Merge "Camera: Fix test pattern data propagation" into sc-dev am: 001d147389
Original change: https://googleplex-android-review.googlesource.com/c/platform/frameworks/av/+/15379651
Change-Id: If63fa3987f54c8dd74618a0f44d8a6989cfd28ea
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 4b08295..f97fe4d 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -565,6 +565,145 @@
};
typedef int32_t aaudio_session_id_t;
+/**
+ * Defines the audio channel mask.
+ * Channel masks are used to describe the samples and their
+ * arrangement in the audio frame. They are also used in the endpoint
+ * (e.g. a USB audio interface, a DAC connected to headphones) to
+ * specify allowable configurations of a particular device.
+ *
+ * Added in API level 32.
+ */
+enum {
+ /**
+ * Invalid channel mask
+ */
+ AAUDIO_CHANNEL_INVALID = -1,
+
+ /**
+ * Output audio channel mask
+ */
+ AAUDIO_CHANNEL_FRONT_LEFT = 1 << 0,
+ AAUDIO_CHANNEL_FRONT_RIGHT = 1 << 1,
+ AAUDIO_CHANNEL_FRONT_CENTER = 1 << 2,
+ AAUDIO_CHANNEL_LOW_FREQUENCY = 1 << 3,
+ AAUDIO_CHANNEL_BACK_LEFT = 1 << 4,
+ AAUDIO_CHANNEL_BACK_RIGHT = 1 << 5,
+ AAUDIO_CHANNEL_FRONT_LEFT_OF_CENTER = 1 << 6,
+ AAUDIO_CHANNEL_FRONT_RIGHT_OF_CENTER = 1 << 7,
+ AAUDIO_CHANNEL_BACK_CENTER = 1 << 8,
+ AAUDIO_CHANNEL_SIDE_LEFT = 1 << 9,
+ AAUDIO_CHANNEL_SIDE_RIGHT = 1 << 10,
+ AAUDIO_CHANNEL_TOP_CENTER = 1 << 11,
+ AAUDIO_CHANNEL_TOP_FRONT_LEFT = 1 << 12,
+ AAUDIO_CHANNEL_TOP_FRONT_CENTER = 1 << 13,
+ AAUDIO_CHANNEL_TOP_FRONT_RIGHT = 1 << 14,
+ AAUDIO_CHANNEL_TOP_BACK_LEFT = 1 << 15,
+ AAUDIO_CHANNEL_TOP_BACK_CENTER = 1 << 16,
+ AAUDIO_CHANNEL_TOP_BACK_RIGHT = 1 << 17,
+ AAUDIO_CHANNEL_TOP_SIDE_LEFT = 1 << 18,
+ AAUDIO_CHANNEL_TOP_SIDE_RIGHT = 1 << 19,
+ AAUDIO_CHANNEL_BOTTOM_FRONT_LEFT = 1 << 20,
+ AAUDIO_CHANNEL_BOTTOM_FRONT_CENTER = 1 << 21,
+ AAUDIO_CHANNEL_BOTTOM_FRONT_RIGHT = 1 << 22,
+ AAUDIO_CHANNEL_LOW_FREQUENCY_2 = 1 << 23,
+ AAUDIO_CHANNEL_FRONT_WIDE_LEFT = 1 << 24,
+ AAUDIO_CHANNEL_FRONT_WIDE_RIGHT = 1 << 25,
+
+ AAUDIO_CHANNEL_MONO = AAUDIO_CHANNEL_FRONT_LEFT,
+ AAUDIO_CHANNEL_STEREO = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT,
+ AAUDIO_CHANNEL_2POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_LOW_FREQUENCY,
+ AAUDIO_CHANNEL_TRI = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_FRONT_CENTER,
+ AAUDIO_CHANNEL_TRI_BACK = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_BACK_CENTER,
+ AAUDIO_CHANNEL_3POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_FRONT_CENTER |
+ AAUDIO_CHANNEL_LOW_FREQUENCY,
+ AAUDIO_CHANNEL_2POINT0POINT2 = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+ AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+ AAUDIO_CHANNEL_2POINT1POINT2 = AAUDIO_CHANNEL_2POINT0POINT2 |
+ AAUDIO_CHANNEL_LOW_FREQUENCY,
+ AAUDIO_CHANNEL_3POINT0POINT2 = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_FRONT_CENTER |
+ AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+ AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+ AAUDIO_CHANNEL_3POINT1POINT2 = AAUDIO_CHANNEL_3POINT0POINT2 |
+ AAUDIO_CHANNEL_LOW_FREQUENCY,
+ AAUDIO_CHANNEL_QUAD = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_BACK_LEFT |
+ AAUDIO_CHANNEL_BACK_RIGHT,
+ AAUDIO_CHANNEL_QUAD_SIDE = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_SIDE_LEFT |
+ AAUDIO_CHANNEL_SIDE_RIGHT,
+ AAUDIO_CHANNEL_SURROUND = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_FRONT_CENTER |
+ AAUDIO_CHANNEL_BACK_CENTER,
+ AAUDIO_CHANNEL_PENTA = AAUDIO_CHANNEL_QUAD |
+ AAUDIO_CHANNEL_FRONT_CENTER,
+ // aka 5POINT1_BACK
+ AAUDIO_CHANNEL_5POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_FRONT_CENTER |
+ AAUDIO_CHANNEL_LOW_FREQUENCY |
+ AAUDIO_CHANNEL_BACK_LEFT |
+ AAUDIO_CHANNEL_BACK_RIGHT,
+ AAUDIO_CHANNEL_5POINT1_SIDE = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_FRONT_CENTER |
+ AAUDIO_CHANNEL_LOW_FREQUENCY |
+ AAUDIO_CHANNEL_SIDE_LEFT |
+ AAUDIO_CHANNEL_SIDE_RIGHT,
+ AAUDIO_CHANNEL_6POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+ AAUDIO_CHANNEL_FRONT_RIGHT |
+ AAUDIO_CHANNEL_FRONT_CENTER |
+ AAUDIO_CHANNEL_LOW_FREQUENCY |
+ AAUDIO_CHANNEL_BACK_LEFT |
+ AAUDIO_CHANNEL_BACK_RIGHT |
+ AAUDIO_CHANNEL_BACK_CENTER,
+ AAUDIO_CHANNEL_7POINT1 = AAUDIO_CHANNEL_5POINT1 |
+ AAUDIO_CHANNEL_SIDE_LEFT |
+ AAUDIO_CHANNEL_SIDE_RIGHT,
+ AAUDIO_CHANNEL_5POINT1POINT2 = AAUDIO_CHANNEL_5POINT1 |
+ AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+ AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+ AAUDIO_CHANNEL_5POINT1POINT4 = AAUDIO_CHANNEL_5POINT1 |
+ AAUDIO_CHANNEL_TOP_FRONT_LEFT |
+ AAUDIO_CHANNEL_TOP_FRONT_RIGHT |
+ AAUDIO_CHANNEL_TOP_BACK_LEFT |
+ AAUDIO_CHANNEL_TOP_BACK_RIGHT,
+ AAUDIO_CHANNEL_7POINT1POINT2 = AAUDIO_CHANNEL_7POINT1 |
+ AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+ AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+ AAUDIO_CHANNEL_7POINT1POINT4 = AAUDIO_CHANNEL_7POINT1 |
+ AAUDIO_CHANNEL_TOP_FRONT_LEFT |
+ AAUDIO_CHANNEL_TOP_FRONT_RIGHT |
+ AAUDIO_CHANNEL_TOP_BACK_LEFT |
+ AAUDIO_CHANNEL_TOP_BACK_RIGHT,
+ AAUDIO_CHANNEL_9POINT1POINT4 = AAUDIO_CHANNEL_7POINT1POINT4 |
+ AAUDIO_CHANNEL_FRONT_WIDE_LEFT |
+ AAUDIO_CHANNEL_FRONT_WIDE_RIGHT,
+ AAUDIO_CHANNEL_9POINT1POINT6 = AAUDIO_CHANNEL_9POINT1POINT4 |
+ AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+ AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+
+ AAUDIO_CHANNEL_FRONT_BACK = AAUDIO_CHANNEL_FRONT_CENTER |
+ AAUDIO_CHANNEL_BACK_CENTER,
+};
+typedef uint32_t aaudio_channel_mask_t;
+
typedef struct AAudioStreamStruct AAudioStream;
typedef struct AAudioStreamBuilderStruct AAudioStreamBuilder;
@@ -699,6 +838,11 @@
* If an exact value is specified then an opened stream will use that value.
* If a stream cannot be opened with the specified value then the open will fail.
*
+ * As the channel count provided here may be different from the corresponding channel count
+ * of channel mask used in {@link AAudioStreamBuilder_setChannelMask}, the last called function
+ * will be respected if both this function and {@link AAudioStreamBuilder_setChannelMask} are
+ * called.
+ *
* Available since API level 26.
*
* @param builder reference provided by AAudio_createStreamBuilder()
@@ -714,6 +858,8 @@
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param samplesPerFrame Number of samples in a frame.
+ *
+ * @deprecated use {@link AAudioStreamBuilder_setChannelCount}
*/
AAUDIO_API void AAudioStreamBuilder_setSamplesPerFrame(AAudioStreamBuilder* builder,
int32_t samplesPerFrame) __INTRODUCED_IN(26);
@@ -1136,6 +1282,32 @@
AAUDIO_API aaudio_result_t AAudioStreamBuilder_delete(AAudioStreamBuilder* builder)
__INTRODUCED_IN(26);
+/**
+ * Set audio channel mask for the stream.
+ *
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
+ * If both channel mask and count are not set, then stereo will then be chosen when the
+ * stream is opened.
+ * After opening a stream with an unspecified value, the application must query for the
+ * actual value, which may vary by device.
+ *
+ * If an exact value is specified then an opened stream will use that value.
+ * If a stream cannot be opened with the specified value then the open will fail.
+ *
+ * As the corresponding channel count of provided channel mask here may be different
+ * from the channel count used in {@link AAudioStreamBuilder_setChannelCount} or
+ * {@link AAudioStreamBuilder_setSamplesPerFrame}, the last called function will be
+ * respected if this function and {@link AAudioStreamBuilder_setChannelCount} or
+ * {@link AAudioStreamBuilder_setSamplesPerFrame} are called.
+ *
+ * Available since API level 32.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param channelMask Audio channel mask desired.
+ */
+AAUDIO_API void AAudioStreamBuilder_setChannelMask(AAudioStreamBuilder* builder,
+ aaudio_channel_mask_t channelMask) __INTRODUCED_IN(32);
+
// ============================================================
// Stream Control
// ============================================================
@@ -1652,6 +1824,18 @@
AAUDIO_API bool AAudioStream_isPrivacySensitive(AAudioStream* stream)
__INTRODUCED_IN(30);
+/**
+ * Return the channel mask for the stream. This will be the mask set using
+ * {@link #AAudioStreamBuilder_setChannelMask}, or {@link #AAUDIO_UNSPECIFIED} otherwise.
+ *
+ * Available since API level 32.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return actual channel mask
+ */
+AAUDIO_API aaudio_channel_mask_t AAudioStream_getChannelMask(AAudioStream* stream)
+ __INTRODUCED_IN(32);
+
#ifdef __cplusplus
}
#endif
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index 2d501ef..dc19bb3 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
@@ -30,7 +30,7 @@
using android::media::audio::common::AudioFormat;
AAudioStreamConfiguration::AAudioStreamConfiguration(const StreamParameters& parcelable) {
- setSamplesPerFrame(parcelable.samplesPerFrame);
+ setChannelMask(parcelable.channelMask);
setSampleRate(parcelable.sampleRate);
setDeviceId(parcelable.deviceId);
static_assert(sizeof(aaudio_sharing_mode_t) == sizeof(parcelable.sharingMode));
@@ -63,7 +63,7 @@
StreamParameters AAudioStreamConfiguration::parcelable() const {
StreamParameters result;
- result.samplesPerFrame = getSamplesPerFrame();
+ result.channelMask = getChannelMask();
result.sampleRate = getSampleRate();
result.deviceId = getDeviceId();
static_assert(sizeof(aaudio_sharing_mode_t) == sizeof(result.sharingMode));
diff --git a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
index b7c4f70..1a971b0 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
@@ -19,7 +19,7 @@
import android.media.audio.common.AudioFormat;
parcelable StreamParameters {
- int samplesPerFrame; // = AAUDIO_UNSPECIFIED;
+ int channelMask; // = AAUDIO_UNSPECIFIED;
int sampleRate; // = AAUDIO_UNSPECIFIED;
int deviceId; // = AAUDIO_UNSPECIFIED;
int /* aaudio_sharing_mode_t */ sharingMode; // = AAUDIO_SHARING_MODE_SHARED;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index cf2abe8..ebb06b2 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -123,9 +123,9 @@
request.getConfiguration().setDeviceId(getDeviceId());
request.getConfiguration().setSampleRate(getSampleRate());
- request.getConfiguration().setSamplesPerFrame(getSamplesPerFrame());
request.getConfiguration().setDirection(getDirection());
request.getConfiguration().setSharingMode(getSharingMode());
+ request.getConfiguration().setChannelMask(getChannelMask());
request.getConfiguration().setUsage(getUsage());
request.getConfiguration().setContentType(getContentType());
@@ -138,7 +138,8 @@
mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
if (mServiceStreamHandle < 0
- && request.getConfiguration().getSamplesPerFrame() == 1 // mono?
+ && (request.getConfiguration().getSamplesPerFrame() == 1
+ || request.getConfiguration().getChannelMask() == AAUDIO_CHANNEL_MONO)
&& getDirection() == AAUDIO_DIRECTION_OUTPUT
&& !isInService()) {
// if that failed then try switching from mono to stereo if OUTPUT.
@@ -146,7 +147,7 @@
// that writes to a stereo MMAP stream.
ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO",
__func__, mServiceStreamHandle);
- request.getConfiguration().setSamplesPerFrame(2); // stereo
+ request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO);
mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
}
if (mServiceStreamHandle < 0) {
@@ -171,9 +172,10 @@
goto error;
}
// Save results of the open.
- if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
- setSamplesPerFrame(configurationOutput.getSamplesPerFrame());
+ if (getChannelMask() == AAUDIO_UNSPECIFIED) {
+ setChannelMask(configurationOutput.getChannelMask());
}
+
mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
setSampleRate(configurationOutput.getSampleRate());
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index d103aca..02e7f5f 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -128,7 +128,8 @@
int32_t samplesPerFrame)
{
AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
- streamBuilder->setSamplesPerFrame(samplesPerFrame);
+ const aaudio_channel_mask_t channelMask = AAudioConvert_channelCountToMask(samplesPerFrame);
+ streamBuilder->setChannelMask(channelMask);
}
AAUDIO_API void AAudioStreamBuilder_setDirection(AAudioStreamBuilder* builder,
@@ -223,6 +224,13 @@
streamBuilder->setFramesPerDataCallback(frames);
}
+AAUDIO_API void AAudioStreamBuilder_setChannelMask(AAudioStreamBuilder* builder,
+ aaudio_channel_mask_t channelMask)
+{
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+ streamBuilder->setChannelMask(channelMask);
+}
+
AAUDIO_API aaudio_result_t AAudioStreamBuilder_openStream(AAudioStreamBuilder* builder,
AAudioStream** streamPtr)
{
@@ -562,3 +570,11 @@
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
return audioStream->isPrivacySensitive();
}
+
+AAUDIO_API aaudio_channel_mask_t AAudioStream_getChannelMask(AAudioStream* stream)
+{
+ AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+ const aaudio_channel_mask_t channelMask = audioStream->getChannelMask();
+ // Do not return channel index masks as they are not public.
+ return AAudio_isChannelIndexMask(channelMask) ? AAUDIO_UNSPECIFIED : channelMask;
+}
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index acfac24..59d94eb 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -49,6 +49,7 @@
mIsPrivacySensitive = other.mIsPrivacySensitive;
mOpPackageName = other.mOpPackageName;
mAttributionTag = other.mAttributionTag;
+ mChannelMask = other.mChannelMask;
}
static aaudio_result_t isFormatValid(audio_format_t format) {
@@ -187,7 +188,94 @@
// break;
}
- return AAUDIO_OK;
+ return validateChannelMask();
+}
+
+bool AAudioStreamParameters::validateChannelMask() const {
+ if (mChannelMask == AAUDIO_UNSPECIFIED) {
+ return AAUDIO_OK;
+ }
+
+ if (mChannelMask & AAUDIO_CHANNEL_BIT_INDEX) {
+ switch (mChannelMask) {
+ case AAUDIO_CHANNEL_INDEX_MASK_1:
+ case AAUDIO_CHANNEL_INDEX_MASK_2:
+ case AAUDIO_CHANNEL_INDEX_MASK_3:
+ case AAUDIO_CHANNEL_INDEX_MASK_4:
+ case AAUDIO_CHANNEL_INDEX_MASK_5:
+ case AAUDIO_CHANNEL_INDEX_MASK_6:
+ case AAUDIO_CHANNEL_INDEX_MASK_7:
+ case AAUDIO_CHANNEL_INDEX_MASK_8:
+ case AAUDIO_CHANNEL_INDEX_MASK_9:
+ case AAUDIO_CHANNEL_INDEX_MASK_10:
+ case AAUDIO_CHANNEL_INDEX_MASK_11:
+ case AAUDIO_CHANNEL_INDEX_MASK_12:
+ case AAUDIO_CHANNEL_INDEX_MASK_13:
+ case AAUDIO_CHANNEL_INDEX_MASK_14:
+ case AAUDIO_CHANNEL_INDEX_MASK_15:
+ case AAUDIO_CHANNEL_INDEX_MASK_16:
+ case AAUDIO_CHANNEL_INDEX_MASK_17:
+ case AAUDIO_CHANNEL_INDEX_MASK_18:
+ case AAUDIO_CHANNEL_INDEX_MASK_19:
+ case AAUDIO_CHANNEL_INDEX_MASK_20:
+ case AAUDIO_CHANNEL_INDEX_MASK_21:
+ case AAUDIO_CHANNEL_INDEX_MASK_22:
+ case AAUDIO_CHANNEL_INDEX_MASK_23:
+ case AAUDIO_CHANNEL_INDEX_MASK_24:
+ return AAUDIO_OK;
+ default:
+ ALOGD("Invalid channel index mask %#x", mChannelMask);
+ return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ }
+ }
+
+ if (getDirection() == AAUDIO_DIRECTION_INPUT) {
+ switch (mChannelMask) {
+ case AAUDIO_CHANNEL_MONO:
+ case AAUDIO_CHANNEL_STEREO:
+ case AAUDIO_CHANNEL_FRONT_BACK:
+ case AAUDIO_CHANNEL_2POINT0POINT2:
+ case AAUDIO_CHANNEL_2POINT1POINT2:
+ case AAUDIO_CHANNEL_3POINT0POINT2:
+ case AAUDIO_CHANNEL_3POINT1POINT2:
+ case AAUDIO_CHANNEL_5POINT1:
+ return AAUDIO_OK;
+ default:
+ ALOGD("Invalid channel mask %#x, IN", mChannelMask);
+ return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ }
+ } else {
+ switch (mChannelMask) {
+ case AAUDIO_CHANNEL_MONO:
+ case AAUDIO_CHANNEL_STEREO:
+ case AAUDIO_CHANNEL_2POINT1:
+ case AAUDIO_CHANNEL_TRI:
+ case AAUDIO_CHANNEL_TRI_BACK:
+ case AAUDIO_CHANNEL_3POINT1:
+ case AAUDIO_CHANNEL_2POINT0POINT2:
+ case AAUDIO_CHANNEL_2POINT1POINT2:
+ case AAUDIO_CHANNEL_3POINT0POINT2:
+ case AAUDIO_CHANNEL_3POINT1POINT2:
+ case AAUDIO_CHANNEL_QUAD:
+ case AAUDIO_CHANNEL_QUAD_SIDE:
+ case AAUDIO_CHANNEL_SURROUND:
+ case AAUDIO_CHANNEL_PENTA:
+ case AAUDIO_CHANNEL_5POINT1:
+ case AAUDIO_CHANNEL_5POINT1_SIDE:
+ case AAUDIO_CHANNEL_5POINT1POINT2:
+ case AAUDIO_CHANNEL_5POINT1POINT4:
+ case AAUDIO_CHANNEL_6POINT1:
+ case AAUDIO_CHANNEL_7POINT1:
+ case AAUDIO_CHANNEL_7POINT1POINT2:
+ case AAUDIO_CHANNEL_7POINT1POINT4:
+ case AAUDIO_CHANNEL_9POINT1POINT4:
+ case AAUDIO_CHANNEL_9POINT1POINT6:
+ return AAUDIO_OK;
+ default:
+ ALOGD("Invalid channel mask %#x. OUT", mChannelMask);
+ return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ }
+ }
}
void AAudioStreamParameters::dump() const {
@@ -195,6 +283,7 @@
ALOGD("mSessionId = %6d", mSessionId);
ALOGD("mSampleRate = %6d", mSampleRate);
ALOGD("mSamplesPerFrame = %6d", mSamplesPerFrame);
+ ALOGD("mChannelMask = %#x", mChannelMask);
ALOGD("mSharingMode = %6d", (int)mSharingMode);
ALOGD("mAudioFormat = %6d", (int)mAudioFormat);
ALOGD("mDirection = %6d", mDirection);
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.h b/media/libaaudio/src/core/AAudioStreamParameters.h
index 5737052..a5c8043 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.h
+++ b/media/libaaudio/src/core/AAudioStreamParameters.h
@@ -49,13 +49,6 @@
return mSamplesPerFrame;
}
- /**
- * This is also known as channelCount.
- */
- void setSamplesPerFrame(int32_t samplesPerFrame) {
- mSamplesPerFrame = samplesPerFrame;
- }
-
audio_format_t getFormat() const {
return mAudioFormat;
}
@@ -153,6 +146,15 @@
mAttributionTag = attributionTag;
}
+ aaudio_channel_mask_t getChannelMask() const {
+ return mChannelMask;
+ }
+
+ void setChannelMask(aaudio_channel_mask_t channelMask) {
+ mChannelMask = channelMask;
+ mSamplesPerFrame = AAudioConvert_channelMaskToCount(channelMask);
+ }
+
/**
* @return bytes per frame of getFormat()
*/
@@ -171,6 +173,8 @@
void dump() const;
private:
+ bool validateChannelMask() const;
+
int32_t mSamplesPerFrame = AAUDIO_UNSPECIFIED;
int32_t mSampleRate = AAUDIO_UNSPECIFIED;
int32_t mDeviceId = AAUDIO_UNSPECIFIED;
@@ -186,6 +190,7 @@
bool mIsPrivacySensitive = false;
std::optional<std::string> mOpPackageName = {};
std::optional<std::string> mAttributionTag = {};
+ aaudio_channel_mask_t mChannelMask = AAUDIO_UNSPECIFIED;
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 09d9535..ffc3b9d 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -76,6 +76,7 @@
// Copy parameters from the Builder because the Builder may be deleted after this call.
// TODO AudioStream should be a subclass of AudioStreamParameters
mSamplesPerFrame = builder.getSamplesPerFrame();
+ mChannelMask = builder.getChannelMask();
mSampleRate = builder.getSampleRate();
mDeviceId = builder.getDeviceId();
mFormat = builder.getFormat();
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 9835c8c..47693f8 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -270,7 +270,8 @@
}
/**
- * This is only valid after setSamplesPerFrame() and setFormat() have been called.
+ * This is only valid after setChannelMask() and setFormat()
+ * have been called.
*/
int32_t getBytesPerFrame() const {
return mSamplesPerFrame * getBytesPerSample();
@@ -284,7 +285,7 @@
}
/**
- * This is only valid after setSamplesPerFrame() and setDeviceFormat() have been called.
+ * This is only valid after setChannelMask() and setDeviceFormat() have been called.
*/
int32_t getBytesPerDeviceFrame() const {
return getSamplesPerFrame() * audio_bytes_per_sample(getDeviceFormat());
@@ -318,6 +319,15 @@
return mFramesPerDataCallback;
}
+ aaudio_channel_mask_t getChannelMask() const {
+ return mChannelMask;
+ }
+
+ void setChannelMask(aaudio_channel_mask_t channelMask) {
+ mChannelMask = channelMask;
+ mSamplesPerFrame = AAudioConvert_channelMaskToCount(channelMask);
+ }
+
/**
* @return true if data callback has been specified
*/
@@ -495,11 +505,6 @@
}
// This should not be called after the open() call.
