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
This commit is contained in:
stefan@webrtc.org
2014-11-05 14:05:29 +00:00
parent a22a628356
commit 0bae1fab4a
26 changed files with 315 additions and 129 deletions

View File

@ -33,16 +33,19 @@ struct RtpStatistics {
int extended_max_sequence_number;
};
struct StreamStats {
StreamStats()
struct SsrcStats {
SsrcStats()
: key_frames(0),
delta_frames(0),
bitrate_bps(0),
total_bitrate_bps(0),
retransmit_bitrate_bps(0),
avg_delay_ms(0),
max_delay_ms(0) {}
uint32_t key_frames;
uint32_t delta_frames;
int32_t bitrate_bps;
// TODO(holmer): Move bitrate_bps out to the webrtc::Call layer.
int total_bitrate_bps;
int retransmit_bitrate_bps;
int avg_delay_ms;
int max_delay_ms;
StreamDataCounters rtp_stats;