Use channel mask instead of channel count for track creation
Record and playback objects (resp AudioRecord and AudioTrack)
are created using a channel mask, but this information is lost
in the mixer because only the channel count is known to
AudioFlinger. A channel count can always be derived from a
channel mask.
The change consists in:
- disambiguiting variable names for channel masks and counts
- passing the mask information from the client to AudioFlinger
and the mixer.
- when using the DIRECT ouput, only verifying the format of
the track is compatible with the output's for PCM.
Change-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index baab2e8..605680a 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -130,7 +130,7 @@
* sampleRate: Track sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
- * channels: Channel mask: see audio_channels_t.
+ * channelMask: Channel mask: see audio_channels_t.
* frameCount: Total size of track PCM buffer in frames. This defines the
* latency of the track.
* flags: A bitmask of acoustic values from enum record_flags. It enables
@@ -151,7 +151,7 @@
AudioRecord(int inputSource,
uint32_t sampleRate = 0,
int format = 0,
- uint32_t channels = AUDIO_CHANNEL_IN_MONO,
+ uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
@@ -177,7 +177,7 @@
status_t set(int inputSource = 0,
uint32_t sampleRate = 0,
int format = 0,
- uint32_t channels = AUDIO_CHANNEL_IN_MONO,
+ uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
@@ -348,8 +348,8 @@
bool processAudioBuffer(const sp<ClientRecordThread>& thread);
status_t openRecord_l(uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
audio_io_handle_t input);
@@ -364,10 +364,10 @@
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
- uint8_t mFormat;
+ uint32_t mFormat;
uint8_t mChannelCount;
uint8_t mInputSource;
- uint8_t mReserved;
+ uint8_t mReserved[2];
status_t mStatus;
uint32_t mLatency;
@@ -382,7 +382,7 @@
uint32_t mNewPosition;
uint32_t mUpdatePeriod;
uint32_t mFlags;
- uint32_t mChannels;
+ uint32_t mChannelMask;
audio_io_handle_t mInput;
int mSessionId;
};
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index de928da..df30e8c 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -69,8 +69,8 @@
MUTE = 0x00000001
};
uint32_t flags;
- int channelCount;
int format;
+ int channelCount; // will be removed in the future, do not use
size_t frameCount;
size_t size;
union {
@@ -129,7 +129,7 @@
* sampleRate: Track sampling rate in Hz.
* format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
- * channels: Channel mask: see audio_channels_t.
+ * channelMask: Channel mask: see audio_channels_t.
* frameCount: Total size of track PCM buffer in frames. This defines the
* latency of the track.
* flags: Reserved for future use.
@@ -143,7 +143,7 @@
AudioTrack( int streamType,
uint32_t sampleRate = 0,
int format = 0,
- int channels = 0,
+ int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
@@ -163,7 +163,7 @@
AudioTrack( int streamType,
uint32_t sampleRate = 0,
int format = 0,
- int channels = 0,
+ int channelMask = 0,
const sp<IMemory>& sharedBuffer = 0,
uint32_t flags = 0,
callback_t cbf = 0,
@@ -187,7 +187,7 @@
status_t set(int streamType =-1,
uint32_t sampleRate = 0,
int format = 0,
- int channels = 0,
+ int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
@@ -438,8 +438,8 @@
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
status_t createTrack_l(int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -459,11 +459,12 @@
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
+ uint32_t mFormat;
uint8_t mStreamType;
- uint8_t mFormat;
uint8_t mChannelCount;
uint8_t mMuted;
- uint32_t mChannels;
+ uint8_t mReserved;
+ uint32_t mChannelMask;
status_t mStatus;
uint32_t mLatency;
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index d8fdc27..4037c46 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -48,8 +48,8 @@
pid_t pid,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -61,8 +61,8 @@
pid_t pid,
int input,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
@@ -73,7 +73,7 @@
*/
virtual uint32_t sampleRate(int output) const = 0;
virtual int channelCount(int output) const = 0;
