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• Modify H.264 Level in MP4 stream

  25 mars 2013, par d33pika

  I have 1000+ videos with wrong Level information, and hence do not play on some devices. They are H.264 baseline with aac audio in mp4 container. I want to correct them but avoid re-encoding. Am I right in assuming that this is possible and will be faster ? How do I identify the level_idc bits in the streams ? Will all streams have them in the same location ? I expect this to be in just one place in the stream, is this assumption correct ?
  Update : I came across a windows executable link that claims to do this. I want to be able to do this commandline on unix, so I can script it.

  More Updates : As per Roman's answer below. I dug up MP4 syntax and found that Profile and Level information are part of avcC atom under stsd. I could see that level was set to 1E, I corrected it to 1F. Before :

  61 76 63 43 01 42 C0 1E FF E1 00 19 67 42 C0 1E
D9

  After :

  61 76 63 43 01 42 C0 1F FF E1 00 19 67 42 C0 1E
D9

  with that the MediaInfo changed to :

  Video
  
  ID                                       : 1
  
  Format                                   : AVC
  
  Format/Info                              : Advanced Video Codec
  
  Format profile                           : Baseline@L3.0
  
  Format settings, CABAC                   : No
  
  Format settings, ReFrames                : 3 frames
  
  Muxing mode                              : Container profile=Baseline@3.1
  
  Codec ID                                 : avc1
  
  Codec ID/Info                            : Advanced Video Coding
  
  Duration                                 : 1mn 4s
  
  Bit rate                                 : 3 000 Kbps
  
  Width                                    : 1 280 pixels
  
  Height                                   : 720 pixels
  
  Display aspect ratio                     : 16:9
  
  Frame rate mode                          : Constant
  
  Frame rate                               : 30.000 fps
  
  Color space                              : YUV
  
  Chroma subsampling                       : 4:2:0
  
  Bit depth                                : 8 bits
  
  Scan type                                : Progressive
  
  Bits/(Pixel*Frame)                       : 0.109
  
  Stream size                              : 22.7 MiB (92%)
  
  Writing library                          : x264 core 120 r2151 a3f4407
  
  Encoding settings                        : cabac=0 / ref=3 / deblock=1:0:0 / analyse=0x1:0x111 / me=hex / subme=7 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=12 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=0 / weightp=0 / keyint=250 / keyint_min=25 / scenecut=40 / intra_refresh=0 / rc_lookahead=40 / rc=abr / mbtree=1 / bitrate=3000 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / ip_ratio=1.40 / aq=1:1.00 

  It introduced a Muxing mode parameter with level 3.1 but alas this stream did not play on the device either. How do I correct the level indicated in Format Profile Is this possible without re-encoding ?

• Audioqueue try to read ADPCM sound

  4 septembre 2013, par Kamax

  I try to read a .asf stream with FFMPEG and the audio part is ADPCM IMA WAV codec.

  The sound i hear when i hear something is only noise. I suppose my AudioStreamBasicDescription are wrong. How can i get the correct values ?

  From ffprobe i have this :

  [STREAM]
  
  index=1
  
  codec_name=adpcm_ima_wav
  
  codec_long_name=ADPCM IMA WAV
  
  codec_type=audio
  
  codec_time_base=1/8000
  
  codec_tag_string=[17][0][0][0]
  
  codec_tag=0x0011
  
  sample_rate=8000.000000 
  
  channels=1
  
  bits_per_sample=4
  
  r_frame_rate=0/0
  
  avg_frame_rate=250/41
  
  time_base=1/1000
  
  start_time=0.000000 
  
  duration=N/A
  
  [/STREAM]

  and

  Stream #0.1: Audio: adpcm_ima_wav, 8000 Hz, 1 channels, s16, 32 kb/s

  my code for the moment is :

  asbd.mSampleRate = 8000;
  
  asbd.mFormatID = kAudioFormatAppleIMA4;
  
  asbd.mBytesPerPacket = 34;
  
  asbd.mFramesPerPacket = 64;
  
  asdb.mbitsPerChannel = 1;

  All the rest is to 0 (flags, byte per frame, bits per channel, reserved).

  Edit :

  I just find that the codec code 17 is maybe for kAudioFormatDVIIntelIMA and not kAudioFormatAppleIMA4. Can someone confirm this ?

  This is from ffprobe :

  [PACKET]
  
  codec_type=audio
  
  stream_index=1
  
  pts=11200
  
  pts_time=11.200000 
  
  dts=11200
  
  dts_time=11.200000 
  
  duration=164
  
  duration_time=0.164000 
  
  size=656.000000 
  
  pos=1171105
  
  flags=K
  
  [/PACKET]

  Is the size 656 for mBytesperpacket ?

  With this value and the half 328 for mFramesPerPacket i can hear something but it's not continuous and has a lot of noise.

  Please help !

• ffmpeg - converting pngs with white background to movie give black background

  14 octobre 2015, par Population Xplosive

  I have a list of files with their names in the form [1-1000].00.png.
  All these are flattened images (graphs) with white background color.
  I am trying to make a movie out of these files using ffmpeg. However, when I use ffmpeg, everything white in the images become black in the movie. And everything black remains the same. The command I used is :

  The command I use is :

  ffmpeg -pix_fmt yuv420p -r 8 -f image2 -pattern_type glob -i
  ’*.00.png’ movie.mp4