Switch AudioRenderSink::Callback to use AudioBus.

content/renderer/media/webrtc_audio_device_impl.cc

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 c3ef160808f2464e57b69a3b5d71614943463d98..0946ab2bd046b3c95825c1edad3e347cff6c0da3 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -174,7 +174,7 @@ int32_t WebRtcAudioDeviceImpl::Release() {
 }
 
 int WebRtcAudioDeviceImpl::Render(
-    const std::vector<float*>& audio_data,
+    media::AudioBus* audio_bus,
     int number_of_frames,
     int audio_delay_milliseconds) {
   DCHECK_LE(number_of_frames, output_buffer_size());
@@ -185,7 +185,7 @@ int WebRtcAudioDeviceImpl::Render(
     output_delay_ms_ = audio_delay_milliseconds;
   }
 
-  const int channels = audio_data.size();
+  const int channels = audio_bus->channels();
   DCHECK_LE(channels, output_channels());
 
   int samples_per_sec = output_sample_rate();
@@ -222,7 +222,7 @@ int WebRtcAudioDeviceImpl::Render(
   for (int channel_index = 0; channel_index < channels; ++channel_index) {
     media::DeinterleaveAudioChannel(
         output_buffer_.get(),
-        audio_data[channel_index],
+        audio_bus->channel(channel_index),
         channels,
         channel_index,
         bytes_per_sample_,
@@ -237,7 +237,7 @@ void WebRtcAudioDeviceImpl::OnRenderError() {
   LOG(ERROR) << "OnRenderError()";
 }
 
-void WebRtcAudioDeviceImpl::Capture(const std::vector<float*>& audio_data,
+void WebRtcAudioDeviceImpl::Capture(media::AudioBus* audio_bus,
                                     int number_of_frames,
                                     int audio_delay_milliseconds,
                                     double volume) {
@@ -261,13 +261,13 @@ void WebRtcAudioDeviceImpl::Capture(const std::vector<float*>& audio_data,
     output_delay_ms = output_delay_ms_;
   }
 
-  const int channels = audio_data.size();
+  const int channels = audio_bus->channels();
   DCHECK_LE(channels, input_channels());
   uint32_t new_mic_level = 0;
 
   // Interleave, scale, and clip input to int and store result in
   // a local byte buffer.
-  media::InterleaveFloatToInt(audio_data,
+  media::InterleaveFloatToInt(*audio_bus,
                               input_buffer_.get(),
                               number_of_frames,
                               input_audio_parameters_.bits_per_sample() / 8);