Give this a shot:

    ffmpeg -i input.flv -f s16le -acodec pcm_s16le output.raw

You can get these options by running:

    ffmpeg -formats

See https://trac.ffmpeg.org/wiki/audio%20types for details

convert `mp4` file to `pcm`

    ffmpeg -y  -i input.mp4  -acodec pcm_s16le -f s16le -ac 1 -ar 16000 output.pcm


you can also use it to convert `mp3` to `pcm`

    ffmpeg -y  -i input.mp3  -acodec pcm_s16le -f s16le -ac 1 -ar 16000 output.pcm


key params means:


    -f s16le … PCM signed 16-bit little-endian samples
    
    -ac 1 … 1 channel (mono)
    
    -ar 16000 … sample rate 16000Hz



Suppose the user enter an array, for example:

    Array = {France, Spain, France, France, Italy, Spain, Spain, Italy}

which I did know the length of it

the `index` array would be:

    index = {0, 1, 2, 3, 4, 5, 6, 7}

Now, after sorting it using `Arrays.sort(Array);`

`newArray` will be like:

    newArray = {France, France, France, Italy, Italy, Spain, Spain, Spain}

and the `newIndex` will be:

    newIndex = {0, 2, 3, 4, 7, 1, 5, 6}


The problem is: how can I find the `newIndex` from the input Array?

Thanks in advance

Get the indices of an array after sorting?

First, What is the difference between Perm space and Heap space (What and how does the JVM choose to use each memory space)?

Second, but most importantly, what sort of ratio would be recommended for a standard MVC type java application?

Perm space vs Heap space

I&#39;m currently using `ffmpeg` to convert `FLV/Speex` to `WAV/pcm_s16le`, successfully.  However, I now need the output format to be RAW, that is, PCM signed 16-bit little endian, without the WAV header.  I tried the following:

    ffmpeg -y -i input.flv -vn -acodec pcm_s16le output.raw

But ffmpeg responds with:

    Unable to find a suitable output format for &#39;output.raw&#39;

I also tried using `output.pcm` and `output` as output file names, with the same result.

I also tried the `-f` flag to specify raw format, but that gives:

    Unknown input or output format: raw

Is this possible with FFmpeg?  If so, how?

Can ffmpeg convert audio to raw PCM? If so, how?

I'm currently using <code>ffmpeg</code> to convert <code>FLV/Speex</code> to <code>WAV/pcm_s16le</code>, successfully. However, I now need the output format to be RAW, that is, PCM signed 16-bit little endian, without the WAV header. I tried the following:
<pre><code class="hljs language-css">ffmpeg -y -i input.flv -vn -acodec pcm_s16le output.raw
</code></pre>
But ffmpeg responds with:
<pre><code class="hljs language-lua">Unable to find a suitable output format for 'output.raw'
</code></pre>
I also tried using <code>output.pcm</code> and <code>output</code> as output file names, with the same result.
I also tried the <code>-f</code> flag to specify raw format, but that gives:
<pre><code class="hljs language-lua">Unknown input or output format: raw
</code></pre>
Is this possible with FFmpeg? If so, how?

Here is the command I am using to combine multiple videos:

&lt;pre&gt;ffmpeg -i 75_540_38HQ2.mp4 -i 76_70_20.mp4 -i 76_173_80.mp4 -i 81_186_35.mp4 -vcodec copy -acodec copy Mux1.mp4&lt;/pre&gt;

The resulting &lt;code&gt;Mux1.mp4&lt;/code&gt; does not contain all videos. Only the first video (&lt;code&gt;75_540_38HQ2.mp4&lt;/code&gt;). The file size of the source and resulting video is below (as you can see, resulting video is slightly larger than first vid):

&lt;pre&gt;
$ ls -lh
-rw-r--r-- 1 dbaker dbaker  42M 2011-03-24 11:59 75_540_38HQ2.mp4
-rw-r--r-- 1 dbaker dbaker 236M 2011-03-24 12:09 76_173_80.mp4
-rw-r--r-- 1 dbaker dbaker  26M 2011-03-24 12:05 76_70_20.mp4
-rw-r--r-- 1 dbaker dbaker  54M 2011-03-24 12:15 81_186_35.mp4
-rw-r--r-- 1 dbaker dbaker  44M 2011-03-24 14:48 Mux1.mp4
&lt;/pre&gt;

Here is the output of the &lt;code&gt;ffmpeg&lt;/code&gt; command. To me it looks ok, showing the multiple source inputs and the single output.