- virtual int format(int output) const = 0;
+ virtual uint32_t format(int output) const = 0;
virtual size_t frameCount(int output) const = 0;
virtual uint32_t latency(int output) const = 0;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 1827c3e..072329d 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -82,7 +82,7 @@
// 16 bit because data is converted to 16 bit before being stored in buffer
uint8_t frameSize;
- uint8_t channelCount;
+ uint8_t pad1;
uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
uint16_t waitTimeMs; // Cumulated wait time
@@ -90,6 +90,7 @@
volatile int32_t flags;
// Cache line boundary (32 bytes)
+
audio_track_cblk_t();
uint32_t stepUser(uint32_t frameCount);
bool stepServer(uint32_t frameCount);
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 446e3df..f6c4cc7 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -88,7 +88,7 @@
int inputSource,
uint32_t sampleRate,
int format,
- uint32_t channels,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
callback_t cbf,
@@ -97,7 +97,7 @@
int sessionId)
: mStatus(NO_INIT), mSessionId(0)
{
- mStatus = set(inputSource, sampleRate, format, channels,
+ mStatus = set(inputSource, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames, sessionId);
}
@@ -121,7 +121,7 @@
int inputSource,
uint32_t sampleRate,
int format,
- uint32_t channels,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
callback_t cbf,
@@ -131,7 +131,7 @@
int sessionId)
{
- LOGV("set(): sampleRate %d, channels %d, frameCount %d",sampleRate, channels, frameCount);
+ LOGV("set(): sampleRate %d, channelMask %d, frameCount %d",sampleRate, channelMask, frameCount);
AutoMutex lock(mLock);
@@ -156,14 +156,14 @@
return BAD_VALUE;
}
- if (!audio_is_input_channel(channels)) {
+ if (!audio_is_input_channel(channelMask)) {
return BAD_VALUE;
}
- int channelCount = popcount(channels);
+ int channelCount = popcount(channelMask);
audio_io_handle_t input = AudioSystem::getInput(inputSource,
- sampleRate, format, channels, (audio_in_acoustics_t)flags);
+ sampleRate, format, channelMask, (audio_in_acoustics_t)flags);
if (input == 0) {
LOGE("Could not get audio input for record source %d", inputSource);
return BAD_VALUE;
@@ -190,7 +190,7 @@
mSessionId = sessionId;
// create the IAudioRecord
- status = openRecord_l(sampleRate, format, channelCount,
+ status = openRecord_l(sampleRate, format, channelMask,
frameCount, flags, input);
if (status != NO_ERROR) {
return status;
@@ -209,7 +209,7 @@
// Update buffer size in case it has been limited by AudioFlinger during track creation
mFrameCount = mCblk->frameCount;
mChannelCount = (uint8_t)channelCount;
- mChannels = channels;
+ mChannelMask = channelMask;
mActive = 0;
mCbf = cbf;
mNotificationFrames = notificationFrames;
@@ -437,8 +437,8 @@
// must be called with mLock held
status_t AudioRecord::openRecord_l(
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
audio_io_handle_t input)
@@ -451,7 +451,7 @@
sp<IAudioRecord> record = audioFlinger->openRecord(getpid(), input,
sampleRate, format,
- channelCount,
+ channelMask,
frameCount,
((uint16_t)flags) << 16,
&mSessionId,
@@ -589,7 +589,7 @@
{
mInput = AudioSystem::getInput(mInputSource,
mCblk->sampleRate,
- mFormat, mChannels,
+ mFormat, mChannelMask,
(audio_in_acoustics_t)mFlags);
return mInput;
}
@@ -756,7 +756,7 @@
// if the new IAudioRecord is created, openRecord_l() will modify the
// following member variables: mAudioRecord, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioRecord and IMemory
- result = openRecord_l(cblk->sampleRate, mFormat, mChannelCount,
+ result = openRecord_l(cblk->sampleRate, mFormat, mChannelMask,
mFrameCount, mFlags, getInput_l());
if (result == NO_ERROR) {
result = mAudioRecord->start();
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7520ed9..ea44f87 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -87,7 +87,7 @@
int streamType,
uint32_t sampleRate,
int format,
- int channels,
+ int channelMask,
int frameCount,
uint32_t flags,
callback_t cbf,
@@ -96,7 +96,7 @@
int sessionId)
: mStatus(NO_INIT)
{
- mStatus = set(streamType, sampleRate, format, channels,
