureRequest->mRotationAndCropUpdated = true;
}
}
}
@@ -4785,6 +4809,12 @@
mStreamIdsToBeDrained = streamIds;
}
+void Camera3Device::RequestThread::resetPipelineDrain() {
+ Mutex::Autolock pl(mPauseLock);
+ mNotifyPipelineDrain = false;
+ mStreamIdsToBeDrained.clear();
+}
+
void Camera3Device::RequestThread::clearPreviousRequest() {
Mutex::Autolock l(mRequestLock);
mPrevRequest.clear();
@@ -5607,7 +5637,7 @@
std::lock_guard<std::mutex> lock(mLock);
mStatusTracker = statusTracker;
- mRequestBufferStatusId = statusTracker->addComponent();
+ mRequestBufferStatusId = statusTracker->addComponent("BufferRequestSM");
return OK;
}
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 8e2c2f9..c579071 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -517,6 +517,19 @@
// overriding of ROTATE_AND_CROP value and adjustment of coordinates
// in several other controls in both the request and the result
bool mRotateAndCropAuto;
+ // Whether this capture request has its zoom ratio set to 1.0x before
+ // the framework overrides it for camera HAL consumption.
+ bool mZoomRatioIs1x;
+
+
+ // Whether this capture request's distortion correction update has
+ // been done.
+ bool mDistortionCorrectionUpdated = false;
+ // Whether this capture request's rotation and crop update has been
+ // done.
+ bool mRotationAndCropUpdated = false;
+ // Whether this capture request's zoom ratio update has been done.
+ bool mZoomRatioUpdated = false;
};
typedef List<sp<CaptureRequest> > RequestList;
@@ -832,6 +845,7 @@
}
void signalPipelineDrain(const std::vector<int>& streamIds);
+ void resetPipelineDrain();
status_t switchToOffline(
const std::vector<int32_t>& streamsToKeep,
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index 20f6168..f208561 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -330,7 +330,8 @@
// Register for idle tracking
sp<StatusTracker> statusTracker = mStatusTracker.promote();
if (statusTracker != 0 && mStatusId == StatusTracker::NO_STATUS_ID) {
- mStatusId = statusTracker->addComponent();
+ std::string name = std::string("Stream ") + std::to_string(mId);
+ mStatusId = statusTracker->addComponent(name.c_str());
}
// Check if the stream configuration is unchanged, and skip reallocation if
diff --git a/services/camera/libcameraservice/device3/StatusTracker.cpp b/services/camera/libcameraservice/device3/StatusTracker.cpp
index 723b5c2..ea1f2c1 100644
--- a/services/camera/libcameraservice/device3/StatusTracker.cpp
+++ b/services/camera/libcameraservice/device3/StatusTracker.cpp
@@ -40,7 +40,7 @@
StatusTracker::~StatusTracker() {
}
-int StatusTracker::addComponent() {
+int StatusTracker::addComponent(std::string componentName) {
int id;
ssize_t err;
{
@@ -49,8 +49,12 @@
ALOGV("%s: Adding new component %d", __FUNCTION__, id);
err = mStates.add(id, IDLE);
- ALOGE_IF(err < 0, "%s: Can't add new component %d: %s (%zd)",
- __FUNCTION__, id, strerror(-err), err);
+ if (componentName.empty()) {
+ componentName = std::to_string(id);
+ }
+ mComponentNames.add(id, componentName);
+ ALOGE_IF(err < 0, "%s: Can't add new component %d (%s): %s (%zd)",
+ __FUNCTION__, id, componentName.c_str(), strerror(-err), err);
}
if (err >= 0) {
@@ -68,6 +72,7 @@
Mutex::Autolock l(mLock);
ALOGV("%s: Removing component %d", __FUNCTION__, id);
idx = mStates.removeItem(id);
+ mComponentNames.removeItem(id);
}
if (idx >= 0) {
@@ -80,6 +85,20 @@
}
+void StatusTracker::dumpActiveComponents() {
+ Mutex::Autolock l(mLock);
+ if (mDeviceState == IDLE) {
+ ALOGI("%s: all components are IDLE", __FUNCTION__);
+ return;
+ }
+ for (size_t i = 0; i < mStates.size(); i++) {
+ if (mStates.valueAt(i) == ACTIVE) {
+ ALOGI("%s: component %d (%s) is active", __FUNCTION__, mStates.keyAt(i),
+ mComponentNames.valueAt(i).c_str());
+ }
+ }
+}
+
void StatusTracker::markComponentIdle(int id, const sp<Fence>& componentFence) {
markComponent(id, IDLE, componentFence);
}
diff --git a/services/camera/libcameraservice/device3/StatusTracker.h b/services/camera/libcameraservice/device3/StatusTracker.h
index 3a1d85c..3741cce 100644
--- a/services/camera/libcameraservice/device3/StatusTracker.h
+++ b/services/camera/libcameraservice/device3/StatusTracker.h
@@ -17,6 +17,7 @@
#ifndef ANDROID_SERVERS_CAMERA3_STATUSTRACKER_H
#define ANDROID_SERVERS_CAMERA3_STATUSTRACKER_H
+#include <string>
#include <utils/Condition.h>
#include <utils/Errors.h>
#include <utils/List.h>
@@ -54,7 +55,7 @@
// Add a component to track; returns non-negative unique ID for the new
// component on success, negative error code on failure.
