Chromium Code Reviews| Index: webrtc/media/base/mediachannel.h |
| diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h |
| index dadb55bfdbae513e4bea0e255ecb377e7faf73b2..40869594e58956c2d3edc34ad5303732b318800a 100644 |
| --- a/webrtc/media/base/mediachannel.h |
| +++ b/webrtc/media/base/mediachannel.h |
| @@ -868,6 +868,8 @@ struct VideoMediaInfo { |
| } |
| std::vector<VideoSenderInfo> senders; |
| std::vector<VideoReceiverInfo> receivers; |
| + // Deprecated. |
| + // TODO(holmer): Remove once upstream projects no longer use this. |
| std::vector<BandwidthEstimationInfo> bw_estimations; |
|
hbos
2017/06/02 10:37:51
You've still removed filling in the bw_estimations
holmer
2017/06/02 11:34:26
Yes, that should be fine according to Noah.
|
| RtpCodecParametersMap send_codecs; |
| RtpCodecParametersMap receive_codecs; |
| @@ -1082,6 +1084,15 @@ class VideoMediaChannel : public MediaChannel { |
| // If SSRC is 0, the sink is used for the 'default' stream. |
| virtual bool SetSink(uint32_t ssrc, |
| rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) = 0; |
| + // This fills the "bitrate parts" (rtx, video bitrate) of the |
| + // BandwidthEstimationInfo, since that part that isn't possible to get |
| + // through webrtc::Call::GetStats, as they are statistics of the send |
| + // streams. |
| + // TODO(holmer): We should change this so that either BWE graphs doesn't |
| + // need access to bitrates of the streams, or change the (RTC)StatsCollector |
| + // so that it's getting the send stream stats separately by calling |
| + // GetStats(), and merges with BandwidthEstimationInfo by itself. |
| + virtual void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0; |
| // Gets quality stats for the channel. |
| virtual bool GetStats(VideoMediaInfo* info) = 0; |
| }; |