Index: webrtc/modules/audio_processing/audio_buffer.cc |
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc |
index 04dcaea799d60af6bbc48d899e9ded8134d6ce03..cc8375e449b1e300b96d616c7d6cd782e4e618e8 100644 |
--- a/webrtc/modules/audio_processing/audio_buffer.cc |
+++ b/webrtc/modules/audio_processing/audio_buffer.cc |
@@ -436,20 +436,28 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) { |
} |
void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const { |
- assert(proc_num_frames_ == output_num_frames_); |
- assert(num_channels_ == num_input_channels_); |
- assert(frame->num_channels_ == num_channels_); |
- assert(frame->samples_per_channel_ == proc_num_frames_); |
frame->vad_activity_ = activity_; |
- |
if (!data_changed) { |
return; |
} |
- Interleave(data_->ibuf()->channels(), |
- proc_num_frames_, |
- num_channels_, |
- frame->data_); |
+ assert(proc_num_frames_ == output_num_frames_); |
+ assert(frame->num_channels_ == num_channels_ || num_channels_ == 1); |
+ assert(frame->samples_per_channel_ == proc_num_frames_); |
+ |
+ if (frame->num_channels_ == num_channels_) { |
+ Interleave(data_->ibuf()->channels(), proc_num_frames_, num_channels_, |
+ frame->data_); |
+ } else { |
+ // Copy single AudioBuffer channel into all AudioFrame channels |
Andrew MacDonald
2015/07/24 23:50:39
Period at the end, and can you note that this is s
ekm
2015/07/29 00:37:19
Done.
|
+ rtc::scoped_ptr<int16_t*> channel_ptr_copies( |
+ new int16_t*[frame->num_channels_]); |
Andrew MacDonald
2015/07/24 23:50:39
Arg, why did you switch to using dynamic allocatio
ekm
2015/07/29 00:37:19
Done.
|
+ for (int i = 0; i < frame->num_channels_; ++i) { |
+ channel_ptr_copies.get()[i] = data_->ibuf()->channels()[0]; |
+ } |
+ Interleave(channel_ptr_copies.get(), proc_num_frames_, num_channels_, |
+ frame->data_); |
+ } |
} |
void AudioBuffer::CopyLowPassToReference() { |