+ mStatus = set(streamType, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0, false, sessionId);
}
@@ -105,7 +105,7 @@
int streamType,
uint32_t sampleRate,
int format,
- int channels,
+ int channelMask,
const sp<IMemory>& sharedBuffer,
uint32_t flags,
callback_t cbf,
@@ -114,7 +114,7 @@
int sessionId)
: mStatus(NO_INIT)
{
- mStatus = set(streamType, sampleRate, format, channels,
+ mStatus = set(streamType, sampleRate, format, channelMask,
0, flags, cbf, user, notificationFrames,
sharedBuffer, false, sessionId);
}
@@ -141,7 +141,7 @@
int streamType,
uint32_t sampleRate,
int format,
- int channels,
+ int channelMask,
int frameCount,
uint32_t flags,
callback_t cbf,
@@ -180,8 +180,8 @@
if (format == 0) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
- if (channels == 0) {
- channels = AUDIO_CHANNEL_OUT_STEREO;
+ if (channelMask == 0) {
+ channelMask = AUDIO_CHANNEL_OUT_STEREO;
}
// validate parameters
@@ -195,15 +195,15 @@
flags |= AUDIO_POLICY_OUTPUT_FLAG_DIRECT;
}
- if (!audio_is_output_channel(channels)) {
+ if (!audio_is_output_channel(channelMask)) {
LOGE("Invalid channel mask");
return BAD_VALUE;
}
- uint32_t channelCount = popcount(channels);
+ uint32_t channelCount = popcount(channelMask);
audio_io_handle_t output = AudioSystem::getOutput(
(audio_stream_type_t)streamType,
- sampleRate,format, channels,
+ sampleRate,format, channelMask,
(audio_policy_output_flags_t)flags);
if (output == 0) {
@@ -222,8 +222,8 @@
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
- format,
- channelCount,
+ (uint32_t)format,
+ (uint32_t)channelMask,
frameCount,
flags,
sharedBuffer,
@@ -245,8 +245,8 @@
mStatus = NO_ERROR;
mStreamType = streamType;
- mFormat = format;
- mChannels = channels;
+ mFormat = (uint32_t)format;
+ mChannelMask = (uint32_t)channelMask;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
@@ -681,7 +681,7 @@
audio_io_handle_t AudioTrack::getOutput_l()
{
return AudioSystem::getOutput((audio_stream_type_t)mStreamType,
- mCblk->sampleRate, mFormat, mChannels, (audio_policy_output_flags_t)mFlags);
+ mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
}
int AudioTrack::getSessionId()
@@ -705,8 +705,8 @@
status_t AudioTrack::createTrack_l(
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -767,6 +767,7 @@
}
} else {
// Ensure that buffer alignment matches channelcount
+ int channelCount = popcount(channelMask);
if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
return BAD_VALUE;
@@ -779,7 +780,7 @@
streamType,
sampleRate,
format,
- channelCount,
+ channelMask,
frameCount,
((uint16_t)flags) << 16,
sharedBuffer,
@@ -1164,7 +1165,7 @@
result = createTrack_l(mStreamType,
cblk->sampleRate,
mFormat,
- mChannelCount,
+ mChannelMask,
mFrameCount,
mFlags,
mSharedBuffer,
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 158d2f5..4a12962 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -82,8 +82,8 @@
pid_t pid,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -98,7 +98,7 @@
data.writeInt32(streamType);
data.writeInt32(sampleRate);
data.writeInt32(format);
- data.writeInt32(channelCount);
+ data.writeInt32(channelMask);
data.writeInt32(frameCount);
data.writeInt32(flags);
data.writeStrongBinder(sharedBuffer->asBinder());
@@ -129,8 +129,8 @@
pid_t pid,
int input,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
@@ -143,7 +143,7 @@
data.writeInt32(input);
data.writeInt32(sampleRate);
data.writeInt32(format);
- data.writeInt32(channelCount);
+ data.writeInt32(channelMask);
data.writeInt32(frameCount);
data.writeInt32(flags);
int lSessionId = 0;
@@ -186,7 +186,7 @@
return reply.readInt32();
}
- virtual int format(int output) const
+ virtual uint32_t format(int output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index d51c946..eae93ff 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -1164,7 +1164,7 @@
mem = new MemoryBase(cache->getHeap(), 0, cache->size());
*pSampleRate = cache->sampleRate();
*pNumChannels = cache->channelCount();
- *pFormat = cache->format();
+ *pFormat = (int)cache->format();
LOGV("return memory @ %p, sampleRate=%u, channelCount = %d, format = %d", mem->pointer(), *pSampleRate, *pNumChannels, *pFormat);
Exit:
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 053854f..f806624 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -360,8 +360,8 @@
pid_t pid,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -429,7 +429,7 @@
LOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
- channelCount, frameCount, sharedBuffer, lSessionId, &lStatus);
+ channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
@@ -477,7 +477,7 @@
return thread->channelCount();
}
-int AudioFlinger::format(int output) const
+uint32_t AudioFlinger::format(int output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -916,7 +916,7 @@
return (int)mChannelCount;
}
-int AudioFlinger::ThreadBase::format() const
+uint32_t AudioFlinger::ThreadBase::format() const
{
return mFormat;
}
@@ -1002,6 +1002,8 @@
result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
result.append(buffer);
+ snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
+ result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
@@ -1075,7 +1077,7 @@
snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
- result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
+ result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
@@ -1086,7 +1088,7 @@
snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
- result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
+ result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
wp<Track> wTrack = mActiveTracks[i];
if (wTrack != 0) {
@@ -1172,8 +1174,8 @@
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -1183,11 +1185,14 @@
status_t lStatus;
if (mType == DIRECT) {
- if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) {
- LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
- sampleRate, format, channelCount, mOutput);
- lStatus = BAD_VALUE;
- goto Exit;
+ if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
+ if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
+ LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
+ "for output %p with format %d",
+ sampleRate, format, channelMask, mOutput, mFormat);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
}
} else {
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
@@ -1224,7 +1229,7 @@
}
track = new Track(this, client, streamType, sampleRate, format,
- channelCount, frameCount, sharedBuffer, sessionId);
+ channelMask, frameCount, sharedBuffer, sessionId);
if (track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
@@ -1373,7 +1378,7 @@
switch (event) {
case AudioSystem::OUTPUT_OPENED:
case AudioSystem::OUTPUT_CONFIG_CHANGED:
- desc.channels = mChannels;
+ desc.channels = mChannelMask;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
@@ -1393,8 +1398,8 @@
void AudioFlinger::PlaybackThread::readOutputParameters()
{
mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
- mChannels = mOutput->stream->common.get_channels(&mOutput->stream->common);
- mChannelCount = (uint16_t)popcount(mChannels);
+ mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
+ mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
@@ -1804,7 +1809,7 @@
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
- AudioMixer::CHANNEL_COUNT, (void *)track->channelCount());
+ AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
@@ -2683,7 +2688,7 @@
this,
mSampleRate,
mFormat,
- mChannelCount,
+ mChannelMask,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
@@ -2751,8 +2756,8 @@
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -2772,6 +2777,7 @@
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
+ uint8_t channelCount = popcount(channelMask);
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
if (sharedBuffer == 0) {
size += bufferSize;
@@ -2786,7 +2792,8 @@
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
- mCblk->channelCount = (uint8_t)channelCount;
+ mChannelCount = channelCount;
+ mChannelMask = channelMask;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
@@ -2810,7 +2817,8 @@
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
- mCblk->channelCount = (uint8_t)channelCount;
+ mChannelCount = channelCount;
+ mChannelMask = channelMask;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
@@ -2877,7 +2885,11 @@
}