&lt;pre&gt;
FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers
  built on Mar 21 2011 18:05:32 with gcc 4.4.5
  configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab
  libavutil     50.36. 0 / 50.36. 0
  libavcore      0.16. 1 /  0.16. 1
  libavcodec    52.108. 0 / 52.108. 0
  libavformat   52.93. 0 / 52.93. 0
  libavdevice   52. 2. 3 / 52. 2. 3
  libavfilter    1.74. 0 /  1.74. 0
  libswscale     0.12. 0 /  0.12. 0
  libpostproc   51. 2. 0 / 51. 2. 0
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#39;75_540_38HQ2.mp4&#39;:
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 1970-01-01 00:00:00
    encoder         : Lavf52.93.0
  Duration: 00:00:29.99, start: 0.000000, bitrate: 11517 kb/s
    Stream #0.0(eng): Video: h264, yuv420p, 1280x960 [PAR 1:1 DAR 4:3], 11575 kb/s, 29.94 fps, 29.97 tbr, 30k tbn, 59.94 tbc
    Metadata:
      creation_time   : 1970-01-01 00:00:00
    Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s
    Metadata:
      creation_time   : 1970-01-01 00:00:00
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from &#39;76_70_20.mp4&#39;:
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 1970-01-01 00:00:00
    encoder         : Lavf52.93.0
  Duration: 00:00:19.98, start: 0.000000, bitrate: 10901 kb/s
    Stream #1.0(eng): Video: h264, yuv420p, 1280x960 [PAR 1:1 DAR 4:3], 10804 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc
    Metadata:
      creation_time   : 1970-01-01 00:00:00
    Stream #1.1(eng): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s
    Metadata:
      creation_time   : 1970-01-01 00:00:00
Input #2, mov,mp4,m4a,3gp,3g2,mj2, from &#39;76_173_80.mp4&#39;:
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 1970-01-01 00:00:00
    encoder         : Lavf52.93.0
  Duration: 00:03:09.99, start: 0.000000, bitrate: 10393 kb/s
    Stream #2.0(eng): Video: h264, yuv420p, 1280x960 [PAR 1:1 DAR 4:3], 10321 kb/s, 29.96 fps, 29.97 tbr, 30k tbn, 59.94 tbc
    Metadata:
      creation_time   : 1970-01-01 00:00:00
    Stream #2.1(eng): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s
    Metadata:
      creation_time   : 1970-01-01 00:00:00

Seems stream 0 codec frame rate differs from container frame rate: 119.88 (120000/1001) -&gt; 30000.00 (30000/1)
Input #3, mov,mp4,m4a,3gp,3g2,mj2, from &#39;81_186_35.mp4&#39;:
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 1970-01-01 00:00:00
    encoder         : Lavf52.93.0
  Duration: 00:00:35.00, start: 0.000000, bitrate: 12700 kb/s
    Stream #3.0(eng): Video: h264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 12620 kb/s, 59.91 fps, 30k tbr, 60k tbn, 119.88 tbc
    Metadata:
      creation_time   : 1970-01-01 00:00:00
    Stream #3.1(eng): Audio: aac, 48000 Hz, stereo, s16, 128 kb/s
    Metadata:
      creation_time   : 1970-01-01 00:00:00
Output #0, mp4, to &#39;Mux1.mp4&#39;:
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    creation_time   : 1970-01-01 00:00:00
    encoder         : Lavf52.93.0
    Stream #0.0(eng): Video: libx264, yuv420p, 1280x960 [PAR 1:1 DAR 4:3], q=2-31, 11575 kb/s, 30k tbn, 29.97 tbc
    Metadata:
      creation_time   : 1970-01-01 00:00:00
    Stream #0.1(eng): Audio: libfaac, 48000 Hz, stereo, 128 kb/s
    Metadata:
      creation_time   : 1970-01-01 00:00:00
Stream mapping:
  Stream #0.0 -&gt; #0.0
  Stream #2.1 -&gt; #0.1
Press [q] to stop encoding
frame=  883 fps=632 q=-1.0 Lsize=   44730kB time=29.40 bitrate=12465.1kbits/s    
video:41678kB audio:2969kB global headers:0kB muxing overhead 0.184548%
&lt;/pre&gt;

Am I doing something blindingly stupid here? 

