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• FFMPEG - Edit Audio

  9 avril 2015, par Tony

  Looking to edit a blip out of the beginning of a video/audio file... How can I use FFMPEG to only edit the first two seconds of the clip so that there’s no longer that blip in the audio ?

  How do I ensure that all 7 audio tracks are affected by the audio filter and then how do I ensure that all 7 are also present in my output file in the same input specs ?

  Complete output -

  ffmpeg -i ~/Downloads/in.mov  -vcodec copy -acodec pcm_s24le -q:a 0 -af "volume=enable='between(t,0,3)':volume=0"  ~/Downloads/out2.mov
  
  ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/2.6.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc --enable-libxvid --enable-vda
  
    libavutil      54. 20.100 / 54. 20.100
  
    libavcodec     56. 26.100 / 56. 26.100
  
    libavformat    56. 25.101 / 56. 25.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 11.102 /  5. 11.102
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Guessed Channel Layout for  Input Stream #0.1 : mono
  
  Guessed Channel Layout for  Input Stream #0.2 : mono
  
  Guessed Channel Layout for  Input Stream #0.3 : mono
  
  Guessed Channel Layout for  Input Stream #0.4 : mono
  
  Guessed Channel Layout for  Input Stream #0.5 : mono
  
  Guessed Channel Layout for  Input Stream #0.6 : mono
  
  Guessed Channel Layout for  Input Stream #0.7 : stereo
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '~/Downloads/in.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537199360
  
      compatible_brands: qt  
  
      creation_time   : 2014-01-16 18:02:27
  
    Duration: 00:00:06.44, start: 0.000000, bitrate: 119128 kb/s
  
      Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le(bt709), 1920x1080, 109843 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 2997 tbn, 2997 tbc (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
        encoder         : Apple ProRes 422 (HQ)
  
        timecode        : 01:00:00;00
  
      Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:2(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:3(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:4(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:5(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:6(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:7(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, 2 channels, s32 (24 bit), 2304 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:8(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:28
  
        handler_name    : Apple Alias Data Handler
  
        timecode        : 01:00:00;00
  
  File '~/Downloads/out2.mov' already exists. Overwrite ? [y/N] y
  
  Output #0, mov, to '~/Downloads/out2.mov':
  
    Metadata:
  
      major_brand     : qt  
  
      minor_version   : 537199360
  
      compatible_brands: qt  
  
      encoder         : Lavf56.25.101
  
      Stream #0:0(eng): Video: prores (apch / 0x68637061), yuv422p10le, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 109843 kb/s, 29.97 fps, 29.97 tbr, 11988 tbn, 2997 tbc (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
        encoder         : Apple ProRes 422 (HQ)
  
        timecode        : 01:00:00;00
  
      Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, stereo, s32 (24 bit), 2304 kb/s (default)
  
      Metadata:
  
        creation_time   : 2014-01-16 18:02:27
  
        handler_name    : Apple Alias Data Handler
  
        encoder         : Lavc56.26.100 pcm_s24le
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
    Stream #0:7 -> #0:1 (pcm_s24le (native) -> pcm_s24le (native))
  
  Press [q] to stop, [?] for help
  
  frame=  193 fps=109 q=-1.0 Lsize=   88166kB time=00:00:06.43 bitrate=112155.7kbits/s    
  
  video:86349kB audio:1811kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.006946%

• How to detect delay or silence in an audio file ?

  15 juillet 2022, par Lynob

  I want to detect silence or delay in audio for a given duration file and remove it. For example, if someone started speaking and then paused for some duration to think.

  


  There's this question but it only detects the silence at the end and doesn't remove it. My colleague suggested sox but I'm not sure if it's the best tool for the job nor how to use it frankly, moreover, the project died in 2015.

  


• How do terminal pipes in Python differ from those in Rust ?