- void setSamplesPerFrame(int32_t samplesPerFrame) {
- mSamplesPerFrame = samplesPerFrame;
- }
-
- // This should not be called after the open() call.
void setFramesPerBurst(int32_t framesPerBurst) {
mFramesPerBurst = framesPerBurst;
}
@@ -633,6 +638,7 @@
// These do not change after open().
int32_t mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+ aaudio_channel_mask_t mChannelMask = AAUDIO_UNSPECIFIED;
int32_t mSampleRate = AAUDIO_UNSPECIFIED;
int32_t mDeviceId = AAUDIO_UNSPECIFIED;
aaudio_sharing_mode_t mSharingMode = AAUDIO_SHARING_MODE_SHARED;
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index e015592..5e1e007 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -268,8 +268,8 @@
void AudioStreamBuilder::logParameters() const {
// This is very helpful for debugging in the future. Please leave it in.
- ALOGI("rate = %6d, channels = %d, format = %d, sharing = %s, dir = %s",
- getSampleRate(), getSamplesPerFrame(), getFormat(),
+ ALOGI("rate = %6d, channels = %d, channelMask = %#x, format = %d, sharing = %s, dir = %s",
+ getSampleRate(), getSamplesPerFrame(), getChannelMask(), getFormat(),
AAudio_convertSharingModeToShortText(getSharingMode()),
AAudio_convertDirectionToText(getDirection()));
ALOGI("device = %6d, sessionId = %d, perfMode = %d, callback: %s with frames = %d",
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index dc66742..fe8fb19 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -65,11 +65,8 @@
const audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
// TODO Support UNSPECIFIED in AudioRecord. For now, use stereo if unspecified.
- int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
- ? 2 : getSamplesPerFrame();
- audio_channel_mask_t channelMask = samplesPerFrame <= 2 ?
- audio_channel_in_mask_from_count(samplesPerFrame) :
- audio_channel_mask_for_index_assignment_from_count(samplesPerFrame);
+ audio_channel_mask_t channelMask =
+ AAudio_getChannelMaskForOpen(getChannelMask(), getSamplesPerFrame(), true /*isInput*/);
size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0
: builder.getBufferCapacity();
@@ -115,7 +112,7 @@
constexpr int32_t kMostLikelySampleRateForFast = 48000;
if (getFormat() == AUDIO_FORMAT_PCM_FLOAT
&& perfMode == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
- && (samplesPerFrame <= 2) // FAST only for mono and stereo
+ && (audio_channel_count_from_in_mask(channelMask) <= 2) // FAST only for mono and stereo
&& (getSampleRate() == kMostLikelySampleRateForFast
|| getSampleRate() == AAUDIO_UNSPECIFIED)) {
setDeviceFormat(AUDIO_FORMAT_PCM_16_BIT);
@@ -228,7 +225,9 @@
.set(AMEDIAMETRICS_PROP_ENCODINGCLIENT, toString(requestedFormat).c_str()).record();
// Get the actual values from the AudioRecord.
- setSamplesPerFrame(mAudioRecord->channelCount());
+ setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+ mAudioRecord->channelMask(), true /*isInput*/,
+ AAudio_isChannelIndexMask(getChannelMask())));
setSampleRate(mAudioRecord->getSampleRate());
setBufferCapacity(getBufferCapacityFromDevice());
setFramesPerBurst(getFramesPerBurstFromDevice());
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 1d412c0..2291ad7 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -66,13 +66,8 @@
const aaudio_session_id_t requestedSessionId = builder.getSessionId();
const audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
- // Try to create an AudioTrack
- // Use stereo if unspecified.
- int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
- ? 2 : getSamplesPerFrame();
- audio_channel_mask_t channelMask = samplesPerFrame <= 2 ?
- audio_channel_out_mask_from_count(samplesPerFrame) :
- audio_channel_mask_for_index_assignment_from_count(samplesPerFrame);
+ audio_channel_mask_t channelMask =
+ AAudio_getChannelMaskForOpen(getChannelMask(), getSamplesPerFrame(), false /*isInput*/);
audio_output_flags_t flags;
aaudio_performance_mode_t perfMode = getPerformanceMode();
@@ -199,7 +194,9 @@
doSetVolume();
// Get the actual values from the AudioTrack.
- setSamplesPerFrame(mAudioTrack->channelCount());
+ setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+ mAudioTrack->channelMask(), false /*isInput*/,
+ AAudio_isChannelIndexMask(getChannelMask())));
setFormat(mAudioTrack->format());
setDeviceFormat(mAudioTrack->format());
setSampleRate(mAudioTrack->getSampleRate());
diff --git a/media/libaaudio/src/libaaudio.map.txt b/media/libaaudio/src/libaaudio.map.txt
index 1dd44d1..8fa9e38 100644
--- a/media/libaaudio/src/libaaudio.map.txt
+++ b/media/libaaudio/src/libaaudio.map.txt
@@ -25,6 +25,7 @@
AAudioStreamBuilder_setPrivacySensitive; # introduced=30
AAudioStreamBuilder_setPackageName; # introduced=31
AAudioStreamBuilder_setAttributionTag; # introduced=31
+ AAudioStreamBuilder_setChannelMask; # introduced=32
AAudioStreamBuilder_openStream;
AAudioStreamBuilder_delete;
AAudioStream_close;
@@ -61,6 +62,7 @@
AAudioStream_isMMapUsed;
AAudioStream_isPrivacySensitive; # introduced=30
AAudioStream_release; # introduced=30
+ AAudioStream_getChannelMask; # introduced=32
local:
*;
};
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index d795725..d829934 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -256,6 +256,248 @@
return privacySensitive ? AUDIO_FLAG_CAPTURE_PRIVATE : AUDIO_FLAG_NONE;
}
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelLayoutMask(
+ aaudio_channel_mask_t channelMask, bool isInput) {
+ if (isInput) {
+ switch (channelMask) {
+ case AAUDIO_CHANNEL_MONO:
+ return AUDIO_CHANNEL_IN_MONO;
+ case AAUDIO_CHANNEL_STEREO:
+ return AUDIO_CHANNEL_IN_STEREO;
+ case AAUDIO_CHANNEL_FRONT_BACK:
+ return AUDIO_CHANNEL_IN_FRONT_BACK;
+ case AAUDIO_CHANNEL_2POINT0POINT2:
+ return AUDIO_CHANNEL_IN_2POINT0POINT2;
+ case AAUDIO_CHANNEL_2POINT1POINT2:
+ return AUDIO_CHANNEL_IN_2POINT1POINT2;
+ case AAUDIO_CHANNEL_3POINT0POINT2:
+ return AUDIO_CHANNEL_IN_3POINT0POINT2;
+ case AAUDIO_CHANNEL_3POINT1POINT2:
+ return AUDIO_CHANNEL_IN_3POINT1POINT2;
+ case AAUDIO_CHANNEL_5POINT1:
+ return AUDIO_CHANNEL_IN_5POINT1;
+ default:
+ ALOGE("%s() %#x unrecognized", __func__, channelMask);
+ return AUDIO_CHANNEL_INVALID;
+ }
+ } else {
+ switch (channelMask) {
+ case AAUDIO_CHANNEL_MONO:
+ return AUDIO_CHANNEL_OUT_MONO;
+ case AAUDIO_CHANNEL_STEREO:
+ return AUDIO_CHANNEL_OUT_STEREO;
+ case AAUDIO_CHANNEL_2POINT1:
+ return AUDIO_CHANNEL_OUT_2POINT1;
+ case AAUDIO_CHANNEL_TRI:
+ return AUDIO_CHANNEL_OUT_TRI;
+ case AAUDIO_CHANNEL_TRI_BACK:
+ return AUDIO_CHANNEL_OUT_TRI_BACK;
+ case AAUDIO_CHANNEL_3POINT1:
+ return AUDIO_CHANNEL_OUT_3POINT1;
+ case AAUDIO_CHANNEL_2POINT0POINT2:
+ return AUDIO_CHANNEL_OUT_2POINT0POINT2;
+ case AAUDIO_CHANNEL_2POINT1POINT2:
+ return AUDIO_CHANNEL_OUT_2POINT1POINT2;
+ case AAUDIO_CHANNEL_3POINT0POINT2:
+ return AUDIO_CHANNEL_OUT_3POINT0POINT2;
+ case AAUDIO_CHANNEL_3POINT1POINT2:
+ return AUDIO_CHANNEL_OUT_3POINT1POINT2;
+ case AAUDIO_CHANNEL_QUAD:
+ return AUDIO_CHANNEL_OUT_QUAD;
+ case AAUDIO_CHANNEL_QUAD_SIDE:
+ return AUDIO_CHANNEL_OUT_QUAD_SIDE;
+ case AAUDIO_CHANNEL_SURROUND:
+ return AUDIO_CHANNEL_OUT_SURROUND;
+ case AAUDIO_CHANNEL_PENTA:
+ return AUDIO_CHANNEL_OUT_PENTA;
+ case AAUDIO_CHANNEL_5POINT1:
+ return AUDIO_CHANNEL_OUT_5POINT1;
+ case AAUDIO_CHANNEL_5POINT1_SIDE:
+ return AUDIO_CHANNEL_OUT_5POINT1_SIDE;
+ case AAUDIO_CHANNEL_5POINT1POINT2:
+ return AUDIO_CHANNEL_OUT_5POINT1POINT2;
+ case AAUDIO_CHANNEL_5POINT1POINT4:
+ return AUDIO_CHANNEL_OUT_5POINT1POINT4;
+ case AAUDIO_CHANNEL_6POINT1:
+ return AUDIO_CHANNEL_OUT_6POINT1;
+ case AAUDIO_CHANNEL_7POINT1:
+ return AUDIO_CHANNEL_OUT_7POINT1;
+ case AAUDIO_CHANNEL_7POINT1POINT2:
+ return AUDIO_CHANNEL_OUT_7POINT1POINT2;
+ case AAUDIO_CHANNEL_7POINT1POINT4:
+ return AUDIO_CHANNEL_OUT_7POINT1POINT4;
+ // TODO: add 9point1point4 and 9point1point6 when they are added in audio-hal-enums.h
+ // case AAUDIO_CHANNEL_9POINT1POINT4:
+ // return AUDIO_CHANNEL_OUT_9POINT1POINT4;
+ // case AAUDIO_CHANNEL_9POINT1POINT6:
+ // return AUDIO_CHANNEL_OUT_9POINT1POINT6;
+ default:
+ ALOGE("%s() %#x unrecognized", __func__, channelMask);
+ return AUDIO_CHANNEL_INVALID;
+ }
+ }
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelLayoutMask(
+ audio_channel_mask_t channelMask, bool isInput) {
+ if (isInput) {
+ switch (channelMask) {
+ case AUDIO_CHANNEL_IN_MONO:
+ return AAUDIO_CHANNEL_MONO;
+ case AUDIO_CHANNEL_IN_STEREO:
+ return AAUDIO_CHANNEL_STEREO;
+ case AUDIO_CHANNEL_IN_FRONT_BACK:
+ return AAUDIO_CHANNEL_FRONT_BACK;
+ case AUDIO_CHANNEL_IN_2POINT0POINT2:
+ return AAUDIO_CHANNEL_2POINT0POINT2;
+ case AUDIO_CHANNEL_IN_2POINT1POINT2:
+ return AAUDIO_CHANNEL_2POINT1POINT2;
+ case AUDIO_CHANNEL_IN_3POINT0POINT2:
+ return AAUDIO_CHANNEL_3POINT0POINT2;
+ case AUDIO_CHANNEL_IN_3POINT1POINT2:
+ return AAUDIO_CHANNEL_3POINT1POINT2;
+ case AUDIO_CHANNEL_IN_5POINT1:
+ return AAUDIO_CHANNEL_5POINT1;
+ default:
+ ALOGE("%s() %#x unrecognized", __func__, channelMask);
+ return AAUDIO_CHANNEL_INVALID;
+ }
+ } else {
+ switch (channelMask) {
+ case AUDIO_CHANNEL_OUT_MONO:
+ return AAUDIO_CHANNEL_MONO;
+ case AUDIO_CHANNEL_OUT_STEREO:
+ return AAUDIO_CHANNEL_STEREO;
+ case AUDIO_CHANNEL_OUT_2POINT1:
+ return AAUDIO_CHANNEL_2POINT1;
+ case AUDIO_CHANNEL_OUT_TRI:
+ return AAUDIO_CHANNEL_TRI;
+ case AUDIO_CHANNEL_OUT_TRI_BACK:
+ return AAUDIO_CHANNEL_TRI_BACK;
+ case AUDIO_CHANNEL_OUT_3POINT1:
+ return AAUDIO_CHANNEL_3POINT1;
+ case AUDIO_CHANNEL_OUT_2POINT0POINT2:
+ return AAUDIO_CHANNEL_2POINT0POINT2;
+ case AUDIO_CHANNEL_OUT_2POINT1POINT2:
+ return AAUDIO_CHANNEL_2POINT1POINT2;
+ case AUDIO_CHANNEL_OUT_3POINT0POINT2:
+ return AAUDIO_CHANNEL_3POINT0POINT2;
+ case AUDIO_CHANNEL_OUT_3POINT1POINT2:
+ return AAUDIO_CHANNEL_3POINT1POINT2;
+ case AUDIO_CHANNEL_OUT_QUAD:
+ return AAUDIO_CHANNEL_QUAD;
+ case AUDIO_CHANNEL_OUT_QUAD_SIDE:
+ return AAUDIO_CHANNEL_QUAD_SIDE;
+ case AUDIO_CHANNEL_OUT_SURROUND:
+ return AAUDIO_CHANNEL_SURROUND;
+ case AUDIO_CHANNEL_OUT_PENTA:
+ return AAUDIO_CHANNEL_PENTA;
+ case AUDIO_CHANNEL_OUT_5POINT1:
+ return AAUDIO_CHANNEL_5POINT1;
+ case AUDIO_CHANNEL_OUT_5POINT1_SIDE:
+ return AAUDIO_CHANNEL_5POINT1_SIDE;
+ case AUDIO_CHANNEL_OUT_5POINT1POINT2:
+ return AAUDIO_CHANNEL_5POINT1POINT2;
+ case AUDIO_CHANNEL_OUT_5POINT1POINT4:
+ return AAUDIO_CHANNEL_5POINT1POINT4;
+ case AUDIO_CHANNEL_OUT_6POINT1:
+ return AAUDIO_CHANNEL_6POINT1;
+ case AUDIO_CHANNEL_OUT_7POINT1:
+ return AAUDIO_CHANNEL_7POINT1;
+ case AUDIO_CHANNEL_OUT_7POINT1POINT2:
+ return AAUDIO_CHANNEL_7POINT1POINT2;
+ case AUDIO_CHANNEL_OUT_7POINT1POINT4:
+ return AAUDIO_CHANNEL_7POINT1POINT4;
+ // TODO: add 9point1point4 and 9point1point6 when they are added in audio-hal-enums.h
+ // case AUDIO_CHANNEL_OUT_9POINT1POINT4:
+ // return AAUDIO_CHANNEL_9POINT1POINT4;
+ // case AUDIO_CHANNEL_OUT_9POINT1POINT6:
+ // return AAUDIO_CHANNEL_9POINT1POINT6;
+ default:
+ ALOGE("%s() %#x unrecognized", __func__, channelMask);
+ return AAUDIO_CHANNEL_INVALID;
+ }
+ }
+}
+
+int32_t AAudioConvert_channelMaskToCount(aaudio_channel_mask_t channelMask) {
+ return __builtin_popcount(channelMask & ~AAUDIO_CHANNEL_BIT_INDEX);
+}
+
+aaudio_channel_mask_t AAudioConvert_channelCountToMask(int32_t channelCount) {
+ if (channelCount < 0 || channelCount > AUDIO_CHANNEL_COUNT_MAX) {
+ return AAUDIO_CHANNEL_INVALID;
+ }
+
+ if (channelCount == 0) {
+ return AAUDIO_UNSPECIFIED;
+ }
+
+ // Return index mask if the channel count is greater than 2.
+ return AAUDIO_CHANNEL_BIT_INDEX | ((1 << channelCount) - 1);
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelIndexMask(
+ audio_channel_mask_t channelMask) {
+ if (audio_channel_mask_get_representation(channelMask) != AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+ ALOGE("%s() %#x not an index mask", __func__, channelMask);
+ return AAUDIO_CHANNEL_INVALID;
+ }
+ return (channelMask & ~AUDIO_CHANNEL_INDEX_HDR) | AAUDIO_CHANNEL_BIT_INDEX;
+}
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelIndexMask(
+ aaudio_channel_mask_t channelMask) {
+ if (!AAudio_isChannelIndexMask(channelMask)) {
+ ALOGE("%s() %#x not an index mask", __func__, channelMask);
+ return AUDIO_CHANNEL_INVALID;
+ }
+ return audio_channel_mask_for_index_assignment_from_count(
+ AAudioConvert_channelMaskToCount(channelMask));
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelMask(
+ audio_channel_mask_t channelMask, bool isInput, bool indexMaskRequired) {
+ if (audio_channel_mask_get_representation(channelMask) == AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+ return AAudioConvert_androidToAAudioChannelIndexMask(channelMask);
+ }
+ if (indexMaskRequired) {
+ // Require index mask, `channelMask` here is a position mask.
+ const int channelCount = isInput ? audio_channel_count_from_in_mask(channelMask)
+ : audio_channel_count_from_out_mask(channelMask);
+ return AAudioConvert_channelCountToMask(channelCount);
+ }
+ return AAudioConvert_androidToAAudioChannelLayoutMask(channelMask, isInput);
+}
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelMask(
+ aaudio_channel_mask_t channelMask, bool isInput) {
+ return AAudio_isChannelIndexMask(channelMask)
+ ? AAudioConvert_aaudioToAndroidChannelIndexMask(channelMask)
+ : AAudioConvert_aaudioToAndroidChannelLayoutMask(channelMask, isInput);
+}
+
+bool AAudio_isChannelIndexMask(aaudio_channel_mask_t channelMask) {
+ return (channelMask & AAUDIO_CHANNEL_BIT_INDEX) == AAUDIO_CHANNEL_BIT_INDEX;
+}
+
+audio_channel_mask_t AAudio_getChannelMaskForOpen(
+ aaudio_channel_mask_t channelMask, int32_t samplesPerFrame, bool isInput) {
+ if (channelMask != AAUDIO_UNSPECIFIED) {
+ if (AAudio_isChannelIndexMask(channelMask) && samplesPerFrame <= 2) {
+ // When it is index mask and the count is less than 3, use position mask
+ // instead of index mask for opening a stream. This may need to be revisited
+ // when making channel index mask public.
+ return isInput ? audio_channel_in_mask_from_count(samplesPerFrame)
+ : audio_channel_out_mask_from_count(samplesPerFrame);
+ }
+ return AAudioConvert_aaudioToAndroidChannelMask(channelMask, isInput);
+ }
+
+ // Return stereo when unspecified.
+ return isInput ? AUDIO_CHANNEL_IN_STEREO : AUDIO_CHANNEL_OUT_STEREO;
+}
+
int32_t AAudioConvert_framesToBytes(int32_t numFrames,
int32_t bytesPerFrame,
int32_t *sizeInBytes) {
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index 82eb77d..5eda30c 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -96,6 +96,33 @@
audio_flags_mask_t AAudioConvert_privacySensitiveToAudioFlagsMask(
bool privacySensitive);
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelLayoutMask(
+ aaudio_channel_mask_t channelMask, bool isInput);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelLayoutMask(
+ audio_channel_mask_t channelMask, bool isInput);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelIndexMask(
+ audio_channel_mask_t channelMask);
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelIndexMask(
+ aaudio_channel_mask_t channelMask);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelMask(
+ audio_channel_mask_t channelMask, bool isInput, bool indexMaskRequired);
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelMask(
+ aaudio_channel_mask_t channelMask, bool isInput);
+
+bool AAudio_isChannelIndexMask(aaudio_channel_mask_t channelMask);
+
+int32_t AAudioConvert_channelMaskToCount(aaudio_channel_mask_t channelMask);
+
+aaudio_channel_mask_t AAudioConvert_channelCountToMask(int32_t channelCount);
+
+audio_channel_mask_t AAudio_getChannelMaskForOpen(
+ aaudio_channel_mask_t channelMask, int32_t samplesPerFrame, bool isInput);
+
// Note that this code may be replaced by Settings or by some other system configuration tool.
/**
@@ -318,4 +345,36 @@
std::atomic<int> mRequested{0};
std::atomic<int> mAcknowledged{0};
};
+
+enum {
+ /**
+ * Audio channel index mask, only used internally.
+ */
+ AAUDIO_CHANNEL_BIT_INDEX = 0x80000000,
+ AAUDIO_CHANNEL_INDEX_MASK_1 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 1) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_2 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 2) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_3 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 3) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_4 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 4) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_5 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 5) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_6 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 6) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_7 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 7) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_8 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 8) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_9 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 9) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_10 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 10) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_11 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 11) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_12 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 12) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_13 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 13) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_14 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 14) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_15 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 15) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_16 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 16) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_17 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 17) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_18 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 18) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_19 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 19) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_20 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 20) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_21 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 21) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_22 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 22) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_23 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 23) - 1,
+ AAUDIO_CHANNEL_INDEX_MASK_24 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 24) - 1,
+};
+
#endif //UTILITY_AAUDIO_UTILITIES_H
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 88e752b..5344756 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -480,6 +480,12 @@
return af->systemReady();
}
+status_t AudioSystem::audioPolicyReady() {
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) return NO_INIT;
+ return af->audioPolicyReady();
+}
+
status_t AudioSystem::getFrameCountHAL(audio_io_handle_t ioHandle,
size_t* frameCount) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index cae81f0..2af1c50 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -715,6 +715,10 @@
return statusTFromBinderStatus(mDelegate->systemReady());
}
+status_t AudioFlingerClientAdapter::audioPolicyReady() {
+ return statusTFromBinderStatus(mDelegate->audioPolicyReady());
+}
+
size_t AudioFlingerClientAdapter::frameCountHAL(audio_io_handle_t ioHandle) const {
auto result = [&]() -> ConversionResult<size_t> {
int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
@@ -1189,6 +1193,11 @@
return Status::fromStatusT(mDelegate->systemReady());
}
+Status AudioFlingerServerAdapter::audioPolicyReady() {
+ mDelegate->audioPolicyReady();
+ return Status::ok();
+}
+
Status AudioFlingerServerAdapter::frameCountHAL(int32_t ioHandle, int64_t* _aidl_return) {
audio_io_handle_t ioHandleLegacy = VALUE_OR_RETURN_BINDER(
aidl2legacy_int32_t_audio_io_handle_t(ioHandle));
diff --git a/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl b/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
index f88fc3c..8538d8a 100644
--- a/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
@@ -24,4 +24,5 @@
int id;
float resonantFrequency;
float qFactor;
+ float maxAmplitude;
}
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index d2cae6d..7ffcc33 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -197,6 +197,9 @@
/* Indicate JAVA services are ready (scheduling, power management ...) */
oneway void systemReady();
+ /* Indicate audio policy service is ready */
+ oneway void audioPolicyReady();
+
// Returns the number of frames per audio HAL buffer.
long frameCountHAL(int /* audio_io_handle_t */ ioHandle);
diff --git a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
index 06b12e9..1541948 100644
--- a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
@@ -17,6 +17,7 @@
package android.media;
import android.media.AudioConfig;
+import android.media.AudioConfigBase;
import android.media.AudioPort;
/**
@@ -25,7 +26,8 @@
parcelable OpenOutputRequest {
/** Interpreted as audio_module_handle_t. */
int module;
- AudioConfig config;
+ AudioConfig halConfig;
+ AudioConfigBase mixerConfig;
/** Type must be DEVICE. */
AudioPort device;
/** Bitmask, indexed by AudioOutputFlag. */
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index d03c6fa..12473fc 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -648,11 +648,15 @@
sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(getValue(&mFdp, kDevices));
audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
+ audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+
media::OpenOutputRequest request{};
media::OpenOutputResponse response{};
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
- request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ request.halConfig = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ request.mixerConfig =
+ VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_base_t_AudioConfigBase(mixerConfig));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
diff --git a/media/libaudioclient/include/media/AudioCommonTypes.h b/media/libaudioclient/include/media/AudioCommonTypes.h
index 5dfe5fc..5f0c590 100644
--- a/media/libaudioclient/include/media/AudioCommonTypes.h
+++ b/media/libaudioclient/include/media/AudioCommonTypes.h
@@ -41,6 +41,42 @@
return !(lhs==rhs);
}
+constexpr bool operator==(const audio_offload_info_t &lhs, const audio_offload_info_t &rhs)
+{
+ return lhs.version == rhs.version && lhs.size == rhs.size &&
+ lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+ lhs.format == rhs.format && lhs.stream_type == rhs.stream_type &&
+ lhs.bit_rate == rhs.bit_rate && lhs.duration_us == rhs.duration_us &&
+ lhs.has_video == rhs.has_video && lhs.is_streaming == rhs.is_streaming &&
+ lhs.bit_width == rhs.bit_width && lhs.offload_buffer_size == rhs.offload_buffer_size &&
+ lhs.usage == rhs.usage && lhs.encapsulation_mode == rhs.encapsulation_mode &&
+ lhs.content_id == rhs.content_id && lhs.sync_id == rhs.sync_id;
+}
+constexpr bool operator!=(const audio_offload_info_t &lhs, const audio_offload_info_t &rhs)
+{
+ return !(lhs==rhs);
+}
+
+constexpr bool operator==(const audio_config_t &lhs, const audio_config_t &rhs)
+{
+ return lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+ lhs.format == rhs.format && lhs.offload_info == rhs.offload_info;
+}
+constexpr bool operator!=(const audio_config_t &lhs, const audio_config_t &rhs)
+{
+ return !(lhs==rhs);
+}
+
+constexpr bool operator==(const audio_config_base_t &lhs, const audio_config_base_t &rhs)
+{
+ return lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+ lhs.format == rhs.format;
+}
+constexpr bool operator!=(const audio_config_base_t &lhs, const audio_config_base_t &rhs)
+{
+ return !(lhs==rhs);
+}
+
enum volume_group_t : uint32_t;
static const volume_group_t VOLUME_GROUP_NONE = static_cast<volume_group_t>(-1);
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 326919a..f17ee3a 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -264,6 +264,7 @@
size_t frameCount() const { return mFrameCount; }
size_t frameSize() const { return mFrameSize; }
audio_source_t inputSource() const { return mAttributes.source; }
+ audio_channel_mask_t channelMask() const { return mChannelMask; }
/*
* Return the period of the notification callback in frames.