// New components start in the idle state.
- int addComponent();
+ int addComponent(std::string componentName);
// Remove existing component from idle tracking. Ignores unknown IDs
void removeComponent(int id);
@@ -68,6 +69,8 @@
// Set the state of a tracked component to be active. Ignores unknown IDs.
void markComponentActive(int id);
+ void dumpActiveComponents();
+
virtual void requestExit();
protected:
@@ -105,6 +108,7 @@
// Current component states
KeyedVector<int, ComponentState> mStates;
+ KeyedVector<int, std::string> mComponentNames;
// Merged fence for all processed state changes
sp<Fence> mIdleFence;
// Current overall device state
diff --git a/services/mediametrics/AudioTypes.cpp b/services/mediametrics/AudioTypes.cpp
index aa44447..5d044bb 100644
--- a/services/mediametrics/AudioTypes.cpp
+++ b/services/mediametrics/AudioTypes.cpp
@@ -76,6 +76,7 @@
{"AUDIO_DEVICE_IN_ECHO_REFERENCE", 1LL << 27},
{"AUDIO_DEVICE_IN_DEFAULT", 1LL << 28},
// R values above.
+ {"AUDIO_DEVICE_IN_BLE_HEADSET", 1LL << 29},
};
return map;
}
@@ -121,6 +122,8 @@
{"AUDIO_DEVICE_OUT_ECHO_CANCELLER", 1LL << 29},
{"AUDIO_DEVICE_OUT_DEFAULT", 1LL << 30},
// R values above.
+ {"AUDIO_DEVICE_OUT_BLE_HEADSET", 1LL << 31},
+ {"AUDIO_DEVICE_OUT_BLE_SPAEKER", 1LL << 32},
};
return map;
}
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index a171cb0..04b906a 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -47,7 +47,11 @@
virtual aaudio_result_t open(const aaudio::AAudioStreamRequest &request) = 0;
- virtual aaudio_result_t close() = 0;
+ /*
+ * Perform any cleanup necessary before deleting the stream.
+ * This might include releasing and closing internal streams.
+ */
+ virtual void close() = 0;
aaudio_result_t registerStream(android::sp<AAudioServiceStreamBase> stream);
diff --git a/services/oboeservice/AAudioServiceEndpointCapture.cpp b/services/oboeservice/AAudioServiceEndpointCapture.cpp
index de36d50..206a264 100644
--- a/services/oboeservice/AAudioServiceEndpointCapture.cpp
+++ b/services/oboeservice/AAudioServiceEndpointCapture.cpp
@@ -36,8 +36,8 @@
using namespace aaudio; // TODO just import names needed
AAudioServiceEndpointCapture::AAudioServiceEndpointCapture(AAudioService &audioService)
- : mStreamInternalCapture(audioService, true) {
- mStreamInternal = &mStreamInternalCapture;
+ : AAudioServiceEndpointShared(
+ (AudioStreamInternal *)(new AudioStreamInternalCapture(audioService, true))) {
}
aaudio_result_t AAudioServiceEndpointCapture::open(const aaudio::AAudioStreamRequest &request) {
diff --git a/services/oboeservice/AAudioServiceEndpointCapture.h b/services/oboeservice/AAudioServiceEndpointCapture.h
index ae5a189..2ca43cf 100644
--- a/services/oboeservice/AAudioServiceEndpointCapture.h
+++ b/services/oboeservice/AAudioServiceEndpointCapture.h
@@ -37,7 +37,6 @@
void *callbackLoop() override;
private:
- AudioStreamInternalCapture mStreamInternalCapture;
std::unique_ptr<uint8_t[]> mDistributionBuffer;
};
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 0843e0b..04c6453 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -226,7 +226,7 @@
return result;
}
-aaudio_result_t AAudioServiceEndpointMMAP::close() {
+void AAudioServiceEndpointMMAP::close() {
if (mMmapStream != nullptr) {
// Needs to be explicitly cleared or CTS will fail but it is not clear why.
mMmapStream.clear();
@@ -235,8 +235,6 @@
// FIXME Make closing synchronous.
