Use audio_channel_mask_t more places
Use it in AudioSystem::getOutput(), AudioSystem::getInput(),
IAudioPolicyService::getOutput(), IAudioPolicyService::getInput(),
and various other places in AudioFlinger.
Not done: AudioTrack and OutputDescriptor.
Change-Id: I70e83455820bd8f05dafd30c63d636c6a47cd172
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 18c9c92..66f4b84 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -188,7 +188,7 @@
static audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
- uint32_t channels = AUDIO_CHANNEL_OUT_STEREO,
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE);
static status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
@@ -200,7 +200,7 @@
static audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
- uint32_t channels = AUDIO_CHANNEL_IN_MONO,
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO,
int sessionId = 0);
static status_t startInput(audio_io_handle_t input);
static status_t stopInput(audio_io_handle_t input);
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index 23fa964..bdc12db 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -51,7 +51,7 @@
virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
- uint32_t channels = 0,
+ audio_channel_mask_t channelMask = 0,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE) = 0;
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
@@ -63,7 +63,7 @@
virtual audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
- uint32_t channels = 0,
+ audio_channel_mask_t channelMask = 0,
int audioSession = 0) = 0;
virtual status_t startInput(audio_io_handle_t input) = 0;
virtual status_t stopInput(audio_io_handle_t input) = 0;
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9c270c8..d880989 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -449,7 +449,7 @@
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %d frameCount %d latency %d",
+ ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %#x frameCount %d latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency);
} break;
case OUTPUT_CLOSED: {
@@ -471,7 +471,7 @@
if (param2 == NULL) break;
desc = (const OutputDescriptor *)param2;
- ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
+ ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %#x frameCount %d latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channels, desc->frameCount, desc->latency);
OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -588,12 +588,12 @@
audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
audio_output_flags_t flags)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
- return aps->getOutput(stream, samplingRate, format, channels, flags);
+ return aps->getOutput(stream, samplingRate, format, channelMask, flags);
}
status_t AudioSystem::startOutput(audio_io_handle_t output,
@@ -624,12 +624,12 @@
audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
int sessionId)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
- return aps->getInput(inputSource, samplingRate, format, channels, sessionId);
+ return aps->getInput(inputSource, samplingRate, format, channelMask, sessionId);
}
status_t AudioSystem::startInput(audio_io_handle_t input)
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 5a4512e..f0c0f2e 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -123,7 +123,7 @@
audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
audio_output_flags_t flags)
{
Parcel data, reply;
@@ -131,7 +131,7 @@
data.writeInt32(static_cast <uint32_t>(stream));
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
- data.writeInt32(channels);
+ data.writeInt32(channelMask);
data.writeInt32(static_cast <uint32_t>(flags));
remote()->transact(GET_OUTPUT, data, &reply);
return static_cast <audio_io_handle_t> (reply.readInt32());
@@ -175,7 +175,7 @@
audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
int audioSession)
{
Parcel data, reply;
@@ -183,7 +183,7 @@
data.writeInt32((int32_t) inputSource);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
- data.writeInt32(channels);
+ data.writeInt32(channelMask);
data.writeInt32(audioSession);
remote()->transact(GET_INPUT, data, &reply);
return static_cast <audio_io_handle_t> (reply.readInt32());
@@ -415,14 +415,14 @@
static_cast <audio_stream_type_t>(data.readInt32());
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- uint32_t channels = data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
audio_output_flags_t flags =
static_cast <audio_output_flags_t>(data.readInt32());
audio_io_handle_t output = getOutput(stream,
samplingRate,
format,
- channels,
+ channelMask,
flags);
reply->writeInt32(static_cast <int>(output));
return NO_ERROR;
@@ -462,12 +462,12 @@
audio_source_t inputSource = (audio_source_t) data.readInt32();
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
- uint32_t channels = data.readInt32();
+ audio_channel_mask_t channelMask = data.readInt32();
int audioSession = data.readInt32();
audio_io_handle_t input = getInput(inputSource,
samplingRate,
format,
- channels,
+ channelMask,
audioSession);
reply->writeInt32(static_cast <int>(input));
return NO_ERROR;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index ad0a533..d41585a 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -432,7 +432,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
IAudioFlinger::track_flags_t flags,