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index a9109c8..626dcbf 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -225,6 +225,9 @@
// Indicate JAVA services are ready (scheduling, power management ...)
static status_t systemReady();
+ // Indicate audio policy service is ready
+ static status_t audioPolicyReady();
+
// Returns the number of frames per audio HAL buffer.
// Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input.
// See also getFrameCount().
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index cb00990..5fea637 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -401,6 +401,7 @@
uint32_t channelCount() const { return mChannelCount; }
size_t frameCount() const { return mFrameCount; }
+ audio_channel_mask_t channelMask() const { return mChannelMask; }
/*
* Return the period of the notification callback in frames.
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 0e059f7..9e5019e 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -329,6 +329,9 @@
/* Indicate JAVA services are ready (scheduling, power management ...) */
virtual status_t systemReady() = 0;
+ // Indicate audio policy service is ready
+ virtual status_t audioPolicyReady() = 0;
+
// Returns the number of frames per audio HAL buffer.
virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const = 0;
@@ -432,6 +435,8 @@
status_t setAudioPortConfig(const struct audio_port_config* config) override;
audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) override;
status_t systemReady() override;
+ status_t audioPolicyReady() override;
+
size_t frameCountHAL(audio_io_handle_t ioHandle) const override;
status_t getMicrophones(std::vector<media::MicrophoneInfo>* microphones) override;
status_t setAudioHalPids(const std::vector<pid_t>& pids) override;
@@ -514,6 +519,7 @@
SET_AUDIO_PORT_CONFIG = media::BnAudioFlingerService::TRANSACTION_setAudioPortConfig,
GET_AUDIO_HW_SYNC_FOR_SESSION = media::BnAudioFlingerService::TRANSACTION_getAudioHwSyncForSession,
SYSTEM_READY = media::BnAudioFlingerService::TRANSACTION_systemReady,
+ AUDIO_POLICY_READY = media::BnAudioFlingerService::TRANSACTION_audioPolicyReady,
FRAME_COUNT_HAL = media::BnAudioFlingerService::TRANSACTION_frameCountHAL,
GET_MICROPHONES = media::BnAudioFlingerService::TRANSACTION_getMicrophones,
SET_MASTER_BALANCE = media::BnAudioFlingerService::TRANSACTION_setMasterBalance,
@@ -624,6 +630,7 @@
Status setAudioPortConfig(const media::AudioPortConfig& config) override;
Status getAudioHwSyncForSession(int32_t sessionId, int32_t* _aidl_return) override;
Status systemReady() override;
+ Status audioPolicyReady() override;
Status frameCountHAL(int32_t ioHandle, int64_t* _aidl_return) override;
Status getMicrophones(std::vector<media::MicrophoneInfoData>* _aidl_return) override;
Status setAudioHalPids(const std::vector<int32_t>& pids) override;
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.cpp b/media/libaudiohal/impl/ConversionHelperHidl.cpp
index 32eaa31..0503698 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.cpp
+++ b/media/libaudiohal/impl/ConversionHelperHidl.cpp
@@ -105,6 +105,15 @@
}
// static
+void ConversionHelperHidl::argsFromHal(
+ const Vector<String16>& args, hidl_vec<hidl_string> *hidlArgs) {
+ hidlArgs->resize(args.size());
+ for (size_t i = 0; i < args.size(); ++i) {
+ (*hidlArgs)[i] = String8(args[i]).c_str();
+ }
+}
+
+// static
status_t ConversionHelperHidl::analyzeResult(const Result& result) {
switch (result) {
case Result::OK: return OK;
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.h b/media/libaudiohal/impl/ConversionHelperHidl.h
index 59122c7..2909013 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.h
+++ b/media/libaudiohal/impl/ConversionHelperHidl.h
@@ -21,6 +21,8 @@
#include <hidl/HidlSupport.h>
#include <system/audio.h>
#include <utils/String8.h>
+#include <utils/String16.h>
+#include <utils/Vector.h>
using ::android::hardware::audio::CPP_VERSION::ParameterValue;
using CoreResult = ::android::hardware::audio::CPP_VERSION::Result;
@@ -37,6 +39,7 @@
static status_t keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys);
static status_t parametersFromHal(const String8& kvPairs, hidl_vec<ParameterValue> *hidlParams);
static void parametersToHal(const hidl_vec<ParameterValue>& parameters, String8 *values);
+ static void argsFromHal(const Vector<String16>& args, hidl_vec<hidl_string> *hidlArgs);
ConversionHelperHidl(const char* className);
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index ca4f663..f86df1e 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -457,11 +457,13 @@
}
#endif
-status_t DeviceHalHidl::dump(int fd) {
+status_t DeviceHalHidl::dump(int fd, const Vector<String16>& args) {
if (mDevice == 0) return NO_INIT;
native_handle_t* hidlHandle = native_handle_create(1, 0);
hidlHandle->data[0] = fd;
- Return<void> ret = mDevice->debug(hidlHandle, {} /* options */);
+ hidl_vec<hidl_string> hidlArgs;
+ argsFromHal(args, &hidlArgs);
+ Return<void> ret = mDevice->debug(hidlHandle, hidlArgs);
native_handle_delete(hidlHandle);
return processReturn("dump", ret);
}
diff --git a/media/libaudiohal/impl/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
index 2c847cf..2694ab3 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -119,7 +119,7 @@
status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
- virtual status_t dump(int fd);
+ status_t dump(int fd, const Vector<String16>& args) override;
private:
friend class DevicesFactoryHalHidl;
diff --git a/media/libaudiohal/impl/DeviceHalLocal.cpp b/media/libaudiohal/impl/DeviceHalLocal.cpp
index af7dc1a..e0304af 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.cpp
+++ b/media/libaudiohal/impl/DeviceHalLocal.cpp
@@ -233,7 +233,7 @@
return INVALID_OPERATION;
}
-status_t DeviceHalLocal::dump(int fd) {
+status_t DeviceHalLocal::dump(int fd, const Vector<String16>& /* args */) {
return mDev->dump(mDev, fd);
}
diff --git a/media/libaudiohal/impl/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
index 46b510b..2fde936 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -112,7 +112,7 @@
status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
- virtual status_t dump(int fd);
+ status_t dump(int fd, const Vector<String16>& args) override;
void closeOutputStream(struct audio_stream_out *stream_out);
void closeInputStream(struct audio_stream_in *stream_in);
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 9c4363c..f75bbf3 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -73,7 +73,9 @@
uint32_t index, effect_descriptor_t *pDescriptor) {
// TODO: We need somehow to track the changes on the server side
// or figure out how to convert everybody to query all the descriptors at once.
- // TODO: check for nullptr
+ if (pDescriptor == nullptr) {
+ return BAD_VALUE;
+ }
if (mLastDescriptors.size() == 0) {
status_t queryResult = queryAllDescriptors();
if (queryResult != OK) return queryResult;
@@ -85,7 +87,9 @@
status_t EffectsFactoryHalHidl::getDescriptor(
const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
- // TODO: check for nullptr
+ if (pDescriptor == nullptr || pEffectUuid == nullptr) {
+ return BAD_VALUE;
+ }
if (mEffectsFactory == 0) return NO_INIT;
Uuid hidlUuid;
UuidUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
@@ -105,6 +109,33 @@
return processReturn(__FUNCTION__, ret);
}
+status_t EffectsFactoryHalHidl::getDescriptors(const effect_uuid_t *pEffectType,
+ std::vector<effect_descriptor_t> *descriptors) {
+ if (pEffectType == nullptr || descriptors == nullptr) {
+ return BAD_VALUE;
+ }
+
+ uint32_t numEffects = 0;
+ status_t status = queryNumberEffects(&numEffects);
+ if (status != NO_ERROR) {
+ ALOGW("%s error %d from FactoryHal queryNumberEffects", __func__, status);
+ return status;
+ }
+
+ for (uint32_t i = 0; i < numEffects; i++) {
+ effect_descriptor_t descriptor;
+ status = getDescriptor(i, &descriptor);
+ if (status != NO_ERROR) {
+ ALOGW("%s error %d from FactoryHal getDescriptor", __func__, status);
+ continue;
+ }
+ if (memcmp(&descriptor.type, pEffectType, sizeof(effect_uuid_t)) == 0) {
+ descriptors->push_back(descriptor);
+ }
+ }
+ return descriptors->empty() ? NAME_NOT_FOUND : NO_ERROR;
+}
+
status_t EffectsFactoryHalHidl::createEffect(
const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
int32_t deviceId __unused, sp<EffectHalInterface> *effect) {
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 5fa85e7..ff26d9f 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -45,6 +45,9 @@
virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
effect_descriptor_t *pDescriptor);
+ virtual status_t getDescriptors(const effect_uuid_t *pEffectType,
+ std::vector<effect_descriptor_t> *descriptors);
+
// Creates an effect engine of the specified type.
// To release the effect engine, it is necessary to release references
// to the returned effect object.
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index 539a149..452f84d 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -152,11 +152,13 @@
return processReturn("standby", mStream->standby());
}
-status_t StreamHalHidl::dump(int fd) {
+status_t StreamHalHidl::dump(int fd, const Vector<String16>& args) {
if (!mStream) return NO_INIT;
native_handle_t* hidlHandle = native_handle_create(1, 0);
hidlHandle->data[0] = fd;
- Return<void> ret = mStream->debug(hidlHandle, {} /* options */);
+ hidl_vec<hidl_string> hidlArgs;
+ argsFromHal(args, &hidlArgs);
+ Return<void> ret = mStream->debug(hidlHandle, hidlArgs);
native_handle_delete(hidlHandle);
mStreamPowerLog.dump(fd);
return processReturn("dump", ret);
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index 970903b..6f5dd04 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -71,7 +71,7 @@
// Put the audio hardware input/output into standby mode.
virtual status_t standby();
- virtual status_t dump(int fd);
+ virtual status_t dump(int fd, const Vector<String16>& args) override;
// Start a stream operating in mmap mode.
virtual status_t start();
diff --git a/media/libaudiohal/impl/StreamHalLocal.cpp b/media/libaudiohal/impl/StreamHalLocal.cpp
index 34bd5df..11fac61 100644
--- a/media/libaudiohal/impl/StreamHalLocal.cpp
+++ b/media/libaudiohal/impl/StreamHalLocal.cpp
@@ -87,7 +87,8 @@
return mStream->standby(mStream);
}
-status_t StreamHalLocal::dump(int fd) {
+status_t StreamHalLocal::dump(int fd, const Vector<String16>& args) {
+ (void) args;
status_t status = mStream->dump(mStream, fd);
mStreamPowerLog.dump(fd);
return status;
diff --git a/media/libaudiohal/impl/StreamHalLocal.h b/media/libaudiohal/impl/StreamHalLocal.h
index b260495..493c521 100644
--- a/media/libaudiohal/impl/StreamHalLocal.h
+++ b/media/libaudiohal/impl/StreamHalLocal.h
@@ -50,7 +50,7 @@
// Put the audio hardware input/output into standby mode.
virtual status_t standby();
- virtual status_t dump(int fd);
+ virtual status_t dump(int fd, const Vector<String16>& args) override;
// Start a stream operating in mmap mode.
virtual status_t start() = 0;
diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index 29ef011..69cbcec 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
@@ -120,7 +120,7 @@
virtual status_t removeDeviceEffect(
audio_port_handle_t device, sp<EffectHalInterface> effect) = 0;
- virtual status_t dump(int fd) = 0;
+ virtual status_t dump(int fd, const Vector<String16>& args) = 0;
protected:
// Subclasses can not be constructed directly by clients.
diff --git a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
index 9fb56ae..3e505bd 100644
--- a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
@@ -37,6 +37,9 @@
virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
effect_descriptor_t *pDescriptor) = 0;
+ virtual status_t getDescriptors(const effect_uuid_t *pEffectType,
+ std::vector<effect_descriptor_t> *descriptors) = 0;
+
// Creates an effect engine of the specified type.
// To release the effect engine, it is necessary to release references
// to the returned effect object.
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index 2be12fb..2b5b2db 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -25,6 +25,7 @@
#include <utils/Errors.h>
#include <utils/RefBase.h>
#include <utils/String8.h>
+#include <utils/Vector.h>
namespace android {
@@ -69,7 +70,7 @@
// Put the audio hardware input/output into standby mode.
virtual status_t standby() = 0;
- virtual status_t dump(int fd) = 0;
+ virtual status_t dump(int fd, const Vector<String16>& args = {}) = 0;
// Start a stream operating in mmap mode.
virtual status_t start() = 0;
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index d85e2e9..e68c002 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -434,6 +434,12 @@
track->mHapticIntensity = hapticIntensity;
}
} break;
+ case HAPTIC_MAX_AMPLITUDE: {
+ const float hapticMaxAmplitude = *reinterpret_cast<float*>(value);
+ if (track->mHapticMaxAmplitude != hapticMaxAmplitude) {
+ track->mHapticMaxAmplitude = hapticMaxAmplitude;
+ }
+ } break;
default:
LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
}
@@ -553,6 +559,7 @@
// haptic
t->mHapticPlaybackEnabled = false;
t->mHapticIntensity = os::HapticScale::NONE;
+ t->mHapticMaxAmplitude = NAN;
t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
t->mMixerHapticChannelCount = 0;
t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
@@ -602,7 +609,8 @@
switch (t->mMixerFormat) {
// Mixer format should be AUDIO_FORMAT_PCM_FLOAT.
case AUDIO_FORMAT_PCM_FLOAT: {
- os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity);
+ os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity,
+ t->mHapticMaxAmplitude);
} break;
default:
LOG_ALWAYS_FATAL("bad mMixerFormat: %#x", t->mMixerFormat);
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index cd47dc6..2988c67 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -17,6 +17,8 @@
#ifndef ANDROID_AUDIO_MIXER_OPS_H
#define ANDROID_AUDIO_MIXER_OPS_H
+#include <audio_utils/channels.h>
+#include <audio_utils/primitives.h>
#include <system/audio.h>
namespace android {
@@ -229,15 +231,26 @@
* complexity of working on interleaved streams is now getting
* too high, and likely limits compiler optimization.
*/
-template <int MIXTYPE, int NCHAN,
+
+// compile-time function.
+constexpr inline bool usesCenterChannel(audio_channel_mask_t mask) {
+ using namespace audio_utils::channels;
+ for (size_t i = 0; i < std::size(kSideFromChannelIdx); ++i) {
+ if ((mask & (1 << i)) != 0 && kSideFromChannelIdx[i] == AUDIO_GEOMETRY_SIDE_CENTER) {
+ return true;
+ }
+ }
+ return false;
+}
+
+/*
+ * Applies stereo volume to the audio data based on proper left right channel affinity
+ * (templated channel MASK parameter).
+ */
+template <int MIXTYPE, audio_channel_mask_t MASK,
typename TO, typename TI, typename TV,
typename F>
-void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
- static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
- static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
- || MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
- || MIXTYPE == MIXTYPE_STEREOEXPAND
- || MIXTYPE == MIXTYPE_MONOEXPAND);
+void stereoVolumeHelperWithChannelMask(TO*& out, const TI*& in, const TV *vol, F f) {
auto proc = [](auto& a, const auto& b) {
if constexpr (MIXTYPE == MIXTYPE_MULTI_STEREOVOL
|| MIXTYPE == MIXTYPE_STEREOEXPAND
@@ -250,59 +263,109 @@
auto inp = [&in]() -> const TI& {
if constexpr (MIXTYPE == MIXTYPE_STEREOEXPAND
|| MIXTYPE == MIXTYPE_MONOEXPAND) {
- return *in;
+ return *in; // note STEREOEXPAND assumes replicated L/R channels (see doc below).
} else {
return *in++;
}
};
- // HALs should only expose the canonical channel masks.
- proc(*out++, f(inp(), vol[0])); // front left
- if constexpr (NCHAN == 1) return;
- proc(*out++, f(inp(), vol[1])); // front right
- if constexpr (NCHAN == 2) return;
- if constexpr (NCHAN == 4) {
- proc(*out++, f(inp(), vol[0])); // back left
- proc(*out++, f(inp(), vol[1])); // back right
- return;
- }
-
- // TODO: Precompute center volume if not ramping.
std::decay_t<TV> center;
- if constexpr (std::is_floating_point_v<TV>) {
- center = (vol[0] + vol[1]) * 0.5; // do not use divide
- } else {
- center = (vol[0] >> 1) + (vol[1] >> 1); // rounds to 0.
- }
- proc(*out++, f(inp(), center)); // center (or 2.1 LFE)
- if constexpr (NCHAN == 3) return;
- if constexpr (NCHAN == 5) {
- proc(*out++, f(inp(), vol[0])); // back left
- proc(*out++, f(inp(), vol[1])); // back right
- return;
- }
-
- proc(*out++, f(inp(), center)); // lfe
- proc(*out++, f(inp(), vol[0])); // back left
- proc(*out++, f(inp(), vol[1])); // back right
- if constexpr (NCHAN == 6) return;
- if constexpr (NCHAN == 7) {
- proc(*out++, f(inp(), center)); // back center
- return;
- }
- // NCHAN == 8
- proc(*out++, f(inp(), vol[0])); // side left
- proc(*out++, f(inp(), vol[1])); // side right
- if constexpr (NCHAN > FCC_8) {
- // Mutes to zero extended surround channels.
- // 7.1.4 has the correct behavior.
- // 22.2 has the behavior that FLC and FRC will be mixed instead
- // of SL and SR and LFE will be center, not left.
- for (int i = 8; i < NCHAN; ++i) {
- // TODO: Consider using android::audio_utils::channels::kSideFromChannelIdx
- proc(*out++, f(inp(), 0.f));
+ constexpr bool USES_CENTER_CHANNEL = usesCenterChannel(MASK);
+ if constexpr (USES_CENTER_CHANNEL) {
+ if constexpr (std::is_floating_point_v<TV>) {
+ center = (vol[0] + vol[1]) * 0.5; // do not use divide
+ } else {
+ center = (vol[0] >> 1) + (vol[1] >> 1); // rounds to 0.
}
}
+
+ using namespace audio_utils::channels;
+
+ // if LFE and LFE2 are both present, they take left and right volume respectively.
+ constexpr unsigned LFE_LFE2 = \
+ AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2;
+ constexpr bool has_LFE_LFE2 = (MASK & LFE_LFE2) == LFE_LFE2;
+
+#pragma push_macro("DO_CHANNEL_POSITION")
+#undef DO_CHANNEL_POSITION
+#define DO_CHANNEL_POSITION(BIT_INDEX) \
+ if constexpr ((MASK & (1 << BIT_INDEX)) != 0) { \
+ constexpr auto side = kSideFromChannelIdx[BIT_INDEX]; \
+ if constexpr (side == AUDIO_GEOMETRY_SIDE_LEFT || \
+ has_LFE_LFE2 && (1 << BIT_INDEX) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) { \
+ proc(*out++, f(inp(), vol[0])); \
+ } else if constexpr (side == AUDIO_GEOMETRY_SIDE_RIGHT || \
+ has_LFE_LFE2 && (1 << BIT_INDEX) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) { \
+ proc(*out++, f(inp(), vol[1])); \
+ } else /* constexpr */ { \
+ proc(*out++, f(inp(), center)); \
+ } \
+ }
+
+ DO_CHANNEL_POSITION(0);
+ DO_CHANNEL_POSITION(1);
+ DO_CHANNEL_POSITION(2);
+ DO_CHANNEL_POSITION(3);
+ DO_CHANNEL_POSITION(4);
+ DO_CHANNEL_POSITION(5);
+ DO_CHANNEL_POSITION(6);
+ DO_CHANNEL_POSITION(7);
+
+ DO_CHANNEL_POSITION(8);
+ DO_CHANNEL_POSITION(9);
+ DO_CHANNEL_POSITION(10);
+ DO_CHANNEL_POSITION(11);
+ DO_CHANNEL_POSITION(12);
+ DO_CHANNEL_POSITION(13);
+ DO_CHANNEL_POSITION(14);
+ DO_CHANNEL_POSITION(15);
+
+ DO_CHANNEL_POSITION(16);
+ DO_CHANNEL_POSITION(17);
+ DO_CHANNEL_POSITION(18);
+ DO_CHANNEL_POSITION(19);
+ DO_CHANNEL_POSITION(20);
+ DO_CHANNEL_POSITION(21);
+ DO_CHANNEL_POSITION(22);
+ DO_CHANNEL_POSITION(23);
+ static_assert(FCC_LIMIT <= FCC_24); // Note: this may need to change.
+#pragma pop_macro("DO_CHANNEL_POSITION")
+}
+
+// These are the channel position masks we expect from the HAL.
+// See audio_channel_out_mask_from_count() but this is constexpr
+constexpr inline audio_channel_mask_t canonicalChannelMaskFromCount(size_t channelCount) {
+ constexpr audio_channel_mask_t canonical[] = {
+ [0] = AUDIO_CHANNEL_NONE,
+ [1] = AUDIO_CHANNEL_OUT_MONO,
+ [2] = AUDIO_CHANNEL_OUT_STEREO,
+ [3] = AUDIO_CHANNEL_OUT_2POINT1,
+ [4] = AUDIO_CHANNEL_OUT_QUAD,
+ [5] = AUDIO_CHANNEL_OUT_PENTA,
+ [6] = AUDIO_CHANNEL_OUT_5POINT1,
+ [7] = AUDIO_CHANNEL_OUT_6POINT1,
+ [8] = AUDIO_CHANNEL_OUT_7POINT1,
+ [12] = AUDIO_CHANNEL_OUT_7POINT1POINT4,
+ [24] = AUDIO_CHANNEL_OUT_22POINT2,
+ };
+ return channelCount < std::size(canonical) ? canonical[channelCount] : AUDIO_CHANNEL_NONE;
+}
+
+template <int MIXTYPE, int NCHAN,
+ typename TO, typename TI, typename TV,
+ typename F>
+void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
+ static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
+ static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
+ || MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
+ || MIXTYPE == MIXTYPE_STEREOEXPAND
+ || MIXTYPE == MIXTYPE_MONOEXPAND);
+ constexpr audio_channel_mask_t MASK{canonicalChannelMaskFromCount(NCHAN)};
+ if constexpr (MASK == AUDIO_CHANNEL_NONE) {
+ ALOGE("%s: Invalid position count %d", __func__, NCHAN);
+ return; // not a valid system mask, ignore.
