Merge "audio policy: make input device selection more consistent" into pi-dev
diff --git a/include/media/MmapStreamCallback.h b/include/media/MmapStreamCallback.h
index 8098e79..31b8eb5 100644
--- a/include/media/MmapStreamCallback.h
+++ b/include/media/MmapStreamCallback.h
@@ -31,8 +31,9 @@
* The mmap stream should be torn down because conditions that permitted its creation with
* the requested parameters have changed and do not allow it to operate with the requested
* constraints any more.
+ * \param[in] handle handle for the client stream to tear down.
*/
- virtual void onTearDown() = 0;
+ virtual void onTearDown(audio_port_handle_t handle) = 0;
/**
* The volume to be applied to the use case specified when opening the stream has changed
diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
index e5ad2d9..c1ff34b 100644
--- a/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor.cpp
@@ -26,23 +26,22 @@
#include "AAudioExampleUtils.h"
#include "AAudioSimpleRecorder.h"
-// TODO support FLOAT
-#define REQUIRED_FORMAT AAUDIO_FORMAT_PCM_I16
#define MIN_FRAMES_TO_READ 48 /* arbitrary, 1 msec at 48000 Hz */
static const int FRAMES_PER_LINE = 20000;
int main(int argc, const char **argv)
{
- AAudioArgsParser argParser;
- aaudio_result_t result;
- AAudioSimpleRecorder recorder;
- int actualSamplesPerFrame;
- int actualSampleRate;
- aaudio_format_t actualDataFormat;
+ AAudioArgsParser argParser;
+ AAudioSimpleRecorder recorder;
+ AAudioStream *aaudioStream = nullptr;
- AAudioStream *aaudioStream = nullptr;
+ aaudio_result_t result;
+ aaudio_format_t actualDataFormat;
aaudio_stream_state_t state;
+
+ int32_t actualSamplesPerFrame;
+ int32_t actualSampleRate;
int32_t framesPerBurst = 0;
int32_t framesPerRead = 0;
int32_t framesToRecord = 0;
@@ -50,18 +49,18 @@
int32_t nextFrameCount = 0;
int32_t frameCount = 0;
int32_t xRunCount = 0;
- int64_t previousFramePosition = -1;
- int16_t *data = nullptr;
- float peakLevel = 0.0;
int32_t deviceId;
+ int16_t *shortData = nullptr;
+ float *floatData = nullptr;
+ float peakLevel = 0.0;
+
// Make printf print immediately so that debug info is not stuck
// in a buffer if we hang or crash.
setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
- printf("%s - Monitor input level using AAudio read, V0.1.2\n", argv[0]);
+ printf("%s - Monitor input level using AAudio read, V0.1.3\n", argv[0]);
- argParser.setFormat(REQUIRED_FORMAT);
if (argParser.parseArgs(argc, argv)) {
return EXIT_FAILURE;
}
@@ -69,6 +68,7 @@
result = recorder.open(argParser);
if (result != AAUDIO_OK) {
fprintf(stderr, "ERROR - recorder.open() returned %d\n", result);
+ printf("IMPORTANT - Did you remember to enter: adb root\n");
goto finish;
}
aaudioStream = recorder.getStream();
@@ -96,17 +96,18 @@
printf("DataFormat: framesPerRead = %d\n",framesPerRead);
actualDataFormat = AAudioStream_getFormat(aaudioStream);
- printf("DataFormat: requested = %d, actual = %d\n",
- REQUIRED_FORMAT, actualDataFormat);
- // TODO handle other data formats
- assert(actualDataFormat == REQUIRED_FORMAT);
// Allocate a buffer for the PCM_16 audio data.
- data = new(std::nothrow) int16_t[framesPerRead * actualSamplesPerFrame];
- if (data == nullptr) {
- fprintf(stderr, "ERROR - could not allocate data buffer\n");
- result = AAUDIO_ERROR_NO_MEMORY;
- goto finish;
+ switch (actualDataFormat) {
+ case AAUDIO_FORMAT_PCM_I16:
+ shortData = new int16_t[framesPerRead * actualSamplesPerFrame];
+ break;
+ case AAUDIO_FORMAT_PCM_FLOAT:
+ floatData = new float[framesPerRead * actualSamplesPerFrame];
+ break;
+ default:
+ fprintf(stderr, "UNEXPECTED FORMAT! %d", actualDataFormat);
+ goto finish;
}
// Start the stream.
@@ -126,7 +127,12 @@
// Read audio data from the stream.
const int64_t timeoutNanos = 1000 * NANOS_PER_MILLISECOND;
int minFrames = (framesToRecord < framesPerRead) ? framesToRecord : framesPerRead;
- int actual = AAudioStream_read(aaudioStream, data, minFrames, timeoutNanos);
+ int actual = 0;
+ if (actualDataFormat == AAUDIO_FORMAT_PCM_I16) {
+ actual = AAudioStream_read(aaudioStream, shortData, minFrames, timeoutNanos);
+ } else if (actualDataFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+ actual = AAudioStream_read(aaudioStream, floatData, minFrames, timeoutNanos);
+ }
if (actual < 0) {
fprintf(stderr, "ERROR - AAudioStream_read() returned %d\n", actual);
result = actual;
@@ -140,7 +146,12 @@
// Peak finder.
for (int frameIndex = 0; frameIndex < actual; frameIndex++) {
- float sample = data[frameIndex * actualSamplesPerFrame] * (1.0/32768);
+ float sample = 0.0f;
+ if (actualDataFormat == AAUDIO_FORMAT_PCM_I16) {
+ sample = shortData[frameIndex * actualSamplesPerFrame] * (1.0/32768);
+ } else if (actualDataFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+ sample = floatData[frameIndex * actualSamplesPerFrame];
+ }
if (sample > peakLevel) {
peakLevel = sample;
}
@@ -151,17 +162,15 @@
displayPeakLevel(peakLevel);
peakLevel = 0.0;
nextFrameCount += FRAMES_PER_LINE;
- }
- // Print timestamps.
