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• why I can't compile x264 or ffmpeg downloaded from git

  4 avril 2013, par Monjura Rumi

  It has been just a few days that I am using x264, ffmpeg and trying to take help from codes in git. I am always having some peculiar problem. It seems like no one or few people faced such problems. Here is one of those problems. If I download x264 or ffmpeg using git then I could not compile that source code. At first terminal will show ./configure: Permission denied. Then If I change permission of configure file using chmod 755 command then it will show ./configure: /bin/bash^M: bad interpreter: No such file or directory. To build x264 and ffmpeg I have followed this link https://github.com/lince/libffmpeg-c/wiki. My attempt to compile x264 has failed. Can you please tell me how can one use source code downloaded from git ?

• Skiping MP4 video

  30 avril 2013, par M.Yazdian

  I like to create the video player to show videos on my website. I made a video player in flash CS5 and actionscript3 to stream the videos on the website.

  My videos are often flv or mp4 format.
  In the flv files I use flvmdi app to inject the metadata to that ,and for skipping the flv file I use xmoov-php (video player can pass the keyframe number from flv metadat to xmoov-php to create the new flv file from passed keyframe to the last flv keyframe, and return the flv video file on the php header for flash video player to show the video ).

  but about mp4 format in Xmoov-php wiki it says :
  this file can't support the mp4 format to skiping...!!!
  Now I need skiping the mp4 files in my video player (I like my viewers select a part of video before buffering all video on their browser and play it ).

  please give me a suggestion to skiping the mp4 file in the php or .net platform
  Many thanks

• CoreAudio : how to retrieve actual sampling rate of raw data ?

  13 août 2014, par jyavenard

  When attempting to play AAC-HE content in an mp4 container, the reported sampling rate found in the mp4 container appears to be half of the actual sampling rate.

  E.g it appears as 24kHz instead of 48kHz.

  Using the FFmpeg AAC decoder, retrieving the actual sampling rate can be done by simply decoding an audio packet using

  avcodec_decode_audio4

  And looking at AVCodecContext::sample_rate which will be updated appropriately. From that it’s easy to adapt the output.

  With CoreAudio decoder, I would use a AudioConverterRef set the input and output AudioStreamBasicDescription
  and call AudioConverterFillComplexBuffer

  As the converter performs all the required internal conversion including resampling it’s fine. But it plays the content after resampling it to 24kHz (as that’s what the input AudioStreamBasicDescription contains.

  Would there be a way to retrieve the actual sampling rate as found be the decoder (rather than the demuxer) in a similar fashion as one can with FFmpeg ?

  Would prefer to avoid losing audio quality if at all possible, and not downmix data

  Thanks