Use new channel count functions for audio masks

Change-Id: Ia658ab4b6320d19fdb50f123c930918724ff0ef3
Signed-off-by: Andy Hung <hunga@google.com>

11 files changed, 31 insertions, 22 deletions

diff --git a/media/libeffects/downmix/EffectDownmix.c b/media/libeffects/downmix/EffectDownmix.c
index 661fde9..6686f27 100644
--- a/media/libeffects/downmix/EffectDownmix.c
+++ b/media/libeffects/downmix/EffectDownmix.c
@@ -118,7 +118,7 @@ void Downmix_testIndexComputation(uint32_t mask) {
         hasBacks = true;
     }
 
-    const int numChan = popcount(mask);
+    const int numChan = audio_channel_count_from_out_mask(mask);
     const bool hasFC = ((mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) == AUDIO_CHANNEL_OUT_FRONT_CENTER);
     const bool hasLFE =
             ((mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY);
@@ -629,7 +629,8 @@ int Downmix_Configure(downmix_module_t *pDwmModule, effect_config_t *pConfig, bo
             ALOGE("Downmix_Configure error: input channel mask can't be 0");
             return -EINVAL;
         }
-        pDownmixer->input_channel_count = popcount(pConfig->inputCfg.channels);
+        pDownmixer->input_channel_count =
+                audio_channel_count_from_out_mask(pConfig->inputCfg.channels);
     }
 
     Downmix_Reset(pDownmixer, init);
@@ -997,7 +998,7 @@ bool Downmix_foldGeneric(
         hasBacks = true;
     }
 
-    const int numChan = popcount(mask);
+    const int numChan = audio_channel_count_from_out_mask(mask);
     const bool hasFC = ((mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) == AUDIO_CHANNEL_OUT_FRONT_CENTER);
     const bool hasLFE =
             ((mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY);
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index a96a703..cf98f56 100644
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -879,8 +879,8 @@ int Session_ReleaseEffect(preproc_session_t *session,
 int Session_SetConfig(preproc_session_t *session, effect_config_t *config)
 {
     uint32_t sr;
-    uint32_t inCnl = popcount(config->inputCfg.channels);
-    uint32_t outCnl = popcount(config->outputCfg.channels);
+    uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
+    uint32_t outCnl = audio_channel_count_from_out_mask(config->outputCfg.channels);
 
     if (config->inputCfg.samplingRate != config->outputCfg.samplingRate ||
         config->inputCfg.format != config->outputCfg.format ||
@@ -1035,7 +1035,7 @@ int Session_SetReverseConfig(preproc_session_t *session, effect_config_t *config
             config->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
         return -EINVAL;
     }
-    uint32_t inCnl = popcount(config->inputCfg.channels);
+    uint32_t inCnl = audio_channel_count_from_out_mask(config->inputCfg.channels);
     int status = session->apm->set_num_reverse_channels(inCnl);
     if (status < 0) {
         return -EINVAL;
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index 47cab62..e5089da 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -207,7 +207,8 @@ int Visualizer_init(VisualizerContext *pContext)
     pContext->mScalingMode = VISUALIZER_SCALING_MODE_NORMALIZED;
 
     // measurement initialization
-    pContext->mChannelCount = popcount(pContext->mConfig.inputCfg.channels);
+    pContext->mChannelCount =
+            audio_channel_count_from_out_mask(pContext->mConfig.inputCfg.channels);
     pContext->mMeasurementMode = MEASUREMENT_MODE_NONE;
     pContext->mMeasurementWindowSizeInBuffers = MEASUREMENT_WINDOW_MAX_SIZE_IN_BUFFERS;
     pContext->mMeasurementBufferIdx = 0;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 97ab8f8..1c808d0 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -51,7 +51,8 @@ status_t AudioRecord::getMinFrameCount(
 
     // We double the size of input buffer for ping pong use of record buffer.
     // Assumes audio_is_linear_pcm(format)
-    if ((*frameCount = (size * 2) / (popcount(channelMask) * audio_bytes_per_sample(format))) == 0) {
+    if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) *
+            audio_bytes_per_sample(format))) == 0) {
         ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
             sampleRate, format, channelMask);
         return BAD_VALUE;
@@ -193,7 +194,7 @@ status_t AudioRecord::set(
         return BAD_VALUE;
     }
     mChannelMask = channelMask;
-    uint32_t channelCount = popcount(channelMask);
+    uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
     mChannelCount = channelCount;
 
     if (audio_is_linear_pcm(format)) {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index aaaa3f1..b4d6d76 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -290,7 +290,7 @@ status_t AudioTrack::set(
         return BAD_VALUE;
     }
     mChannelMask = channelMask;
-    uint32_t channelCount = popcount(channelMask);
+    uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
     mChannelCount = channelCount;
 