- int64_t framePosition = 0;
- int64_t frameTime = 0;
- aaudio_result_t timeResult;
- timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
- &framePosition, &frameTime);
+ // Print timestamps.
+ int64_t framePosition = 0;
+ int64_t frameTime = 0;
+ aaudio_result_t timeResult;
+ timeResult = AAudioStream_getTimestamp(aaudioStream, CLOCK_MONOTONIC,
+ &framePosition, &frameTime);
- if (timeResult == AAUDIO_OK) {
- if (framePosition > (previousFramePosition + FRAMES_PER_LINE)) {
+ if (timeResult == AAUDIO_OK) {
int64_t realTime = getNanoseconds();
int64_t framesRead = AAudioStream_getFramesRead(aaudioStream);
@@ -175,11 +184,15 @@
(long long) framePosition,
(long long) frameTime,
latencyMillis);
- previousFramePosition = framePosition;
+ } else {
+ printf("WARNING - AAudioStream_getTimestamp() returned %d\n", timeResult);
}
}
}
+ state = AAudioStream_getState(aaudioStream);
+ printf("after loop, state = %s\n", AAudio_convertStreamStateToText(state));
+
xRunCount = AAudioStream_getXRunCount(aaudioStream);
printf("AAudioStream_getXRunCount %d\n", xRunCount);
@@ -192,7 +205,8 @@
finish:
recorder.close();
- delete[] data;
+ delete[] shortData;
+ delete[] floatData;
printf("exiting - AAudio result = %d = %s\n", result, AAudio_convertResultToText(result));
return (result != AAUDIO_OK) ? EXIT_FAILURE : EXIT_SUCCESS;
}
diff --git a/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
index 893795b..d10f812 100644
--- a/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
+++ b/media/libaaudio/examples/input_monitor/src/input_monitor_callback.cpp
@@ -26,29 +26,39 @@
#include "AAudioExampleUtils.h"
#include "AAudioSimpleRecorder.h"
-#define NUM_SECONDS 5
-
-int main(int argc, char **argv)
+int main(int argc, const char **argv)
{
- (void)argc; // unused
- AAudioSimpleRecorder recorder;
- PeakTrackerData_t myData = {0.0};
- aaudio_result_t result;
+ AAudioArgsParser argParser;
+ AAudioSimpleRecorder recorder;
+ PeakTrackerData_t myData = {0.0};
+ AAudioStream *aaudioStream = nullptr;
+ aaudio_result_t result;
aaudio_stream_state_t state;
+
+ int loopsNeeded = 0;
const int displayRateHz = 20; // arbitrary
- const int loopsNeeded = NUM_SECONDS * displayRateHz;
// Make printf print immediately so that debug info is not stuck
// in a buffer if we hang or crash.
setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
- printf("%s - Display audio input using an AAudio callback, V0.1.2\n", argv[0]);
+ printf("%s - Display audio input using an AAudio callback, V0.1.3\n", argv[0]);
- result = recorder.open(2, 48000, AAUDIO_FORMAT_PCM_I16,
- SimpleRecorderDataCallbackProc, SimpleRecorderErrorCallbackProc, &myData);
+ if (argParser.parseArgs(argc, argv)) {
+ return EXIT_FAILURE;
+ }
+
+ result = recorder.open(argParser,
+ SimpleRecorderDataCallbackProc,
+ SimpleRecorderErrorCallbackProc,
+ &myData);
if (result != AAUDIO_OK) {
fprintf(stderr, "ERROR - recorder.open() returned %d\n", result);
+ printf("IMPORTANT - Did you remember to enter: adb root\n");
goto error;
}
+ aaudioStream = recorder.getStream();
+ argParser.compareWithStream(aaudioStream);
+
printf("recorder.getFramesPerSecond() = %d\n", recorder.getFramesPerSecond());
printf("recorder.getSamplesPerFrame() = %d\n", recorder.getSamplesPerFrame());
@@ -58,7 +68,9 @@
goto error;
}
- printf("Sleep for %d seconds while audio record in a callback thread.\n", NUM_SECONDS);
+ printf("Sleep for %d seconds while audio record in a callback thread.\n",
+ argParser.getDurationSeconds());
+ loopsNeeded = argParser.getDurationSeconds() * displayRateHz;
for (int i = 0; i < loopsNeeded; i++)
{
const struct timespec request = { .tv_sec = 0,
@@ -67,7 +79,7 @@
printf("%08d: ", (int)recorder.getFramesRead());
displayPeakLevel(myData.peakLevel);
- result = AAudioStream_waitForStateChange(recorder.getStream(),
+ result = AAudioStream_waitForStateChange(aaudioStream,
AAUDIO_STREAM_STATE_CLOSED,
&state,
0);
@@ -93,7 +105,8 @@
goto error;
}
- printf("Sleep for %d seconds while audio records in a callback thread.\n", NUM_SECONDS);
+ printf("Sleep for %d seconds while audio records in a callback thread.\n",
+ argParser.getDurationSeconds());
for (int i = 0; i < loopsNeeded; i++)
{
const struct timespec request = { .tv_sec = 0,
