I am trying to combining audio and video for streaming from a Raspberry Pi. The audio leads the video by a few seconds, whether I stream the output or write it to a file.

Command line:

raspivid -o - -t 0 -n -w 1280 -h 720 -fps 25 | ffmpeg -report -probesize 32000 -thread_queue_size 1024 -y -f h264 -r 25 -framerate 25 -i - -thread_queue_size 512 -f alsa -ac 1 -channel_layout mono -i hw:1 -vcodec copy -acodec mp3 -f mpegts udp://

ffmpeg report/log:

ffmpeg started on 2015-06-15 at 19:14:26
Report written to "ffmpeg-20150615-191426.log"
Command line:
ffmpeg -report -probesize 32000 -thread_queue_size 1024 -y -f h264 -r 25 -framerate 25 -i - -thread_queue_size 512 -f alsa -ac 1 -channel_layout mono -i hw:1 -vcodec copy -acodec mp3 -f mpegts udp://
ffmpeg version N-72834-gfd004e1 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.6 (Debian 4.6.3-14+rpi1)
  configuration: --enable-libmp3lame
  libavutil      54. 27.100 / 54. 27.100
  libavcodec     56. 41.100 / 56. 41.100
  libavformat    56. 36.100 / 56. 36.100
  libavdevice    56.  4.100 / 56.  4.100
  libavfilter     5. 16.101 /  5. 16.101
  libswscale      3.  1.101 /  3.  1.101
  libswresample   1.  2.100 /  1.  2.100
Splitting the commandline.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Reading option '-probesize' ... matched as AVOption 'probesize' with argument '32000'.
Reading option '-thread_queue_size' ... matched as option 'thread_queue_size' (set the maximum number of queued packets from the demuxer) with argument '1024'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'h264'.
Reading option '-r' ... matched as option 'r' (set frame rate (Hz value, fraction or abbreviation)) with argument '25'.
Reading option '-framerate' ... matched as AVOption 'framerate' with argument '25'.
Reading option '-i' ... matched as input file with argument '-'.
Reading option '-thread_queue_size' ... matched as option 'thread_queue_size' (set the maximum number of queued packets from the demuxer) with argument '512'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'alsa'.
Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '1'.
Reading option '-channel_layout' ... matched as option 'channel_layout' (set channel layout) with argument 'mono'.
Reading option '-i' ... matched as input file with argument 'hw:1'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'mp3'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'mpegts'.
Reading option 'udp://' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option report (generate a report) with argument 1.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file -.
Applying option thread_queue_size (set the maximum number of queued packets from the demuxer) with argument 1024.
Applying option f (force format) with argument h264.
Applying option r (set frame rate (Hz value, fraction or abbreviation)) with argument 25.
Successfully parsed a group of options.
Opening an input file: -.
[h264 @ 0x12a83c0] Before avformat_find_stream_info() pos: 0 bytes read:4096 seeks:0
[h264 @ 0x12a83c0] Probe buffer size limit of 32000 bytes reached
[h264 @ 0x12a83c0] Stream #0: not enough frames to estimate rate; consider increasing probesize
[h264 @ 0x12a83c0] decoding for stream 0 failed
[h264 @ 0x12a83c0] After avformat_find_stream_info() pos: 34816 bytes read:53248 seeks:0 frames:3
Input #0, h264, from 'pipe:':
  Duration: N/A, bitrate: N/A
    Stream #0:0, 3, 1/1200000: Video: h264 (High), yuv420p, 1280x720, 25 fps, 25 tbr, 1200k tbn, 50 tbc
Successfully opened the file.
Parsing a group of options: input file hw:1.
Applying option thread_queue_size (set the maximum number of queued packets from the demuxer) with argument 512.
Applying option f (force format) with argument alsa.
Applying option ac (set number of audio channels) with argument 1.
Applying option channel_layout (set channel layout) with argument mono.
Successfully parsed a group of options.
Opening an input file: hw:1.
[alsa @ 0x12cee80] All info found
Input #1, alsa, from 'hw:1':
  Duration: N/A, start: 1434392072.070301, bitrate: 768 kb/s
    Stream #1:0, 1, 1/1000000: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s
Successfully opened the file.
Parsing a group of options: output file udp://
Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument mp3.
Applying option f (force format) with argument mpegts.
Successfully parsed a group of options.
Opening an output file: udp://
Matched encoder 'libmp3lame' for codec 'mp3'.
Successfully opened the file.
detected 4 logical cores
[graph 0 input from stream 1:0 @ 0x12a82a0] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 1:0 @ 0x12a82a0] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 1:0 @ 0x12a82a0] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 1:0 @ 0x12a82a0] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 1:0 @ 0x12a82a0] tb:1/48000 samplefmt:s16 samplerate:48000 chlayout:0x4
[audio format for output stream 0:1 @ 0x12a0390] Setting 'sample_fmts' to value 's32p|fltp|s16p'
[audio format for output stream 0:1 @ 0x12a0390] Setting 'sample_rates' to value '44100|48000|32000|22050|24000|16000|11025|12000|8000'
[audio format for output stream 0:1 @ 0x12a0390] Setting 'channel_layouts' to value '0x4|0x3'
[audio format for output stream 0:1 @ 0x12a0390] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
[AVFilterGraph @ 0x1320b60] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x1321c60] ch:1 chl:mono fmt:s16 r:48000Hz -> ch:1 chl:mono fmt:s16p r:48000Hz
[mpegts @ 0x12bdbf0] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts
Output #0, mpegts, to 'udp://':
    encoder         : Lavf56.36.100
    Stream #0:0, 0, 1/90000: Video: h264, yuv420p, 1280x720, q=2-31, 25 fps, 25 tbr, 90k tbn, 25 tbc
    Stream #0:1, 0, 1/90000: Audio: mp3 (libmp3lame), 48000 Hz, mono, s16p
      encoder         : Lavc56.41.100 libmp3lame
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #1:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
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frame= 1724 fps= 25 q=-1.0 size=   33797kB time=00:01:08.96 bitrate=4014.9kbits/s    
frame= 1737 fps= 25 q=-1.0 size=   34091kB time=00:01:09.48 bitrate=4019.5kbits/s    
frame= 1750 fps= 25 q=-1.0 size=   34451kB time=00:01:10.00 bitrate=4031.8kbits/s    
frame= 1763 fps= 25 q=-1.0 size=   34750kB time=00:01:10.52 bitrate=4036.7kbits/s    
frame= 1775 fps= 25 q=-1.0 size=   35010kB time=00:01:11.00 bitrate=4039.5kbits/s    
[libmp3lame @ 0x1320770] Trying to remove 47 more samples than there are in the queue
frame= 1779 fps= 25 q=-1.0 Lsize=   35116kB time=00:01:11.28 bitrate=4035.7kbits/s    
video:31803kB audio:557kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 8.515517%
Input file #0 (pipe:):
  Input stream #0:0 (video): 1779 packets read (32566370 bytes); 
  Total: 1779 packets (32566370 bytes) demuxed
Input file #1 (hw:1):
  Input stream #1:0 (audio): 57720 packets read (6842992 bytes); 57720 frames decoded (3421496 samples); 
  Total: 57720 packets (6842992 bytes) demuxed
Output file #0 (udp://
  Output stream #0:0 (video): 1779 packets muxed (32566370 bytes); 
  Output stream #0:1 (audio): 2970 frames encoded (3421440 samples); 2971 packets muxed (570432 bytes); 
  Total: 4750 packets (33136802 bytes) muxed
57720 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x1321210] Statistics: 0 seeks, 25491 writeouts
[AVIOContext @ 0x12a7e40] Statistics: 35211617 bytes read, 0 seeks
Received signal 2: terminating.
  • Are you running your Pi headless or do you have a monitor attached? I ask because I've found that H264 video from webcams has a tendency to lag by a couple of seconds when viewing in real time (i.e., lip sync's off when one's speaking into the camera), and you'd need a monitor to verify whether or not this is the source of the problem. – Mr. What Jun 22 '15 at 4:52
  • I'm not sure what tool I would use to test this. How would I know that the problem was not with the tool? – Neik Jul 2 '15 at 16:33
  • If your Pi has a monitor, using the raspivid preview function (e.g., raspivid -t 0 -p '0,0,1280,720') should allow you to check output sync in real time. If you're speaking into the camera, for instance, you'll immediately know if the camera's lagging behind. If you're running headless, trying to diagnose sync problems by way of a viewable stream isn't going to help much. You can, however, attempt to delay the audio by using an -itsoffset option. Details coming shortly in a proposed solution. – Mr. What Jul 2 '15 at 20:05