+ }
+ stereoVolumeHelperWithChannelMask<MIXTYPE, MASK, TO, TI, TV, F>(out, in, vol, f);
}
/*
diff --git a/media/libaudioprocessing/include/media/AudioMixer.h b/media/libaudioprocessing/include/media/AudioMixer.h
index 70eafe3..5a9fa07 100644
--- a/media/libaudioprocessing/include/media/AudioMixer.h
+++ b/media/libaudioprocessing/include/media/AudioMixer.h
@@ -50,6 +50,7 @@
// for haptic
HAPTIC_ENABLED = 0x4007, // Set haptic data from this track should be played or not.
HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
+ HAPTIC_MAX_AMPLITUDE = 0x4009, // Set the max amplitude allowed for haptic data.
// for target TIMESTRETCH
PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name;
// parameter 'value' is a pointer to the new playback rate.
@@ -145,6 +146,7 @@
// Haptic
bool mHapticPlaybackEnabled;
os::HapticScale mHapticIntensity;
+ float mHapticMaxAmplitude;
audio_channel_mask_t mHapticChannelMask;
uint32_t mHapticChannelCount;
audio_channel_mask_t mMixerHapticChannelMask;
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
index 3856817..ad402db 100644
--- a/media/libaudioprocessing/tests/Android.bp
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -76,6 +76,7 @@
//
cc_binary {
name: "mixerops_objdump",
+ header_libs: ["libaudioutils_headers"],
srcs: ["mixerops_objdump.cpp"],
}
@@ -84,6 +85,16 @@
//
cc_benchmark {
name: "mixerops_benchmark",
+ header_libs: ["libaudioutils_headers"],
srcs: ["mixerops_benchmark.cpp"],
static_libs: ["libgoogle-benchmark"],
}
+
+//
+// mixerops unit test
+//
+cc_test {
+ name: "mixerops_tests",
+ defaults: ["libaudioprocessing_test_defaults"],
+ srcs: ["mixerops_tests.cpp"],
+}
diff --git a/media/libaudioprocessing/tests/mixerops_benchmark.cpp b/media/libaudioprocessing/tests/mixerops_benchmark.cpp
index 7a4c5c7..f866b1a 100644
--- a/media/libaudioprocessing/tests/mixerops_benchmark.cpp
+++ b/media/libaudioprocessing/tests/mixerops_benchmark.cpp
@@ -16,11 +16,9 @@
#include <inttypes.h>
#include <type_traits>
-#include "../../../../system/media/audio_utils/include/audio_utils/primitives.h"
#define LOG_ALWAYS_FATAL(...)
#include <../AudioMixerOps.h>
-
#include <benchmark/benchmark.h>
using namespace android;
diff --git a/media/libaudioprocessing/tests/mixerops_tests.cpp b/media/libaudioprocessing/tests/mixerops_tests.cpp
new file mode 100644
index 0000000..2500ba9
--- /dev/null
+++ b/media/libaudioprocessing/tests/mixerops_tests.cpp
@@ -0,0 +1,175 @@
+/*
+ * Copyright (C) 2021 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 "mixerop_tests"
+#include <log/log.h>
+
+#include <inttypes.h>
+#include <type_traits>
+
+#include <../AudioMixerOps.h>
+#include <gtest/gtest.h>
+
+using namespace android;
+
+// Note: gtest templated tests require typenames, not integers.
+template <int MIXTYPE, int NCHAN>
+class MixerOpsBasicTest {
+public:
+ static void testStereoVolume() {
+ using namespace android::audio_utils::channels;
+
+ constexpr size_t FRAME_COUNT = 1000;
+ constexpr size_t SAMPLE_COUNT = FRAME_COUNT * NCHAN;
+
+ const float in[SAMPLE_COUNT] = {[0 ... (SAMPLE_COUNT - 1)] = 1.f};
+
+ AUDIO_GEOMETRY_SIDE sides[NCHAN];
+ size_t i = 0;
+ unsigned channel = canonicalChannelMaskFromCount(NCHAN);
+ constexpr unsigned LFE_LFE2 =
+ AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2;
+ bool has_LFE_LFE2 = (channel & LFE_LFE2) == LFE_LFE2;
+ while (channel != 0) {
+ const int index = __builtin_ctz(channel);
+ if (has_LFE_LFE2 && (1 << index) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) {
+ sides[i++] = AUDIO_GEOMETRY_SIDE_LEFT; // special case
+ } else if (has_LFE_LFE2 && (1 << index) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) {
+ sides[i++] = AUDIO_GEOMETRY_SIDE_RIGHT; // special case
+ } else {
+ sides[i++] = sideFromChannelIdx(index);
+ }
+ channel &= ~(1 << index);
+ }
+
+ float vola[2] = {1.f, 0.f}; // left volume at max.
+ float out[SAMPLE_COUNT]{};
+ float aux[FRAME_COUNT]{};
+ float volaux = 0.5;
+ {
+ volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, vola, volaux);
+ const float *outp = out;
+ const float *auxp = aux;
+ const float left = vola[0];
+ const float center = (vola[0] + vola[1]) * 0.5;
+ const float right = vola[1];
+ for (size_t i = 0; i < FRAME_COUNT; ++i) {
+ for (size_t j = 0; j < NCHAN; ++j) {
+ const float audio = *outp++;
+ if (sides[j] == AUDIO_GEOMETRY_SIDE_LEFT) {
+ EXPECT_EQ(left, audio);
+ } else if (sides[j] == AUDIO_GEOMETRY_SIDE_CENTER) {
+ EXPECT_EQ(center, audio);
+ } else {
+ EXPECT_EQ(right, audio);
+ }
+ }
+ EXPECT_EQ(volaux, *auxp++); // works if all channels contain 1.f
+ }
+ }
+ float volb[2] = {0.f, 0.5f}; // right volume at half max.
+ {
+ // this accumulates into out, aux.
+ // float out[SAMPLE_COUNT]{};
+ // float aux[FRAME_COUNT]{};
+ volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, volb, volaux);
+ const float *outp = out;
+ const float *auxp = aux;
+ const float left = vola[0] + volb[0];
+ const float center = (vola[0] + vola[1] + volb[0] + volb[1]) * 0.5;
+ const float right = vola[1] + volb[1];
+ for (size_t i = 0; i < FRAME_COUNT; ++i) {
+ for (size_t j = 0; j < NCHAN; ++j) {
+ const float audio = *outp++;
+ if (sides[j] == AUDIO_GEOMETRY_SIDE_LEFT) {
+ EXPECT_EQ(left, audio);
+ } else if (sides[j] == AUDIO_GEOMETRY_SIDE_CENTER) {
+ EXPECT_EQ(center, audio);
+ } else {
+ EXPECT_EQ(right, audio);
+ }
+ }
+ // aux is accumulated so 2x the amplitude
+ EXPECT_EQ(volaux * 2.f, *auxp++); // works if all channels contain 1.f
+ }
+ }
+
+ { // test aux as derived from out.
+ // AUX channel is the weighted sum of all of the output channels prior to volume
+ // adjustment. We must set L and R to the same volume to allow computation
+ // of AUX from the output values.
+ const float volmono = 0.25f;
+ const float vollr[2] = {volmono, volmono}; // all the same.
+ float out[SAMPLE_COUNT]{};
+ float aux[FRAME_COUNT]{};
+ volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, vollr, volaux);
+ const float *outp = out;
+ const float *auxp = aux;
+ for (size_t i = 0; i < FRAME_COUNT; ++i) {
+ float accum = 0.f;
+ for (size_t j = 0; j < NCHAN; ++j) {
+ accum += *outp++;
+ }
+ EXPECT_EQ(accum / NCHAN * volaux / volmono, *auxp++);
+ }
+ }
+ }
+};
+
+TEST(mixerops, stereovolume_1) { // Note: mono not used for output sinks yet.
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 1>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_2) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 2>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_3) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 3>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_4) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 4>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_5) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 5>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_6) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 6>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_7) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 7>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_8) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 8>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_12) {
+ if constexpr (FCC_LIMIT >= 12) { // NOTE: FCC_LIMIT is an enum, so can't #if
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 12>::testStereoVolume();
+ }
+}
+TEST(mixerops, stereovolume_24) {
+ if constexpr (FCC_LIMIT >= 24) {
+ MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 24>::testStereoVolume();
+ }
+}
+TEST(mixerops, channel_equivalence) {
+ // we must match the constexpr function with the system determined channel mask from count.
+ for (size_t i = 0; i < FCC_LIMIT; ++i) {
+ const audio_channel_mask_t actual = canonicalChannelMaskFromCount(i);
+ const audio_channel_mask_t system = audio_channel_out_mask_from_count(i);
+ if (system == AUDIO_CHANNEL_INVALID) continue;
+ EXPECT_EQ(system, actual);
+ }
+}
diff --git a/media/libeffects/hapticgenerator/Android.bp b/media/libeffects/hapticgenerator/Android.bp
index a660957..03ce329 100644
--- a/media/libeffects/hapticgenerator/Android.bp
+++ b/media/libeffects/hapticgenerator/Android.bp
@@ -45,6 +45,7 @@
shared_libs: [
"libaudioutils",
+ "libbase",
"libbinder",
"liblog",
"libutils",
diff --git a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
index 65a20a7..3137e13 100644
--- a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
+++ b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
@@ -22,12 +22,15 @@
#include <algorithm>
#include <memory>
+#include <string>
#include <utility>
#include <errno.h>
#include <inttypes.h>
#include <math.h>
+#include <android-base/parsedouble.h>
+#include <android-base/properties.h>
#include <audio_effects/effect_hapticgenerator.h>
#include <audio_utils/format.h>
#include <system/audio.h>
@@ -35,6 +38,7 @@
static constexpr float DEFAULT_RESONANT_FREQUENCY = 150.0f;
static constexpr float DEFAULT_BSF_ZERO_Q = 8.0f;
static constexpr float DEFAULT_BSF_POLE_Q = 4.0f;
+static constexpr float DEFAULT_DISTORTION_OUTPUT_GAIN = 1.5f;
// This is the only symbol that needs to be exported
__attribute__ ((visibility ("default")))
@@ -81,6 +85,15 @@
namespace {
+float getFloatProperty(const std::string& key, float defaultValue) {
+ float result;
+ std::string value = android::base::GetProperty(key, "");
+ if (!value.empty() && android::base::ParseFloat(value, &result)) {
+ return result;
+ }
+ return defaultValue;
+}
+
int HapticGenerator_Init(struct HapticGeneratorContext *context) {
context->itfe = &gHapticGeneratorInterface;
@@ -114,7 +127,9 @@
context->param.distortionCornerFrequency = 300.0f;
context->param.distortionInputGain = 0.3f;
context->param.distortionCubeThreshold = 0.1f;
- context->param.distortionOutputGain = 1.5f;
+ context->param.distortionOutputGain = getFloatProperty(
+ "vendor.audio.hapticgenerator.distortion.output.gain", DEFAULT_DISTORTION_OUTPUT_GAIN);
+ ALOGD("Using distortion output gain as %f", context->param.distortionOutputGain);
context->state = HAPTICGENERATOR_STATE_INITIALIZED;
return 0;
@@ -287,15 +302,17 @@
break;
}
case HG_PARAM_VIBRATOR_INFO: {
- if (value == nullptr || size != 2 * sizeof(float)) {
+ if (value == nullptr || size != 3 * sizeof(float)) {
return -EINVAL;
}
const float resonantFrequency = *(float*) value;
const float qFactor = *((float *) value + 1);
+ const float maxAmplitude = *((float *) value + 2);
context->param.resonantFrequency =
isnan(resonantFrequency) ? DEFAULT_RESONANT_FREQUENCY : resonantFrequency;
context->param.bsfZeroQ = isnan(qFactor) ? DEFAULT_BSF_POLE_Q : qFactor;
context->param.bsfPoleQ = context->param.bsfZeroQ / 2.0f;
+ context->param.maxHapticAmplitude = maxAmplitude;
if (context->processorsRecord.bpf != nullptr) {
context->processorsRecord.bpf->setCoefficients(
@@ -448,7 +465,8 @@
float* hapticOutBuffer = HapticGenerator_runProcessingChain(
context->processingChain, context->inputBuffer.data(),
context->outputBuffer.data(), inBuffer->frameCount);
- os::scaleHapticData(hapticOutBuffer, hapticSampleCount, context->param.maxHapticIntensity);
+ os::scaleHapticData(hapticOutBuffer, hapticSampleCount, context->param.maxHapticIntensity,
+ context->param.maxHapticAmplitude);
// For haptic data, the haptic playback thread will copy the data from effect input buffer,
// which contains haptic data at the end of the buffer, directly to sink buffer.
diff --git a/media/libeffects/hapticgenerator/EffectHapticGenerator.h b/media/libeffects/hapticgenerator/EffectHapticGenerator.h
index 96b744a..85e961f 100644
--- a/media/libeffects/hapticgenerator/EffectHapticGenerator.h
+++ b/media/libeffects/hapticgenerator/EffectHapticGenerator.h
@@ -51,6 +51,7 @@
// A map from track id to haptic intensity.
std::map<int, os::HapticScale> id2Intensity;
os::HapticScale maxHapticIntensity; // max intensity will be used to scale haptic data.
+ float maxHapticAmplitude; // max amplitude will be used to limit haptic data absolute values.
float resonantFrequency;
float bpfQ;
diff --git a/media/ndk/NdkMediaFormat.cpp b/media/ndk/NdkMediaFormat.cpp
index c1793ce..b035e5a 100644
--- a/media/ndk/NdkMediaFormat.cpp
+++ b/media/ndk/NdkMediaFormat.cpp
@@ -351,6 +351,11 @@
EXPORT const char* AMEDIAFORMAT_KEY_MIME = "mime";
EXPORT const char* AMEDIAFORMAT_KEY_MPEG_USER_DATA = "mpeg-user-data";
EXPORT const char* AMEDIAFORMAT_KEY_MPEG2_STREAM_HEADER = "mpeg2-stream-header";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS = "mpegh-compatible-sets";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION =
+ "mpegh-profile-level-indication";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT =
+ "mpegh-reference-channel-layout";
EXPORT const char* AMEDIAFORMAT_KEY_OPERATING_RATE = "operating-rate";
EXPORT const char* AMEDIAFORMAT_KEY_PCM_ENCODING = "pcm-encoding";
EXPORT const char* AMEDIAFORMAT_KEY_PRIORITY = "priority";
diff --git a/media/ndk/include/media/NdkMediaFormat.h b/media/ndk/include/media/NdkMediaFormat.h
index fbd855d..2d2fcc0 100644
--- a/media/ndk/include/media/NdkMediaFormat.h
+++ b/media/ndk/include/media/NdkMediaFormat.h
@@ -320,6 +320,34 @@
extern const char* AMEDIAFORMAT_VIDEO_QP_P_MAX __INTRODUCED_IN(31);
extern const char* AMEDIAFORMAT_VIDEO_QP_P_MIN __INTRODUCED_IN(31);
+/**
+ * MPEG-H audio profile and level compatibility.
+ *
+ * See FDAmd_2 of ISO_IEC_23008-3;2019 MHAProfileAndLevelCompatibilitySetBox.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS __INTRODUCED_IN(32);
+
+/**
+ * MPEG-H audio profile level indication.
+ *
+ * See ISO_IEC_23008-3;2019 MHADecoderConfigurationRecord mpegh3daProfileLevelIndication.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION __INTRODUCED_IN(32);
+
+/**
+ * MPEG-H audio reference channel layout.
+ *
+ * See ISO_IEC_23008-3;2019 MHADecoderConfigurationRecord referenceChannelLayout
+ * and ISO_IEC_23001‐8 ChannelConfiguration value.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT __INTRODUCED_IN(32);
+
__END_DECLS
#endif // _NDK_MEDIA_FORMAT_H
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index 7e9e57e..6f275c7 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -126,6 +126,9 @@
AMEDIAFORMAT_KEY_MIME; # var introduced=21
AMEDIAFORMAT_KEY_MPEG_USER_DATA; # var introduced=28
AMEDIAFORMAT_KEY_MPEG2_STREAM_HEADER; # var introduced=29
+ AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS; # var introduced=32
+ AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION; # var introduced=32
+ AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT; # var introduced=32
AMEDIAFORMAT_KEY_OPERATING_RATE; # var introduced=28
AMEDIAFORMAT_KEY_PCM_BIG_ENDIAN; # var introduced=29
AMEDIAFORMAT_KEY_PCM_ENCODING; # var introduced=28
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index bfe73d5..73c4e3b 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -81,6 +81,36 @@
export_include_dirs: ["include"],
}
+cc_library {
+ name: "libmediautils_vendor",
+ vendor_available: true, // required for platform/hardware/interfaces
+ srcs: [
+ "MemoryLeakTrackUtil.cpp",
+ ],
+
+ cflags: [
+ "-Wall",
+ "-Wextra",
+ "-Werror",
+ ],
+ shared_libs: [
+ "liblog",
+ "libutils",
+ ],
+
+ static_libs: [
+ "libc_malloc_debug_backtrace",
+ ],
+
+ header_libs: [
+ "bionic_libc_platform_headers",
+ ],
+
+ local_include_dirs: ["include"],
+ export_include_dirs: ["include"],
+}
+
+
cc_library_headers {
name: "libmediautils_headers",
vendor_available: true, // required for platform/hardware/interfaces
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 65a163f..a130d0a 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -336,11 +336,11 @@
}
// getDefaultVibratorInfo_l must be called with AudioFlinger lock held.
-const media::AudioVibratorInfo* AudioFlinger::getDefaultVibratorInfo_l() {
+std::optional<media::AudioVibratorInfo> AudioFlinger::getDefaultVibratorInfo_l() {
if (mAudioVibratorInfos.empty()) {
- return nullptr;
+ return {};
}
- return &mAudioVibratorInfos.front();
+ return mAudioVibratorInfos.front();
}
AudioFlinger::~AudioFlinger()
@@ -695,7 +695,7 @@
// dump all hardware devs
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
- dev->dump(fd);
+ dev->dump(fd, args);
}
mPatchPanel.dump(fd);
@@ -2455,6 +2455,10 @@
ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
thread->systemReady();
}
+ for (size_t i = 0; i < mMmapThreads.size(); i++) {
+ ThreadBase *thread = (ThreadBase *)mMmapThreads.valueAt(i).get();
+ thread->systemReady();
+ }
return NO_ERROR;
}
@@ -2501,7 +2505,8 @@
sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig __unused,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags)
@@ -2529,16 +2534,16 @@
// Check only for Normal Mixing mode
if (kEnableExtendedPrecision) {
// Specify format (uncomment one below to choose)
- //config->format = AUDIO_FORMAT_PCM_FLOAT;
- //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
- //config->format = AUDIO_FORMAT_PCM_32_BIT;
- //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
- // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
+ //halConfig->format = AUDIO_FORMAT_PCM_FLOAT;
+ //halConfig->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+ //halConfig->format = AUDIO_FORMAT_PCM_32_BIT;
+ //halConfig->format = AUDIO_FORMAT_PCM_8_24_BIT;
+ // ALOGV("openOutput_l() upgrading format to %#08x", halConfig->format);
}
if (kEnableExtendedChannels) {
// Specify channel mask (uncomment one below to choose)
- //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
- //config->channel_mask = audio_channel_mask_from_representation_and_bits(
+ //halConfig->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
+ //halConfig->channel_mask = audio_channel_mask_from_representation_and_bits(
// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
}
}
@@ -2549,7 +2554,7 @@
*output,
deviceType,
flags,
- config,
+ halConfig,
address.string());
mHardwareStatus = AUDIO_HW_IDLE;
@@ -2564,13 +2569,20 @@
return thread;
} else {
sp<PlaybackThread> thread;
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+ if (flags == (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST
+ | AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
+ thread = new VirtualizerStageThread(this, outputStream, *output,
+ mSystemReady, mixerConfig);
+ ALOGD("openOutput_l() created virtualizer output: ID %d thread %p",
+ *output, thread.get());
+ } else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created offload output: ID %d thread %p",
*output, thread.get());
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
- || !isValidPcmSinkFormat(config->format)
- || !isValidPcmSinkChannelMask(config->channel_mask)) {
+ || !isValidPcmSinkFormat(halConfig->format)
+ || !isValidPcmSinkChannelMask(halConfig->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
@@ -2597,8 +2609,10 @@
{
audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
aidl2legacy_int32_t_audio_module_handle_t(request.module));
- audio_config_t config = VALUE_OR_RETURN_STATUS(
- aidl2legacy_AudioConfig_audio_config_t(request.config));
+ audio_config_t halConfig = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioConfig_audio_config_t(request.halConfig));
+ audio_config_base_t mixerConfig = VALUE_OR_RETURN_STATUS(
+ aidl2legacy_AudioConfigBase_audio_config_base_t(request.mixerConfig));
sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
aidl2legacy_DeviceDescriptorBase(request.device));
audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
@@ -2611,9 +2625,9 @@
"Channels %#x, flags %#x",
this, module,
device->toString().c_str(),
- config.sample_rate,
- config.format,
- config.channel_mask,
+ halConfig.sample_rate,
+ halConfig.format,
+ halConfig.channel_mask,
flags);
audio_devices_t deviceType = device->type();
@@ -2625,7 +2639,8 @@
Mutex::Autolock _l(mLock);
- sp<ThreadBase> thread = openOutput_l(module, &output, &config, deviceType, address, flags);
+ sp<ThreadBase> thread = openOutput_l(module, &output, &halConfig,
+ &mixerConfig, deviceType, address, flags);
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
@@ -2650,7 +2665,8 @@
mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
- response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+ response->config =
+ VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(halConfig));
response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
response->flags = VALUE_OR_RETURN_STATUS(
legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
@@ -3797,7 +3813,8 @@
io = mPlaybackThreads.keyAt(0);
}
ALOGV("createEffect() got io %d for effect %s", io, descOut.name);
- } else if (checkPlaybackThread_l(io) != nullptr) {
+ } else if (checkPlaybackThread_l(io) != nullptr
+ && sessionId != AUDIO_SESSION_OUTPUT_STAGE) {
// allow only one effect chain per sessionId on mPlaybackThreads.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
const audio_io_handle_t checkIo = mPlaybackThreads.keyAt(i);
@@ -4178,6 +4195,7 @@
case TransactionCode::LIST_AUDIO_PATCHES:
case TransactionCode::SET_AUDIO_PORT_CONFIG:
case TransactionCode::SET_RECORD_SILENCED:
+ case TransactionCode::AUDIO_POLICY_READY:
ALOGW("%s: transaction %d received from PID %d",
__func__, code, IPCThreadState::self()->getCallingPid());
// return status only for non void methods
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index fff61f8..8fcd6e4 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -269,6 +269,9 @@
/* Indicate JAVA services are ready (scheduling, power management ...) */
virtual status_t systemReady();
+ virtual status_t audioPolicyReady() { mAudioPolicyReady.store(true); return NO_ERROR; }
+ bool isAudioPolicyReady() const { return mAudioPolicyReady.load(); }
+
virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
@@ -309,7 +312,7 @@
void updateDownStreamPatches_l(const struct audio_patch *patch,
const std::set<audio_io_handle_t> streams);
- const media::AudioVibratorInfo* getDefaultVibratorInfo_l();
+ std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l();
private:
// FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed.