@@ -102,13 +115,14 @@
printf("%08d: ", (int)recorder.getFramesRead());
displayPeakLevel(myData.peakLevel);
- state = AAudioStream_getState(recorder.getStream());
+ state = AAudioStream_getState(aaudioStream);
if (state != AAUDIO_STREAM_STATE_STARTING && state != AAUDIO_STREAM_STATE_STARTED) {
printf("Stream state is %d %s!\n", state, AAudio_convertStreamStateToText(state));
break;
}
}
printf("Woke up now.\n");
+ argParser.compareWithStream(aaudioStream);
result = recorder.stop();
if (result != AAUDIO_OK) {
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 39d079e..026ff0f 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -151,8 +151,7 @@
static void MyErrorCallbackProc(
AAudioStream *stream __unused,
void *userData __unused,
- aaudio_result_t error)
-{
+ aaudio_result_t error) {
printf("Error Callback, error: %d\n",(int)error);
LoopbackData *myData = (LoopbackData *) userData;
myData->outputError = error;
diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index eb6925a..88d7401 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h
@@ -87,7 +87,6 @@
return;
}
-// TODO use this as a base class within AAudio
class AAudioParameters {
public:
@@ -262,6 +261,9 @@
case 'd':
setDeviceId(atoi(&arg[2]));
break;
+ case 'f':
+ setFormat(atoi(&arg[2]));
+ break;
case 'i':
setInputPreset(atoi(&arg[2]));
break;
@@ -326,6 +328,10 @@
printf(" -b{bufferCapacity} frames\n");
printf(" -c{channels} for example 2 for stereo\n");
printf(" -d{deviceId} default is %d\n", AAUDIO_UNSPECIFIED);
+ printf(" -f{0|1|2} set format\n");
+ printf(" 0 = UNSPECIFIED\n");
+ printf(" 1 = PCM_I16\n");
+ printf(" 2 = FLOAT\n");
printf(" -i{inputPreset} eg. 5 for AAUDIO_INPUT_PRESET_CAMCORDER\n");
printf(" -m{0|1|2|3} set MMAP policy\n");
printf(" 0 = _UNSPECIFIED, use aaudio.mmap_policy system property, default\n");
diff --git a/media/libaaudio/examples/write_sine/src/write_sine.cpp b/media/libaaudio/examples/write_sine/src/write_sine.cpp
index 38e1e4c..8e33a31 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine.cpp
@@ -57,7 +57,7 @@
// in a buffer if we hang or crash.
setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
- printf("%s - Play a sine wave using AAudio V0.1.2\n", argv[0]);
+ printf("%s - Play a sine wave using AAudio V0.1.3\n", argv[0]);
if (argParser.parseArgs(argc, argv)) {
return EXIT_FAILURE;
diff --git a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
index e167773..e33e9f8 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
@@ -204,7 +204,7 @@
AAudioArgsParser::usage();
printf(" -l{count} loopCount start/stop, every other one is silent\n");
printf(" -t{msec} play a high pitched tone at the beginning\n");
- printf(" -f force periodic underruns by sleeping in callback\n");
+ printf(" -z force periodic underruns by sleeping in callback\n");
}
int main(int argc, const char **argv)
@@ -219,7 +219,7 @@
// in a buffer if we hang or crash.
setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
- printf("%s - Play a sine sweep using an AAudio callback V0.1.3\n", argv[0]);
+ printf("%s - Play a sine sweep using an AAudio callback V0.1.4\n", argv[0]);
for (int i = 1; i < argc; i++) {
const char *arg = argv[i];
@@ -234,8 +234,8 @@
case 't':
prefixToneMsec = atoi(&arg[2]);
break;
- case 'f':
- forceUnderruns = true;
+ case 'z':
+ forceUnderruns = true; // Zzzzzzz
break;
default:
usage();
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index e40a6cd..2207cb8c 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -146,6 +146,8 @@
* to make more refined volume or routing decisions.
*
* Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ *
+ * Added in API level 28.
*/
enum {
/**
@@ -220,6 +222,8 @@
* enforce audio focus.
*
* Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ *
+ * Added in API level 28.
*/
enum {
@@ -252,6 +256,8 @@
* configuration.
*
* Note that these match the equivalent values in MediaRecorder.AudioSource in the Android Java API.
+ *
+ * Added in API level 28.
*/
enum {
/**
@@ -288,6 +294,8 @@
* Do not allocate a session ID.
* Effects cannot be used with this stream.
* Default.
+ *
+ * Added in API level 28.
*/
AAUDIO_SESSION_ID_NONE = -1,
@@ -297,6 +305,8 @@
* Note that the use of this flag may result in higher latency.
