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• How can I capture audio AND video simultenaous with ffmpeg from an USB capture device

  22 octobre 2018, par oban

  I’m capturing a video by means of an USB Terratec Grabster AV350 (which is based on the em2860 chip).

  I don’t succeed to get the audio when it is played . If I play the captured video with vlc or with ffplay I got only 3 seconds sound and then a silence for the rest of the video ...

  During the capturing I don’t get any errors. At the end it indicates the size of the video and audio captured ....

  I’m using the ffmpeg command for this :

  ffmpeg -f alsa -ac 2 -i hw:3 -f video4linux2 -i /dev/video0 -acodec ac3 -ab 128k -vcodec mpeg4 -b 6000k -r 25 test5.avi

  The log is :

  [alsa @ 0x9bcd420]Estimating duration from bitrate, this may be inaccurate
  
  Input #0, alsa, from 'hw:3':
  
  Duration: N/A, start: 69930.998994, bitrate: N/A
  
  Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  [video4linux2 @ 0x9bf5d30]Estimating duration from bitrate, this may be inaccurate
  
  Input #1, video4linux2, from '/dev/video0':
  
  Duration: N/A, start: 1307111377.654173, bitrate: -2147483 kb/s
  
  Stream #1.0: Video: rawvideo, yuyv422, 720x576, -2147483 kb/s, 1000k tbr, 1000k tbn, 1000k tbc
  
  [ac3 @ 0x9bf9590]No channel layout specified. The encoder will guess the layout, but it might be incorrect.
  
  Output #0, avi, to 'test5.avi':
  
  Metadata:
  
  ISFT : Lavf52.64.2
  
  Stream #0.0: Video: mpeg4, yuv420p, 720x576, q=2-31, 6000 kb/s, 25 tbn, 25 tbc
  
  Stream #0.1: Audio: ac3, 44100 Hz, stereo, s16, 128 kb/s
  
  Stream mapping:
  
  Stream #1.0 -> #0.0
  
  Stream #0.0 -> #0.1
  
  Press [q] to stop encoding
  
  frame= 1283 fps= 25 q=2.3 Lsize= 38677kB time=51.32 bitrate=6173.9kbits/s
  
  **video:37755kB audio:846kB** global headers:0kB muxing overhead 0.198922%

  If I reduce the command for only capturing audio, then the audio file can be played successfully :

  ffmpeg -f alsa -ac 2 -i hw:3,0 -acodec ac3 -ab 128k test5.avi

  [alsa @ 0x8ede420]Estimating duration from bitrate, this may be inaccurate
  
  Input #0, alsa, from 'hw:3,0':
  
  Duration: N/A, start: 70395.998935, bitrate: N/A
  
  Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  [ac3 @ 0x8eebac0]No channel layout specified. The encoder will guess the layout, but it might be incorrect.
  
  Output #0, avi, to 'test5.avi':
  
  Metadata:
  
  ISFT : Lavf52.64.2
  
  Stream #0.0: Audio: ac3, 44100 Hz, stereo, s16, 128 kb/s
  
  Stream mapping:
  
  Stream #0.0 -> #0.0
  
  Press [q] to stop encoding
  
  size= 227kB time=13.62 bitrate= 136.8kbits/s
  
  **video:0kB audio:213kB** global headers:0kB muxing overhead 6.902375%

  If I run the command for only video capturing then vlc or ffplay can play the video successfully :

  ffmpeg -f video4linux2 -i /dev/video0 -vcodec mpeg4 -b 12000k -r 25 test5.avi

  [video4linux2 @ 0x91d6420]Estimating duration from bitrate, this may be inaccurate
  
  Input #0, video4linux2, from '/dev/video0':
  
  Duration: N/A, start: 1307112044.025687, bitrate: -2147483 kb/s
  
  Stream #0.0: Video: rawvideo, yuyv422, 720x576, -2147483 kb/s, 1000k tbr, 1000k tbn, 1000k tbc
  
  Output #0, avi, to 'test5.avi':
  
  Metadata:
  
