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Add interface to propagate audio capture timestamp to the renderer.

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BUG=3111
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
wu@webrtc.org
2014-05-19 17:39:11 +00:00
parent ebb467fdc8
commit cb711f77d2
15 changed files with 133 additions and 23 deletions
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6
webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
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@@ -473,6 +473,12 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
previous_audio_activity_ = audio_frame->vad_activity_;
call_stats_.DecodedByNetEq(audio_frame->speech_type_);
// Computes the RTP timestamp of the first sample in |audio_frame| from
// |PlayoutTimestamp|, which is the timestamp of the last sample of
// |audio_frame|.
audio_frame->timestamp_ =
PlayoutTimestamp() - audio_frame->samples_per_channel_;
return 0;
}