*
* Note that this matches the value of AudioManager.AUDIO_SESSION_ID_GENERATE.
+ *
+ * Added in API level 28.
*/
AAUDIO_SESSION_ID_ALLOCATE = 0,
};
@@ -481,6 +491,8 @@
*
* The default, if you do not call this function, is AAUDIO_USAGE_MEDIA.
*
+ * Added in API level 28.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param usage the desired usage, eg. AAUDIO_USAGE_GAME
*/
@@ -496,6 +508,8 @@
*
* The default, if you do not call this function, is AAUDIO_CONTENT_TYPE_MUSIC.
*
+ * Added in API level 28.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param contentType the type of audio data, eg. AAUDIO_CONTENT_TYPE_SPEECH
*/
@@ -514,6 +528,8 @@
* That is because VOICE_RECOGNITION is the preset with the lowest latency
* on many platforms.
*
+ * Added in API level 28.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param inputPreset the desired configuration for recording
*/
@@ -540,6 +556,8 @@
*
* Allocated session IDs will always be positive and nonzero.
*
+ * Added in API level 28.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sessionId an allocated sessionID or AAUDIO_SESSION_ID_ALLOCATE
*/
@@ -1059,6 +1077,8 @@
*
* The sessionID for a stream should not change once the stream has been opened.
*
+ * Added in API level 28.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return session ID or AAUDIO_SESSION_ID_NONE
*/
@@ -1094,6 +1114,8 @@
/**
* Return the use case for the stream.
*
+ * Added in API level 28.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames read
*/
@@ -1102,6 +1124,8 @@
/**
* Return the content type for the stream.
*
+ * Added in API level 28.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return content type, for example AAUDIO_CONTENT_TYPE_MUSIC
*/
@@ -1110,6 +1134,8 @@
/**
* Return the input preset for the stream.
*
+ * Added in API level 28.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return input preset, for example AAUDIO_INPUT_PRESET_CAMCORDER
*/
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index 3352b33..8bbb9d9 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -179,19 +179,17 @@
int64_t localPosition;
status_t status = extendedTimestamp->getBestTimestamp(&localPosition, timeNanoseconds,
timebase, &location);
- // use MonotonicCounter to prevent retrograde motion.
- mTimestampPosition.update32((int32_t)localPosition);
- *framePosition = mTimestampPosition.get();
+ if (status == OK) {
+ // use MonotonicCounter to prevent retrograde motion.
+ mTimestampPosition.update32((int32_t) localPosition);
+ *framePosition = mTimestampPosition.get();
+ }
// ALOGD("getBestTimestamp() fposition: server = %6lld, kernel = %6lld, location = %d",
// (long long) extendedTimestamp->mPosition[ExtendedTimestamp::Location::LOCATION_SERVER],
// (long long) extendedTimestamp->mPosition[ExtendedTimestamp::Location::LOCATION_KERNEL],
// (int)location);
- if (status == WOULD_BLOCK) {
- return AAUDIO_ERROR_INVALID_STATE;
- } else {
- return AAudioConvert_androidToAAudioResult(status);
- }
+ return AAudioConvert_androidToAAudioResult(status);
}
void AudioStreamLegacy::onAudioDeviceUpdate(audio_port_handle_t deviceId)
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 14ffb1d..0a1bdfe 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1289,7 +1289,8 @@
ALOGV("Tear down audio with reason %d.", reason);
if (reason == Renderer::kDueToTimeout && !(mPaused && mOffloadAudio)) {
// TimeoutWhenPaused is only for offload mode.
- ALOGW("Receive a stale message for teardown.");
+ ALOGW("Received a stale message for teardown, mPaused(%d), mOffloadAudio(%d)",
+ mPaused, mOffloadAudio);
break;
}
int64_t positionUs;
@@ -1789,6 +1790,8 @@
void NuPlayer::restartAudio(
int64_t currentPositionUs, bool forceNonOffload, bool needsToCreateAudioDecoder) {
+ ALOGD("restartAudio timeUs(%lld), dontOffload(%d), createDecoder(%d)",
+ (long long)currentPositionUs, forceNonOffload, needsToCreateAudioDecoder);
if (mAudioDecoder != NULL) {
mAudioDecoder->pause();
mAudioDecoder.clear();
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 63c887b..3e5bdd6 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -669,6 +669,11 @@
notifyListener_l(MEDIA_STOPPED);
}
+ if (property_get_bool("persist.debug.sf.stats", false)) {
+ Vector<String16> args;
+ dump(-1, args);
+ }
+
mState = STATE_RESET_IN_PROGRESS;
mPlayer->resetAsync();
diff --git a/media/libstagefright/VideoFrameScheduler.cpp b/media/libstagefright/VideoFrameScheduler.cpp
index 03226c7..6819bba 100644
--- a/media/libstagefright/VideoFrameScheduler.cpp
+++ b/media/libstagefright/VideoFrameScheduler.cpp
@@ -129,6 +129,11 @@
numSamplesToUse = mNumSamples;
}
+ if ((period >> kPrecision) == 0 ) {
+ ALOGW("Period is 0, or after including precision is 0 - would cause div0, returning");
+ return false;
+ }
+
int64_t sumX = 0;
int64_t sumXX = 0;
int64_t sumXY = 0;
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index 28524b0..fb56694 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -108,6 +108,7 @@
AMediaCodec_queueInputBuffer;
AMediaCodec_queueSecureInputBuffer;
AMediaCodec_releaseCrypto; # introduced=28
+ AMediaCodec_releaseName; # introduced=28
AMediaCodec_releaseOutputBuffer;
AMediaCodec_releaseOutputBufferAtTime;
AMediaCodec_setAsyncNotifyCallback; # introduced=28
diff --git a/packages/MediaComponents/Android.mk b/packages/MediaComponents/Android.mk