     // AudioFlinger does not currently support 8-bit data in shared memory
diff --git a/media/libnbaio/AudioStreamInSource.cpp b/media/libnbaio/AudioStreamInSource.cpp
index af8f365..6aab48a 100644
--- a/media/libnbaio/AudioStreamInSource.cpp
+++ b/media/libnbaio/AudioStreamInSource.cpp
@@ -46,7 +46,8 @@ ssize_t AudioStreamInSource::negotiate(const NBAIO_Format offers[], size_t numOf
         uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common);
         audio_channel_mask_t channelMask =
                 (audio_channel_mask_t) mStream->common.get_channels(&mStream->common);
-        mFormat = Format_from_SR_C(sampleRate, popcount(channelMask), streamFormat);
+        mFormat = Format_from_SR_C(sampleRate,
+                audio_channel_count_from_in_mask(channelMask), streamFormat);
         mFrameSize = Format_frameSize(mFormat);
     }
     return NBAIO_Source::negotiate(offers, numOffers, counterOffers, numCounterOffers);
diff --git a/media/libnbaio/AudioStreamOutSink.cpp b/media/libnbaio/AudioStreamOutSink.cpp
index c28d34d..0d5f935 100644
--- a/media/libnbaio/AudioStreamOutSink.cpp
+++ b/media/libnbaio/AudioStreamOutSink.cpp
@@ -43,7 +43,8 @@ ssize_t AudioStreamOutSink::negotiate(const NBAIO_Format offers[], size_t numOff
         uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common);
         audio_channel_mask_t channelMask =
                 (audio_channel_mask_t) mStream->common.get_channels(&mStream->common);
-        mFormat = Format_from_SR_C(sampleRate, popcount(channelMask), streamFormat);
+        mFormat = Format_from_SR_C(sampleRate,
+                audio_channel_count_from_out_mask(channelMask), streamFormat);
         mFrameSize = Format_frameSize(mFormat);
     }
     return NBAIO_Sink::negotiate(offers, numOffers, counterOffers, numCounterOffers);
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 11170c2..45e17f8 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1837,7 +1837,8 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
     if (status == BAD_VALUE &&
         reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
         (config.sample_rate <= 2 * reqSamplingRate) &&
-        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
+        (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) &&
+        (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) {
         // FIXME describe the change proposed by HAL (save old values so we can log them here)
         ALOGV("openInput() reopening with proposed sampling rate and channel mask");
         inStream = NULL;
@@ -1857,7 +1858,8 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
             TEE_SINK_OLD,   // copy input using an existing pipe
         } kind;
         NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
-                                        popcount(inStream->common.get_channels(&inStream->common)));
+                audio_channel_count_from_in_mask(
+                        inStream->common.get_channels(&inStream->common)));
         if (!mTeeSinkInputEnabled) {
             kind = TEE_SINK_NO;
         } else if (!Format_isValid(format)) {
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 2d67efb..d32f4d1 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -215,7 +215,7 @@ void AudioMixer::invalidateState(uint32_t mask)
 
 status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackNum, audio_channel_mask_t mask)
 {
-    uint32_t channelCount = popcount(mask);
+    uint32_t channelCount = audio_channel_count_from_out_mask(mask);
     ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
     status_t status = OK;
     if (channelCount > MAX_NUM_CHANNELS) {
@@ -410,7 +410,7 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
             audio_channel_mask_t mask =
                 static_cast<audio_channel_mask_t>(reinterpret_cast<uintptr_t>(value));
             if (track.channelMask != mask) {
-                uint32_t channelCount = popcount(mask);
+                uint32_t channelCount = audio_channel_count_from_out_mask(mask);
                 ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
                 track.channelMask = mask;
                 track.channelCount = channelCount;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2d4e025..8d9ccd0 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1730,7 +1730,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
         LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; "
                 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask);
     }
-    mChannelCount = popcount(mChannelMask);
+    mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
     mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
     if (!audio_is_valid_format(mFormat)) {
         LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
@@ -5564,8 +5564,8 @@ bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValueP
                 reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
                 (mInput->stream->common.get_sample_rate(&mInput->stream->common)
                         <= (2 * samplingRate)) &&
-                popcount(mInput->stream->common.get_channels(&mInput->stream->common))
-                        <= FCC_2 &&
+                audio_channel_count_from_in_mask(
+                        mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
                 (channelMask == AUDIO_CHANNEL_IN_MONO ||
                         channelMask == AUDIO_CHANNEL_IN_STEREO)) {
                 status = NO_ERROR;
@@ -5619,7 +5619,7 @@ void AudioFlinger::RecordThread::readInputParameters_l()
 {
     mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
     mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
-    mChannelCount = popcount(mChannelMask);
+    mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
     mFormat = mInput->stream->common.get_format(&mInput->stream->common);
     if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
         ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6dc7f30..1d46f87 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -81,7 +81,9 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
         mSampleRate(sampleRate),
         mFormat(format),
         mChannelMask(channelMask),
-        mChannelCount(popcount(channelMask)),
+        mChannelCount(isOut ?
+                audio_channel_count_from_out_mask(channelMask) :
+                audio_channel_count_from_in_mask(channelMask)),
         mFrameSize(audio_is_linear_pcm(format) ?
                 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
         mFrameCount(frameCount),
@@ -1849,7 +1851,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
 
     mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
 
-    uint32_t channelCount = popcount(channelMask);
+    uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
     // FIXME I don't understand either of the channel count checks
     if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 &&
             channelCount <= FCC_2) {