The source videos came from a video camera, and are small snippets taken with &lt;code&gt;ffmpeg -i bigfile.mp4 -ss 20 -t 10 -vcodec copy etc..&lt;/code&gt;

Thanks heaps!!
Dave


&lt;hr&gt;

Edit: couldn&#39;t solve it so I just use avidemux GUI tool. It seemed to append the MP4&#39;s just fine. 

Must be a problem with MP4&#39;s or just the ones that come off a gopro camera.


ffmpeg: Combine/merge multiple mp4 videos not working, output only contains the first video

I wish to split a large avi video into two smaller consecutive videos. I am using ffmpeg.

One way is to run ffmpeg two times:

    ffmpeg -i input.avi -vcodec copy -acodec copy -ss 00:00:00 -t 00:30:00 output1.avi
    ffmpeg -i input.avi -vcodec copy -acodec copy -ss 00:30:00 -t 00:30:00 output2.avi

But according to manpage of ffmpeg, I can make more than one ouput file from one input file using just **one** line:

    ffmpeg -i input.avi -vcodec copy -acodec copy -ss 00:00:00 -t 00:30:00 output1.avi \
       -vcodec copy -acodec copy -ss 00:30:00 -t 00:30:00 output2.avi

My question is, does the later approach save computation time and memory?


FFmpeg: How to split video efficiently?

I have a *.mp4 video file(MPEG4 video codec) and I am trying to convert this to a H264 video codec format(raw h.264 format) using ffmpeg on Linux(Version - FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1, Copyright (c) 2000-2009 Fabrice Bellard,) using command line as shown below, 

    ffmpeg -i input .mp4 output.h264 

but I get an error saying - 

    Unsupported codec for output stream #0.0

Then when i try this option:

    ffmpeg -i input .mp4 -formats h264 output.h264 

it still does not work, and gives -

    Seems stream 0 codec frame rate differs from container frame rate: 59.94 (5994/100) -&gt; 29.97 (30000/1001)
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from &#39;Rapture.mp4&#39;:
      Duration: 00:02:06.44, start: 0.000000, bitrate: 26574 kb/s
        Stream #0.0(eng): Video: h264, yuv420p, 1920x1080, 29.97 tbr, 29.97 tbn, 59.94 tbc
        Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16

And then it prints out help on the formats which we get when we do `ffmpeg -formats`

When I checked the help, `ffmpeg -formats`, I see below information related to H264 file format and codec: 

    File format : 
    
    DE h264            raw H.264 video format
    
    Codecs:
    
    D V D  h264         H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10

My questions :

1. How can I convert the video to a H264 encoded video (raw H264 video format)

2. When I do `ffmpeg -formats`, I see many acronyms for the codecs supported, I see many acronyms before the codec name/type such as - D V D S E A, what do they stand for?

3. How to use the ffmpeg options `-vcodec` and `-formats`?


ffmpeg usage to encode a video to H264 codec format

How do you convert an entire directory/folder with ffmpeg via command line or with a batch script?

How do you convert an entire directory with ffmpeg?

 - In simple terms what are pts and dts values?
 - Why are they important while transcoding [decode-encode] videos ?

What does this code bit do in [**ffmpeg.c**][1] , what is its purpose?

    01562    ist-&gt;next_pts = ist-&gt;pts = picture.best_effort_timestamp;
    01563    if (ist-&gt;st-&gt;codec-&gt;time_base.num != 0) {
    01564        int ticks= ist-&gt;st-&gt;parser ? ist-&gt;st-&gt;parser-&gt;repeat_pict+1 : ist-&gt;st-&gt;codec-&gt;ticks_per_frame;
    01565        ist-&gt;next_pts += ((int64_t)AV_TIME_BASE *
    01566                         ist-&gt;st-&gt;codec-&gt;time_base.num * ticks) /
    01567                         ist-&gt;st-&gt;codec-&gt;time_base.den;
    01568    }


  [1]: http://cekirdek.pardus.org.tr/~ismail/ffmpeg-docs/ffmpeg_8c-source.html

ffmpeg.c what are pts and dts ? what does this code block do in ffmpeg.c?

