Roll webrtc/libjingle to 6187:6189.

BUG=
R=wjia@chromium.org

Review URL: https://codereview.chromium.org/298433003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@271491 0039d316-1c4b-4281-b951-d872f2087c98

DEPS

@@ -55,7 +55,7 @@ vars = {
   # Three lines of non-changing comments so that
   # the commit queue can handle CLs rolling WebRTC
   # and V8 without interference from each other.
-  "webrtc_revision": "6187",
+  "webrtc_revision": "6189",
   "jsoncpp_revision": "248",
   "nss_revision": "267366",
   # Three lines of non-changing comments so that

content/renderer/media/webrtc_audio_device_impl.cc

@@ -150,6 +150,8 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
   int16* audio_data = &render_buffer_[0];
   while (accumulated_audio_frames < audio_bus->frames()) {
     // Get 10ms and append output to temporary byte buffer.
+    uint32_t rtp_ts = 0;
+    int64_t ntp_ts = 0;
     if (is_audio_track_processing_enabled_) {
       // When audio processing is enabled in the audio track, we use
       // PullRenderData() instead of NeedMorePlayData() to avoid passing the
@@ -160,7 +162,9 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
                                                 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
@@ -170,7 +174,9 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
                                                   audio_bus->channels(),
                                                   sample_rate,
                                                   audio_data,
-                                                  num_audio_frames);
+                                                  num_audio_frames,
+                                                  &rtp_ts,
+                                                  &ntp_ts);
       accumulated_audio_frames += num_audio_frames;
     }
 

third_party/libjingle/README.chromium

 Name: libjingle
 URL: http://code.google.com/p/webrtc/
 Version: unknown
-Revision: 6174
+Revision: 6189
 License: BSD
 License File: source/talk/COPYING
 Security Critical: yes