英文:
Pion WebRTC Audio stream cutting out while video works
问题
我正在尝试通过Pion WebRTC将一个MP4视频发送到浏览器。
使用FFmpeg，我将其分割为Opus OGG流和Annex-B H.264视频流。视频工作正常，但音频一直在中断和恢复。它正常播放几秒钟，然后停止一秒钟，然后继续播放。
这是我用于音频的FFmpeg命令：
ffmpeg -i demo.mp4 -c:a libopus -vn -page_duration 20000 demo.ogg
这是我的传输器（缩短版）：
var lastGranule uint64
for {
pageData, pageHeader, err := ogg.ParseNextPage() // 使用Pion OggReader
// 参考自play-from-disk示例
sampleCount := float64(pageHeader.GranulePosition - lastGranule)
lastGranule = pageHeader.GranulePosition
sampleDuration := time.Duration((sampleCount/48000)*1000) * time.Millisecond
err = audioTrack.WriteSample(media.Sample{Data: pageData, Duration: sampleDuration})
util.HandleError(err)
time.Sleep(sampleDuration)
}
我尝试将延迟硬编码为15毫秒，这样可以解决音频中断的问题，但是它会随机播放得太快或跳过部分内容。由于在更新FFmpeg命令之前我的视频存在故障（添加关键帧和删除B帧），我认为这也是编码器的问题。
可能的原因是什么？
更新：使用Chrome中的WebRTC日志记录，我发现经常出现以下日志行：
[27216:21992:0809/141533.175:WARNING:rtcp_receiver.cc(452)] 30 RTCP blocks were skipped due to being malformed or of unrecognized/unsupported type, during the past 10 second period.
这可能是中断的原因，尽管我无法弄清楚为什么会接收到格式错误的数据。
英文:
I am trying to send an MP4 video through Pion WebRTC to the browser.
Using FFmpeg, I split it into an Opus OGG stream and an Annex-B H.264 video stream. While the video works fine, the audio keeps cutting in and out. It plays fine for a few seconds, then stops for a second, and continues.
This is the FFmpeg command I use for audio:
ffmpeg -i demo.mp4 -c:a libopus -vn -page_duration 20000 demo.ogg
And this is my transmitter (shortened):
var lastGranule uint64
for {
pageData, pageHeader, err := ogg.ParseNextPage() // Uses Pion OggReader
// Taken from the play-from-disk example
sampleCount := float64(pageHeader.GranulePosition - lastGranule)
lastGranule = pageHeader.GranulePosition
sampleDuration := time.Duration((sampleCount/48000)*1000) * time.Millisecond
err = audioTrack.WriteSample(media.Sample{Data: pageData, Duration: sampleDuration})
util.HandleError(err)
time.Sleep(sampleDuration)
}
I tried hardcoding the delay to 15ms, which fixes the issue that it's cutting out, but then it randomly plays way too fast or starts skipping. Since I had glitchy video before updating my FFmpeg command (add keyframes and remove b-frames), I assume this is also an encoder problem.
What could be the cause for this?
Update: Using WebRTC logging in Chrome, I discovered the following log lines that occurred frequently:
[27216:21992:0809/141533.175:WARNING:rtcp_receiver.cc(452)] 30 RTCP blocks were skipped due to being malformed or of unrecognized/unsupported type, during the past 10 second period.
This is probably the reason for the cutouts, although I can't figure out why it receives malformed data.
答案1
得分: 1
问题最终是由于Go本身的问题#44343导致的Sleep时间不准确。这导致样本不是以恒定的速率发送，而是以随机的速率偏离5到15毫秒，从而导致流媒体不连续。
Sean DuBois和我在Pion存储库的最新play-from-disk和play-from-disk-h264示例中通过用Ticker替换for循环和Sleep()来修复了这个问题，Ticker更准确。
英文:
The problem in the end was an inaccuracy in the Sleep time caused by issue #44343 in Go itself. It caused the samples not to be sent at a constant rate, but at a rate that randomly was between 5 and 15ms off, resulting in a choppy stream.
Sean DuBois and me fixed this in the latest play-from-disk and play-from-disk-h264 examples in the Pion repository by replacing the for-loop and Sleep() with a Ticker, which is more accurate.
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