index b0d8e7d..def9dc5 100644
--- a/packages/MediaComponents/Android.mk
+++ b/packages/MediaComponents/Android.mk
@@ -14,59 +14,59 @@
# limitations under the License.
#
-LOCAL_PATH := $(call my-dir)
-
-ifneq ($(TARGET_BUILD_PDK),true)
-# Build MediaComponents only if this is not a PDK build. MediaComponents won't
-# build in PDK builds because frameworks/base/core/java is not available but
-# IMediaSession2.aidl and IMediaController2.aidl are using classes from
-# frameworks/base/core/java.
-
-include $(CLEAR_VARS)
-
-LOCAL_PACKAGE_NAME := MediaComponents
-LOCAL_MODULE_OWNER := google
-
-# TODO: create a separate key for this package.
-LOCAL_CERTIFICATE := platform
-
-# TODO: Use System SDK once public APIs are approved
-# LOCAL_SDK_VERSION := system_current
-LOCAL_PRIVATE_PLATFORM_APIS := true
-
-LOCAL_SRC_FILES := \
- $(call all-java-files-under, src) \
- $(call all-Iaidl-files-under, src)
-
-LOCAL_PROGUARD_FLAG_FILES := proguard.cfg
-
-LOCAL_MULTILIB := first
-
-LOCAL_JAVA_LIBRARIES += android-support-annotations
-
-# To embed native libraries in package, uncomment the lines below.
-#LOCAL_MODULE_TAGS := samples
-#LOCAL_JNI_SHARED_LIBRARIES := \
-# libaacextractor \
-# libamrextractor \
-# libflacextractor \
-# libmidiextractor \
-# libmkvextractor \
-# libmp3extractor \
-# libmp4extractor \
-# libmpeg2extractor \
-# liboggextractor \
-# libwavextractor \
-
-# TODO: Remove dependency with other support libraries.
-LOCAL_STATIC_ANDROID_LIBRARIES += \
- android-support-v4 \
- android-support-v7-appcompat \
- android-support-v7-palette
-LOCAL_USE_AAPT2 := true
-
-include $(BUILD_PACKAGE)
-
-endif # ifneq ($(TARGET_BUILD_PDK),true)
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
+# This package is excluded from build for now since APIs using this apk became hidden.
+#
+#LOCAL_PATH := $(call my-dir)
+#ifneq ($(TARGET_BUILD_PDK),true) # Build MediaComponents only if this is not a PDK build. MediaComponents won't
+## build in PDK builds because frameworks/base/core/java is not available but
+## IMediaSession2.aidl and IMediaController2.aidl are using classes from
+## frameworks/base/core/java.
+#
+#include $(CLEAR_VARS)
+#
+#LOCAL_PACKAGE_NAME := MediaComponents
+#LOCAL_MODULE_OWNER := google
+#
+## TODO: create a separate key for this package.
+#LOCAL_CERTIFICATE := platform
+#
+## TODO: Use System SDK once public APIs are approved
+## LOCAL_SDK_VERSION := system_current
+#LOCAL_PRIVATE_PLATFORM_APIS := true
+#
+#LOCAL_SRC_FILES := \
+# $(call all-java-files-under, src) \
+# $(call all-Iaidl-files-under, src)
+#
+#LOCAL_PROGUARD_FLAG_FILES := proguard.cfg
+#
+#LOCAL_MULTILIB := first
+#
+#LOCAL_JAVA_LIBRARIES += android-support-annotations
+#
+## To embed native libraries in package, uncomment the lines below.
+##LOCAL_MODULE_TAGS := samples
+##LOCAL_JNI_SHARED_LIBRARIES := \
+## libaacextractor \
+## libamrextractor \
+## libflacextractor \
+## libmidiextractor \
+## libmkvextractor \
+## libmp3extractor \
+## libmp4extractor \
+## libmpeg2extractor \
+## liboggextractor \
+## libwavextractor \
+#
+## TODO: Remove dependency with other support libraries.
