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• vp9/x86 : idct_add_16x16_ssse3.

  14 décembre 2013, par Ronald S. Bultje

  vp9/x86 : idct_add_16x16_ssse3.
  Currently only dc-only and full 16x16. Other subforms will follow in the
  
  near future. Total decoding time of ped1080p.webm goes from 9.7 to 9.3
  
  seconds. DC-only goes from 957 -> 131 cycles, and the full IDCT goes
  
  from 4050 to 745 cycles.
  

  • [DH] libavcodec/x86/vp9dsp_init.c
  • [DH] libavcodec/x86/vp9itxfm.asm

• Fluent-ffmpeg "not a suitable output format"

  5 septembre 2015, par J4G

  I’m using the fluent-ffmpeg module for Node.js to convert audio files. I have a .mp3 file that I’d like to convert to .wma

  Here’s what that looks like :

  var proc = new ffmpeg({
  
     source: 'file.mp3',
  
     nolog: false
  
  }).toFormat('wma')                                                
  
    .saveToFile('file.wma', function(stdout, stderr)
  
  {                                                 
  
      console.log(stderr);
  
  });

  Unfortunately, I get the error :

  Requested output format 'wma' is not a suitable output format

  This is the entire error log :

  ffmpeg version 0.8.9-4:0.8.9-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
  
    built on Nov  9 2013 19:25:10 with gcc 4.6.3
  
  *** THIS PROGRAM IS DEPRECATED ***
  
  This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
  
  Input #0, mp3, from 'song_downloads/You Suffer.mp3':
  
    Metadata:
  
      title           : You Suffer
  
      artist          : Napalm Death
  
      album           : Scum
  
      genre           : Death Metal
  
      track           : 12
  
      date            : 1987
  
    Duration: 00:00:04.98, start: 0.000000, bitrate: 381 kb/s
  
      Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 193 kb/s
  
  Requested output format 'wma' is not a suitable output format

  I know this isn’t an ffmpeg issue because

  ffmpeg -i file.mp3 file.wma

  Works fine. Any ideas ?

• Stream RTP to FFMPEG using SDP

  9 avril 2021, par Johnathan Kanarek

  I get RTP stream from WebRTC server (I used mediasoup) using node.js and I get the decrypted RTP packets raw data from the stream.
I want to forward this RTP data to ffmpeg and from there I can save it to file, or push it as RTMP stream to other media servers.
I guess that the best way would be to create SDP file that describes both the audio and video streams and send the packets through new sockets.

  



  The ffmpeg command is :

  



  ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i test.sdp -vcodec libx264 -acodec aac -y output.mp4

  



  I tried to send the packets through UDP :

  



  v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 RTP/AVP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=sendrecv
m=video 33302 RTP/AVP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=sendrecv


  



  But I always get (removed the boring parts) :

  



  Opening an input file: test.sdp.

[sdp @ 0x103dea0]
Format sdp probed with size=2048 and score=50
[sdp @ 0x103dea0] audio codec set to: (null)
[sdp @ 0x103dea0] audio samplerate set to: 44100
[sdp @ 0x103dea0] audio channels set to: 1
[sdp @ 0x103dea0] video codec set to: (null)
[udp @ 0x10402e0] end receive buffer size reported is 131072
[udp @ 0x10400c0] end receive buffer size reported is 131072
[sdp @ 0x103dea0] setting jitter buffer size to 500
[udp @ 0x1040740] bind failed: Address already in use
[AVIOContext @ 0x1046980] Statistics: 473 bytes read, 0 seeks
test.sdp: Invalid data found when processing input


  



  Note that I get it even if I don't open socket at all or send anything to this port, as if the ffmpeg itself tries to open these ports more than once.

  



  I tried also to open two (video and audio) TCP servers and define SDP with TCP :

  



  v=0
o=mediasoup 7199daf55e496b370e36cd1d25b1ef5b9dff6858 0 IN IP4 192.168.193.182
s=7199daf55e496b370e36cd1d25b1ef5b9dff6858
c=IN IP4 192.168.193.182
t=0 0
m=audio 33301 TCP 111
a=rtpmap:111 /opus/48000
a=fmtp:111 minptime=10;useinbandfec=1
a=rtcp-fb:111 transport-cc
a=setup:active
a=connection:new
a=sendrecv
m=video 33302 TCP 100
a=rtpmap:100 /VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=setup:active
a=connection:new
a=sendrecv


  



  However I don't see any incoming connection into my TCP servers and I get the following from ffmpeg :

  



  Opening an input file: test.sdp.

