| 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 8931f75e40aa51a73d4b5078e8b26e3083873fd4..9ee3da787e2e700d6b6995c9d38b327636319a66 100644
|
| --- a/content/renderer/media/webrtc_audio_device_impl.cc
|
| +++ b/content/renderer/media/webrtc_audio_device_impl.cc
|
| @@ -90,7 +90,7 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
|
| // webrtc::AudioTransport source. Keep reading until our internal buffer
|
| // is full.
|
| int accumulated_audio_frames = 0;
|
| - int16* audio_data = &render_buffer_[0];
|
| + int16_t* audio_data = &render_buffer_[0];
|
| while (accumulated_audio_frames < audio_bus->frames()) {
|
| // Get 10ms and append output to temporary byte buffer.
|
| int64_t elapsed_time_ms = -1;
|
|
|