[Chromecast] Support mono input in mixer

Bug: internal b/77586397

Change-Id: I1a40b631bc3666e38008a367458737a4564f93a7
Reviewed-on: https://chromium-review.googlesource.com/994135Reviewed-by: Stephen Lanham <slan@chromium.org>
Commit-Queue: Kenneth MacKay <kmackay@chromium.org>
Cr-Commit-Position: refs/heads/master@{#548224}

chromecast/media/cma/backend/direct_mixer_source.cc

@@ -15,8 +15,6 @@ namespace media {
 
 namespace {
 
-const int kNumOutputChannels = 2;
-
 std::string AudioContentTypeToString(media::AudioContentType type) {
   switch (type) {
     case media::AudioContentType::kAlarm:
@@ -34,11 +32,9 @@ std::string AudioContentTypeToString(media::AudioContentType type) {
 DirectAudioSourceToken* CastMediaShlib::AddDirectAudioSource(
     DirectAudioSource* source,
     const MediaPipelineDeviceParams& params,
-    int source_sample_rate,
     int playout_channel) {
   DCHECK(source);
-  return new DirectMixerSource(source, params, source_sample_rate,
-                               playout_channel);
+  return new DirectMixerSource(source, params, playout_channel);
 }
 
 // static
@@ -49,11 +45,10 @@ void CastMediaShlib::RemoveDirectAudioSource(DirectAudioSourceToken* token) {
 
 DirectMixerSource::DirectMixerSource(DirectAudioSource* direct_source,
                                      const MediaPipelineDeviceParams& params,
-                                     int source_sample_rate,
                                      int playout_channel)
     : source_(direct_source),
-      num_channels_(kNumOutputChannels),
-      input_samples_per_second_(source_sample_rate),
+      num_channels_(source_->GetNumChannels()),
+      input_samples_per_second_(source_->GetSampleRate()),
       primary_(params.audio_type !=
                MediaPipelineDeviceParams::kAudioStreamSoundEffects),
       device_id_(params.device_id),
@@ -61,7 +56,6 @@ DirectMixerSource::DirectMixerSource(DirectAudioSource* direct_source,
       playout_channel_(playout_channel),
       mixer_(StreamMixer::Get()),
       channel_vector_(num_channels_) {
-  DCHECK(source_);
   LOG(INFO) << "Create " << device_id_ << " (" << this
             << "), content type = " << AudioContentTypeToString(content_type_);
   DCHECK(source_);

chromecast/media/cma/backend/direct_mixer_source.h

@@ -35,7 +35,6 @@ class DirectMixerSource : public MixerInput::Source,
 
   DirectMixerSource(DirectAudioSource* direct_source,
                     const MediaPipelineDeviceParams& params,
-                    int source_sample_rate,
                     int playout_channel);
 
   // Sets the volume multiplier for this stream. If |multiplier| < 0, sets the

chromecast/media/cma/backend/filter_group.cc

@@ -62,6 +62,9 @@ bool FilterGroup::CanProcessInput(const std::string& input_device_id) {
 
 void FilterGroup::AddInput(MixerInput* input) {
   active_inputs_.insert(input);
+  if (mixed_) {
+    AddTempBuffer(input->num_channels(), mixed_->frames());
+  }
 }
 