+#LOCAL_STATIC_ANDROID_LIBRARIES += \
+# android-support-v4 \
+# android-support-v7-appcompat \
+# android-support-v7-palette
+#LOCAL_USE_AAPT2 := true
+#
+#include $(BUILD_PACKAGE)
+#
+#endif # ifneq ($(TARGET_BUILD_PDK),true)
+#
+#include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/packages/MediaComponents/AndroidManifest.xml b/packages/MediaComponents/AndroidManifest.xml
index 061ae44..50fdca1 100644
--- a/packages/MediaComponents/AndroidManifest.xml
+++ b/packages/MediaComponents/AndroidManifest.xml
@@ -8,6 +8,7 @@
android:label="Media Components Update"
android:multiArch="true"
android:allowBackup="false"
+ android:hasCode="false"
android:extractNativeLibs="false">
</application>
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index b38d37f..54121cd 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1000,14 +1000,12 @@
{
ALOGV("AudioFlinger::setRecordSilenced(uid:%d, silenced:%d)", uid, silenced);
- // TODO: Notify MmapThreads
-
AutoMutex lock(mLock);
for (size_t i = 0; i < mRecordThreads.size(); i++) {
- sp<RecordThread> thread = mRecordThreads.valueAt(i);
- if (thread != 0) {
- thread->setRecordSilenced(uid, silenced);
- }
+ mRecordThreads[i]->setRecordSilenced(uid, silenced);
+ }
+ for (size_t i = 0; i < mMmapThreads.size(); i++) {
+ mMmapThreads[i]->setRecordSilenced(uid, silenced);
}
}
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index a210a1b..6f546c3 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -43,6 +43,15 @@
static void appendDumpHeader(String8& result);
void appendDump(String8& result, bool active);
+ // protected by MMapThread::mLock
+ void setSilenced_l(bool silenced) { mSilenced = silenced;
+ mSilencedNotified = false;}
+ // protected by MMapThread::mLock
+ bool isSilenced_l() const { return mSilenced; }
+ // protected by MMapThread::mLock
+ bool getAndSetSilencedNotified_l() { bool silencedNotified = mSilencedNotified;
+ mSilencedNotified = true;
+ return silencedNotified; }
private:
friend class MmapThread;
@@ -58,5 +67,7 @@
virtual void onTimestamp(const ExtendedTimestamp ×tamp);
pid_t mPid;
+ bool mSilenced; // protected by MMapThread::mLock
+ bool mSilencedNotified; // protected by MMapThread::mLock
}; // end of Track
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 43a8b50..1517d11 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -4395,7 +4395,7 @@
float volume = masterVolume
* mStreamTypes[track->streamType()].volume
* vh;
- track->mCachedVolume = masterVolume;
+ track->mCachedVolume = volume;
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
float vlf = volume * float_from_gain(gain_minifloat_unpack_left(vlr));
float vrf = volume * float_from_gain(gain_minifloat_unpack_right(vlr));
@@ -4807,6 +4807,18 @@
track->reset();
}
+ // Track destruction may occur outside of threadLoop once it is removed from active tracks.
+ // Ensure the AudioMixer doesn't have a raw "buffer provider" pointer to the track if
+ // it ceases to be active, to allow safe removal from the AudioMixer at the start
+ // of prepareTracks_l(); this releases any outstanding buffer back to the track.
+ // See also the implementation of destroyTrack_l().
+ for (const auto &track : *tracksToRemove) {
+ const int name = track->name();
+ if (mAudioMixer->exists(name)) { // Normal tracks here, fast tracks in FastMixer.
+ mAudioMixer->setBufferProvider(name, nullptr /* bufferProvider */);
+ }
+ }
+
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
@@ -7886,7 +7898,7 @@
mSessionId(AUDIO_SESSION_NONE),
mDeviceId(AUDIO_PORT_HANDLE_NONE), mPortId(AUDIO_PORT_HANDLE_NONE),
mHalStream(stream), mHalDevice(hwDev->hwDevice()), mAudioHwDev(hwDev),
- mActiveTracks(&this->mLocalLog)
+ mActiveTracks(&this->mLocalLog), mNoCallbackWarningCount(0)
{
mStandby = true;
readHalParameters_l();
@@ -7904,7 +7916,14 @@
void AudioFlinger::MmapThread::disconnect()
{
- for (const sp<MmapTrack> &t : mActiveTracks) {
+ ActiveTracks<MmapTrack> activeTracks;
+ {
+ Mutex::Autolock _l(mLock);
+ for (const sp<MmapTrack> &t : mActiveTracks) {
+ activeTracks.add(t);
+ }
+ }
+ for (const sp<MmapTrack> &t : activeTracks) {
stop(t->portId());
}
// This will decrement references and may cause the destruction of this thread.
@@ -7949,6 +7968,17 @@
return mHalStream->getMmapPosition(position);
}
+status_t AudioFlinger::MmapThread::exitStandby()
+{
+ status_t ret = mHalStream->start();
+ if (ret != NO_ERROR) {
+ ALOGE("%s: error mHalStream->start() = %d for first track", __FUNCTION__, ret);
+ return ret;
+ }
+ mStandby = false;
+ return NO_ERROR;
+}
+
status_t AudioFlinger::MmapThread::start(const AudioClient& client,
audio_port_handle_t *handle)
{
@@ -7962,13 +7992,7 @@
if (*handle == mPortId) {
// for the first track, reuse portId and session allocated when the stream was opened
- ret = mHalStream->start();
- if (ret != NO_ERROR) {
- ALOGE("%s: error mHalStream->start() = %d for first track", __FUNCTION__, ret);
- return ret;
- }
- mStandby = false;
- return NO_ERROR;
+ return exitStandby();
}
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
@@ -8016,33 +8040,43 @@
return BAD_VALUE;
}
+ bool silenced = false;
if (isOutput()) {
ret = AudioSystem::startOutput(mId, streamType(), mSessionId);
} else {
- // TODO: Block recording for idle UIDs (b/72134552)
- bool silenced;
ret = AudioSystem::startInput(portId, &silenced);
}
+ Mutex::Autolock _l(mLock);
// abort if start is rejected by audio policy manager
if (ret != NO_ERROR) {
ALOGE("%s: error start rejected by AudioPolicyManager = %d", __FUNCTION__, ret);
if (mActiveTracks.size() != 0) {
+ mLock.unlock();
if (isOutput()) {
AudioSystem::releaseOutput(mId, streamType(), mSessionId);
} else {
AudioSystem::releaseInput(portId);
}
+ mLock.lock();
} else {
mHalStream->stop();
}
return PERMISSION_DENIED;
}
+ if (!isOutput() && !silenced) {
+ for (const sp<MmapTrack> &track : mActiveTracks) {
+ if (track->isSilenced_l() && track->uid() != client.clientUid)
+ track->invalidate();
+ }
+ }
+
// Given that MmapThread::mAttr is mutable, should a MmapTrack have attributes ?