  ISFT : Lavf52.64.2
  
  Stream #0.0: Video: mpeg4, yuv420p, 720x576, q=2-31, 12000 kb/s, 25 tbn, 25 tbc
  
  Stream mapping:
  
  Stream #0.0 -> #0.0
  
  Press [q] to stop encoding
  
  frame= 388 fps= 25 q=2.0 Lsize= 12963kB time=15.52 bitrate=6842.5kbits/s
  
  **video:12949kB audio:0kB** global headers:0kB muxing overhead 0.114584%

  Strange behaviour I noticed is that when I tried capturing video and audio, I can not capture the audio afterwards any more,
  unless I unplug the AV350 first.

  The G350 is located at card 3 :

  htpc@htpc-01:/proc/asound/G350/pcm0c$ more info
  
  card: 3
  
  device: 0
  
  subdevice: 0
  
  stream: CAPTURE
  
  id: USB Audio
  
  name: USB Audio
  
  subname: subdevice #0
  
  class: 0
  
  subclass: 0
  
  subdevices_count: 1
  
  subdevices_avail: 1

  The OS is a Linux 2.6.38-8-generic with the Ubuntu Natty Narwhal version

  Any help on how to tackle this issue would be great ....

  Thanks !

• ffmpeg disconnects with different audio sample rate

  11 juillet 2017, par HRG

  Am using ffmpeg transcoder tool to convert a streaming audio input to another format.The raw audio data comes from a input device in wav format (.wmv) which can be accessed via server port (ex.8080) . The audio format is in wav with 16bits/sample and sample rate of 6000 hz.
  When I use ffmpeg to read this audio input with above specs and convert to another format it works fine.

  Ex.
  ffmpeg -f s16le -ar 6000 -ac 1 -i http://local:8080/sampleaudio.wmv -f webm out.webm

  But if i use fmmpeg to read the audio input at sample rate of 4000hz (other than the actual rate) ,it disconnects from the server always .

  Ex.
  ffmpeg -f s16le -ar 4000 -ac 1 -i http://local:8080/sampleaudio.wmv -f webm out.webm

  The error in the ffmpeg console is "Stream ends prematurely at 194604, should be 4800044"

  I know the input rate at which ffmpeg is reading and the actual audio output is different .But am curious to know

  why ffmpeg is disconnecting from the server ,what is happening behind the scene.

  Also how to enable logs for network connections(http,sockets..)

  Thanks

• Why does avcodec_fill_audio_frame return -22 when only sample count is different ?

  24 juillet 2017, par N0un

  My problem is very fast to explain : I have to encode audio samples using FFmpeg (raw PCM to G.711 mu-law). This is the guilty part of my code (I put raw parameters in this example to be explicit) :

  AVFrame* frame = av_frame_alloc();
  
  frame->nb_samples = 8000;
  
  frame->format = AV_SAMPLE_FMT_S16;
  
  frame->channels = 1;
  
  frame->channel_layout = AV_CH_LAYOUT_MONO;
  
  frame->sample_rate = 8000;
  
  frame->quality = 1;
  
  
  
  int res = avcodec_fill_audio_frame(frame, 1, AV_SAMPLE_FMT_S16, /*my samples data*/, 16000, 0);
  
  
  
  // If res >= 0, continue with avcodec_encode_audio2

  And it works :) ... Well, I mean...

  When my input is 8000 audio samples (S16 format so 16000 bytes), it works. But when I have 6000 audio samples (still S16 format so 12000 bytes), it fails with a -22 (invalid parameters). Any idea ?

  PRECISION : This sample count is not dynamically changing. I have sessions with data always composed by 8000 sample (and it works), and other sessions with data always composed by 6000 sample (and it fails). Sample count and data size are the only parameters that are not the same between these sessions.

  EDIT : If I set or not the frame_size field in AVCodecContext, it returns to 0 after avcodec_open2 but the mu-law encoder selected has the AV_CODEC_CAP_VARIABLE_FRAME_SIZE capability so it sounds normal.