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; |