 void FilterGroup::RemoveInput(MixerInput* input) {
@@ -114,10 +117,18 @@ float FilterGroup::MixAndFilter(
   // Mix InputQueues
   mixed_->ZeroFramesPartial(0, num_frames);
   for (MixerInput* input : active_inputs_) {
-    int filled = input->FillAudioData(num_frames, rendering_delay, temp_.get());
-    for (int c = 0; c < num_channels_; ++c) {
-      input->VolumeScaleAccumulate(c != 0, temp_->channel(c), filled,
-                                   mixed_->channel(c));
+    DCHECK_LT(input->num_channels(), static_cast<int>(temp_buffers_.size()));
+    DCHECK(temp_buffers_[input->num_channels()]);
+    ::media::AudioBus* temp = temp_buffers_[input->num_channels()].get();
+    int filled = input->FillAudioData(num_frames, rendering_delay, temp);
+    int in_c = 0;
+    for (int out_c = 0; out_c < num_channels_; ++out_c) {
+      input->VolumeScaleAccumulate(out_c != 0, temp->channel(in_c), filled,
+                                   mixed_->channel(out_c));
+      ++in_c;
+      if (in_c >= input->num_channels()) {
+        in_c = 0;
+      }
     }
     volume = std::max(volume, input->InstantaneousVolume());
     content_type = std::max(content_type, input->content_type());
@@ -197,13 +208,26 @@ bool FilterGroup::ResizeBuffersIfNecessary(int num_frames) {
     return false;
   }
   mixed_ = ::media::AudioBus::Create(num_channels_, num_frames);
-  temp_ = ::media::AudioBus::Create(num_channels_, num_frames);
+  temp_buffers_.clear();
+  for (MixerInput* input : active_inputs_) {
+    AddTempBuffer(input->num_channels(), num_frames);
+  }
   interleaved_.reset(static_cast<float*>(
       base::AlignedAlloc(num_frames * num_channels_ * sizeof(float),
                          ::media::AudioBus::kChannelAlignment)));
   return true;
 }
 
+void FilterGroup::AddTempBuffer(int num_channels, int num_frames) {
+  if (static_cast<int>(temp_buffers_.size()) <= num_channels) {
+    temp_buffers_.resize(num_channels + 1);
+  }
+  if (!temp_buffers_[num_channels]) {
+    temp_buffers_[num_channels] =
+        ::media::AudioBus::Create(num_channels, num_frames);
+  }
+}
+
 void FilterGroup::SetPostProcessorConfig(const std::string& name,
                                          const std::string& config) {
   post_processing_pipeline_->SetPostProcessorConfig(name, config);

chromecast/media/cma/backend/filter_group.h

@@ -120,6 +120,7 @@ class FilterGroup {
   // Resizes temp_ and mixed_ if they are too small to hold |num_frames| frames.
   // Returns |true| if |num_frames| is larger than all previous |num_frames|.
   bool ResizeBuffersIfNecessary(int num_frames);
+  void AddTempBuffer(int num_channels, int num_frames);
 
   const int num_channels_;
   const GroupType type_;
@@ -139,7 +140,7 @@ class FilterGroup {
   // Buffers that hold audio data while it is mixed.
   // These are kept as members of this class to minimize copies and
   // allocations.
-  std::unique_ptr<::media::AudioBus> temp_;
+  std::vector<std::unique_ptr<::media::AudioBus>> temp_buffers_;
   std::unique_ptr<::media::AudioBus> mixed_;
 
   // Interleaved data must be aligned to 16 bytes.

chromecast/public/cast_media_shlib.h

@@ -145,7 +145,6 @@ class CHROMECAST_EXPORT CastMediaShlib {
   static DirectAudioSourceToken* AddDirectAudioSource(
       DirectAudioSource* source,
       const MediaPipelineDeviceParams& params,
-      int source_sample_rate,
       int playout_channel) __attribute__((__weak__));
 
   // Removes a direct audio source, given the |token| that was returned by

chromecast/public/media/direct_audio_source.h

@@ -21,6 +21,15 @@ class DirectAudioSource {
  public:
   using RenderingDelay = MediaPipelineBackend::AudioDecoder::RenderingDelay;
 
+  // Returns the sample rate of audio provided by the source, in samples per
+  // second.
+  virtual int GetSampleRate() = 0;
+
+  // Returns the number of audio channels provided by the source. This is the
+  // number of channels that will be requested when FillAudioPlaybackFrames()
+  // is called.
+  virtual int GetNumChannels() = 0;
+
   // Returns the desired playback buffer size in frames. This is the desired
   // value for |num_frames| when FillAudioPlaybackFrames(); it affects the
   // playback latency (larger value = higher latency). The backend may choose a