@@ -735,7 +738,8 @@
const String8& outputDeviceAddress);
sp<ThreadBase> openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags);
@@ -986,6 +990,7 @@
DeviceEffectManager mDeviceEffectManager;
bool mSystemReady;
+ std::atomic_bool mAudioPolicyReady{};
mediautils::UidInfo mUidInfo;
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
index a05f5fe..c222de8 100644
--- a/services/audioflinger/DeviceEffectManager.h
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -163,8 +163,13 @@
bool isOffloadOrMmap() const override { return false; }
uint32_t sampleRate() const override { return 0; }
- audio_channel_mask_t channelMask() const override { return AUDIO_CHANNEL_NONE; }
- uint32_t channelCount() const override { return 0; }
+ audio_channel_mask_t inChannelMask(int id __unused) const override {
+ return AUDIO_CHANNEL_NONE;
+ }
+ uint32_t inChannelCount(int id __unused) const override { return 0; }
+ audio_channel_mask_t outChannelMask() const override { return AUDIO_CHANNEL_NONE; }
+ uint32_t outChannelCount() const override { return 0; }
+
audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
size_t frameCount() const override { return 0; }
uint32_t latency() const override { return 0; }
@@ -190,6 +195,10 @@
wp<EffectChain> chain() const override { return nullptr; }
+ bool isAudioPolicyReady() const override {
+ return mManager.audioFlinger().isAudioPolicyReady();
+ }
+
int newEffectId() { return mManager.audioFlinger().nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); }
status_t addEffectToHal(audio_port_handle_t deviceId,
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index b267d88..bd661f9 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -242,6 +242,12 @@
{
Mutex::Autolock _l(mLock);
+
+ if ((isInternal_l() && !mPolicyRegistered)
+ || !getCallback()->isAudioPolicyReady()) {
+ return NO_ERROR;
+ }
+
// register effect when first handle is attached and unregister when last handle is removed
if (mPolicyRegistered != mHandles.size() > 0) {
doRegister = true;
@@ -875,9 +881,9 @@
// similar to output EFFECT_FLAG_TYPE_INSERT/REPLACE,
// in which case input channel masks should be used here.
callback = getCallback();
- channelMask = callback->channelMask();
+ channelMask = callback->inChannelMask(mId);
mConfig.inputCfg.channels = channelMask;
- mConfig.outputCfg.channels = channelMask;
+ mConfig.outputCfg.channels = callback->outChannelMask();
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
if (mConfig.inputCfg.channels != AUDIO_CHANNEL_OUT_MONO) {
@@ -1600,7 +1606,7 @@
return status;
}
-status_t AudioFlinger::EffectModule::setVibratorInfo(const media::AudioVibratorInfo* vibratorInfo)
+status_t AudioFlinger::EffectModule::setVibratorInfo(const media::AudioVibratorInfo& vibratorInfo)
{
if (mStatus != NO_ERROR) {
return mStatus;
@@ -1610,15 +1616,17 @@
return INVALID_OPERATION;
}
+ const size_t paramCount = 3;
std::vector<uint8_t> request(
- sizeof(effect_param_t) + sizeof(int32_t) + 2 * sizeof(float));
+ sizeof(effect_param_t) + sizeof(int32_t) + paramCount * sizeof(float));
effect_param_t *param = (effect_param_t*) request.data();
param->psize = sizeof(int32_t);
- param->vsize = 2 * sizeof(float);
+ param->vsize = paramCount * sizeof(float);
*(int32_t*)param->data = HG_PARAM_VIBRATOR_INFO;
float* vibratorInfoPtr = reinterpret_cast<float*>(param->data + sizeof(int32_t));
- vibratorInfoPtr[0] = vibratorInfo->resonantFrequency;
- vibratorInfoPtr[1] = vibratorInfo->qFactor;
+ vibratorInfoPtr[0] = vibratorInfo.resonantFrequency;
+ vibratorInfoPtr[1] = vibratorInfo.qFactor;
+ vibratorInfoPtr[2] = vibratorInfo.maxAmplitude;
std::vector<uint8_t> response;
status_t status = command(EFFECT_CMD_SET_PARAM, request, sizeof(int32_t), &response);
if (status == NO_ERROR) {
@@ -2048,11 +2056,11 @@
mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX),
mEffectCallback(new EffectCallback(wp<EffectChain>(this), thread))
{
- mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
sp<ThreadBase> p = thread.promote();
if (p == nullptr) {
return;
}
+ mStrategy = p->getStrategyForStream(AUDIO_STREAM_MUSIC);
mMaxTailBuffers = ((kProcessTailDurationMs * p->sampleRate()) / 1000) /
p->frameCount();
}
@@ -2125,8 +2133,8 @@
if (mInBuffer == NULL) {
return;
}
- const size_t frameSize =
- audio_bytes_per_sample(EFFECT_BUFFER_FORMAT) * mEffectCallback->channelCount();
+ const size_t frameSize = audio_bytes_per_sample(EFFECT_BUFFER_FORMAT)
+ * mEffectCallback->inChannelCount(mEffects[0]->id());
memset(mInBuffer->audioBuffer()->raw, 0, mEffectCallback->frameCount() * frameSize);
mInBuffer->commit();
@@ -2236,6 +2244,9 @@
numSamples * sizeof(int32_t), &halBuffer);
#endif
if (result != OK) return result;
+
+ effect->configure();
+
effect->setInBuffer(halBuffer);
// auxiliary effects output samples to chain input buffer for further processing
// by insert effects
@@ -2303,6 +2314,10 @@
}
}
+ mEffects.insertAt(effect, idx_insert);
+
+ effect->configure();
+
// always read samples from chain input buffer
effect->setInBuffer(mInBuffer);
@@ -2310,14 +2325,13 @@
// output buffer, otherwise to chain input buffer
if (idx_insert == size) {
if (idx_insert != 0) {
- mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
mEffects[idx_insert-1]->configure();
+ mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
}
effect->setOutBuffer(mOutBuffer);
} else {
effect->setOutBuffer(mInBuffer);
}
- mEffects.insertAt(effect, idx_insert);
ALOGV("addEffect_l() effect %p, added in chain %p at rank %zu", effect.get(), this,
idx_insert);
@@ -2350,14 +2364,21 @@
if (type != EFFECT_FLAG_TYPE_AUXILIARY) {
if (i == size - 1 && i != 0) {
- mEffects[i - 1]->setOutBuffer(mOutBuffer);
mEffects[i - 1]->configure();
+ mEffects[i - 1]->setOutBuffer(mOutBuffer);
}
}
mEffects.removeAt(i);
+
+ // make sure the input buffer configuration for the new first effect in the chain
+ // is updated if needed (can switch from HAL channel mask to mixer channel mask)
+ if (i == 0 && size > 1) {
+ mEffects[0]->configure();
+ mEffects[0]->setInBuffer(mInBuffer);
+ }
+
ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %zu", effect.get(),
this, i);
-
break;
}
}
@@ -2932,7 +2953,43 @@
return t->sampleRate();
}
-audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::inChannelMask(int id) const {
+ sp<ThreadBase> t = thread().promote();
+ if (t == nullptr) {
+ return AUDIO_CHANNEL_NONE;
+ }
+ sp<EffectChain> c = chain().promote();
+ if (c == nullptr) {
+ return AUDIO_CHANNEL_NONE;
+ }
+
+ if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+ || c->isFirstEffect(id)) {
+ return t->mixerChannelMask();
+ } else {
+ return t->channelMask();
+ }
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::inChannelCount(int id) const {
+ sp<ThreadBase> t = thread().promote();
+ if (t == nullptr) {
+ return 0;
+ }
+ sp<EffectChain> c = chain().promote();
+ if (c == nullptr) {
+ return 0;
+ }
+
+ if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+ || c->isFirstEffect(id)) {
+ return audio_channel_count_from_out_mask(t->mixerChannelMask());
+ } else {
+ return t->channelCount();
+ }
+}
+
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::outChannelMask() const {
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_CHANNEL_NONE;
@@ -2940,7 +2997,7 @@
return t->channelMask();
}
-uint32_t AudioFlinger::EffectChain::EffectCallback::channelCount() const {
+uint32_t AudioFlinger::EffectChain::EffectCallback::outChannelCount() const {
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
return 0;
@@ -3364,7 +3421,8 @@
return proxy->sampleRate();
}
-audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelMask(
+ int id __unused) const {
sp<DeviceEffectProxy> proxy = mProxy.promote();
if (proxy == nullptr) {
return AUDIO_CHANNEL_OUT_STEREO;
@@ -3372,7 +3430,23 @@
return proxy->channelMask();
}
-uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelCount() const {
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelCount(int id __unused) const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return 2;
+ }
+ return proxy->channelCount();
+}
+
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelMask() const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return AUDIO_CHANNEL_OUT_STEREO;
+ }
+ return proxy->channelMask();
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelCount() const {
sp<DeviceEffectProxy> proxy = mProxy.promote();
if (proxy == nullptr) {
return 2;
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index a727e04..1d0d00d 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -34,8 +34,10 @@
virtual bool isOffloadOrDirect() const = 0;
virtual bool isOffloadOrMmap() const = 0;
virtual uint32_t sampleRate() const = 0;
- virtual audio_channel_mask_t channelMask() const = 0;
- virtual uint32_t channelCount() const = 0;
+ virtual audio_channel_mask_t inChannelMask(int id) const = 0;
+ virtual uint32_t inChannelCount(int id) const = 0;
+ virtual audio_channel_mask_t outChannelMask() const = 0;
+ virtual uint32_t outChannelCount() const = 0;
virtual audio_channel_mask_t hapticChannelMask() const = 0;
virtual size_t frameCount() const = 0;
@@ -64,6 +66,8 @@
virtual void resetVolume() = 0;
virtual wp<EffectChain> chain() const = 0;
+
+ virtual bool isAudioPolicyReady() const = 0;
};
// EffectBase(EffectModule) and EffectChain classes both have their own mutex to protect
@@ -164,6 +168,16 @@
void dump(int fd, const Vector<String16>& args);
+protected:
+ bool isInternal_l() const {
+ for (auto handle : mHandles) {
+ if (handle->client() != nullptr) {
+ return false;
+ }
+ }
+ return true;
+ }
+
private:
friend class AudioFlinger; // for mHandles
bool mPinned = false;
@@ -259,7 +273,7 @@
bool isHapticGenerator() const;
status_t setHapticIntensity(int id, int intensity);
- status_t setVibratorInfo(const media::AudioVibratorInfo* vibratorInfo);
+ status_t setVibratorInfo(const media::AudioVibratorInfo& vibratorInfo);
void dump(int fd, const Vector<String16>& args);
@@ -342,6 +356,8 @@
android::binder::Status disconnect() override;
android::binder::Status getCblk(media::SharedFileRegion* _aidl_return) override;
+ sp<Client> client() const { return mClient; }
+
private:
void disconnect(bool unpinIfLast);
@@ -511,6 +527,8 @@
sp<EffectCallbackInterface> effectCallback() const { return mEffectCallback; }
wp<ThreadBase> thread() const { return mEffectCallback->thread(); }
+ bool isFirstEffect(int id) const { return !mEffects.isEmpty() && id == mEffects[0]->id(); }
+
void dump(int fd, const Vector<String16>& args);
private:
@@ -544,8 +562,10 @@
bool isOffloadOrMmap() const override;
uint32_t sampleRate() const override;
- audio_channel_mask_t channelMask() const override;
- uint32_t channelCount() const override;
+ audio_channel_mask_t inChannelMask(int id) const override;
+ uint32_t inChannelCount(int id) const override;
+ audio_channel_mask_t outChannelMask() const override;
+ uint32_t outChannelCount() const override;
audio_channel_mask_t hapticChannelMask() const override;
size_t frameCount() const override;
uint32_t latency() const override;
@@ -566,6 +586,10 @@
wp<EffectChain> chain() const override { return mChain; }
+ bool isAudioPolicyReady() const override {
+ return mAudioFlinger.isAudioPolicyReady();
+ }
+
wp<ThreadBase> thread() const { return mThread.load(); }
void setThread(const wp<ThreadBase>& thread) {
@@ -694,8 +718,10 @@
bool isOffloadOrMmap() const override { return false; }
uint32_t sampleRate() const override;
- audio_channel_mask_t channelMask() const override;
- uint32_t channelCount() const override;
+ audio_channel_mask_t inChannelMask(int id) const override;
+ uint32_t inChannelCount(int id) const override;
+ audio_channel_mask_t outChannelMask() const override;
+ uint32_t outChannelCount() const override;
audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
size_t frameCount() const override { return 0; }
uint32_t latency() const override { return 0; }
@@ -716,6 +742,10 @@
wp<EffectChain> chain() const override { return nullptr; }
+ bool isAudioPolicyReady() const override {
+ return mManagerCallback->isAudioPolicyReady();
+ }
+
int newEffectId();
private:
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 88d4eaf..fc34d95 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -204,6 +204,8 @@
(void *)(uintptr_t)fastTrack->mHapticPlaybackEnabled);
mMixer->setParameter(index, AudioMixer::TRACK, AudioMixer::HAPTIC_INTENSITY,
(void *)(uintptr_t)fastTrack->mHapticIntensity);
+ mMixer->setParameter(index, AudioMixer::TRACK, AudioMixer::HAPTIC_MAX_AMPLITUDE,
+ (void *)(&(fastTrack->mHapticMaxAmplitude)));
mMixer->enable(index);
break;
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index 857d3de..ce3cc14 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -17,6 +17,8 @@
#ifndef ANDROID_AUDIO_FAST_MIXER_STATE_H
#define ANDROID_AUDIO_FAST_MIXER_STATE_H
+#include <math.h>
+
#include <audio_utils/minifloat.h>
#include <system/audio.h>
#include <media/AudioMixer.h>
@@ -51,6 +53,7 @@
int mGeneration; // increment when any field is assigned
bool mHapticPlaybackEnabled = false; // haptic playback is enabled or not
os::HapticScale mHapticIntensity = os::HapticScale::MUTE; // intensity of haptic data
+ float mHapticMaxAmplitude = NAN; // max amplitude allowed for haptic data
};
// Represents a single state of the fast mixer
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index a381c7d..93118b8 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -258,6 +258,7 @@
reinterpret_cast<PlaybackThread*>(thread.get()), false /*closeThread*/);
} else {
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
@@ -276,6 +277,7 @@
patch->sinks[0].ext.device.hw_module,
&output,
&config,
+ &mixerConfig,
outputDevice,
outputDeviceAddress,
flags);
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 0929055..e9e98ca 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -19,6 +19,8 @@
#error This header file should only be included from AudioFlinger.h
#endif
+#include <math.h>
+
// Checks and monitors OP_PLAY_AUDIO
class OpPlayAudioMonitor : public RefBase {
public:
@@ -161,6 +163,8 @@
}
/** Return at what intensity to play haptics, used in mixer. */
os::HapticScale getHapticIntensity() const { return mHapticIntensity; }
+ /** Return the maximum amplitude allowed for haptics data, used in mixer. */
+ float getHapticMaxAmplitude() const { return mHapticMaxAmplitude; }
/** Set intensity of haptic playback, should be set after querying vibrator service. */
void setHapticIntensity(os::HapticScale hapticIntensity) {
if (os::isValidHapticScale(hapticIntensity)) {
@@ -168,6 +172,12 @@
setHapticPlaybackEnabled(mHapticIntensity != os::HapticScale::MUTE);
}
}
+ /** Set maximum amplitude allowed for haptic data, should be set after querying
+ * vibrator service.
+ */
+ void setHapticMaxAmplitude(float maxAmplitude) {
+ mHapticMaxAmplitude = maxAmplitude;
+ }
sp<os::ExternalVibration> getExternalVibration() const { return mExternalVibration; }
void setTeePatches(TeePatches teePatches);
@@ -282,6 +292,8 @@
bool mHapticPlaybackEnabled = false; // indicates haptic playback enabled or not
// intensity to play haptic data
os::HapticScale mHapticIntensity = os::HapticScale::MUTE;
+ // max amplitude allowed for haptic data
+ float mHapticMaxAmplitude = NAN;
class AudioVibrationController : public os::BnExternalVibrationController {
public:
explicit AudioVibrationController(Track* track) : mTrack(track) {}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b9cdab8..7292527 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -50,8 +50,10 @@
#include <audio_utils/format.h>
#include <audio_utils/minifloat.h>
#include <audio_utils/safe_math.h>
-#include <system/audio_effects/effect_ns.h>
#include <system/audio_effects/effect_aec.h>
+#include <system/audio_effects/effect_downmix.h>
+#include <system/audio_effects/effect_ns.h>
+#include <system/audio_effects/effect_virtualizer_stage.h>
#include <system/audio.h>
// NBAIO implementations
@@ -507,6 +509,8 @@
return "MMAP_PLAYBACK";
case MMAP_CAPTURE:
return "MMAP_CAPTURE";
+ case VIRTUALIZER_STAGE:
+ return "VIRTUALIZER_STAGE";
default:
return "unknown";
}
@@ -722,6 +726,19 @@
sendConfigEvent_l(configEvent);
}
+void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent()
+{
+ Mutex::Autolock _l(mLock);
+ sendCheckOutputStageEffectsEvent_l();
+}
+
+void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent_l()
+{
+ sp<ConfigEvent> configEvent =
+ (ConfigEvent *)new CheckOutputStageEffectsEvent();
+ sendConfigEvent_l(configEvent);
+}
+
// post condition: mConfigEvents.isEmpty()
void AudioFlinger::ThreadBase::processConfigEvents_l()
{
@@ -784,6 +801,11 @@
(ResizeBufferConfigEventData *)event->mData.get();
resizeInputBuffer_l(data->mMaxSharedAudioHistoryMs);
} break;
+
+ case CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS: {
+ setCheckOutputStageEffects();
+ } break;
+
default:
ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
break;
@@ -1008,6 +1030,8 @@
return String16("MmapPlayback");
case MMAP_CAPTURE:
return String16("MmapCapture");
+ case VIRTUALIZER_STAGE:
+ return String16("AudioVirt");
default:
ALOG_ASSERT(false);
return String16("AudioUnknown");
@@ -1401,6 +1425,13 @@
return BAD_VALUE;
}
break;
+ case VIRTUALIZER_STAGE:
+ if (!audio_is_global_session(sessionId)) {
+ ALOGW("checkEffectCompatibility_l(): non global effect %s on VIRTUALIZER_STAGE"
+ " thread %s", desc->name, mThreadName);
+ return BAD_VALUE;
+ }
+ break;
default:
LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
}
@@ -1477,11 +1508,11 @@
if (effect->isHapticGenerator()) {
// TODO(b/184194057): Use the vibrator information from the vibrator that will be used
// for the HapticGenerator.
- const media::AudioVibratorInfo* defaultVibratorInfo =
- mAudioFlinger->getDefaultVibratorInfo_l();
- if (defaultVibratorInfo != nullptr) {
+ const std::optional<media::AudioVibratorInfo> defaultVibratorInfo =
+ std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+ if (defaultVibratorInfo) {
// Only set the vibrator info when it is a valid one.
- effect->setVibratorInfo(defaultVibratorInfo);
+ effect->setVibratorInfo(*defaultVibratorInfo);
}
}
// create effect handle and connect it to effect module
@@ -1489,6 +1520,7 @@
lStatus = handle->initCheck();
if (lStatus == OK) {
lStatus = effect->addHandle(handle.get());
+ sendCheckOutputStageEffectsEvent_l();
}
if (enabled != NULL) {
*enabled = (int)effect->isEnabled();
@@ -1531,6 +1563,7 @@
if (remove) {
removeEffect_l(effect, true);
}
+ sendCheckOutputStageEffectsEvent_l();
}
if (remove) {
mAudioFlinger->updateOrphanEffectChains(effect);
@@ -1888,6 +1921,14 @@
item->selfrecord();
}
+product_strategy_t AudioFlinger::ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
+{
+ if (!mAudioFlinger->isAudioPolicyReady()) {
+ return PRODUCT_STRATEGY_NONE;
+ }
+ return AudioSystem::getStrategyForStream(stream);
+}
+
// ----------------------------------------------------------------------------
// Playback
// ----------------------------------------------------------------------------
@@ -1896,15 +1937,16 @@
AudioStreamOut* output,
audio_io_handle_t id,
type_t type,
- bool systemReady)
+ bool systemReady,
+ audio_config_base_t *mixerConfig)
: ThreadBase(audioFlinger, id, type, systemReady, true /* isOut */),
mNormalFrameCount(0), mSinkBuffer(NULL),
- mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
+ mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision || type == VIRTUALIZER_STAGE),
mMixerBuffer(NULL),
mMixerBufferSize(0),
mMixerBufferFormat(AUDIO_FORMAT_INVALID),
mMixerBufferValid(false),
- mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
+ mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision || type == VIRTUALIZER_STAGE),
mEffectBuffer(NULL),
mEffectBufferSize(0),
mEffectBufferFormat(AUDIO_FORMAT_INVALID),
@@ -1956,8 +1998,18 @@
mOutput->audioHwDev->moduleName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0;
}
+ if (mixerConfig != nullptr && mixerConfig->channel_mask != AUDIO_CHANNEL_NONE) {
+ mMixerChannelMask = mixerConfig->channel_mask;
+ }
+
readOutputParameters_l();
+ if (mType != VIRTUALIZER_STAGE
+ && mMixerChannelMask != mChannelMask) {
+ LOG_ALWAYS_FATAL("HAL channel mask %#x does not match mixer channel mask %#x",
+ mChannelMask, mMixerChannelMask);
+ }
+
// TODO: We may also match on address as well as device type for
// AUDIO_DEVICE_OUT_BUS, AUDIO_DEVICE_OUT_ALL_A2DP, AUDIO_DEVICE_OUT_REMOTE_SUBMIX
if (type == MIXER || type == DIRECT || type == OFFLOAD) {
@@ -2080,10 +2132,12 @@
write(fd, result.string(), result.size());
}
-void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
dprintf(fd, " Master volume: %f\n", mMasterVolume);
dprintf(fd, " Master mute: %s\n", mMasterMute ? "on" : "off");
+ dprintf(fd, " Mixer channel Mask: %#x (%s)\n",
+ mMixerChannelMask, channelMaskToString(mMixerChannelMask, true /* output */).c_str());
if (mHapticChannelMask != AUDIO_CHANNEL_NONE) {
dprintf(fd, " Haptic channel mask: %#x (%s)\n", mHapticChannelMask,
channelMaskToString(mHapticChannelMask, true /* output */).c_str());
@@ -2109,7 +2163,7 @@
}
if (output != nullptr) {
dprintf(fd, " Hal stream dump:\n");
- (void)output->stream->dump(fd);
+ (void)output->stream->dump(fd, args);
}
}
@@ -2397,11 +2451,11 @@
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
- product_strategy_t strategy = AudioSystem::getStrategyForStream(streamType);
+ product_strategy_t strategy = getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0 && t->isExternalTrack()) {
- product_strategy_t actual = AudioSystem::getStrategyForStream(t->streamType());
+ product_strategy_t actual = getStrategyForStream(t->streamType());
if (sessionId == t->sessionId() && strategy != actual) {
ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
strategy, actual);
@@ -2445,7 +2499,7 @@
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
- chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
+ chain->setStrategy(getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
@@ -2613,8 +2667,19 @@
mLock.unlock();
const int intensity = AudioFlinger::onExternalVibrationStart(
track->getExternalVibration());
+ std::optional<media::AudioVibratorInfo> vibratorInfo;
+ {
+ // TODO(b/184194780): Use the vibrator information from the vibrator that will be
+ // used to play this track.
+ Mutex::Autolock _l(mAudioFlinger->mLock);
+ vibratorInfo = std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+ }
mLock.lock();
track->setHapticIntensity(static_cast<os::HapticScale>(intensity));
+ if (vibratorInfo) {
+ track->setHapticMaxAmplitude(vibratorInfo->maxAmplitude);
+ }
+
// Haptic playback should be enabled by vibrator service.
if (track->getHapticPlaybackEnabled()) {
// Disable haptic playback of all active track to ensure only
@@ -2814,14 +2879,20 @@
if (!audio_is_output_channel(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
}
- if ((mType == MIXER || mType == DUPLICATING)
- && !isValidPcmSinkChannelMask(mChannelMask)) {
+ if (hasMixer() && !isValidPcmSinkChannelMask(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
mChannelMask);
}
+
+ if (mMixerChannelMask == AUDIO_CHANNEL_NONE) {
+ mMixerChannelMask = mChannelMask;
+ }
+
mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
mBalance.setChannelMask(mChannelMask);
+ uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mMixerChannelMask);
+
// Get actual HAL format.
status_t result = mOutput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result);
@@ -2831,8 +2902,7 @@
if (!audio_is_valid_format(mFormat)) {
LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
}
- if ((mType == MIXER || mType == DUPLICATING)
- && !isValidPcmSinkFormat(mFormat)) {
+ if (hasMixer() && !isValidPcmSinkFormat(mFormat)) {
LOG_FATAL("HAL format %#x not supported for mixed output",
mFormat);
}
@@ -2841,7 +2911,7 @@
LOG_ALWAYS_FATAL_IF(result != OK,
"Error when retrieving output stream buffer size: %d", result);
mFrameCount = mBufferSize / mFrameSize;
- if ((mType == MIXER || mType == DUPLICATING) && (mFrameCount & 15)) {
+ if (hasMixer() && (mFrameCount & 15)) {
ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
mFrameCount);
}
@@ -2914,7 +2984,7 @@
}
mNormalFrameCount = multiplier * mFrameCount;
// round up to nearest 16 frames to satisfy AudioMixer
- if (mType == MIXER || mType == DUPLICATING) {
+ if (hasMixer()) {
mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
}
ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
@@ -2941,7 +3011,7 @@
mMixerBuffer = NULL;
if (mMixerBufferEnabled) {
mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // no longer valid: AUDIO_FORMAT_PCM_16_BIT.
