diff options
Diffstat (limited to 'services')
-rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 120 | ||||
-rw-r--r-- | services/audioflinger/AudioFlinger.h | 44 | ||||
-rw-r--r-- | services/audioflinger/AudioMixer.cpp | 20 | ||||
-rw-r--r-- | services/audioflinger/AudioMixer.h | 3 |
4 files changed, 108 insertions, 79 deletions
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 @@ 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 +429,7 @@ 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 +477,7 @@ 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 +916,7 @@ int AudioFlinger::ThreadBase::channelCount() const return (int)mChannelCount; } -int AudioFlinger::ThreadBase::format() const +uint32_t AudioFlinger::ThreadBase::format() const { return mFormat; } @@ -1002,6 +1002,8 @@ 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 +1077,7 @@ 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 +1088,7 @@ 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 +1174,8 @@ 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 +1185,14 @@ 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 +1229,7 @@ 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 +1378,7 @@ 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 +1398,8 @@ 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 +1809,7 @@ 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 +2688,7 @@ 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 +2756,8 @@ 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 +2777,7 @@ 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 +2792,8 @@ 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 +2817,8 @@ 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 +2885,11 @@ 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 +2901,9 @@ 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 +2918,12 @@ 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 +2943,7 @@ 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 +2991,12 @@ 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 +3231,21 @@ 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 +3325,10 @@ 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 +3344,10 @@ 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 +3358,10 @@ 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 +3396,7 @@ 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 +3681,8 @@ 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 +3731,7 @@ 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 +4079,7 @@ 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 +4233,7 @@ 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 +4256,8 @@ 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); 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 @@ public: 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 @@ public: 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 @@ public: 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 @@ private: 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 @@ private: 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 @@ private: // 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 @@ private: 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 @@ private: 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 @@ private: 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 @@ private: 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 @@ private: 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 @@ private: 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 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate) 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 @@ status_t AudioMixer::setParameter(int target, int name, void *value) 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 @@ public: // set Parameter names // for target TRACK - CHANNEL_COUNT = 0x4000, + CHANNEL_MASK = 0x4000, FORMAT = 0x4001, MAIN_BUFFER = 0x4002, AUX_BUFFER = 0x4003, @@ -150,6 +150,7 @@ private: uint8_t enabled : 1; uint8_t reserved0 : 3; uint8_t format; + uint32_t channelMask; AudioBufferProvider* bufferProvider; mutable AudioBufferProvider::Buffer buffer; |