Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats. BUG=1788 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -33,16 +33,19 @@ struct RtpStatistics {
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int extended_max_sequence_number;
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};
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struct StreamStats {
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StreamStats()
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struct SsrcStats {
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SsrcStats()
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: key_frames(0),
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delta_frames(0),
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bitrate_bps(0),
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total_bitrate_bps(0),
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retransmit_bitrate_bps(0),
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avg_delay_ms(0),
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max_delay_ms(0) {}
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uint32_t key_frames;
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uint32_t delta_frames;
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int32_t bitrate_bps;
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// TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
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int total_bitrate_bps;
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int retransmit_bitrate_bps;
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int avg_delay_ms;
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int max_delay_ms;
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StreamDataCounters rtp_stats;
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