[sdp @ 0xdddea0]
Format sdp probed with size=2048 and score=50

[sdp @ 0xdddea0]
audio codec set to: (null)

[sdp @ 0xdddea0]
audio samplerate set to: 44100
[sdp @ 0xdddea0] audio channels set to: 1
[sdp @ 0xdddea0] video codec set to: (null)
[udp @ 0xde02e0] end receive buffer size reported is 131072
[udp @ 0xde00c0] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[udp @ 0xde0740] end receive buffer size reported is 131072

[udp @ 0xde0180] end receive buffer size reported is 131072
[sdp @ 0xdddea0] setting jitter buffer size to 500
[sdp @ 0xdddea0] Before avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 nb_streams:2
[libvpx @ 0xdeea80] v1.3.0
[libvpx @ 0xdeea80] --target=x86_64-linux-gcc --enable-pic --disable-install-srcs --as=nasm --enable-shared --prefix=/usr --libdir=/usr/lib64

[sdp @ 0xdddea0] Could not find codec parameters for stream 1 (Video: vp8, 1 reference frame, none): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[sdp @ 0xdddea0] After avformat_find_stream_info() pos: 593 bytes read:593 seeks:0 frames:0
Input #0, sdp, from 'test.sdp':
  Metadata:
    title           : 7199daf55e496b370e36cd1d25b1ef5b9dff6858
  Duration: N/A, bitrate: N/A
    Stream #0:0, 0, 1/90000: Audio: opus, 48000 Hz, mono, fltp
    Stream #0:1, 0, 1/90000: Video: vp8, 1 reference frame, none, 90k tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output file output.mp4.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 0xde3660] Setting default whitelist 'file,crypto'
Successfully opened the file.

detected 1 logical cores
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'time_base' to value '1/48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_rate' to value '48000'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'sample_fmt' to value 'fltp'
[graph 0 input from stream 0:0 @ 0xde3940] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:0 @ 0xde3940] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x4
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_fmts' to value 'fltp'
[audio format for output stream 0:0 @ 0xe37900] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 0xde0220] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed

Output #0, mp4, to 'output.mp4':

  Metadata:

    title           :
7199daf55e496b370e36cd1d25b1ef5b9dff6858


    encoder         :
Lavf57.56.100


    Stream #0:0
, 0, 1/48000
: Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, mono, fltp, delay 1024, 69 kb/s


    Metadata:

      encoder         :
Lavc57.64.100 aac


Stream mapping:

  Stream #0:0 -> #0:0 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)

test.sdp: Connection timed out
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[output stream 0:0 @ 0xde3b40] EOF on sink link output stream 0:0:default.
No more output streams to write to, finishing.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
[mp4 @ 0xe6a540] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
[mp4 @ 0xe6a540] Encoder did not produce proper pts, making some up.
[aac @ 0xde2b00] Trying to remove 1024 samples, but the queue is empty
[aac @ 0xde2b00] Trying to remove 1024 more samples than there are in the queue
size=       1kB time=00:00:00.04 bitrate= 157.9kbits/s speed=0.00426x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 3268.000000%
Input file #0 (test.sdp):
  Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
  Input stream #0:1 (video): 0 packets read (0 bytes);
  Total: 0 packets (0 bytes) demuxed
Output file #0 (output.mp4):
  Output stream #0:0 (audio): 0 frames encoded (0 samples); 2 packets muxed (25 bytes);
  Total: 2 packets (25 bytes) muxed
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0xde37a0] Statistics: 30 seeks, 25 writeouts
[aac @ 0xde2b00] Qavg: 47249.418

[AVIOContext @ 0xde6980] Statistics: 593 bytes read, 0 seeks


  



  Note to the "Connection timed out" in the log above.

  



  I guess that both my SDPs are wrong, any suggestions ?

  



  Alternatives to SDP are also most welcomed.