int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
- return (int)mCblk->channelCount;
+ return (const int)mChannelCount;
+}
+
+uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
+ return mChannelMask;
}
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -2889,9 +2901,9 @@
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
- server %d, serverBase %d, user %d, userBase %d, channelCount %d",
+ server %d, serverBase %d, user %d, userBase %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
- cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);
+ cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
return 0;
}
@@ -2906,12 +2918,12 @@
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
- : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId),
+ : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
@@ -2931,7 +2943,7 @@
mStreamType = streamType;
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
- mCblk->frameSize = audio_is_linear_pcm(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
+ mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(int8_t);
}
}
@@ -2979,12 +2991,12 @@
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
- snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
+ snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
mFormat,
- mCblk->channelCount,
+ mChannelMask,
mSessionId,
mFrameCount,
mState,
@@ -3219,21 +3231,21 @@
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
- channelCount, frameCount, flags, 0, sessionId),
+ channelMask, frameCount, flags, 0, sessionId),
mOverflow(false)
{
if (mCblk != NULL) {
LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
if (format == AUDIO_FORMAT_PCM_16_BIT) {
- mCblk->frameSize = channelCount * sizeof(int16_t);
+ mCblk->frameSize = mChannelCount * sizeof(int16_t);
} else if (format == AUDIO_FORMAT_PCM_8_BIT) {
- mCblk->frameSize = channelCount * sizeof(int8_t);
+ mCblk->frameSize = mChannelCount * sizeof(int8_t);
} else {
mCblk->frameSize = sizeof(int8_t);
}
@@ -3313,10 +3325,10 @@
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
- snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n",
+ snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n",
(mClient == NULL) ? getpid() : mClient->pid(),
mFormat,
- mCblk->channelCount,
+ mChannelMask,
mSessionId,
mFrameCount,
mState,
@@ -3332,10 +3344,10 @@
const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount)
- : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelCount, frameCount, NULL, 0),
+ : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
@@ -3346,8 +3358,10 @@
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
- LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
- mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);
+ LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
+ "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
+ mCblk, mBuffer, mCblk->buffers,
+ mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
} else {
LOGW("Error creating output track on thread %p", playbackThread);
}
@@ -3382,7 +3396,7 @@
{
Buffer *pInBuffer;
Buffer inBuffer;
- uint32_t channelCount = mCblk->channelCount;
+ uint32_t channelCount = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
@@ -3667,8 +3681,8 @@
pid_t pid,
int input,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
@@ -3717,7 +3731,7 @@
}
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
- format, channelCount, frameCount, flags, lSessionId);
+ format, channelMask, frameCount, flags, lSessionId);
}
if (recordTrack->getCblk() == NULL) {
// remove local strong reference to Client before deleting the RecordTrack so that the Client
@@ -4065,7 +4079,7 @@
if (mActiveTrack != 0) {
result.append("Active Track:\n");
- result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n");
+ result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n");
mActiveTrack->dump(buffer, SIZE);
result.append(buffer);
@@ -4219,7 +4233,7 @@
switch (event) {
case AudioSystem::INPUT_OPENED:
case AudioSystem::INPUT_CONFIG_CHANGED:
- desc.channels = mChannels;
+ desc.channels = mChannelMask;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
@@ -4242,8 +4256,8 @@