Here's what I'd suggest: If you're able to test the output sync of your camera by using a raspvid preview window and you find the camera to be lagging behind, then I'd say the camera's h264 encoder is to blame. If it's in sync, however, then FFmpeg is doing a number on the piped input.

If the camera's h264 encoder is the problem and the video lag behind the audio is of a consistent duration (i.e., you observe a 2.5-second lag every time), you can add an -itsoffset parameter to the audio input to delay it by that duration. E.g:

raspivid -o - -t 0 -n -w 1280 -h 720 -fps 25 | \
ffmpeg -report -y \
-f h264 -r 25 -i - \
-itsoffset 2.5 -f alsa -ac 1 -i hw:1 \
-vcodec copy -acodec libmp3lame -ac 1 -ar 44100 -b:a 128k \
-f mpegts udp://

If there's no lag in the camera's preview output, then I'd suggest installing the camera's V4L2 driver so your FFmpeg command can pull frames from the camera directly.

ffmpeg -report -y \
-f v4l2 -input_format h264 -r 25 -video_size 1280x720 -i /dev/video0 \
-f alsa -ac 1 -i hw:1 \
-vcodec copy -acodec libmp3lame -ac 1 -ar 44100 -b:a 128k \
-f mpegts udp://

However, from what I've gathered online, folks have complaints about the performance of the V4L2 driver of this camera, so you might end up sticking with the devil you know and using -itsoffset on the audio input regardless.

Command Notes: I've trimmed off the probesize option from your command since you're specifying the input format and framerate. Also, you shouldn't specify both -r 25 (which 'forces' the framerate to specified value) and -framerate 25 (which would attempt to use the 'native' input framerate). I've left -r 25 in because I've found that it works better for live video inputs. The MP3 encoder settings are more specific just so FFmpeg doesn't have to "guess" about them. -thread_queue_size params were trimmed off for line-length, so you can add them back in if absolutely necessary.


try to restream audio on rtp:// and combine the video in mp4 container.

  • How do you do this? What will it achieve? How does it solve the askers issue? As it stands this could be useful but it is very low quality. – Mokubai Jun 21 '15 at 7:02

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