| Index: content/renderer/media/webrtc_audio_device_impl.cc
|
| diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
|
| index dc6d2ba91b4e369c7e06250275c79757c43c21d9..b6b29ecb68decad91c39214f3e8d42ed139930b2 100644
|
| --- a/content/renderer/media/webrtc_audio_device_impl.cc
|
| +++ b/content/renderer/media/webrtc_audio_device_impl.cc
|
| @@ -11,8 +11,8 @@
|
| #include "content/renderer/media/audio_device_factory.h"
|
| #include "content/renderer/media/audio_hardware.h"
|
| #include "content/renderer/render_thread_impl.h"
|
| -#include "media/audio/audio_util.h"
|
| #include "media/audio/audio_parameters.h"
|
| +#include "media/audio/audio_util.h"
|
| #include "media/audio/sample_rates.h"
|
|
|
| using content::AudioDeviceFactory;
|
| @@ -218,15 +218,8 @@ int WebRtcAudioDeviceImpl::Render(
|
|
|
| // Deinterleave each channel and convert to 32-bit floating-point
|
| // with nominal range -1.0 -> +1.0 to match the callback format.
|
| - for (int channel_index = 0; channel_index < channels; ++channel_index) {
|
| - media::DeinterleaveAudioChannel(
|
| - output_buffer_.get(),
|
| - audio_bus->channel(channel_index),
|
| - channels,
|
| - channel_index,
|
| - bytes_per_sample_,
|
| - audio_bus->frames());
|
| - }
|
| + audio_bus->FromInterleaved(output_buffer_.get(), audio_bus->frames(),
|
| + bytes_per_sample_);
|
| return audio_bus->frames();
|
| }
|
|
|
| @@ -265,10 +258,9 @@ void WebRtcAudioDeviceImpl::Capture(media::AudioBus* audio_bus,
|
|
|
| // Interleave, scale, and clip input to int and store result in
|
| // a local byte buffer.
|
| - media::InterleaveFloatToInt(audio_bus,
|
| - input_buffer_.get(),
|
| - audio_bus->frames(),
|
| - input_audio_parameters_.bits_per_sample() / 8);
|
| + audio_bus->ToInterleaved(audio_bus->frames(),
|
| + input_audio_parameters_.bits_per_sample() / 8,
|
| + input_buffer_.get());
|
|
|
| int samples_per_sec = input_sample_rate();
|
| if (samples_per_sec == 44100) {
|
|
|