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• How to concatenate variables and strings as a full path for the "output file" of an ffmpeg command in a bash script

  12 mai 2022, par djspatule

  I'm trying to learn to bash scripting and I tried to use variables and arguments, etc. to specify a complex output file name to a ffmpeg command as follows :

  


  for file in "$1"/*; do
    #get the name of each file in the directory (without the path nor the extension)
    filename=$(basename $file .mp3)
    #use mimic for Text To Speech. Difficult to install but good and natural voices.
    ~/Desktop/mimic1/mimic -t "$filename" -o $1/wavefile.wav
    #converts the wav file outputed by mimic into mp3
    ffmpeg -i $1/wavefile.wav -f mp3 "${1}/${filename} (title).mp3"
done



  


  But the "${1}/${filename} (title).mp3" part in particular really doesn't seem to work...

  


  Indeed, if I run script.sh ./, I get a file called (title).mp3

  


  Can you help me figure out what it is I'm doing wrong ?
Thanks a million in advance.
Best,

  


  P.S. : i also get earlier in the terminal's output basename: extra operand ‘.mp3’...like my whole code is wrong....?

  


• How to modify pitch in audio of video file with singing voices

  9 janvier 2020, par Keith Bennett

  I have some karaoke .mp4 video files (legally obtained) for Thai songs, and want to convert the pitch downward to fit my singing range. I’ve gotten most of the way there thanks to https://superuser.com/questions/292833/how-to-change-audio-frequency/1076762#1076762
  using a command line like this :

  ffmpeg -i in.mp4 -af 'asetrate=35280.0,atempo=1.25' out.mp4

  ...but the human singing voices don’t sound natural at the modified pitch.

  Is there a better way to change the pitch ? I know some commercial products can do this.

  By the way, I wrote a Ruby script to simplify this ffmpeg call ; it’s at https://gist.github.com/keithrbennett/9ba7043792bfb2fcc92d615076a8413f. It enables you to specify a single factor, and modifies both pitch and tempo accordingly.

• Is FFmpegAudioDecoder supposed to reinitialize upon append of new init segment

  14 novembre 2023, par martin

  I am attempting to switch audio tracks but when switching the FFmpegAudioDecoder never reinitializes like it does with video tracks of differing resolutions. I am not certain if this is the intended behavior of FFmpegAudioDecoder and would love to learn more about the expected behavior.

  


  When switching audio tracks I end up calling the following operations :

  


  if sourceBuffer.getIsUpdate() {sourceBuffer.abort()}
sourceBuffer.remove(0-videoDuration)
initSegmentDataStream = fetch init segment of new audio representation
sourceBuffer.appendBuffer(initSegmentDataStream)


  


  These are the Media tab messages from initial video load

  


  ChunkDemuxer
Selected FFmpegAudioDecoder for audio decoding, config: codec: aac, profile: unknown, bytes_per_channel: 2, channel_layout: STEREO, channels: 2, samples_per_second: 48000, sample_format: Signed 16-bit, bytes_per_frame: 4, seek_preroll: 0us, codec_delay: 0, has extra data: false, encryption scheme: Unencrypted, discard decoder delay: false, target_output_channel_layout: STEREO, target_output_sample_format: Unknown sample format, has aac extra data: true
Cannot select DecryptingVideoDecoder for video decoding
Cannot select VDAVideoDecoder for video decoding
Cannot select VpxVideoDecoder for video decoding
Selected Dav1dVideoDecoder for video decoding, config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1280,720], visible rect: [0,0,1280,720], natural size: [1280,720], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}
Dropping audio frame (DTS 0us PTS -105375us,-62709us) that is outside append window [0us,9223372036854775807us).
Dropping audio frame (DTS 42666us PTS -62708us,-20042us) that is outside append window [0us,9223372036854775807us).
Truncating audio buffer which overlaps append window start. PTS -20041us frame_end_timestamp 22625us append_window_start 0us
Effective playback rate changed from 0 to 1


  


  For comparison this is what I get when appending the init segment of a different video resolution / track

  


  video decoder config changed midstream, new config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1920,1080], visible rect: [0,0,1920,1080], natural size: [1920,1080], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}



  


  Chrome version : Version 119.0.6045.123 (Official Build)

  


  When appending the new init segment of an audio track I was expecting the FFmpegAudioDecoder to be reinitialized like the Dav1dVideoDecoder does for video tracks