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WebRTC: Fix no audio and video issue for Firefox. (#3079)

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* Remove extern SrsPps* duplicate declarations

* fix(rtmp2rtc): fix video payload type for rtmp to rtc bridge (#3041)

* Revert changes not belongs to this PR.

* Fix naming issue, follow SRS style.

* Use srs_assert instead of assert.

* Fix firefox no audio issue.

Co-authored-by: winlin <winlin@vip.126.com>
This commit is contained in:
johzzy 2022-11-21 22:01:01 +08:00 • committed by GitHub
parent 62ab1a1c69
commit 6eb10afca2
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2 changed files with 13 additions and 6 deletions
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trunk/src/app/srs_app_rtc_source.hpp
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@ -261,6 +261,8 @@ private:
uint16_t video_sequence;
uint32_t audio_ssrc;
uint32_t video_ssrc;
uint8_t audio_payload_type_;
uint8_t video_payload_type_;
public:
SrsRtcFromRtmpBridge(SrsRtcSource* source);
virtual ~SrsRtcFromRtmpBridge();