mResampler = 0;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
- mChannels = mInput->stream->common.get_channels(&mInput->stream->common);
- mChannelCount = (uint16_t)popcount(mChannels);
+ mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
+ mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 0698dcb..f3371bf 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -76,8 +76,8 @@
pid_t pid,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -87,7 +87,7 @@
virtual uint32_t sampleRate(int output) const;
virtual int channelCount(int output) const;
- virtual int format(int output) const;
+ virtual uint32_t format(int output) const;
virtual size_t frameCount(int output) const;
virtual uint32_t latency(int output) const;
@@ -189,8 +189,8 @@
pid_t pid,
int input,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
@@ -301,8 +301,8 @@
TrackBase(const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
@@ -329,12 +329,14 @@
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
- int format() const {
+ uint32_t format() const {
return mFormat;
}
int channelCount() const ;
+ uint32_t channelMask() const;
+
int sampleRate() const;
void* getBuffer(uint32_t offset, uint32_t frames) const;
@@ -360,9 +362,11 @@
// we don't really need a lock for these
int mState;
int mClientTid;
- uint8_t mFormat;
+ uint32_t mFormat;
uint32_t mFlags;
int mSessionId;
+ uint8_t mChannelCount;
+ uint32_t mChannelMask;
};
class ConfigEvent {
@@ -375,7 +379,7 @@
uint32_t sampleRate() const;
int channelCount() const;
- int format() const;
+ uint32_t format() const;
size_t frameCount() const;
void wakeUp() { mWaitWorkCV.broadcast(); }
void exit();
@@ -406,10 +410,10 @@
sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
- uint32_t mChannels;
+ uint32_t mChannelMask;
uint16_t mChannelCount;
uint16_t mFrameSize;
- int mFormat;
+ uint32_t mFormat;
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
@@ -442,8 +446,8 @@
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
@@ -530,8 +534,8 @@
OutputTrack( const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount);
~OutputTrack();
@@ -583,8 +587,8 @@
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -829,8 +833,8 @@
RecordTrack(const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
- int format,
- int channelCount,
+ uint32_t format,
+ uint32_t channelMask,
int frameCount,
uint32_t flags,
int sessionId);
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 50dcda7..6e9319d 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -26,6 +26,10 @@
#include <utils/Errors.h>
#include <utils/Log.h>
+#include <cutils/bitops.h>
+
+#include <system/audio.h>
+
#include "AudioMixer.h"
namespace android {
@@ -61,6 +65,7 @@
t->channelCount = 2;
t->enabled = 0;
t->format = 16;
+ t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->buffer.raw = 0;
t->bufferProvider = 0;
t->hook = 0;
@@ -180,13 +185,18 @@
switch (target) {
case TRACK:
- if (name == CHANNEL_COUNT) {
- if ((uint32_t(valueInt) <= MAX_NUM_CHANNELS) && (valueInt)) {
- if (mState.tracks[ mActiveTrack ].channelCount != valueInt) {
- mState.tracks[ mActiveTrack ].channelCount = valueInt;
- LOGV("setParameter(TRACK, CHANNEL_COUNT, %d)", valueInt);
+ if (name == CHANNEL_MASK) {
+ uint32_t mask = (uint32_t)value;
+ if (mState.tracks[ mActiveTrack ].channelMask != mask) {
+ uint8_t channelCount = popcount(mask);
+ if ((channelCount <= MAX_NUM_CHANNELS) && (channelCount)) {
+ mState.tracks[ mActiveTrack ].channelMask = mask;
+ mState.tracks[ mActiveTrack ].channelCount = channelCount;
+ LOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
invalidateState(1<<mActiveTrack);
+ return NO_ERROR;
}
+ } else {
return NO_ERROR;
}
}
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 88408a7..75c9170 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -61,7 +61,7 @@
// set Parameter names
// for target TRACK
- CHANNEL_COUNT = 0x4000,
+ CHANNEL_MASK = 0x4000,
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
@@ -150,6 +150,7 @@
uint8_t enabled : 1;
uint8_t reserved0 : 3;
uint8_t format;
+ uint32_t channelMask;
AudioBufferProvider* bufferProvider;
mutable AudioBufferProvider::Buffer buffer;