Index: content/renderer/media/media_stream_audio_processor.cc |
diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc |
index ac41187c832214d49f44e0a759d87c07361b6d8e..cc804192622ef8d5195fae9780a9cd93febbe746 100644 |
--- a/content/renderer/media/media_stream_audio_processor.cc |
+++ b/content/renderer/media/media_stream_audio_processor.cc |
@@ -50,6 +50,7 @@ AudioProcessing::ChannelLayout MapLayout(media::ChannelLayout media_layout) { |
} |
} |
+// This is only used for playout data where only max two channels is supported. |
AudioProcessing::ChannelLayout ChannelsToLayout(int num_channels) { |
switch (num_channels) { |
case 1: |
@@ -114,20 +115,27 @@ class MediaStreamAudioBus { |
// Wraps AudioFifo to provide a cleaner interface to MediaStreamAudioProcessor. |
// It avoids the FIFO when the source and destination frames match. All methods |
-// are called on one of the capture or render audio threads exclusively. |
+// are called on one of the capture or render audio threads exclusively. If |
+// |source_channels| is larger than |destination_channels|, only the first |
+// |destination_channels| are kept from the source. |
class MediaStreamAudioFifo { |
public: |
- MediaStreamAudioFifo(int channels, int source_frames, |
+ MediaStreamAudioFifo(int source_channels, |
+ int destination_channels, |
+ int source_frames, |
int destination_frames) |
- : source_frames_(source_frames), |
- destination_(new MediaStreamAudioBus(channels, destination_frames)), |
+ : source_channels_(source_channels), |
+ source_frames_(source_frames), |
+ destination_( |
+ new MediaStreamAudioBus(destination_channels, destination_frames)), |
data_available_(false) { |
+ DCHECK_GE(source_channels, destination_channels); |
if (source_frames != destination_frames) { |
// Since we require every Push to be followed by as many Consumes as |
// possible, twice the larger of the two is a (probably) loose upper bound |
// on the FIFO size. |
const int fifo_frames = 2 * std::max(source_frames, destination_frames); |
- fifo_.reset(new media::AudioFifo(channels, fifo_frames)); |
+ fifo_.reset(new media::AudioFifo(destination_channels, fifo_frames)); |
} |
// May be created in the main render thread and used in the audio threads. |
@@ -136,13 +144,29 @@ class MediaStreamAudioFifo { |
void Push(const media::AudioBus* source) { |
DCHECK(thread_checker_.CalledOnValidThread()); |
- DCHECK_EQ(source->channels(), destination_->bus()->channels()); |
+ DCHECK_EQ(source->channels(), source_channels_); |
DCHECK_EQ(source->frames(), source_frames_); |
+ const media::AudioBus* source_to_push = source; |
+ |
+ if (source->channels() > destination_->bus()->channels()) { |
+ if (!audio_source_intermediate_) { |
+ audio_source_intermediate_ = |
no longer working on chromium
2014/09/26 11:20:41
You can set up this audio_source_intermediate_ in
Henrik Grunell
2014/09/30 07:46:54
Done.
|
+ media::AudioBus::CreateWrapper(destination_->bus()->channels()); |
+ } |
+ for (int i = 0; i < destination_->bus()->channels(); ++i) { |
+ audio_source_intermediate_->SetChannelData( |
+ i, |
+ const_cast<float*>(source->channel(i))); |
+ } |
+ audio_source_intermediate_->set_frames(source->frames()); |
+ source_to_push = audio_source_intermediate_.get(); |
+ } |
+ |
if (fifo_) { |
- fifo_->Push(source); |
+ fifo_->Push(source_to_push); |
} else { |
- source->CopyTo(destination_->bus()); |
+ source_to_push->CopyTo(destination_->bus()); |
data_available_ = true; |
} |
} |
@@ -171,7 +195,9 @@ class MediaStreamAudioFifo { |
private: |
base::ThreadChecker thread_checker_; |
+ const int source_channels_; // For a DCHECK. |
const int source_frames_; // For a DCHECK. |
+ scoped_ptr<media::AudioBus> audio_source_intermediate_; |
scoped_ptr<MediaStreamAudioBus> destination_; |
scoped_ptr<media::AudioFifo> fifo_; |
// Only used when the FIFO is disabled; |
@@ -466,10 +492,24 @@ void MediaStreamAudioProcessor::InitializeCaptureFifo( |
// format it would prefer. |
const int output_sample_rate = audio_processing_ ? |
kAudioProcessingSampleRate : input_format.sample_rate(); |
- const media::ChannelLayout output_channel_layout = audio_processing_ ? |
+ media::ChannelLayout output_channel_layout = audio_processing_ ? |
media::GuessChannelLayout(kAudioProcessingNumberOfChannels) : |
input_format.channel_layout(); |
+ // The output channels from the fifo is normally the same as input. |
+ int fifo_output_channels = input_format.channels(); |
+ |
+ // Special case for if we have a keyboard mic channel on the input and no |
+ // audio processing is used. We will then have the fifo strip away that |
+ // channel. So we use stereo as output layout, and also change the output |
+ // channels for the fifo. |
+ if (input_format.channel_layout() == |
+ media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC && |
+ !audio_processing_) { |
+ output_channel_layout = media::CHANNEL_LAYOUT_STEREO; |
+ fifo_output_channels = ChannelLayoutToChannelCount(output_channel_layout); |
+ } |
+ |
// webrtc::AudioProcessing requires a 10 ms chunk size. We use this native |
// size when processing is enabled. When disabled we use the same size as |
// the source if less than 10 ms. |
@@ -496,6 +536,7 @@ void MediaStreamAudioProcessor::InitializeCaptureFifo( |
capture_fifo_.reset( |
new MediaStreamAudioFifo(input_format.channels(), |
+ fifo_output_channels, |
input_format.frames_per_buffer(), |
processing_frames)); |
@@ -528,6 +569,7 @@ void MediaStreamAudioProcessor::InitializeRenderFifoIfNeeded( |
const int analysis_frames = sample_rate / 100; // 10 ms chunks. |
render_fifo_.reset( |
new MediaStreamAudioFifo(number_of_channels, |
+ number_of_channels, |
frames_per_buffer, |
analysis_frames)); |
} |