Roll webrtc/libjingle to 6189.

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 517b3e79e7bced23c2b06530adef628cc77a38e9..282edfe8c9809c1e70818e240ecbb925a3687e37 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -156,21 +156,29 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
       // render data to the APM in WebRTC as reference signal for echo
       // cancellation.
       static const int kBitsPerByte = 8;
+      uint32_t rtp_ts = 0;
+      int64_t ntp_ts = 0;
       audio_transport_callback_->PullRenderData(bytes_per_sample * kBitsPerByte,
                                                 sample_rate,
                                                 audio_bus->channels(),
                                                 frames_per_10_ms,
-                                                audio_data);
+                                                audio_data,
+                                                &rtp_ts,
+                                                &ntp_ts);
       accumulated_audio_frames += frames_per_10_ms;
     } else {
       // TODO(xians): Remove the following code after the APM in WebRTC is
       // deprecated.
+      uint32_t rtp_ts = 0;
+      int64_t ntp_ts = 0;

[wjia(left Chromium), 2014/05/19 17:56:26]

How about moving these 2 lines above "if" line to avoid dup?

[Ronghua Wu (Left Chromium), 2014/05/19 17:58:11]

Good idea. Done

       audio_transport_callback_->NeedMorePlayData(frames_per_10_ms,
                                                   bytes_per_sample,
                                                   audio_bus->channels(),
                                                   sample_rate,
                                                   audio_data,
-                                                  num_audio_frames);
+                                                  num_audio_frames,
+                                                  &rtp_ts,
+                                                  &ntp_ts);
       accumulated_audio_frames += num_audio_frames;
     }