- mMixerBufferSize = mNormalFrameCount * mChannelCount
+ mMixerBufferSize = mNormalFrameCount * mixerChannelCount
* audio_bytes_per_sample(mMixerBufferFormat);
(void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
}
@@ -2949,7 +3019,7 @@
mEffectBuffer = NULL;
if (mEffectBufferEnabled) {
mEffectBufferFormat = EFFECT_BUFFER_FORMAT;
- mEffectBufferSize = mNormalFrameCount * mChannelCount
+ mEffectBufferSize = mNormalFrameCount * mixerChannelCount
* audio_bytes_per_sample(mEffectBufferFormat);
(void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
}
@@ -2958,6 +3028,7 @@
mChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & ~mHapticChannelMask);
mHapticChannelCount = audio_channel_count_from_out_mask(mHapticChannelMask);
mChannelCount -= mHapticChannelCount;
+ mMixerChannelMask = static_cast<audio_channel_mask_t>(mMixerChannelMask & ~mHapticChannelMask);
// force reconfiguration of effect chains and engines to take new buffer size and audio
// parameters into account
@@ -3051,15 +3122,15 @@
// session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
// it is moved to correct output by audio policy manager when A2DP is connected or disconnected
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
- return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+ return getStrategyForStream(AUDIO_STREAM_MUSIC);
}
for (size_t i = 0; i < mTracks.size(); i++) {
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() && !track->isInvalid()) {
- return AudioSystem::getStrategyForStream(track->streamType());
+ return getStrategyForStream(track->streamType());
}
}
- return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+ return getStrategyForStream(AUDIO_STREAM_MUSIC);
}
@@ -3348,7 +3419,8 @@
// Only one effect chain can be present in direct output thread and it uses
// the sink buffer as input
if (mType != DIRECT) {
- size_t numSamples = mNormalFrameCount * (mChannelCount + mHapticChannelCount);
+ size_t numSamples = mNormalFrameCount
+ * (audio_channel_count_from_out_mask(mMixerChannelMask) + mHapticChannelCount);
status_t result = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
numSamples * sizeof(effect_buffer_t),
&halInBuffer);
@@ -3531,6 +3603,8 @@
audio_patch_handle_t lastDownstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+ sendCheckOutputStageEffectsEvent();
+
// loopCount is used for statistics and diagnostics.
for (int64_t loopCount = 0; !exitPending(); ++loopCount)
{
@@ -3587,11 +3661,18 @@
}
}
+ if (mCheckOutputStageEffects.exchange(false)) {
+ checkOutputStageEffects();
+ }
+
{ // scope for mLock
Mutex::Autolock _l(mLock);
processConfigEvents_l();
+ if (mCheckOutputStageEffects.load()) {
+ continue;
+ }
// See comment at declaration of logString for why this is done under mLock
if (logString != NULL) {
@@ -3757,6 +3838,8 @@
if (mMixerBufferValid) {
void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
+ uint32_t channelCount = mEffectBufferValid ?
+ audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
// mono blend occurs for mixer threads only (not direct or offloaded)
// and is handled here if we're going directly to the sink.
@@ -3774,7 +3857,7 @@
}
memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
- mNormalFrameCount * (mChannelCount + mHapticChannelCount));
+ mNormalFrameCount * (channelCount + mHapticChannelCount));
// If we're going directly to the sink and there are haptic channels,
// we should adjust channels as the sample data is partially interleaved
@@ -4448,8 +4531,8 @@
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, bool systemReady, type_t type)
- : PlaybackThread(audioFlinger, output, id, type, systemReady),
+ audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t *mixerConfig)
+ : PlaybackThread(audioFlinger, output, id, type, systemReady, mixerConfig),
// mAudioMixer below
// mFastMixer below
mFastMixerFutex(0),
@@ -4566,6 +4649,7 @@
fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
fastTrack->mHapticPlaybackEnabled = mHapticChannelMask != AUDIO_CHANNEL_NONE;
fastTrack->mHapticIntensity = os::HapticScale::NONE;
+ fastTrack->mHapticMaxAmplitude = NAN;
fastTrack->mGeneration++;
state->mFastTracksGen++;
state->mTrackMask = 1;
@@ -5103,6 +5187,7 @@
fastTrack->mFormat = track->mFormat;
fastTrack->mHapticPlaybackEnabled = track->getHapticPlaybackEnabled();
fastTrack->mHapticIntensity = track->getHapticIntensity();
+ fastTrack->mHapticMaxAmplitude = track->getHapticMaxAmplitude();
fastTrack->mGeneration++;
state->mTrackMask |= 1 << j;
didModify = true;
@@ -5356,7 +5441,7 @@
trackId,
AudioMixer::TRACK,
AudioMixer::MIXER_CHANNEL_MASK,
- (void *)(uintptr_t)(mChannelMask | mHapticChannelMask));
+ (void *)(uintptr_t)(mMixerChannelMask | mHapticChannelMask));
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
uint32_t reqSampleRate = proxy->getSampleRate();
@@ -5425,6 +5510,10 @@
trackId,
AudioMixer::TRACK,
AudioMixer::HAPTIC_INTENSITY, (void *)(uintptr_t)track->getHapticIntensity());
+ mAudioMixer->setParameter(
+ trackId,
+ AudioMixer::TRACK,
+ AudioMixer::HAPTIC_MAX_AMPLITUDE, (void *)(&(track->mHapticMaxAmplitude)));
// reset retry count
track->mRetryCount = kMaxTrackRetries;
@@ -5575,7 +5664,8 @@
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
- if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
+ if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0 ||
+ getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE) != 0) {
mEffectBufferValid = true;
}
@@ -6971,6 +7061,69 @@
MixerThread::cacheParameters_l();
}
+// ----------------------------------------------------------------------------
+
+AudioFlinger::VirtualizerStageThread::VirtualizerStageThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output,
+ audio_io_handle_t id,
+ bool systemReady,
+ audio_config_base_t *mixerConfig)
+ : MixerThread(audioFlinger, output, id, systemReady, VIRTUALIZER_STAGE, mixerConfig)
+{
+}
+
+void AudioFlinger::VirtualizerStageThread::checkOutputStageEffects()
+{
+ bool hasVirtualizer = false;
+ bool hasDownMixer = false;
+ sp<EffectHandle> finalDownMixer;
+ {
+ Mutex::Autolock _l(mLock);
+ sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE);
+ if (chain != 0) {
+ hasVirtualizer = chain->getEffectFromType_l(FX_IID_VIRTUALIZER_STAGE) != nullptr;
+ hasDownMixer = chain->getEffectFromType_l(EFFECT_UIID_DOWNMIX) != nullptr;
+ }
+
+ finalDownMixer = mFinalDownMixer;
+ mFinalDownMixer.clear();
+ }
+
+ if (hasVirtualizer) {
+ if (finalDownMixer != nullptr) {
+ int32_t ret;
+ finalDownMixer->disable(&ret);
+ }
+ finalDownMixer.clear();
+ } else if (!hasDownMixer) {
+ std::vector<effect_descriptor_t> descriptors;
+ status_t status = mAudioFlinger->mEffectsFactoryHal->getDescriptors(
+ EFFECT_UIID_DOWNMIX, &descriptors);
+ if (status != NO_ERROR) {
+ return;
+ }
+ ALOG_ASSERT(!descriptors.empty(),
+ "%s getDescriptors() returned no error but empty list", __func__);
+
+ finalDownMixer = createEffect_l(nullptr /*client*/, nullptr /*effectClient*/,
+ 0 /*priority*/, AUDIO_SESSION_OUTPUT_STAGE, &descriptors[0], nullptr /*enabled*/,
+ &status, false /*pinned*/, false /*probe*/);
+
+ if (finalDownMixer == nullptr || (status != NO_ERROR && status != ALREADY_EXISTS)) {
+ ALOGW("%s error creating downmixer %d", __func__, status);
+ finalDownMixer.clear();
+ } else {
+ int32_t ret;
+ finalDownMixer->enable(&ret);
+ }
+ }
+
+ {
+ Mutex::Autolock _l(mLock);
+ mFinalDownMixer = finalDownMixer;
+ }
+}
+
// ----------------------------------------------------------------------------
// Record
@@ -9278,7 +9431,7 @@
mActiveTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(mSessionId);
if (chain != 0) {
- chain->setStrategy(AudioSystem::getStrategyForStream(streamType()));
+ chain->setStrategy(getStrategyForStream(streamType()));
chain->incTrackCnt();
chain->incActiveTrackCnt();
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 16082a9..3001863 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -32,6 +32,7 @@
OFFLOAD, // Thread class is OffloadThread
MMAP_PLAYBACK, // Thread class for MMAP playback stream
MMAP_CAPTURE, // Thread class for MMAP capture stream
+ VIRTUALIZER_STAGE, //
// If you add any values here, also update ThreadBase::threadTypeToString()
};
@@ -53,7 +54,8 @@
CFG_EVENT_CREATE_AUDIO_PATCH,
CFG_EVENT_RELEASE_AUDIO_PATCH,
CFG_EVENT_UPDATE_OUT_DEVICE,
- CFG_EVENT_RESIZE_BUFFER
+ CFG_EVENT_RESIZE_BUFFER,
+ CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS
};
class ConfigEventData: public RefBase {
@@ -87,7 +89,13 @@
public:
virtual ~ConfigEvent() {}
- void dump(char *buffer, size_t size) { mData->dump(buffer, size); }
+ void dump(char *buffer, size_t size) {
+ snprintf(buffer, size, "Event type: %d\n", mType);
+ if (mData != nullptr) {
+ snprintf(buffer, size, "Data:\n");
+ mData->dump(buffer, size);
+ }
+ }
const int mType; // event type e.g. CFG_EVENT_IO
Mutex mLock; // mutex associated with mCond
@@ -110,7 +118,7 @@
mEvent(event), mPid(pid), mPortId(portId) {}
virtual void dump(char *buffer, size_t size) {
- snprintf(buffer, size, "IO event: event %d\n", mEvent);
+ snprintf(buffer, size, "- IO event: event %d\n", mEvent);
}
const audio_io_config_event mEvent;
@@ -133,7 +141,7 @@
mPid(pid), mTid(tid), mPrio(prio), mForApp(forApp) {}
virtual void dump(char *buffer, size_t size) {
- snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d, for app? %d\n",
+ snprintf(buffer, size, "- Prio event: pid %d, tid %d, prio %d, for app? %d\n",
mPid, mTid, mPrio, mForApp);
}
@@ -158,7 +166,7 @@
mKeyValuePairs(keyValuePairs) {}
virtual void dump(char *buffer, size_t size) {
- snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string());
+ snprintf(buffer, size, "- KeyValue: %s\n", mKeyValuePairs.string());
}
const String8 mKeyValuePairs;
@@ -181,7 +189,7 @@
mPatch(patch), mHandle(handle) {}
virtual void dump(char *buffer, size_t size) {
- snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+ snprintf(buffer, size, "- Patch handle: %u\n", mHandle);
}
const struct audio_patch mPatch;
@@ -205,7 +213,7 @@
mHandle(handle) {}
virtual void dump(char *buffer, size_t size) {
- snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+ snprintf(buffer, size, "- Patch handle: %u\n", mHandle);
}
audio_patch_handle_t mHandle;
@@ -227,7 +235,7 @@
mOutDevices(outDevices) {}
virtual void dump(char *buffer, size_t size) {
- snprintf(buffer, size, "Devices: %s", android::toString(mOutDevices).c_str());
+ snprintf(buffer, size, "- Devices: %s", android::toString(mOutDevices).c_str());
}
DeviceDescriptorBaseVector mOutDevices;
@@ -249,7 +257,7 @@
mMaxSharedAudioHistoryMs(maxSharedAudioHistoryMs) {}
virtual void dump(char *buffer, size_t size) {
- snprintf(buffer, size, "mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
+ snprintf(buffer, size, "- mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
}
int32_t mMaxSharedAudioHistoryMs;
@@ -265,6 +273,16 @@
virtual ~ResizeBufferConfigEvent() {}
};
+ class CheckOutputStageEffectsEvent : public ConfigEvent {
+ public:
+ CheckOutputStageEffectsEvent() :
+ ConfigEvent(CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS) {
+ }
+
+ virtual ~CheckOutputStageEffectsEvent() {}
+ };
+
+
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
@@ -290,8 +308,11 @@
// dynamic externally-visible
uint32_t sampleRate() const { return mSampleRate; }
audio_channel_mask_t channelMask() const { return mChannelMask; }
+ virtual audio_channel_mask_t mixerChannelMask() const { return mChannelMask; }
+
audio_format_t format() const { return mHALFormat; }
uint32_t channelCount() const { return mChannelCount; }
+
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
// and returns the [normal mix] buffer's frame count.
virtual size_t frameCount() const = 0;
@@ -330,7 +351,11 @@
status_t sendUpdateOutDeviceConfigEvent(
const DeviceDescriptorBaseVector& outDevices);
void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs);
+ void sendCheckOutputStageEffectsEvent();
+ void sendCheckOutputStageEffectsEvent_l();
+
void processConfigEvents_l();
+ virtual void setCheckOutputStageEffects() {}
virtual void cacheParameters_l() = 0;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle) = 0;
@@ -574,6 +599,8 @@
return INVALID_OPERATION;
}
+ product_strategy_t getStrategyForStream(audio_stream_type_t stream) const;
+
virtual void dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
{ }
virtual void dumpTracks_l(int fd __unused, const Vector<String16>& args __unused) { }
@@ -824,7 +851,8 @@
static const nsecs_t kMaxNextBufferDelayNs = 100000000;
PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, type_t type, bool systemReady);
+ audio_io_handle_t id, type_t type, bool systemReady,
+ audio_config_base_t *mixerConfig = nullptr);
virtual ~PlaybackThread();
// Thread virtuals
@@ -881,6 +909,8 @@
mActiveTracks.updatePowerState(this, true /* force */);
}
+ virtual void checkOutputStageEffects() {}
+
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
@@ -973,6 +1003,10 @@
virtual size_t frameCount() const { return mNormalFrameCount; }
+ audio_channel_mask_t mixerChannelMask() const override {
+ return mMixerChannelMask;
+ }
+
status_t getTimestamp_l(AudioTimestamp& timestamp);
void addPatchTrack(const sp<PatchTrack>& track);
@@ -1015,6 +1049,9 @@
PlaybackThread::Track* getTrackById_l(audio_port_handle_t trackId);
+ bool hasMixer() const {
+ return mType == MIXER || mType == DUPLICATING || mType == VIRTUALIZER_STAGE;
+ }
protected:
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
@@ -1101,6 +1138,9 @@
// haptic playback.
audio_channel_mask_t mHapticChannelMask = AUDIO_CHANNEL_NONE;
uint32_t mHapticChannelCount = 0;
+
+ audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
+
private:
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
// PlaybackThread needs to find out if master-muted, it checks it's local
@@ -1134,6 +1174,9 @@
// Cache various calculated values, at threadLoop() entry and after a parameter change
virtual void cacheParameters_l();
+ void setCheckOutputStageEffects() override {
+ mCheckOutputStageEffects.store(true);
+ }
virtual uint32_t correctLatency_l(uint32_t latency) const;
@@ -1314,6 +1357,8 @@
// audio patch used by the downstream software patch.
// Only used if ThreadBase::mIsMsdDevice is true.
struct audio_patch mDownStreamPatch;
+
+ std::atomic_bool mCheckOutputStageEffects{};
};
class MixerThread : public PlaybackThread {
@@ -1322,7 +1367,8 @@
AudioStreamOut* output,
audio_io_handle_t id,
bool systemReady,
- type_t type = MIXER);
+ type_t type = MIXER,
+ audio_config_base_t *mixerConfig = nullptr);
virtual ~MixerThread();
// Thread virtuals
@@ -1611,6 +1657,24 @@
}
};
+class VirtualizerStageThread : public MixerThread {
+public:
+ VirtualizerStageThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output,
+ audio_io_handle_t id,
+ bool systemReady,
+ audio_config_base_t *mixerConfig);
+ ~VirtualizerStageThread() override {}
+
+ bool hasFastMixer() const override { return false; }
+
+protected:
+ void checkOutputStageEffects() override;
+
+private:
+ sp<EffectHandle> mFinalDownMixer;
+};
+
// record thread
class RecordThread : public ThreadBase
{
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 2e49e71..6b11d9a 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -359,7 +359,8 @@
// The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 1f9b535..7c7f02d 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -362,7 +362,8 @@
const struct audio_port_config *srcConfig = NULL) const;
virtual void toAudioPort(struct audio_port_v7 *port) const;
- status_t open(const audio_config_t *config,
+ status_t open(const audio_config_t *halConfig,
+ const audio_config_base_t *mixerConfig,
const DeviceVector &devices,
audio_stream_type_t stream,
audio_output_flags_t flags,
@@ -423,6 +424,7 @@
uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
audio_session_t mDirectClientSession; // session id of the direct output client
bool mPendingReopenToQueryProfiles = false;
+ audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
};
// Audio output driven by an input device directly.
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index cf1f64c..9837336 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -202,6 +202,20 @@
{AUDIO_FORMAT_AC4, {}}};
}
+ //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+ // until then, use DEEP_BUFFER+FAST flag combo to indicate the virtualizer stage output profile
+ void convertVirtualizerStageFlag()
+ {
+ for (const auto& hwModule : mHwModules) {
+ for (const auto& curProfile : hwModule->getOutputProfiles()) {
+ if (curProfile->getFlags()
+ == (AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
+ curProfile->setFlags(AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE);
+ }
+ }
+ }
+ }
+
private:
static const constexpr char* const kDefaultEngineLibraryNameSuffix = "default";
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 20b4044..58d05c6 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -168,6 +168,10 @@
DeviceVector getDevicesFromDeviceTypeAddrVec(
const AudioDeviceTypeAddrVector& deviceTypeAddrVector) const;
+ // Return the device vector that contains device descriptor whose AudioDeviceTypeAddr appears
+ // in the given AudioDeviceTypeAddrVector
+ AudioDeviceTypeAddrVector toTypeAddrVector() 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/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 6b08f7c..6c3386f 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -491,7 +491,8 @@
return true;
}
-status_t SwAudioOutputDescriptor::open(const audio_config_t *config,
+status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
+ const audio_config_base_t *mixerConfig,
const DeviceVector &devices,
audio_stream_type_t stream,
audio_output_flags_t flags,
@@ -504,45 +505,62 @@
"with the requested devices, all device types: %s",
__func__, dumpDeviceTypes(devices.types()).c_str());
- audio_config_t lConfig;
- if (config == nullptr) {
- lConfig = AUDIO_CONFIG_INITIALIZER;
- lConfig.sample_rate = mSamplingRate;
- lConfig.channel_mask = mChannelMask;
- lConfig.format = mFormat;
+ audio_config_t lHalConfig;
+ if (halConfig == nullptr) {
+ lHalConfig = AUDIO_CONFIG_INITIALIZER;
+ lHalConfig.sample_rate = mSamplingRate;
+ lHalConfig.channel_mask = mChannelMask;
+ lHalConfig.format = mFormat;
} else {
- lConfig = *config;
+ lHalConfig = *halConfig;
}
// if the selected profile is offloaded and no offload info was specified,
// create a default one
if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
- lConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
+ lHalConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- lConfig.offload_info = AUDIO_INFO_INITIALIZER;
- lConfig.offload_info.sample_rate = lConfig.sample_rate;
- lConfig.offload_info.channel_mask = lConfig.channel_mask;
- lConfig.offload_info.format = lConfig.format;
- lConfig.offload_info.stream_type = stream;
- lConfig.offload_info.duration_us = -1;
- lConfig.offload_info.has_video = true; // conservative
- lConfig.offload_info.is_streaming = true; // likely
- lConfig.offload_info.encapsulation_mode = lConfig.offload_info.encapsulation_mode;
- lConfig.offload_info.content_id = lConfig.offload_info.content_id;
- lConfig.offload_info.sync_id = lConfig.offload_info.sync_id;
+ lHalConfig.offload_info = AUDIO_INFO_INITIALIZER;
+ lHalConfig.offload_info.sample_rate = lHalConfig.sample_rate;
+ lHalConfig.offload_info.channel_mask = lHalConfig.channel_mask;
+ lHalConfig.offload_info.format = lHalConfig.format;
+ lHalConfig.offload_info.stream_type = stream;
+ lHalConfig.offload_info.duration_us = -1;
+ lHalConfig.offload_info.has_video = true; // conservative
+ lHalConfig.offload_info.is_streaming = true; // likely
+ lHalConfig.offload_info.encapsulation_mode = lHalConfig.offload_info.encapsulation_mode;
+ lHalConfig.offload_info.content_id = lHalConfig.offload_info.content_id;
+ lHalConfig.offload_info.sync_id = lHalConfig.offload_info.sync_id;
+ }
+
+ audio_config_base_t lMixerConfig;
+ if (mixerConfig == nullptr) {
+ lMixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+ lMixerConfig.sample_rate = lHalConfig.sample_rate;
+ lMixerConfig.channel_mask = lHalConfig.channel_mask;
+ lMixerConfig.format = lHalConfig.format;
+ } else {
+ lMixerConfig = *mixerConfig;
}
mFlags = (audio_output_flags_t)(mFlags | flags);
+ //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+ audio_output_flags_t halFlags = mFlags;
+ if ((mFlags & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0) {
+ halFlags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+ }
+
ALOGV("opening output for device %s profile %p name %s",
mDevices.toString().c_str(), mProfile.get(), mProfile->getName().c_str());
status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
output,
- &lConfig,
+ &lHalConfig,
+ &lMixerConfig,
device,
&mLatency,
- mFlags);
+ halFlags);
if (status == NO_ERROR) {
LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
@@ -550,9 +568,10 @@
"selected device %s for opening",
__FUNCTION__, *output, devices.toString().c_str(),
device->toString().c_str());
- mSamplingRate = lConfig.sample_rate;
- mChannelMask = lConfig.channel_mask;
- mFormat = lConfig.format;
+ mSamplingRate = lHalConfig.sample_rate;
+ mChannelMask = lHalConfig.channel_mask;
+ mFormat = lHalConfig.format;
+ mMixerChannelMask = lMixerConfig.channel_mask;
mId = PolicyAudioPort::getNextUniqueId();
mIoHandle = *output;
mProfile->curOpenCount++;
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index a92d31e..1722032 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -451,6 +451,14 @@
return devices;
}
+AudioDeviceTypeAddrVector DeviceVector::toTypeAddrVector() const {
+ AudioDeviceTypeAddrVector result;
+ for (const auto& device : *this) {
+ result.push_back(AudioDeviceTypeAddr(device->type(), device->address()));
+ }
+ return result;
+}
+
void DeviceVector::replaceDevicesByType(
audio_devices_t typeToRemove, const DeviceVector &devicesToAdd) {
DeviceVector devicesToRemove = getDevicesFromType(typeToRemove);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index cc2d8e8..a20612e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -246,9 +246,11 @@
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
// close unused outputs after device disconnection or direct outputs that have
// been opened by checkOutputsForDevice() to query dynamic parameters
- if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
- (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
- (desc->mDirectOpenCount == 0))) {
+ if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)
+ || (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+ (desc->mDirectOpenCount == 0))
+ || (((desc->mFlags & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0) &&
+ (desc != mVirtualizerStageOutput))) {
clearAudioSourcesForOutput(output);
closeOutput(output);
}
@@ -925,6 +927,36 @@
return profile;
}
+sp<IOProfile> AudioPolicyManager::getVirtualizerStageOutputProfile(
+ const audio_config_t *config __unused, const AudioDeviceTypeAddrVector &devices,
+ bool forOpening) const
+{
+ for (const auto& hwModule : mHwModules) {
+ for (const auto& curProfile : hwModule->getOutputProfiles()) {
+ if (curProfile->getFlags() != AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) {
+ continue;
+ }
+ // reject profiles not corresponding to a device currently available
+ DeviceVector supportedDevices = curProfile->getSupportedDevices();
+ if (!mAvailableOutputDevices.containsAtLeastOne(supportedDevices)) {
+ continue;
+ }
+ if (!devices.empty()) {
+ if (supportedDevices.getDevicesFromDeviceTypeAddrVec(devices).size()
+ != devices.size()) {
+ continue;
+ }
+ }
+ if (forOpening && !curProfile->canOpenNewIo()) {
+ continue;
+ }
+ ALOGV("%s found profile %s", __func__, curProfile->getName().c_str());
+ return curProfile;
+ }
+ }
+ return nullptr;
+}
+
audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream)
{
DeviceVector devices = mEngine->getOutputDevicesForStream(stream, false /*fromCache*/);
@@ -1094,7 +1126,7 @@
*output = AUDIO_IO_HANDLE_NONE;
if (!msdDevices.isEmpty()) {
- *output = getOutputForDevices(msdDevices, session, *stream, config, flags);
+ *output = getOutputForDevices(msdDevices, session, resultAttr, config, flags);
if (*output != AUDIO_IO_HANDLE_NONE && setMsdOutputPatches(&outputDevices) == NO_ERROR) {
ALOGV("%s() Using MSD devices %s instead of devices %s",
__func__, msdDevices.toString().c_str(), outputDevices.toString().c_str());
@@ -1103,7 +1135,7 @@
}
}
if (*output == AUDIO_IO_HANDLE_NONE) {
- *output = getOutputForDevices(outputDevices, session, *stream, config,
+ *output = getOutputForDevices(outputDevices, session, resultAttr, config,
flags, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
}
if (*output == AUDIO_IO_HANDLE_NONE) {
@@ -1265,7 +1297,8 @@
// all MSD patches to prioritize this request over any active output on MSD.
releaseMsdOutputPatches(devices);
- status_t status = outputDesc->open(config, devices, stream, flags, output);
+ status_t status =
+ outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, flags, output);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
@@ -1300,7 +1333,7 @@
audio_io_handle_t AudioPolicyManager::getOutputForDevices(
const DeviceVector &devices,
audio_session_t session,
- audio_stream_type_t stream,
+ const audio_attributes_t *attr,
const audio_config_t *config,
audio_output_flags_t *flags,
bool forceMutingHaptic)
@@ -1322,6 +1355,9 @@
if ((*flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
*flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
+
+ audio_stream_type_t stream = mEngine->getStreamTypeForAttributes(*attr);
+
// only allow deep buffering for music stream type
if (stream != AUDIO_STREAM_MUSIC) {
*flags = (audio_output_flags_t)(*flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
@@ -1341,6 +1377,11 @@
ALOGV("Set VoIP and Direct output flags for PCM format");
}
+ if (mVirtualizerStageOutput != nullptr
+ && canBeVirtualized(attr, config, devices.toTypeAddrVector())) {
+ return mVirtualizerStageOutput->mIoHandle;
+ }
+
audio_config_t directConfig = *config;
directConfig.channel_mask = channelMask;
status_t status = openDirectOutput(stream, session, &directConfig, *flags, devices, &output);
@@ -4802,6 +4843,136 @@
return source;
}
+bool AudioPolicyManager::canBeVirtualized(const audio_attributes_t *attr,
+ const audio_config_t *config,
+ const AudioDeviceTypeAddrVector &devices) const
+{
+ // The caller can have the audio attributes criteria ignored by either passing a null ptr or
+ // the AUDIO_ATTRIBUTES_INITIALIZER value.