AudioClock::sleepForNanos(100 * AAUDIO_NANOS_PER_MILLISECOND);
}
-
- return AAUDIO_OK;
}
aaudio_result_t AAudioServiceEndpointMMAP::startStream(sp<AAudioServiceStreamBase> stream,
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.h b/services/oboeservice/AAudioServiceEndpointMMAP.h
index 3d10861..b6003b6 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.h
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.h
@@ -50,7 +50,7 @@
aaudio_result_t open(const aaudio::AAudioStreamRequest &request) override;
- aaudio_result_t close() override;
+ void close() override;
aaudio_result_t startStream(android::sp<AAudioServiceStreamBase> stream,
audio_port_handle_t *clientHandle) override;
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.cpp b/services/oboeservice/AAudioServiceEndpointPlay.cpp
index 1603e41..730d161 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.cpp
+++ b/services/oboeservice/AAudioServiceEndpointPlay.cpp
@@ -42,8 +42,8 @@
#define BURSTS_PER_BUFFER_DEFAULT 2
AAudioServiceEndpointPlay::AAudioServiceEndpointPlay(AAudioService &audioService)
- : mStreamInternalPlay(audioService, true) {
- mStreamInternal = &mStreamInternalPlay;
+ : AAudioServiceEndpointShared(
+ (AudioStreamInternal *)(new AudioStreamInternalPlay(audioService, true))) {
}
aaudio_result_t AAudioServiceEndpointPlay::open(const aaudio::AAudioStreamRequest &request) {
diff --git a/services/oboeservice/AAudioServiceEndpointPlay.h b/services/oboeservice/AAudioServiceEndpointPlay.h
index 981e430..160a1de 100644
--- a/services/oboeservice/AAudioServiceEndpointPlay.h
+++ b/services/oboeservice/AAudioServiceEndpointPlay.h
@@ -45,7 +45,6 @@
void *callbackLoop() override;
private:
- AudioStreamInternalPlay mStreamInternalPlay; // for playing output of mixer
bool mLatencyTuningEnabled = false; // TODO implement tuning
AAudioMixer mMixer; //
};
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index dc21886..f5de59f 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -40,6 +40,9 @@
// This is the maximum size in frames. The effective size can be tuned smaller at runtime.
#define DEFAULT_BUFFER_CAPACITY (48 * 8)
+AAudioServiceEndpointShared::AAudioServiceEndpointShared(AudioStreamInternal *streamInternal)
+ : mStreamInternal(streamInternal) {}
+
std::string AAudioServiceEndpointShared::dump() const {
std::stringstream result;
@@ -84,8 +87,8 @@
return result;
}
-aaudio_result_t AAudioServiceEndpointShared::close() {
- return getStreamInternal()->releaseCloseFinal();
+void AAudioServiceEndpointShared::close() {
+ getStreamInternal()->releaseCloseFinal();
}
// Glue between C and C++ callbacks.
diff --git a/services/oboeservice/AAudioServiceEndpointShared.h b/services/oboeservice/AAudioServiceEndpointShared.h
index bfc1744..020b926 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.h
+++ b/services/oboeservice/AAudioServiceEndpointShared.h
@@ -35,12 +35,13 @@
class AAudioServiceEndpointShared : public AAudioServiceEndpoint {
public:
+ explicit AAudioServiceEndpointShared(AudioStreamInternal *streamInternal);
std::string dump() const override;
aaudio_result_t open(const aaudio::AAudioStreamRequest &request) override;
- aaudio_result_t close() override;
+ void close() override;
aaudio_result_t startStream(android::sp<AAudioServiceStreamBase> stream,
audio_port_handle_t *clientHandle) override;
@@ -57,15 +58,15 @@
protected:
AudioStreamInternal *getStreamInternal() const {
- return mStreamInternal;
+ return mStreamInternal.get();
};
aaudio_result_t startSharingThread_l();
aaudio_result_t stopSharingThread();
- // pointer to object statically allocated in subclasses
- AudioStreamInternal *mStreamInternal = nullptr;
+ // An MMAP stream that is shared by multiple clients.
+ android::sp<AudioStreamInternal> mStreamInternal;
std::atomic<bool> mCallbackEnabled{false};
diff --git a/services/oboeservice/SharedRingBuffer.cpp b/services/oboeservice/SharedRingBuffer.cpp
index 0a9196a..c1d4e16 100644
--- a/services/oboeservice/SharedRingBuffer.cpp
+++ b/services/oboeservice/SharedRingBuffer.cpp
@@ -60,16 +60,18 @@
return AAUDIO_ERROR_INTERNAL; // TODO convert errno to a better AAUDIO_ERROR;
}
- // Map the fd to memory addresses.
- mSharedMemory = (uint8_t *) mmap(0, mSharedMemorySizeInBytes,
+ // Map the fd to memory addresses. Use a temporary pointer to keep the mmap result and update
+ // it to `mSharedMemory` only when mmap operate successfully.
+ uint8_t* tmpPtr = (uint8_t *) mmap(0, mSharedMemorySizeInBytes,
PROT_READ|PROT_WRITE,
MAP_SHARED,
mFileDescriptor.get(), 0);
- if (mSharedMemory == MAP_FAILED) {
+ if (tmpPtr == MAP_FAILED) {
ALOGE("allocate() mmap() failed %d", errno);
mFileDescriptor.reset();
return AAUDIO_ERROR_INTERNAL; // TODO convert errno to a better AAUDIO_ERROR;
}
+ mSharedMemory = tmpPtr;
// Get addresses for our counters and data from the shared memory.
fifo_counter_t *readCounterAddress =