const sp<IMemory>& sharedBuffer,
@@ -1652,7 +1652,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -1707,7 +1707,7 @@
frameCount, mFrameCount);
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d "
- "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d "
+ "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d "
"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
isTimed, sharedBuffer.get(), frameCount, mFrameCount, format,
audio_is_linear_pcm(format),
@@ -2192,7 +2192,7 @@
// mNormalSink below
{
ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type);
- ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
+ ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, "
"mFrameCount=%d, mNormalFrameCount=%d",
mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
mNormalFrameCount);
@@ -3466,7 +3466,7 @@
readOutputParameters();
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
- int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask);
+ int name = getTrackName_l(mTracks[i]->mChannelMask);
if (name < 0) break;
mTracks[i]->mName = name;
// limit track sample rate to 2 x new output sample rate
@@ -4061,7 +4061,7 @@
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
@@ -4236,7 +4236,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -4262,7 +4262,7 @@
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
// to avoid leaking a track name, do not allocate one unless there is an mCblk
- mName = thread->getTrackName_l((audio_channel_mask_t)channelMask);
+ mName = thread->getTrackName_l(channelMask);
mCblk->mName = mName;
if (mName < 0) {
ALOGE("no more track names available");
@@ -4781,7 +4781,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId) {
@@ -4799,7 +4799,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
@@ -5294,7 +5294,7 @@
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
@@ -5406,7 +5406,7 @@
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount)
: Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
NULL, 0, IAudioFlinger::TRACK_DEFAULT),
@@ -5794,7 +5794,7 @@
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
IAudioFlinger::track_flags_t flags,
int *sessionId,
@@ -5899,13 +5899,13 @@
AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
uint32_t sampleRate,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
audio_io_handle_t id,
uint32_t device) :
ThreadBase(audioFlinger, id, device, RECORD),
mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
// mRsmpInIndex and mInputBytes set by readInputParameters()
- mReqChannelCount(popcount(channels)),
+ mReqChannelCount(popcount(channelMask)),
mReqSampleRate(sampleRate)
// mBytesRead is only meaningful while active, and so is cleared in start()
// (but might be better to also clear here for dump?)
@@ -6149,7 +6149,7 @@
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
int sessionId,
status_t *status)
@@ -6929,7 +6929,7 @@
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
- uint32_t *pChannelMask)
+ audio_channel_mask_t *pChannelMask)
{
status_t status;
RecordThread *thread = NULL;
@@ -6972,7 +6972,7 @@
reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
(config.sample_rate <= 2 * reqSamplingRate) &&
(popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
- ALOGV("openInput() reopening with proposed sampling rate and channels");
+ ALOGV("openInput() reopening with proposed sampling rate and channel mask");
inStream = NULL;
status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream);
}
@@ -8193,7 +8193,6 @@
status_t AudioFlinger::EffectModule::configure()
{
- uint32_t channels;
if (mEffectInterface == NULL) {
return NO_INIT;
}
@@ -8204,18 +8203,14 @@
}
// TODO: handle configuration of effects replacing track process
- if (thread->channelCount() == 1) {
- channels = AUDIO_CHANNEL_OUT_MONO;
- } else {
- channels = AUDIO_CHANNEL_OUT_STEREO;
- }
+ audio_channel_mask_t channelMask = thread->channelMask();
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
} else {
- mConfig.inputCfg.channels = channels;
+ mConfig.inputCfg.channels = channelMask;
}
- mConfig.outputCfg.channels = channels;
+ mConfig.outputCfg.channels = channelMask;
mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
mConfig.inputCfg.samplingRate = thread->sampleRate();
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 08c727d..47ceefe 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -381,7 +381,7 @@
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
@@ -413,7 +413,7 @@
int channelCount() const { return mChannelCount; }
- uint32_t channelMask() const { return mChannelMask; }
+ audio_channel_mask_t channelMask() const { return mChannelMask; }
int sampleRate() const; // FIXME inline after cblk sr moved
@@ -463,7 +463,7 @@
bool mStepServerFailed;
const int mSessionId;
uint8_t mChannelCount;
- uint32_t mChannelMask;
+ audio_channel_mask_t mChannelMask;
Vector < sp<SyncEvent> >mSyncEvents;
};
@@ -499,6 +499,7 @@
// dynamic externally-visible
uint32_t sampleRate() const { return mSampleRate; }
int channelCount() const { return mChannelCount; }
+ audio_channel_mask_t channelMask() const { return mChannelMask; }
audio_format_t format() const { return mFormat; }
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
// and returns the normal mix buffer's frame count. No API for HAL frame count.