+ // If attributes are specified, current policy is to only allow virtualization for media
+ // and game usages.
+ if (attr != nullptr && *attr != AUDIO_ATTRIBUTES_INITIALIZER &&
+ attr->usage != AUDIO_USAGE_MEDIA && attr->usage != AUDIO_USAGE_GAME) {
+ return false;
+ }
+
+ // The caller can have the devices criteria ignored by passing and empty vector, and
+ // getVirtualizerStageOutputProfile() will ignore the devices when looking for a match.
+ // Otherwise an output profile supporting a virtualizer stage effect that can be routed
+ // to the specified devices must exist.
+ sp<IOProfile> profile =
+ getVirtualizerStageOutputProfile(config, devices, false /*forOpening*/);
+ if (profile == nullptr) {
+ return false;
+ }
+
+ // The caller can have the audio config criteria ignored by either passing a null ptr or
+ // the AUDIO_CONFIG_INITIALIZER value.
+ // If an audio config is specified, current policy is to only allow virtualization for
+ // 5.1, 7.1and 7.1.4 audio.
+ // If the virtualizer stage output is already opened, only channel masks included in the
+ // virtualizer stage output mixer channel mask are allowed.
+ if (config != nullptr && *config != AUDIO_CONFIG_INITIALIZER) {
+ if (config->channel_mask != AUDIO_CHANNEL_OUT_5POINT1
+ && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1
+ && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1POINT4) {
+ return false;
+ }
+ if (mVirtualizerStageOutput != nullptr) {
+ if ((config->channel_mask & mVirtualizerStageOutput->mMixerChannelMask)
+ != config->channel_mask) {
+ return false;
+ }
+ }
+ }
+
+ return true;
+}
+
+void AudioPolicyManager::checkVirtualizerClientRoutes() {
+ std::set<audio_stream_type_t> streamsToInvalidate;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const sp<SwAudioOutputDescriptor>& outputDescriptor = mOutputs[i];
+ for (const sp<TrackClientDescriptor>& client : outputDescriptor->getClientIterable()) {
+ audio_attributes_t attr = client->attributes();
+ DeviceVector devices = mEngine->getOutputDevicesForAttributes(attr, nullptr, false);
+ AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
+ audio_config_base_t clientConfig = client->config();
+ audio_config_t config = audio_config_initializer(&clientConfig);
+ if (canBeVirtualized(&attr, &config, devicesTypeAddress)) {
+ streamsToInvalidate.insert(client->stream());
+ }
+ }
+ }
+
+ for (audio_stream_type_t stream : streamsToInvalidate) {
+ mpClientInterface->invalidateStream(stream);
+ }
+}
+
+status_t AudioPolicyManager::getVirtualizerStageOutput(const audio_config_base_t *mixerConfig,
+ const audio_attributes_t *attr,
+ audio_io_handle_t *output) {
+ *output = AUDIO_IO_HANDLE_NONE;
+
+ if (mVirtualizerStageOutput != nullptr) {
+ return INVALID_OPERATION;
+ }
+
+ DeviceVector devices = mEngine->getOutputDevicesForAttributes(*attr, nullptr, false);
+ AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
+ audio_config_t *configPtr = nullptr;
+ audio_config_t config;
+ if (mixerConfig != nullptr) {
+ config = audio_config_initializer(mixerConfig);
+ configPtr = &config;
+ }
+ if (!canBeVirtualized(attr, configPtr, devicesTypeAddress)) {
+ return BAD_VALUE;
+ }
+
+ sp<IOProfile> profile =
+ getVirtualizerStageOutputProfile(configPtr, devicesTypeAddress, true /*forOpening*/);
+ if (profile == nullptr) {
+ return BAD_VALUE;
+ }
+
+ mVirtualizerStageOutput = new SwAudioOutputDescriptor(profile, mpClientInterface);
+ status_t status = mVirtualizerStageOutput->open(nullptr, mixerConfig, devices,
+ mEngine->getStreamTypeForAttributes(*attr),
+ AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE, output);
+ if (status != NO_ERROR) {
+ ALOGV("%s failed opening output: status %d, output %d", __func__, status, *output);
+ if (*output != AUDIO_IO_HANDLE_NONE) {
+ mVirtualizerStageOutput->close();
+ }
+ mVirtualizerStageOutput.clear();
+ *output = AUDIO_IO_HANDLE_NONE;
+ return status;
+ }
+
+ checkVirtualizerClientRoutes();
+
+ addOutput(*output, mVirtualizerStageOutput);
+ mPreviousOutputs = mOutputs;
+ mpClientInterface->onAudioPortListUpdate();
+
+ ALOGV("%s returns new virtualizer stage output %d", __func__, *output);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseVirtualizerStageOutput(audio_io_handle_t output) {
+ if (mVirtualizerStageOutput == nullptr) {
+ return INVALID_OPERATION;
+ }
+ if (mVirtualizerStageOutput->mIoHandle != output) {
+ return BAD_VALUE;
+ }
+ closeOutput(output);
+ mVirtualizerStageOutput.clear();
+ return NO_ERROR;
+}
+
// ----------------------------------------------------------------------------
// AudioPolicyManager
// ----------------------------------------------------------------------------
@@ -4851,6 +5022,8 @@
ALOGE("could not load audio policy configuration file, setting defaults");
getConfig().setDefault();
}
+ //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+ getConfig().convertVirtualizerStageFlag();
}
status_t AudioPolicyManager::initialize() {
@@ -4990,7 +5163,8 @@
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = outputDesc->open(nullptr, DeviceVector(supportedDevice),
+ status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */,
+ DeviceVector(supportedDevice),
AUDIO_STREAM_DEFAULT,
AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
@@ -5012,7 +5186,8 @@
outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
mPrimaryOutput = outputDesc;
}
- if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+ if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0
+ || (outProfile->getFlags() & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0 ) {
outputDesc->close();
} else {
addOutput(output, outputDesc);
@@ -6995,7 +7170,7 @@
}
sp<SwAudioOutputDescriptor> desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = desc->open(nullptr, devices,
+ status_t status = desc->open(nullptr /* halConfig */, nullptr /* mixerConfig */, devices,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
return nullptr;
@@ -7025,7 +7200,7 @@
config.offload_info.channel_mask = config.channel_mask;
config.offload_info.format = config.format;
- status = desc->open(&config, devices,
+ status = desc->open(&config, nullptr /* mixerConfig */, devices,
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
return nullptr;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 98f96d1..8668f5e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -356,6 +356,16 @@
BAD_VALUE : NO_ERROR;
}
+ virtual bool canBeVirtualized(const audio_attributes_t *attr,
+ const audio_config_t *config,
+ const AudioDeviceTypeAddrVector &devices) const;
+
+ virtual status_t getVirtualizerStageOutput(const audio_config_base_t *config,
+ const audio_attributes_t *attr,
+ audio_io_handle_t *output);
+
+ virtual status_t releaseVirtualizerStageOutput(audio_io_handle_t output);
+
bool isCallScreenModeSupported() override;
void onNewAudioModulesAvailable() override;
@@ -797,6 +807,8 @@
sp<SwAudioOutputDescriptor> mPrimaryOutput; // primary output descriptor
// list of descriptors for outputs currently opened
+ sp<SwAudioOutputDescriptor> mVirtualizerStageOutput;
+
SwAudioOutputCollection mOutputs;
// copy of mOutputs before setDeviceConnectionState() opens new outputs
// reset to mOutputs when updateDevicesAndOutputs() is called.
@@ -933,7 +945,7 @@
audio_io_handle_t getOutputForDevices(
const DeviceVector &devices,
audio_session_t session,
- audio_stream_type_t stream,
+ const audio_attributes_t *attr,
const audio_config_t *config,
audio_output_flags_t *flags,
bool forceMutingHaptic = false);
@@ -948,6 +960,13 @@
audio_output_flags_t flags,
const DeviceVector &devices,
audio_io_handle_t *output);
+
+ sp<IOProfile> getVirtualizerStageOutputProfile(const audio_config_t *config,
+ const AudioDeviceTypeAddrVector &devices,
+ bool forOpening) const;
+
+ void checkVirtualizerClientRoutes();
+
/**
* @brief getInputForDevice selects an input handle for a given input device and
* requester context
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index cd53073..79252d4 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -40,7 +40,8 @@
status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags)
@@ -55,14 +56,17 @@
media::OpenOutputResponse response;
request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
- request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
+ request.halConfig = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*halConfig));
+ request.mixerConfig =
+ VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_base_t_AudioConfigBase(*mixerConfig));
request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
status_t status = af->openOutput(request, &response);
if (status == OK) {
*output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
- *config = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
+ *halConfig =
+ VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
*latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<uint32_t>(response.latencyMs));
}
return status;
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 4d0e1f1..3c757b3 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -127,6 +127,7 @@
loadAudioPolicyManager();
mAudioPolicyManager = mCreateAudioPolicyManager(mAudioPolicyClient);
}
+
// load audio processing modules
sp<AudioPolicyEffects> audioPolicyEffects = new AudioPolicyEffects();
sp<UidPolicy> uidPolicy = new UidPolicy(this);
@@ -139,6 +140,8 @@
}
uidPolicy->registerSelf();
sensorPrivacyPolicy->registerSelf();
+
+ AudioSystem::audioPolicyReady();
}
void AudioPolicyService::unloadAudioPolicyManager()
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 3b77ed8..c0fbde5 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -659,7 +659,8 @@
// The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t *config,
+ audio_config_t *halConfig,
+ audio_config_base_t *mixerConfig,
const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags);
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index f7b0565..84b40d2 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -37,7 +37,8 @@
status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
- audio_config_t * /*config*/,
+ audio_config_t * /*halConfig*/,
+ audio_config_base_t * /*mixerConfig*/,
const sp<DeviceDescriptorBase>& /*device*/,
uint32_t * /*latencyMs*/,
audio_output_flags_t /*flags*/) override {
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index 1384864..4e0735b 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -30,7 +30,8 @@
}
status_t openOutput(audio_module_handle_t /*module*/,
audio_io_handle_t* /*output*/,
- audio_config_t* /*config*/,
+ audio_config_t* /*halConfig*/,
+ audio_config_base_t* /*mixerConfig*/,
const sp<DeviceDescriptorBase>& /*device*/,
uint32_t* /*latencyMs*/,
audio_output_flags_t /*flags*/) override { return NO_INIT; }
diff --git a/services/mediametrics/Android.bp b/services/mediametrics/Android.bp
index 5989181..0351d2d 100644
--- a/services/mediametrics/Android.bp
+++ b/services/mediametrics/Android.bp
@@ -148,7 +148,8 @@
"statsd_mediaparser.cpp",
"statsd_nuplayer.cpp",
"statsd_recorder.cpp",
- "StringUtils.cpp"
+ "StringUtils.cpp",
+ "ValidateId.cpp",
],
proto: {
diff --git a/services/mediametrics/AudioAnalytics.cpp b/services/mediametrics/AudioAnalytics.cpp
index 11ec993..a0dcb55 100644
--- a/services/mediametrics/AudioAnalytics.cpp
+++ b/services/mediametrics/AudioAnalytics.cpp
@@ -28,6 +28,7 @@
#include "AudioTypes.h" // string to int conversions
#include "MediaMetricsService.h" // package info
#include "StringUtils.h"
+#include "ValidateId.h"
#define PROP_AUDIO_ANALYTICS_CLOUD_ENABLED "persist.audio.analytics.cloud.enabled"
@@ -562,7 +563,7 @@
const auto flagsForStats = types::lookup<types::INPUT_FLAG, short_enum_type_t>(flags);
const auto sourceForStats = types::lookup<types::SOURCE_TYPE, short_enum_type_t>(source);
// Android S
- const auto logSessionIdForStats = stringutils::sanitizeLogSessionId(logSessionId);
+ const auto logSessionIdForStats = ValidateId::get()->validateId(logSessionId);
LOG(LOG_LEVEL) << "key:" << key
<< " id:" << id
@@ -717,7 +718,7 @@
types::lookup<types::TRACK_TRAITS, short_enum_type_t>(traits);
const auto usageForStats = types::lookup<types::USAGE, short_enum_type_t>(usage);
// Android S
- const auto logSessionIdForStats = stringutils::sanitizeLogSessionId(logSessionId);
+ const auto logSessionIdForStats = ValidateId::get()->validateId(logSessionId);
LOG(LOG_LEVEL) << "key:" << key
<< " id:" << id
diff --git a/services/mediametrics/LruSet.h b/services/mediametrics/LruSet.h
new file mode 100644
index 0000000..1f0ab60
--- /dev/null
+++ b/services/mediametrics/LruSet.h
@@ -0,0 +1,123 @@
+/*
+ * Copyright (C) 2021 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
+
+#include <list>
+#include <sstream>
+#include <unordered_map>
+
+namespace android::mediametrics {
+
+/**
+ * LruSet keeps a set of the last "Size" elements added or accessed.
+ *
+ * (Lru stands for least-recently-used eviction policy).
+ *
+ * Runs in O(1) time for add, remove, and check. Internally implemented
+ * with an unordered_map and a list. In order to remove elements,
+ * a list iterator is stored in the unordered_map
+ * (noting that std::list::erase() contractually
+ * does not affect iterators other than the one erased).
+ */
+
+template <typename T>
+class LruSet {
+ const size_t mMaxSize;
+ std::list<T> mAccessOrder; // front is the most recent, back is the oldest.
+ // item T with its access order iterator.
+ std::unordered_map<T, typename std::list<T>::iterator> mMap;
+
+public:
+ /**
+ * Constructs a LruSet which checks whether the element was
+ * accessed or added recently.
+ *
+ * The parameter maxSize is used to cap growth of LruSet;
+ * eviction is based on least recently used LRU.
+ * If maxSize is zero, the LruSet contains no elements
+ * and check() always returns false.
+ *
+ * \param maxSize the maximum number of elements that are tracked.
+ */
+ explicit LruSet(size_t maxSize) : mMaxSize(maxSize) {}
+
+ /**
+ * Returns the number of entries in the LruSet.
+ *
+ * This is a number between 0 and maxSize.
+ */
+ size_t size() const {
+ return mMap.size();
+ }
+
+ /** Clears the container contents. */
+ void clear() {
+ mMap.clear();
+ mAccessOrder.clear();
+ }
+
+ /** Returns a string dump of the last n entries. */
+ std::string dump(size_t n) const {
+ std::stringstream ss;
+ auto it = mAccessOrder.cbegin();
+ for (size_t i = 0; i < n && it != mAccessOrder.cend(); ++i) {
+ ss << *it++ << "\n";
+ }
+ return ss.str();
+ }
+
+ /** Adds a new item to the set. */
+ void add(const T& t) {
+ if (mMaxSize == 0) return;
+ auto it = mMap.find(t);
+ if (it != mMap.end()) { // already exists.
+ mAccessOrder.erase(it->second); // move-to-front on the chronologically ordered list.
+ } else if (mAccessOrder.size() >= mMaxSize) {
+ const T last = mAccessOrder.back();
+ mAccessOrder.pop_back();
+ mMap.erase(last);
+ }
+ mAccessOrder.push_front(t);
+ mMap[t] = mAccessOrder.begin();
+ }
+
+ /**
+ * Removes an item from the set.
+ *
+ * \param t item to be removed.
+ * \return false if the item doesn't exist.
+ */
+ bool remove(const T& t) {
+ auto it = mMap.find(t);
+ if (it == mMap.end()) return false;
+ mAccessOrder.erase(it->second);
+ mMap.erase(it);
+ return true;
+ }
+
+ /** Returns true if t is present (and moves the access order of t to the front). */
+ bool check(const T& t) { // not const, as it adjusts the least-recently-used order.
+ auto it = mMap.find(t);
+ if (it == mMap.end()) return false;
+ mAccessOrder.erase(it->second);
+ mAccessOrder.push_front(it->first);
+ it->second = mAccessOrder.begin();
+ return true;
+ }
+};
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/MediaMetricsService.cpp b/services/mediametrics/MediaMetricsService.cpp
index 1d64878..35e0ae4 100644
--- a/services/mediametrics/MediaMetricsService.cpp
+++ b/services/mediametrics/MediaMetricsService.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include "MediaMetricsService.h"
+#include "ValidateId.h"
#include "iface_statsd.h"
#include <pwd.h> //getpwuid
@@ -204,6 +205,15 @@
// now attach either the item or its dup to a const shared pointer
std::shared_ptr<const mediametrics::Item> sitem(release ? item : item->dup());
+ // register log session ids with singleton.
+ if (startsWith(item->getKey(), "metrics.manager")) {
+ std::string logSessionId;
+ if (item->get("logSessionId", &logSessionId)
+ && mediametrics::stringutils::isLogSessionId(logSessionId.c_str())) {
+ mediametrics::ValidateId::get()->registerId(logSessionId);
+ }
+ }
+
(void)mAudioAnalytics.submit(sitem, isTrusted);
(void)dump2Statsd(sitem, mStatsdLog); // failure should be logged in function.
@@ -309,6 +319,9 @@
result << "-- some lines may be truncated --\n";
}
+ result << "LogSessionId:\n"
+ << mediametrics::ValidateId::get()->dump();
+
// Dump the statsd atoms we sent out.
result << "Statsd atoms:\n"
<< mStatsdLog->dumpToString(" " /* prefix */,
diff --git a/services/mediametrics/ValidateId.cpp b/services/mediametrics/ValidateId.cpp
new file mode 100644
index 0000000..0cc8593
--- /dev/null
+++ b/services/mediametrics/ValidateId.cpp
@@ -0,0 +1,65 @@
+/*
+ * Copyright (C) 2021 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 "MediaMetricsService" // not ValidateId
+#include <utils/Log.h>
+
+#include "ValidateId.h"
+
+namespace android::mediametrics {
+
+std::string ValidateId::dump() const
+{
+ std::stringstream ss;
+ ss << "Entries:" << mIdSet.size() << " InvalidIds:" << mInvalidIds << "\n";
+ ss << mIdSet.dump(10);
+ return ss.str();
+}
+
+void ValidateId::registerId(const std::string& id)
+{
+ if (id.empty()) return;
+ if (!mediametrics::stringutils::isLogSessionId(id.c_str())) {
+ ALOGW("%s: rejecting malformed id %s", __func__, id.c_str());
+ return;
+ }
+ ALOGV("%s: registering %s", __func__, id.c_str());
+ mIdSet.add(id);
+}
+
+const std::string& ValidateId::validateId(const std::string& id)
+{
+ static const std::string empty{};
+ if (id.empty()) return empty;
+
+ // reject because the id is malformed
+ if (!mediametrics::stringutils::isLogSessionId(id.c_str())) {
+ ALOGW("%s: rejecting malformed id %s", __func__, id.c_str());
+ ++mInvalidIds;
+ return empty;
+ }
+
+ // reject because the id is unregistered
+ if (!mIdSet.check(id)) {
+ ALOGW("%s: rejecting unregistered id %s", __func__, id.c_str());
+ ++mInvalidIds;
+ return empty;
+ }
+ return id;
+}
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/ValidateId.h b/services/mediametrics/ValidateId.h
new file mode 100644
index 0000000..166b39a
--- /dev/null
+++ b/services/mediametrics/ValidateId.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (C) 2021 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
+
+#include "LruSet.h"
+#include "StringUtils.h"
+#include "Wrap.h"
+
+namespace android::mediametrics {
+
+/*
+ * ValidateId is used to check whether the log session id is properly formed
+ * and has been registered (i.e. from the Java MediaMetricsManagerService).
+ *
+ * The default memory window to track registered ids is set to SINGLETON_LRU_SET_SIZE.
+ *
+ * This class is not thread-safe, but the singleton returned by get() uses LockWrap<>
+ * to ensure thread-safety.
+ */
+class ValidateId {
+ mediametrics::LruSet<std::string> mIdSet;
+ size_t mInvalidIds = 0; // count invalid ids encountered.
+public:
+ /** Creates a ValidateId object with size memory window. */
+ explicit ValidateId(size_t size) : mIdSet{size} {}
+
+ /** Returns a string dump of recent contents and stats. */
+ std::string dump() const;
+
+ /**
+ * Registers the id string.
+ *
+ * If id string is malformed (not 16 Base64Url chars), it is ignored.
+ * Once registered, calling validateId() will return id (instead of the empty string).
+ * ValidateId may "forget" the id after not encountering it within the past N ids,
+ * where N is the size set in the constructor.
+ *
+ * param id string (from MediaMetricsManagerService).
+ */
+ void registerId(const std::string& id);
+
+ /**
+ * Returns the empty string if id string is malformed (not 16 Base64Url chars)
+ * or if id string has not been seen (in the recent size ids);
+ * otherwise it returns the same id parameter.
+ *
+ * \param id string (to be sent to statsd).
+ */
+ const std::string& validateId(const std::string& id);
+
+ /** Singleton set size */
+ static inline constexpr size_t SINGLETON_LRU_SET_SIZE = 2000;
+
+ using LockedValidateId = mediametrics::LockWrap<ValidateId>;
+ /**
+ * Returns a singleton locked ValidateId object that is thread-safe using LockWrap<>.
+ *
+ * The Singleton ValidateId object is created with size LRU_SET_SIZE (during first call).
