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RTC2RTMP: Fix screen sharing stutter caused by packet loss. v5.0.216 v6.0.157 v7.0.18 (#4160)
## How to reproduce? 1. Refer this commit, which contains the web demo to capture screen as video stream through RTC. 2. Copy the `trunk/research/players/whip.html` and `trunk/research/players/js/srs.sdk.js` to replace the `develop` branch source code. 3. `./configure && make` 4. `./objs/srs -c conf/rtc2rtmp.conf` 5. open `http://localhost:8080/players/whip.html?schema=http` 6. check `Screen` radio option. 7. click `publish`, then check the screen to share. 8. play the rtmp live stream: `rtmp://localhost/live/livestream` 9. check the video stuttering. ## Cause When capture screen by the chrome web browser, which send RTP packet with empty payload frequently, then all the cached RTP packets are dropped before next key frame arrive in this case. The OBS screen stream and camera stream do not have such problem. ## Add screen stream to WHIP demo ><img width="581" alt="Screenshot 2024-08-28 at 2 49 46 PM" src="https://github.com/user-attachments/assets/9557dbd2-c799-4dfd-b336-5bbf2e4f8fb8"> --------- Co-authored-by: winlin <winlinvip@gmail.com>
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7 changed files with 67 additions and 20 deletions
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@ -1775,6 +1775,23 @@ srs_error_t SrsRtcFrameBuilder::packet_video_rtmp(const uint16_t start, const ui
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if (0 == nb_payload) {
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srs_warn("empty nalu");
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// The chrome web browser send RTP packet with empty payload frequently,
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// reset header_sn_, lost_sn_ and continue to found next frame in this case,
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// otherwise, all the cached RTP packets are dropped before next key frame arrive.
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header_sn_ = end + 1;
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uint16_t tail_sn = 0;
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int sn = find_next_lost_sn(header_sn_, tail_sn);
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if (-1 == sn) {
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if (check_frame_complete(header_sn_, tail_sn)) {
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err = packet_video_rtmp(header_sn_, tail_sn);
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}
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} else if (-2 == sn) {
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return srs_error_new(ERROR_RTC_RTP_MUXER, "video cache is overflow");
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} else {
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lost_sn_ = sn;
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}
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return err;
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}
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