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• avcodec/aacsbr_template : Remove pointless runtime initialization

  27 décembre 2020, par Andreas Rheinhardt

  avcodec/aacsbr_template : Remove pointless runtime initialization
  The sbr_qmf_window_us array is basically symmetric around its middle
  
  element and therefore the latter half is currently initialized from the
  
  first half at runtime. Yet because the first half is initialized, the
  
  array can't be placed in .bss at all, so that one gains nothing from not
  
  already initializing the whole array statically. Therefore this commit
  
  does exactly this.
  (There are two exceptions to the symmetry : Elements 384 and 512 are the
  
  negations of their mirror element ; for the fixed-point decoder, Q31(-x)
  
  does not equal -Q31(x). In order to keep the array exactly the same, the
  
  latter form has been used for these two elements.)
  Reviewed-by : Lynne <dev@lynne.ee>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavcodec/aacsbr_template.c
  • [DH] libavcodec/aacsbrdata.h

• FFMPEG : Timestamps are unset in a packet for stream 0.Non-monotonous DTS in output stream 0:0

  22 février 2018, par Lily Priya

  I’m using FFMPEG to connect an RTSP and create video files on the fly that can be viewed in a mpeg-dash compatible browser using HTML5 video element and dash.js

  ffmpeg to connect to your ip camera

  ffmpeg -i rtsp ://194.168.26.234:554 -r 64k -an -c:v copy -b:v 2000k
  -fflags nobuffer -f dash -window_size 4 -extra_window_size 0 -min_seg_duration 2000000 -remove_at_exit 1 ..\www\manifest.mpd

  When I Run the command,it streaming properly but in the middle I’m getting error on Timestamps and also in Non-monotonous DTS. And also it was 3-15 sec delay Here My error,

  [dash @ 000000d62f752f20] 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 timest amps properly
  [dash @ 000000d62f752f20] Non-monotonous DTS in output stream 0:0 ;
  previous : 0, current : 0 ;changing to 1. This may result in incorrect
  timestamps in the output file.

  [mp4 @ 000000d6302824e0] Application provided duration : -64 /
  timestamp : 1275009 is out of range for mov/mp4 format

  frame= 513 fps= 34 q=-1.0 size=N/A time=00:00:20.40 bitrate=N/A
  speed=1.35x frame= 527 fps= 34 q=-1.0 size=N/A time=00:00:20.96
  bitrate=N/A speed=1.34x

  [dash @ 000000d62f752f20] Non-monotonous DTS in output stream 0:0 ; previous : 134 1505, current : 1341505 ; changing to 1341506. This may result in incorrect timest amps in the output file.

• How can I encode and segment audio files without having gaps (or audio pops) between segments when I reconstruct it ?

  16 mai 2013, par fenduru

  I'm working on a web application that requires streaming and synchronization of multiple audio files. For this, I am using the Web Audio API over HTML5 audio tags because of the importance of timing audio.

  Currently, I'm using FFMPEG's segmentation feature to encode and segment the audio files into smaller chunks. The reason I am segmenting them is so I can start streaming from the middle of the file instead of starting from the beginning (otherwise I would've just split the files using UNIX split, as shown here. The problem is that when I string the audio segments back together, I get an audio pop between segments.

  If I encode the segments using a PCM encoding (pcm_s24le) in a .wav file, the playback is seamless, which leads me to believe that the encoder is padding either the beginning or the end of the file. Since I will be dealing with many different audio files, using .wav would require far too much bandwidth.

  I'm looking to one of the following solutions to the problem :

  • How can I segment encoded audio files seamlessly,
  • How can I force an encoder to NOT pad audio frames using ffmpeg (or another utility), or
  • What is a better way to stream audio (starting at an arbitrary track time) without using an audio tag ?

  System Information

  • Custom node.js server
  • Upon upload of an audio file, node.js pipes the data into ffmpeg's encoder
  • Need to use HTML5 Web Audio API supported encoding
  • Server sends audio chunks 1 at a time through a WebSockets socket

  Thanks in advance. I've tried to be as clear as possible but if you need clarification I'd be more than willing to provide it.