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-rw-r--r--services/audioflinger/AudioFlinger.cpp127
1 files changed, 67 insertions, 60 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 053854f..d4a8ddb 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -53,13 +53,6 @@
#include <media/EffectVisualizerApi.h>
// ----------------------------------------------------------------------------
-// the sim build doesn't have gettid
-
-#ifndef HAVE_GETTID
-# define gettid getpid
-#endif
-
-// ----------------------------------------------------------------------------
namespace android {
@@ -360,8 +353,8 @@ sp<IAudioTrack> AudioFlinger::createTrack(
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 +422,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
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 +470,7 @@ int AudioFlinger::channelCount(int output) const
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 +909,7 @@ int AudioFlinger::ThreadBase::channelCount() const
return (int)mChannelCount;
}
-int AudioFlinger::ThreadBase::format() const
+uint32_t AudioFlinger::ThreadBase::format() const
{
return mFormat;
}
@@ -1002,6 +995,8 @@ status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args
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 +1070,7 @@ status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>
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 +1081,7 @@ status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>
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 +1167,8 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
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 +1178,14 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
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 +1222,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
}
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 +1371,7 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
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 +1391,8 @@ void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
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 +1802,7 @@ uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track
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 +2681,7 @@ void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
this,
mSampleRate,
mFormat,
- mChannelCount,
+ mChannelMask,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
@@ -2751,8 +2749,8 @@ AudioFlinger::ThreadBase::TrackBase::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,
@@ -2772,6 +2770,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// 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 +2785,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// 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 +2810,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// 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 +2878,11 @@ int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
}
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 +2894,9 @@ void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t f
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 +2911,12 @@ AudioFlinger::PlaybackThread::Track::Track(
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 +2936,7 @@ AudioFlinger::PlaybackThread::Track::Track(
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 +2984,12 @@ void AudioFlinger::PlaybackThread::Track::destroy()
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 +3224,21 @@ AudioFlinger::RecordThread::RecordTrack::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)
: 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 +3318,10 @@ void AudioFlinger::RecordThread::RecordTrack::stop()
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 +3337,10 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
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 +3351,10 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
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 +3389,7 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr
{
Buffer *pInBuffer;
Buffer inBuffer;
- uint32_t channelCount = mCblk->channelCount;
+ uint32_t channelCount = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
@@ -3667,8 +3674,8 @@ sp<IAudioRecord> AudioFlinger::openRecord(
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 +3724,7 @@ sp<IAudioRecord> AudioFlinger::openRecord(
}
// 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 +4072,7 @@ status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
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 +4226,7 @@ void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
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 +4249,8 @@ void AudioFlinger::RecordThread::readInputParameters()
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);
generated by cgit v1.1 at 2025-08-16 09:56:49 +0000