+ */
+ static inline LockedValidateId& get() {
+ static LockedValidateId privateSet{SINGLETON_LRU_SET_SIZE};
+ return privateSet;
+ }
+};
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/statsd_audiorecord.cpp b/services/mediametrics/statsd_audiorecord.cpp
index 41efcaa..c53b6f3 100644
--- a/services/mediametrics/statsd_audiorecord.cpp
+++ b/services/mediametrics/statsd_audiorecord.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
#include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
#include "iface_statsd.h"
@@ -143,8 +143,7 @@
// log_session_id (string)
std::string logSessionId;
(void)item->getString("android.media.audiorecord.logSessionId", &logSessionId);
- const auto log_session_id =
- mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+ const auto log_session_id = mediametrics::ValidateId::get()->validateId(logSessionId);
android::util::BytesField bf_serialized( serialized.c_str(), serialized.size());
int result = android::util::stats_write(android::util::MEDIAMETRICS_AUDIORECORD_REPORTED,
diff --git a/services/mediametrics/statsd_audiotrack.cpp b/services/mediametrics/statsd_audiotrack.cpp
index 59627ae..707effd 100644
--- a/services/mediametrics/statsd_audiotrack.cpp
+++ b/services/mediametrics/statsd_audiotrack.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
#include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
#include "iface_statsd.h"
@@ -137,8 +137,7 @@
// log_session_id (string)
std::string logSessionId;
(void)item->getString("android.media.audiotrack.logSessionId", &logSessionId);
- const auto log_session_id =
- mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+ const auto log_session_id = mediametrics::ValidateId::get()->validateId(logSessionId);
android::util::BytesField bf_serialized( serialized.c_str(), serialized.size());
int result = android::util::stats_write(android::util::MEDIAMETRICS_AUDIOTRACK_REPORTED,
diff --git a/services/mediametrics/statsd_codec.cpp b/services/mediametrics/statsd_codec.cpp
index 46cbdc8..8581437 100644
--- a/services/mediametrics/statsd_codec.cpp
+++ b/services/mediametrics/statsd_codec.cpp
@@ -34,7 +34,7 @@
#include "cleaner.h"
#include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
#include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
#include "iface_statsd.h"
@@ -228,7 +228,7 @@
std::string sessionId;
if (item->getString("android.media.mediacodec.log-session-id", &sessionId)) {
- sessionId = mediametrics::stringutils::sanitizeLogSessionId(sessionId);
+ sessionId = mediametrics::ValidateId::get()->validateId(sessionId);
metrics_proto.set_log_session_id(sessionId);
}
AStatsEvent_writeString(event, codec.c_str());
diff --git a/services/mediametrics/statsd_extractor.cpp b/services/mediametrics/statsd_extractor.cpp
index bcf2e0a..a8bfeaa 100644
--- a/services/mediametrics/statsd_extractor.cpp
+++ b/services/mediametrics/statsd_extractor.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
#include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
#include "iface_statsd.h"
@@ -86,7 +86,7 @@
std::string log_session_id;
if (item->getString("android.media.mediaextractor.logSessionId", &log_session_id)) {
- log_session_id = mediametrics::stringutils::sanitizeLogSessionId(log_session_id);
+ log_session_id = mediametrics::ValidateId::get()->validateId(log_session_id);
metrics_proto.set_log_session_id(log_session_id);
}
diff --git a/services/mediametrics/statsd_mediaparser.cpp b/services/mediametrics/statsd_mediaparser.cpp
index 921b320..67ca874b 100644
--- a/services/mediametrics/statsd_mediaparser.cpp
+++ b/services/mediametrics/statsd_mediaparser.cpp
@@ -31,7 +31,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
#include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
#include "iface_statsd.h"
@@ -81,7 +81,7 @@
std::string logSessionId;
item->getString("android.media.mediaparser.logSessionId", &logSessionId);
- logSessionId = mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+ logSessionId = mediametrics::ValidateId::get()->validateId(logSessionId);
int result = android::util::stats_write(android::util::MEDIAMETRICS_MEDIAPARSER_REPORTED,
timestamp_nanos,
diff --git a/services/mediametrics/statsd_recorder.cpp b/services/mediametrics/statsd_recorder.cpp
index b29ad73..5f54a68 100644
--- a/services/mediametrics/statsd_recorder.cpp
+++ b/services/mediametrics/statsd_recorder.cpp
@@ -32,7 +32,7 @@
#include <statslog.h>
#include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
#include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
#include "iface_statsd.h"
@@ -59,7 +59,7 @@
// string kRecorderLogSessionId = "android.media.mediarecorder.log-session-id";
std::string log_session_id;
if (item->getString("android.media.mediarecorder.log-session-id", &log_session_id)) {
- log_session_id = mediametrics::stringutils::sanitizeLogSessionId(log_session_id);
+ log_session_id = mediametrics::ValidateId::get()->validateId(log_session_id);
metrics_proto.set_log_session_id(log_session_id);
}
// string kRecorderAudioMime = "android.media.mediarecorder.audio.mime";
diff --git a/services/mediametrics/tests/mediametrics_tests.cpp b/services/mediametrics/tests/mediametrics_tests.cpp
index 2336d6f..69ec947 100644
--- a/services/mediametrics/tests/mediametrics_tests.cpp
+++ b/services/mediametrics/tests/mediametrics_tests.cpp
@@ -28,6 +28,7 @@
#include "AudioTypes.h"
#include "StringUtils.h"
+#include "ValidateId.h"
using namespace android;
@@ -1127,3 +1128,100 @@
validId2[3] = '!';
ASSERT_EQ("", mediametrics::stringutils::sanitizeLogSessionId(validId2));
}
+
+TEST(mediametrics_tests, LruSet) {
+ constexpr size_t LRU_SET_SIZE = 2;
+ mediametrics::LruSet<std::string> lruSet(LRU_SET_SIZE);
+
+ // test adding a couple strings.
+ lruSet.add("abc");
+ ASSERT_EQ(1u, lruSet.size());
+ ASSERT_TRUE(lruSet.check("abc"));
+ lruSet.add("def");
+ ASSERT_EQ(2u, lruSet.size());
+
+ // now adding the third string causes eviction of the oldest.
+ lruSet.add("ghi");
+ ASSERT_FALSE(lruSet.check("abc"));
+ ASSERT_TRUE(lruSet.check("ghi"));
+ ASSERT_TRUE(lruSet.check("def")); // "def" is most recent.
+ ASSERT_EQ(2u, lruSet.size()); // "abc" is correctly discarded.
+
+ // adding another string will evict the oldest.
+ lruSet.add("foo");
+ ASSERT_FALSE(lruSet.check("ghi")); // note: "ghi" discarded when "foo" added.
+ ASSERT_TRUE(lruSet.check("foo"));
+ ASSERT_TRUE(lruSet.check("def"));
+
+ // manual removing of a string works, too.
+ ASSERT_TRUE(lruSet.remove("def"));
+ ASSERT_FALSE(lruSet.check("def")); // we manually removed "def".
+ ASSERT_TRUE(lruSet.check("foo")); // "foo" is still there.
+ ASSERT_EQ(1u, lruSet.size());
+
+ // you can't remove a string that has not been added.
+ ASSERT_FALSE(lruSet.remove("bar")); // Note: "bar" doesn't exist, so remove returns false.
+ ASSERT_EQ(1u, lruSet.size());
+
+ lruSet.add("foo"); // adding "foo" (which already exists) doesn't change size.
+ ASSERT_EQ(1u, lruSet.size());
+ lruSet.add("bar"); // add "bar"
+ ASSERT_EQ(2u, lruSet.size());
+ lruSet.add("glorp"); // add "glorp" evicts "foo".
+ ASSERT_EQ(2u, lruSet.size());
+ ASSERT_TRUE(lruSet.check("bar"));
+ ASSERT_TRUE(lruSet.check("glorp"));
+ ASSERT_FALSE(lruSet.check("foo"));
+}
+
+TEST(mediametrics_tests, LruSet0) {
+ constexpr size_t LRU_SET_SIZE = 0;
+ mediametrics::LruSet<std::string> lruSet(LRU_SET_SIZE);
+
+ lruSet.add("a");
+ ASSERT_EQ(0u, lruSet.size());
+ ASSERT_FALSE(lruSet.check("a"));
+ ASSERT_FALSE(lruSet.remove("a")); // never added.
+ ASSERT_EQ(0u, lruSet.size());
+}
+
+// Returns a 16 Base64Url string representing the decimal representation of value
+// (with leading 0s) e.g. 0000000000000000, 0000000000000001, 0000000000000002, ...
+static std::string generateId(size_t value)
+{
+ char id[16 + 1]; // to be filled with 16 Base64Url chars (and zero termination)
+ char *sptr = id + 16; // start at the end.
+ *sptr-- = 0; // zero terminate.
+ // output the digits from least significant to most significant.
+ while (value) {
+ *sptr-- = value % 10;
+ value /= 10;
+ }
+ // add leading 0's
+ while (sptr > id) {
+ *sptr-- = '0';
+ }
+ return std::string(id);
+}
+
+TEST(mediametrics_tests, ValidateId) {
+ constexpr size_t LRU_SET_SIZE = 3;
+ constexpr size_t IDS = 10;
+ static_assert(IDS > LRU_SET_SIZE); // IDS must be greater than LRU_SET_SIZE.
+ mediametrics::ValidateId validateId(LRU_SET_SIZE);
+
+
+ // register IDs as integer strings counting from 0.
+ for (size_t i = 0; i < IDS; ++i) {
+ validateId.registerId(generateId(i));
+ }
+
+ // only the last LRU_SET_SIZE exist.
+ for (size_t i = 0; i < IDS - LRU_SET_SIZE; ++i) {
+ ASSERT_EQ("", validateId.validateId(generateId(i)));
+ }
+ for (size_t i = IDS - LRU_SET_SIZE; i < IDS; ++i) {
+ const std::string id = generateId(i);
+ ASSERT_EQ(id, validateId.validateId(id));
+ }
+}
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index 13dd3d3..340076e 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -59,6 +59,7 @@
result << " Device Id: " << getDeviceId() << "\n";
result << " Sample Rate: " << getSampleRate() << "\n";
result << " Channel Count: " << getSamplesPerFrame() << "\n";
+ result << " Channel Mask: 0x" << std::hex << getChannelMask() << std::dec << "\n";
result << " Format: " << getFormat() << "\n";
result << " Frames Per Burst: " << mFramesPerBurst << "\n";
result << " Usage: " << getUsage() << "\n";
@@ -164,6 +165,10 @@
configuration.getSamplesPerFrame() != getSamplesPerFrame()) {
return false;
}
+ if (configuration.getChannelMask() != AAUDIO_UNSPECIFIED &&
+ configuration.getChannelMask() != getChannelMask()) {
+ return false;
+ }
return true;
}
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index a08098c..35a0890 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -126,20 +126,15 @@
}
config.sample_rate = aaudioSampleRate;
- int32_t aaudioSamplesPerFrame = getSamplesPerFrame();
-
const aaudio_direction_t direction = getDirection();
+ config.channel_mask = AAudio_getChannelMaskForOpen(
+ getChannelMask(), getSamplesPerFrame(), direction == AAUDIO_DIRECTION_INPUT);
+
if (direction == AAUDIO_DIRECTION_OUTPUT) {
- config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
- ? AUDIO_CHANNEL_OUT_STEREO
- : audio_channel_out_mask_from_count(aaudioSamplesPerFrame);
mHardwareTimeOffsetNanos = OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at DAC later
} else if (direction == AAUDIO_DIRECTION_INPUT) {
- config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
- ? AUDIO_CHANNEL_IN_STEREO
- : audio_channel_in_mask_from_count(aaudioSamplesPerFrame);
mHardwareTimeOffsetNanos = INPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at ADC earlier
} else {
@@ -225,9 +220,9 @@
}
// Get information about the stream and pass it back to the caller.
- setSamplesPerFrame((direction == AAUDIO_DIRECTION_OUTPUT)
- ? audio_channel_count_from_out_mask(config.channel_mask)
- : audio_channel_count_from_in_mask(config.channel_mask));
+ setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+ config.channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
+ AAudio_isChannelIndexMask(config.channel_mask)));
// AAudio creates a copy of this FD and retains ownership of the copy.
// Assume that AudioFlinger will close the original shared_memory_fd.
@@ -247,9 +242,9 @@
setFormat(config.format);
setSampleRate(config.sample_rate);
- ALOGD("%s() actual rate = %d, channels = %d"
- ", deviceId = %d, capacity = %d\n",
- __func__, getSampleRate(), getSamplesPerFrame(), deviceId, getBufferCapacity());
+ ALOGD("%s() actual rate = %d, channels = %d channelMask = %#x, deviceId = %d, capacity = %d\n",
+ __func__, getSampleRate(), getSamplesPerFrame(), getChannelMask(),
+ deviceId, getBufferCapacity());
ALOGD("%s() format = 0x%08x, frame size = %d, burst size = %d",
__func__, getFormat(), calculateBytesPerFrame(), mFramesPerBurst);
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index 5fbcadb..5af0a91 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -78,7 +78,7 @@
result = mStreamInternal->open(builder);
setSampleRate(mStreamInternal->getSampleRate());
- setSamplesPerFrame(mStreamInternal->getSamplesPerFrame());
+ setChannelMask(mStreamInternal->getChannelMask());
setDeviceId(mStreamInternal->getDeviceId());
setSessionId(mStreamInternal->getSessionId());
setFormat(AUDIO_FORMAT_PCM_FLOAT); // force for mixer
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 34ddd4d..4ffc127 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -73,7 +73,8 @@
}
std::string AAudioServiceStreamBase::dumpHeader() {
- return std::string(" T Handle UId Port Run State Format Burst Chan Capacity");
+ return std::string(
+ " T Handle UId Port Run State Format Burst Chan Mask Capacity");
}
std::string AAudioServiceStreamBase::dump() const {
@@ -88,6 +89,7 @@
result << std::setw(7) << getFormat();
result << std::setw(6) << mFramesPerBurst;
result << std::setw(5) << getSamplesPerFrame();
+ result << std::setw(8) << std::hex << getChannelMask() << std::dec;
result << std::setw(9) << getBufferCapacity();
return result.str();
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index c665cda..ad06d97 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -164,11 +164,11 @@
goto error;
}
- setSamplesPerFrame(configurationInput.getSamplesPerFrame());
- if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
- setSamplesPerFrame(endpoint->getSamplesPerFrame());
+ setChannelMask(configurationInput.getChannelMask());
+ if (getChannelMask() == AAUDIO_UNSPECIFIED) {
+ setChannelMask(endpoint->getChannelMask());
} else if (getSamplesPerFrame() != endpoint->getSamplesPerFrame()) {
- ALOGD("%s() mSamplesPerFrame = %d, need %d",
+ ALOGD("%s() mSamplesPerFrame = %#x, need %#x",
__func__, getSamplesPerFrame(), endpoint->getSamplesPerFrame());
result = AAUDIO_ERROR_OUT_OF_RANGE;
goto error;
diff --git a/services/oboeservice/fuzzer/README.md b/services/oboeservice/fuzzer/README.md
index 00b85df..ae7af3eb 100644
--- a/services/oboeservice/fuzzer/README.md
+++ b/services/oboeservice/fuzzer/README.md
@@ -15,7 +15,7 @@
4. InService
5. DeviceId
6. SampleRate
-7. SamplesPerFrame
+7. ChannelMask
8. Direction
9. SharingMode
10. Usage
@@ -31,7 +31,7 @@
| `InService` | `bool` | Value obtained from FuzzedDataProvider |
| `DeviceId` | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
| `SampleRate` | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
-| `SamplesPerFrame` | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
+| `ChannelMask` | `AAUDIO_UNSPECIFIED`, `AAUDIO_CHANNEL_INDEX_MASK_1`, `AAUDIO_CHANNEL_INDEX_MASK_2`, `AAUDIO_CHANNEL_INDEX_MASK_3`, `AAUDIO_CHANNEL_INDEX_MASK_4`, `AAUDIO_CHANNEL_INDEX_MASK_5`, `AAUDIO_CHANNEL_INDEX_MASK_6`, `AAUDIO_CHANNEL_INDEX_MASK_7`, `AAUDIO_CHANNEL_INDEX_MASK_8`, `AAUDIO_CHANNEL_INDEX_MASK_9`, `AAUDIO_CHANNEL_INDEX_MASK_10`, `AAUDIO_CHANNEL_INDEX_MASK_11`, `AAUDIO_CHANNEL_INDEX_MASK_12`, `AAUDIO_CHANNEL_INDEX_MASK_13`, `AAUDIO_CHANNEL_INDEX_MASK_14`, `AAUDIO_CHANNEL_INDEX_MASK_15`, `AAUDIO_CHANNEL_INDEX_MASK_16`, `AAUDIO_CHANNEL_INDEX_MASK_17`, `AAUDIO_CHANNEL_INDEX_MASK_18`, `AAUDIO_CHANNEL_INDEX_MASK_19`, `AAUDIO_CHANNEL_INDEX_MASK_20`, `AAUDIO_CHANNEL_INDEX_MASK_21`, `AAUDIO_CHANNEL_INDEX_MASK_22`, `AAUDIO_CHANNEL_INDEX_MASK_23`, `AAUDIO_CHANNEL_INDEX_MASK_24`, `AAUDIO_CHANNEL_MONO`, `AAUDIO_CHANNEL_STEREO`, `AAUDIO_CHANNEL_FRONT_BACK`, `AAUDIO_CHANNEL_2POINT0POINT2`, `AAUDIO_CHANNEL_2POINT1POINT2`, `AAUDIO_CHANNEL_3POINT0POINT2`, `AAUDIO_CHANNEL_3POINT1POINT2`, `AAUDIO_CHANNEL_5POINT1`, `AAUDIO_CHANNEL_MONO`, `AAUDIO_CHANNEL_STEREO`, `AAUDIO_CHANNEL_2POINT1`, `AAUDIO_CHANNEL_TRI`, `AAUDIO_CHANNEL_TRI_BACK`, `AAUDIO_CHANNEL_3POINT1`, `AAUDIO_CHANNEL_2POINT0POINT2`, `AAUDIO_CHANNEL_2POINT1POINT2`, `AAUDIO_CHANNEL_3POINT0POINT2`, `AAUDIO_CHANNEL_3POINT1POINT2`, `AAUDIO_CHANNEL_QUAD`, `AAUDIO_CHANNEL_QUAD_SIDE`, `AAUDIO_CHANNEL_SURROUND`, `AAUDIO_CHANNEL_PENTA`, `AAUDIO_CHANNEL_5POINT1`, `AAUDIO_CHANNEL_5POINT1_SIDE`, `AAUDIO_CHANNEL_5POINT1POINT2`, `AAUDIO_CHANNEL_5POINT1POINT4`, `AAUDIO_CHANNEL_6POINT1`, `AAUDIO_CHANNEL_7POINT1`, `AAUDIO_CHANNEL_7POINT1POINT2`, `AAUDIO_CHANNEL_7POINT1POINT4`, `AAUDIO_CHANNEL_9POINT1POINT4`, `AAUDIO_CHANNEL_9POINT1POINT6` | Value obtained from FuzzedDataProvider |
| `Direction` | `AAUDIO_DIRECTION_OUTPUT`, `AAUDIO_DIRECTION_INPUT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
| `SharingMode` | `AAUDIO_SHARING_MODE_EXCLUSIVE`, `AAUDIO_SHARING_MODE_SHARED` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
| `Usage` | `AAUDIO_USAGE_MEDIA`, `AAUDIO_USAGE_VOICE_COMMUNICATION`, `AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING`, `AAUDIO_USAGE_ALARM`, `AAUDIO_USAGE_NOTIFICATION`, `AAUDIO_USAGE_NOTIFICATION_RINGTONE`, `AAUDIO_USAGE_NOTIFICATION_EVENT`, `AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY`, `AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE`, `AAUDIO_USAGE_ASSISTANCE_SONIFICATION`, `AAUDIO_USAGE_GAME`, `AAUDIO_USAGE_ASSISTANT`, `AAUDIO_SYSTEM_USAGE_EMERGENCY`, `AAUDIO_SYSTEM_USAGE_SAFETY`, `AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS`, `AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
diff --git a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
index 4bc661c..17e8d36 100644
--- a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
+++ b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
@@ -68,10 +68,71 @@
AAUDIO_INPUT_PRESET_UNPROCESSED, AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE,
};
+aaudio_channel_mask_t kAAudioChannelMasks[] = {
+ AAUDIO_UNSPECIFIED,
+ AAUDIO_CHANNEL_INDEX_MASK_1,
+ AAUDIO_CHANNEL_INDEX_MASK_2,
+ AAUDIO_CHANNEL_INDEX_MASK_3,
+ AAUDIO_CHANNEL_INDEX_MASK_4,
+ AAUDIO_CHANNEL_INDEX_MASK_5,
+ AAUDIO_CHANNEL_INDEX_MASK_6,
+ AAUDIO_CHANNEL_INDEX_MASK_7,
+ AAUDIO_CHANNEL_INDEX_MASK_8,
+ AAUDIO_CHANNEL_INDEX_MASK_9,
+ AAUDIO_CHANNEL_INDEX_MASK_10,
+ AAUDIO_CHANNEL_INDEX_MASK_11,
+ AAUDIO_CHANNEL_INDEX_MASK_12,
+ AAUDIO_CHANNEL_INDEX_MASK_13,
+ AAUDIO_CHANNEL_INDEX_MASK_14,
+ AAUDIO_CHANNEL_INDEX_MASK_15,
+ AAUDIO_CHANNEL_INDEX_MASK_16,
+ AAUDIO_CHANNEL_INDEX_MASK_17,
+ AAUDIO_CHANNEL_INDEX_MASK_18,
+ AAUDIO_CHANNEL_INDEX_MASK_19,
+ AAUDIO_CHANNEL_INDEX_MASK_20,
+ AAUDIO_CHANNEL_INDEX_MASK_21,
+ AAUDIO_CHANNEL_INDEX_MASK_22,
+ AAUDIO_CHANNEL_INDEX_MASK_23,
+ AAUDIO_CHANNEL_INDEX_MASK_24,
+ AAUDIO_CHANNEL_MONO,
+ AAUDIO_CHANNEL_STEREO,
+ AAUDIO_CHANNEL_FRONT_BACK,
+ AAUDIO_CHANNEL_2POINT0POINT2,
+ AAUDIO_CHANNEL_2POINT1POINT2,
+ AAUDIO_CHANNEL_3POINT0POINT2,
+ AAUDIO_CHANNEL_3POINT1POINT2,
+ AAUDIO_CHANNEL_5POINT1,
+ AAUDIO_CHANNEL_MONO,
+ AAUDIO_CHANNEL_STEREO,
+ AAUDIO_CHANNEL_2POINT1,
+ AAUDIO_CHANNEL_TRI,
+ AAUDIO_CHANNEL_TRI_BACK,
+ AAUDIO_CHANNEL_3POINT1,
+ AAUDIO_CHANNEL_2POINT0POINT2,
+ AAUDIO_CHANNEL_2POINT1POINT2,
+ AAUDIO_CHANNEL_3POINT0POINT2,
+ AAUDIO_CHANNEL_3POINT1POINT2,
+ AAUDIO_CHANNEL_QUAD,
+ AAUDIO_CHANNEL_QUAD_SIDE,
+ AAUDIO_CHANNEL_SURROUND,
+ AAUDIO_CHANNEL_PENTA,
+ AAUDIO_CHANNEL_5POINT1,
+ AAUDIO_CHANNEL_5POINT1_SIDE,
+ AAUDIO_CHANNEL_5POINT1POINT2,
+ AAUDIO_CHANNEL_5POINT1POINT4,
+ AAUDIO_CHANNEL_6POINT1,
+ AAUDIO_CHANNEL_7POINT1,
+ AAUDIO_CHANNEL_7POINT1POINT2,
+ AAUDIO_CHANNEL_7POINT1POINT4,
+ AAUDIO_CHANNEL_9POINT1POINT4,
+ AAUDIO_CHANNEL_9POINT1POINT6,
+};
+
const size_t kNumAAudioFormats = std::size(kAAudioFormats);
const size_t kNumAAudioUsages = std::size(kAAudioUsages);
const size_t kNumAAudioContentTypes = std::size(kAAudioContentTypes);
const size_t kNumAAudioInputPresets = std::size(kAAudioInputPresets);
+const size_t kNumAAudioChannelMasks = std::size(kAAudioChannelMasks);
class FuzzAAudioClient : public virtual RefBase, public AAudioServiceInterface {
public:
@@ -305,7 +366,11 @@
request.getConfiguration().setDeviceId(fdp.ConsumeIntegral<int32_t>());
request.getConfiguration().setSampleRate(fdp.ConsumeIntegral<int32_t>());
- request.getConfiguration().setSamplesPerFrame(fdp.ConsumeIntegral<int32_t>());
+ request.getConfiguration().setChannelMask((aaudio_channel_mask_t)(
+ fdp.ConsumeBool()
+ ? fdp.ConsumeIntegral<int32_t>()
+ : kAAudioChannelMasks[fdp.ConsumeIntegralInRange<int32_t>(
+ 0, kNumAAudioChannelMasks - 1)]));
request.getConfiguration().setDirection(
fdp.ConsumeBool() ? fdp.ConsumeIntegral<int32_t>()
: (fdp.ConsumeBool() ? AAUDIO_DIRECTION_OUTPUT : AAUDIO_DIRECTION_INPUT));