sp<MmapTrack> track = new MmapTrack(this, mAttr, mSampleRate, mFormat, mChannelMask, mSessionId,
client.clientUid, client.clientPid, portId);
+ track->setSilenced_l(silenced);
mActiveTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(mSessionId);
if (chain != 0) {
@@ -8072,6 +8106,8 @@
return NO_ERROR;
}
+ Mutex::Autolock _l(mLock);
+
sp<MmapTrack> track;
for (const sp<MmapTrack> &t : mActiveTracks) {
if (handle == t->portId()) {
@@ -8085,6 +8121,7 @@
mActiveTracks.remove(track);
+ mLock.unlock();
if (isOutput()) {
AudioSystem::stopOutput(mId, streamType(), track->sessionId());
AudioSystem::releaseOutput(mId, streamType(), track->sessionId());
@@ -8092,6 +8129,7 @@
AudioSystem::stopInput(track->portId());
AudioSystem::releaseInput(track->portId());
}
+ mLock.lock();
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
@@ -8518,9 +8556,11 @@
if (track->isInvalid()) {
sp<MmapStreamCallback> callback = mCallback.promote();
if (callback != 0) {
- callback->onTearDown();
+ callback->onTearDown(track->portId());
+ } else if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) {
+ ALOGW("Could not notify MMAP stream tear down: no onTearDown callback!");
+ mNoCallbackWarningCount++;
}
- break;
}
}
}
@@ -8575,7 +8615,6 @@
mStreamVolume(1.0),
mStreamMute(false),
mHalVolFloat(-1.0f), // Initialize to illegal value so it always gets set properly later.
- mNoCallbackWarningCount(0),
mOutput(output)
{
snprintf(mThreadName, kThreadNameLength, "AudioMmapOut_%X", id);
@@ -8780,6 +8819,12 @@
mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
}
+status_t AudioFlinger::MmapCaptureThread::exitStandby()
+{
+ mInput->stream->setGain(1.0f);
+ return MmapThread::exitStandby();
+}
+
AudioFlinger::AudioStreamIn* AudioFlinger::MmapCaptureThread::clearInput()
{
Mutex::Autolock _l(mLock);
@@ -8788,6 +8833,34 @@
return input;
}
+
+void AudioFlinger::MmapCaptureThread::processVolume_l()
+{
+ bool changed = false;
+ bool silenced = false;
+
+ sp<MmapStreamCallback> callback = mCallback.promote();
+ if (callback == 0) {
+ if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) {
+ ALOGW("Could not set MMAP stream silenced: no onStreamSilenced callback!");
+ mNoCallbackWarningCount++;
+ }
+ }
+
+ // After a change occurred in track silenced state, mute capture in audio DSP if at least one
+ // track is silenced and unmute otherwise
+ for (size_t i = 0; i < mActiveTracks.size() && !silenced; i++) {
+ if (!mActiveTracks[i]->getAndSetSilencedNotified_l()) {
+ changed = true;
+ silenced = mActiveTracks[i]->isSilenced_l();
+ }
+ }
+
+ if (changed) {
+ mInput->stream->setGain(silenced ? 0.0f: 1.0f);
+ }
+}
+
void AudioFlinger::MmapCaptureThread::updateMetadata_l()
{
if (mInput == nullptr || mInput->stream == nullptr ||
@@ -8805,4 +8878,15 @@
mInput->stream->updateSinkMetadata(metadata);
}
+void AudioFlinger::MmapCaptureThread::setRecordSilenced(uid_t uid, bool silenced)
+{
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mActiveTracks.size() ; i++) {
+ if (mActiveTracks[i]->uid() == uid) {
+ mActiveTracks[i]->setSilenced_l(silenced);
+ broadcast_l();
+ }
+ }
+}
+
} // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 5a5961a..bc4a534 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1589,6 +1589,7 @@
virtual void threadLoop_exit();
virtual void threadLoop_standby();
virtual bool shouldStandby_l() { return false; }
+ virtual status_t exitStandby();
virtual status_t initCheck() const { return (mHalStream == 0) ? NO_INIT : NO_ERROR; }
virtual size_t frameCount() const { return mFrameCount; }
@@ -1621,6 +1622,9 @@
virtual void invalidateTracks(audio_stream_type_t streamType __unused) {}
+ // Sets the UID records silence
+ virtual void setRecordSilenced(uid_t uid __unused, bool silenced __unused) {}
+
void dump(int fd, const Vector<String16>& args);
virtual void dumpInternals(int fd, const Vector<String16>& args);
void dumpTracks(int fd, const Vector<String16>& args);
@@ -1637,6 +1641,9 @@
sp<DeviceHalInterface> mHalDevice;
AudioHwDevice* const mAudioHwDev;
ActiveTracks<MmapTrack> mActiveTracks;
+
+ int32_t mNoCallbackWarningCount;
+ static constexpr int32_t kMaxNoCallbackWarnings = 5;
};
class MmapPlaybackThread : public MmapThread, public VolumeInterface