  5 octobre 2022, par rust_convert

  To work on learning Rust (in a Tauri project) I am converting a Python 2 program that uses ffmpeg to create a custom video format from a GUI. The video portion converts successfully, but I am unable to get the audio to work. With the debugging I have done for the past few days, it looks like I am not able to read in the audio data in Rust correctly from the terminal pipe - what is working to read in the video data is not working for the audio. I have tried reading in the audio data as a string and then converting it to bytes but then the byte array appears empty. I have been researching the 'Pipe'-ing of data from the rust documentation and python documentation and am unsure how the Rust pipe could be empty or incorrect if it's working for the video.

  


  From this python article and this rust stack overflow exchange, it looks like the python stdout pipe is equivalent to the rust stdin pipe ?

  


  The python code snippet for video and audio conversion :

  


  output=open(self.outputFile, 'wb')
devnull = open(os.devnull, 'wb')

vidcommand = [ FFMPEG_BIN,
            '-i', self.inputFile,
            '-f', 'image2pipe',
            '-r', '%d' % (self.outputFrameRate),
            '-vf', scaleCommand,
            '-vcodec', 'rawvideo',
            '-pix_fmt', 'bgr565be',
            '-f', 'rawvideo', '-']
        
vidPipe = '';
if os.name=='nt' :
    startupinfo = sp.STARTUPINFO()
    startupinfo.dwFlags |= sp.STARTF_USESHOWWINDOW
    vidPipe=sp.Popen(vidcommand, stdin = sp.PIPE, stdout = sp.PIPE, stderr = devnull, bufsize=self.inputVidFrameBytes*10, startupinfo=startupinfo)
else:
    vidPipe=sp.Popen(vidcommand, stdin = sp.PIPE, stdout = sp.PIPE, stderr = devnull, bufsize=self.inputVidFrameBytes*10)

vidFrame = vidPipe.stdout.read(self.inputVidFrameBytes)

audioCommand = [ FFMPEG_BIN,
    '-i', self.inputFile,
    '-f', 's16le',
    '-acodec', 'pcm_s16le',
    '-ar', '%d' % (self.outputAudioSampleRate),
    '-ac', '1',
    '-']

audioPipe=''
if (self.audioEnable.get() == 1):
    if os.name=='nt' :
        startupinfo = sp.STARTUPINFO()
        startupinfo.dwFlags |= sp.STARTF_USESHOWWINDOW
        audioPipe = sp.Popen(audioCommand, stdin = sp.PIPE, stdout=sp.PIPE, stderr = devnull, bufsize=self.audioFrameBytes*10, startupinfo=startupinfo)
    else:
        audioPipe = sp.Popen(audioCommand, stdin = sp.PIPE, stdout=sp.PIPE, stderr = devnull, bufsize=self.audioFrameBytes*10)

    audioFrame = audioPipe.stdout.read(self.audioFrameBytes) 

currentFrame=0;

while len(vidFrame)==self.inputVidFrameBytes:
    currentFrame+=1
    if(currentFrame%30==0):
        self.progressBarVar.set(100.0*(currentFrame*1.0)/self.totalFrames)
    if (self.videoBitDepth.get() == 16):
        output.write(vidFrame)
    else:
        b16VidFrame=bytearray(vidFrame)
        b8VidFrame=[]
        for p in range(self.outputVidFrameBytes):
            b8VidFrame.append(((b16VidFrame[(p*2)+0]>>0)&0xE0)|((b16VidFrame[(p*2)+0]<<2)&0x1C)|((b16VidFrame[(p*2)+1]>>3)&0x03))
        output.write(bytearray(b8VidFrame))

    vidFrame = vidPipe.stdout.read(self.inputVidFrameBytes) # Read where vidframe is to match up with audio frame and output?
    if (self.audioEnable.get() == 1):


        if len(audioFrame)==self.audioFrameBytes:
            audioData=bytearray(audioFrame) 

            for j in range(int(round(self.audioFrameBytes/2))):
                sample = ((audioData[(j*2)+1]<<8) | audioData[j*2]) + 0x8000
                sample = (sample>>(16-self.outputAudioSampleBitDepth)) & (0x0000FFFF>>(16-self.outputAudioSampleBitDepth))

                audioData[j*2] = sample & 0xFF
                audioData[(j*2)+1] = sample>>8

            output.write(audioData)
            audioFrame = audioPipe.stdout.read(self.audioFrameBytes)

        else:
            emptySamples=[]
            for samples in range(int(round(self.audioFrameBytes/2))):
                emptySamples.append(0x00)
                emptySamples.append(0x00)
            output.write(bytearray(emptySamples))

self.progressBarVar.set(100.0)

vidPipe.terminate()
vidPipe.stdout.close()
vidPipe.wait()

if (self.audioEnable.get() == 1):
    audioPipe.terminate()
    audioPipe.stdout.close()
    audioPipe.wait()

output.close()


