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Change default SSRC for RTCP receiver reports to not collide with video.

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BUG=chromium:547661
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1438183002

Cr-Commit-Position: refs/heads/master@{#10621}
This commit is contained in:
solenberg 2015-11-12 06:02:30 -08:00 committed by Commit bot
parent dfe434e20e
commit 8093d5442e
1 changed files with 3 additions and 2 deletions
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5
talk/media/webrtc/webrtcvoiceengine.h
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@ -315,9 +315,10 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
int64_t default_recv_ssrc_ = -1;
// Volume for unsignalled stream, which may be set before the stream exists.
double default_recv_volume_ = 1.0;
// SSRC to use for RTCP receiver reports; default to 1 in case of no signaled
// Default SSRC to use for RTCP receiver reports in case of no signaled
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
uint32_t receiver_reports_ssrc_ = 1;
// and https://code.google.com/p/chromium/issues/detail?id=547661
uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
class WebRtcAudioSendStream;
std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;