@@ -1670,7 +1677,7 @@
virtual audio_stream_type_t streamType() { return mStreamType; }
virtual void checkSilentMode_l();
- virtual void processVolume_l();
+ void processVolume_l() override;
virtual void dumpInternals(int fd, const Vector<String16>& args);
@@ -1686,8 +1693,6 @@
bool mMasterMute;
bool mStreamMute;
float mHalVolFloat;
- int32_t mNoCallbackWarningCount;
- static constexpr int32_t kMaxNoCallbackWarnings = 5;
AudioStreamOut* mOutput;
};
@@ -1702,9 +1707,12 @@
AudioStreamIn* clearInput();
+ status_t exitStandby() override;
virtual bool isOutput() const override { return false; }
void updateMetadata_l() override;
+ void processVolume_l() override;
+ void setRecordSilenced(uid_t uid, bool silenced) override;
protected:
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index ee9ce84..aff1239 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1782,14 +1782,14 @@
/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
{
- result.append("Active Client Session S Flags Format Chn mask SRate Server FrmCnt\n");
+ result.append("Active Client Session S Flags Format Chn mask SRate Server FrmCnt Sil\n");
}
void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active)
{
result.appendFormat("%c%5s %6u %7u %2s 0x%03X "
"%08X %08X %6u "
- "%08X %6zu\n",
+ "%08X %6zu %3c\n",
isFastTrack() ? 'F' : ' ',
active ? "yes" : "no",
(mClient == 0) ? getpid_cached : mClient->pid(),
@@ -1802,7 +1802,8 @@
mSampleRate,
mCblk->mServer,
- mFrameCount
+ mFrameCount,
+ isSilenced() ? 's' : 'n'
);
}
@@ -1945,7 +1946,7 @@
sessionId, uid, false /* isOut */,
ALLOC_NONE,
TYPE_DEFAULT, portId),
- mPid(pid)
+ mPid(pid), mSilenced(false), mSilencedNotified(false)
{
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 947529e..0d36266 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -1468,14 +1468,19 @@
}
// For MMAP mode, the first call to getInputForAttr() is made on behalf of audioflinger.
// The second call is for the first active client and sets the UID. Any further call
- // corresponds to a new client and is only permitted from the same UId.
+ // corresponds to a new client and is only permitted from the same UID.
+ // If the first UID is silenced, allow a new UID connection and replace with new UID
if (audioSession->openCount() == 1) {
audioSession->setUid(uid);
} else if (audioSession->uid() != uid) {
- ALOGW("getInputForAttr() bad uid %d for session %d uid %d",
- uid, session, audioSession->uid());
- status = INVALID_OPERATION;
- goto error;
+ if (!audioSession->isSilenced()) {
+ ALOGW("getInputForAttr() bad uid %d for session %d uid %d",
+ uid, session, audioSession->uid());
+ status = INVALID_OPERATION;
+ goto error;
+ }
+ audioSession->setUid(uid);
+ audioSession->setSilenced(false);
}
audioSession->changeOpenCount(1);
*inputType = API_INPUT_LEGACY;
diff --git a/services/mediaextractor/Android.mk b/services/mediaextractor/Android.mk
index 5b7571c..d505cfe 100644
--- a/services/mediaextractor/Android.mk
+++ b/services/mediaextractor/Android.mk
@@ -32,7 +32,6 @@
libmpeg2extractor \
liboggextractor \
libwavextractor \
- MediaComponents \
LOCAL_SRC_FILES := main_extractorservice.cpp
LOCAL_SHARED_LIBRARIES := libmedia libmediaextractorservice libbinder libutils \
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 52990da..713ce60 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -343,8 +343,9 @@
}
-void AAudioServiceEndpointMMAP::onTearDown() {
+void AAudioServiceEndpointMMAP::onTearDown(audio_port_handle_t handle __unused) {
ALOGD("%s(%p) called", __func__, this);
+ //TODO: disconnect only stream corresponding to handle received
disconnectRegisteredStreams();
};
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.h b/services/oboeservice/AAudioServiceEndpointMMAP.h
index 16b6269..c4c943d 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.h
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.h
@@ -68,7 +68,7 @@
aaudio_result_t getTimestamp(int64_t *positionFrames, int64_t *timeNanos) override;
// -------------- Callback functions for MmapStreamCallback ---------------------
- void onTearDown() override;
+ void onTearDown(audio_port_handle_t handle) override;
void onVolumeChanged(audio_channel_mask_t channels,
android::Vector<float> values) override;