  


  The Rust snippet that should accomplish the same goals :

  


  let output_file = OpenOptions::new()
    .create(true)
    .truncate(true)
    .write(true)
    .open(&output_path)
    .unwrap();
let mut writer = BufWriter::with_capacity(
    options.video_frame_bytes.max(options.audio_frame_bytes),
    output_file,
);
let ffmpeg_path = sidecar_path("ffmpeg");
#[cfg(debug_assertions)]
let timer = Instant::now();

let mut video_cmd = Command::new(&ffmpeg_path);
#[rustfmt::skip]
video_cmd.args([
    "-i", options.path,
    "-f", "image2pipe",
    "-r", options.frame_rate,
    "-vf", options.scale,
    "-vcodec", "rawvideo",
    "-pix_fmt", "bgr565be",
    "-f", "rawvideo",
    "-",
])
.stdin(Stdio::null())
.stdout(Stdio::piped())
.stderr(Stdio::null());

// windows creation flag CREATE_NO_WINDOW: stops the process from creating a CMD window
// https://docs.microsoft.com/en-us/windows/win32/procthread/process-creation-flags
#[cfg(windows)]
video_cmd.creation_flags(0x08000000);

let mut video_child = video_cmd.spawn().unwrap();
let mut video_stdout = video_child.stdout.take().unwrap();
let mut video_frame = vec![0; options.video_frame_bytes];

let mut audio_cmd = Command::new(&ffmpeg_path);
#[rustfmt::skip]
audio_cmd.args([
    "-i", options.path,
    "-f", "s16le",
    "-acodec", "pcm_s16le",
    "-ar", options.sample_rate,
    "-ac", "1",
    "-",
])
.stdin(Stdio::null())
.stdout(Stdio::piped())
.stderr(Stdio::null());

#[cfg(windows)]
audio_cmd.creation_flags(0x08000000);

let mut audio_child = audio_cmd.spawn().unwrap();
let mut audio_stdout = audio_child.stdout.take().unwrap();
let mut audio_frame = vec![0; options.audio_frame_bytes];

while video_stdout.read_exact(&mut video_frame).is_ok() {
    writer.write_all(&video_frame).unwrap();

    if audio_stdout.read_to_end(&mut audio_frame).is_ok() {
        if audio_frame.len() == options.audio_frame_bytes {
            for i in 0..options.audio_frame_bytes / 2 {
                let temp_sample = ((u32::from(audio_frame[(i * 2) + 1]) << 8)
                    | u32::from(audio_frame[i * 2]))
                    + 0x8000;
                let sample = (temp_sample >> (16 - 10)) & (0x0000FFFF >> (16 - 10));

                audio_frame[i * 2] = (sample & 0xFF) as u8;
                audio_frame[(i * 2) + 1] = (sample >> 8) as u8;
            }
        } else {
            audio_frame.fill(0x00);
        }
    }
    writer.write_all(&audio_frame).unwrap();
}


video_child.wait().unwrap();
audio_child.wait().unwrap();

#[cfg(debug_assertions)]
{
    let elapsed = timer.elapsed();
    dbg!(elapsed);
}

writer.flush().unwrap();


  


  I have looked at the hex data of the files using HxD - regardless of how I alter the Rust program, I am unable to get data different from what is previewed in the attached image - so the audio pipe is incorrectly interfaced. I included a screenshot of the hex data from the working python program that converts the video and audio correctly.

  


  HxD Python program hex output :

  


  

  


  HxD Rust program hex output :