@@ -633,7 +634,7 @@
uint32_t mSampleRate;
size_t mFrameCount; // output HAL, direct output, record
size_t mNormalFrameCount; // normal mixer and effects
- uint32_t mChannelMask;
+ audio_channel_mask_t mChannelMask;
uint16_t mChannelCount;
size_t mFrameSize;
audio_format_t mFormat;
@@ -715,7 +716,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -839,7 +840,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
@@ -882,7 +883,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
@@ -931,7 +932,7 @@
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount);
virtual ~OutputTrack();
@@ -1010,7 +1011,7 @@
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
@@ -1355,7 +1356,7 @@
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
- uint32_t channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
int sessionId);
virtual ~RecordTrack();
@@ -1386,7 +1387,7 @@
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
uint32_t sampleRate,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
audio_io_handle_t id,
uint32_t device);
virtual ~RecordThread();
@@ -1403,7 +1404,7 @@
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
- int channelMask,
+ audio_channel_mask_t channelMask,
int frameCount,
int sessionId,
status_t *status);
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 0c8b3ce..3a8c54d 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -412,7 +412,7 @@
case TRACK:
switch (param) {
case CHANNEL_MASK: {
- uint32_t mask = (uint32_t)value;
+ audio_channel_mask_t mask = (audio_channel_mask_t) value;
if (track.channelMask != mask) {
uint32_t channelCount = popcount(mask);
ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 8b644ab..f07fe8e 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -223,7 +223,7 @@
audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
audio_output_flags_t flags)
{
if (mpAudioPolicy == NULL) {
@@ -231,7 +231,7 @@
}
ALOGV("getOutput() tid %d", gettid());
Mutex::Autolock _l(mLock);
- return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channels, flags);
+ return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate, format, channelMask, flags);
}
status_t AudioPolicyService::startOutput(audio_io_handle_t output,
@@ -271,7 +271,7 @@
audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
uint32_t samplingRate,
audio_format_t format,
- uint32_t channels,
+ audio_channel_mask_t channelMask,
int audioSession)
{
if (mpAudioPolicy == NULL) {
@@ -284,7 +284,7 @@
Mutex::Autolock _l(mLock);
// the audio_in_acoustics_t parameter is ignored by get_input()
audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
- format, channels, (audio_in_acoustics_t) 0);
+ format, channelMask, (audio_in_acoustics_t) 0);
if (input == 0) {
return input;
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index b4924fc..7c6fa74 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -64,7 +64,7 @@
virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
- uint32_t channels = 0,
+ audio_channel_mask_t channelMask = 0,
audio_output_flags_t flags =
AUDIO_OUTPUT_FLAG_NONE);
virtual status_t startOutput(audio_io_handle_t output,
@@ -77,7 +77,7 @@
virtual audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
audio_format_t format = AUDIO_FORMAT_DEFAULT,
- uint32_t channels = 0,
+ audio_channel_mask_t channelMask = 0,
int audioSession = 0);
virtual status_t startInput(audio_io_handle_t input);
virtual status_t stopInput(audio_io_handle_t input);