ef048b0d65
for issue #4418, #4151, #4076 .DVR Missing First Few Seconds of Audio/Video ### Root Cause When recording WebRTC streams to FLV files using DVR, the first 4-6 seconds of audio/video are missing. This occurs because: 1. **Packets are discarded before A/V sync is available**: The RTC-to-RTMP conversion pipeline actively discards all RTP packets when avsync_time <= 0. 2. **Original algorithm requires 2 RTCP SR packets**: The previous implementation needed to receive two RTCP Sender Report (SR) packets before it could calculate the rate for audio/video synchronization timestamp conversion. 3. **Delay causes packet loss**: Since RTCP SR packets typically arrive every 2-3 seconds, waiting for 2 SRs means 4-6 seconds of packets are discarded before A/V sync becomes available. 4. **Audio SR arrives slower than video SR**: As reported in the issue, video RTCP SR packets arrive much faster than audio SR packets. This asymmetry causes audio packets to be discarded for a longer period, resulting in the audio loss observed in DVR recordings. ### Solution 1. **Initialize rate from SDP**: Use the sample rate from SDP (Session Description Protocol) to calculate the initial rate immediately when the track is created. Audio (Opus): 48000 Hz → rate = 48 (RTP units per millisecond) Video (H.264/H.265): 90000 Hz → rate = 90 (RTP units per millisecond) 2. **Enable immediate A/V sync:** With the SDP rate available, cal_avsync_time() can calculate valid timestamps from the very first RTP packet, eliminating packet loss. 3. **Smooth transition to precise rate**: After receiving the 2nd RTCP SR, update to the precisely calculated rate based on actual RTP/NTP timestamp mapping. ## Configuration Added new configuration option `init_rate_from_sdp` in the RTC vhost section: ```nginx vhost rtc.vhost.srs.com { rtc { # Whether initialize RTP rate from SDP sample rate for immediate A/V sync. # When enabled, the RTP rate (units per millisecond) is initialized from the SDP # sample rate (e.g., 90 for video 90kHz, 48 for audio 48kHz) before receiving # 2 RTCP SR packets. This allows immediate audio/video synchronization. # The rate will be updated to a more precise value after receiving the 2nd SR. # Overwrite by env SRS_VHOST_RTC_INIT_RATE_FROM_SDP for all vhosts. # Default: off init_rate_from_sdp off; } } ``` **⚠️ Important Note**: This config defaults to **off** because: - ✅ When **enabled**: Fixes the audio loss problem (no missing first 4-6 seconds) - ❌ When **enabled**: VLC on macOS cannot play the video properly - ✅ Other platforms work fine (Windows, Linux) - ✅ FFplay works fine on all platforms Users experiencing audio loss in DVR recordings can enable this option if they don't need VLC macOS compatibility. We're investigating the VLC macOS issue to make this feature safe to enable by default in the future. --------- Co-authored-by: winlin <winlinvip@gmail.com> Co-authored-by: OSSRS-AI <winlinam@gmail.com>
33 lines
1.3 KiB
C++
33 lines
1.3 KiB
C++
/**
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* The MIT License (MIT)
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*
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* Copyright (c) 2013-2025 Winlin
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
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* the Software, and to permit persons to whom the Software is furnished to do so,
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* subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
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* FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
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* COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
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* IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
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* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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*/
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#ifndef SRS_UTEST_WORKFLOW_RTMP2RTC_HPP
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#define SRS_UTEST_WORKFLOW_RTMP2RTC_HPP
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/*
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#include <srs_utest_workflow_rtmp2rtc.hpp>
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*/
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#include <srs_utest.hpp>
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#endif
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