This question seems to be asked few times over last few years but none has answer for that. I am trying to process PCM data from HLS and I have to use AVPlayer. 

this post taps the local files
https://chritto.wordpress.com/2013/01/07/processing-avplayers-audio-with-mtaudioprocessingtap/

and this tap work with remote files but not with .m3u8 hls files. 
http://venodesigns.net/2014/01/08/recording-live-audio-streams-on-ios/

I can play first two tracks in the playlist but it doesn&#39;t start the needed callbacks to get the pcm, when the file is local or remote(not stream) I can still get the pcm but it is the hls is not working and I need HLS working

here is my code

    //avplayer tap try
    - (void)viewDidLoad {
        [super viewDidLoad];
        
        NSURL*testUrl= [NSURL URLWithString:@&quot;http://playlists.ihrhls.com/c5/1469/playlist.m3u8&quot;];
        
        AVPlayerItem *item = [AVPlayerItem playerItemWithURL:testUrl];
        self.player = [AVPlayer playerWithPlayerItem:item];
        
        // Watch the status property - when this is good to go, we can access the
        // underlying AVAssetTrack we need.
        [item addObserver:self forKeyPath:@&quot;status&quot; options:0 context:nil];
        
    }
    
    -(void)observeValueForKeyPath:(NSString *)keyPath
    ofObject:(id)object
    change:(NSDictionary *)change
    context:(void *)context
    {
        if(![keyPath isEqualToString:@&quot;status&quot;])
            return;
        
        AVPlayerItem *item = (AVPlayerItem *)object;
        if(item.status != AVPlayerItemStatusReadyToPlay)
            return;
        
        NSArray *tracks = [self.player.currentItem tracks];
        for(AVPlayerItemTrack *track in tracks) {
            if([track.assetTrack.mediaType isEqualToString:AVMediaTypeAudio]) {
                NSLog(@&quot;GOT DAT FUCKER&quot;);
                [self beginRecordingAudioFromTrack:track.assetTrack];
                [self.player play];
            }
        }
    }
    
    - (void)beginRecordingAudioFromTrack:(AVAssetTrack *)audioTrack
    {
        // Configure an MTAudioProcessingTap to handle things.
        MTAudioProcessingTapRef tap;
        MTAudioProcessingTapCallbacks callbacks;
        callbacks.version = kMTAudioProcessingTapCallbacksVersion_0;
        callbacks.clientInfo = (__bridge void *)(self);
        callbacks.init = init;
        callbacks.prepare = prepare;
        callbacks.process = process;
        callbacks.unprepare = unprepare;
        callbacks.finalize = finalize;
        
        OSStatus err = MTAudioProcessingTapCreate(
                                                  kCFAllocatorDefault,
                                                  &amp;callbacks,
                                                  kMTAudioProcessingTapCreationFlag_PostEffects,
                                                  &amp;tap
                                                  );
        
        if(err) {
            NSLog(@&quot;Unable to create the Audio Processing Tap %d&quot;, (int)err);
            return;
        }
        
        // Create an AudioMix and assign it to our currently playing &quot;item&quot;, which
        // is just the stream itself.
        AVMutableAudioMix *audioMix = [AVMutableAudioMix audioMix];
        AVMutableAudioMixInputParameters *inputParams = [AVMutableAudioMixInputParameters
                                                         audioMixInputParametersWithTrack:audioTrack];
        
        inputParams.audioTapProcessor = tap;
        audioMix.inputParameters = @[inputParams];
        self.player.currentItem.audioMix = audioMix;
    }
    
    void process(MTAudioProcessingTapRef tap, CMItemCount numberFrames,
                 MTAudioProcessingTapFlags flags, AudioBufferList *bufferListInOut,
                 CMItemCount *numberFramesOut, MTAudioProcessingTapFlags *flagsOut)
    {
        OSStatus err = MTAudioProcessingTapGetSourceAudio(tap, numberFrames, bufferListInOut,
                                                          flagsOut, NULL, numberFramesOut);
        if (err) NSLog(@&quot;Error from GetSourceAudio: %d&quot;, (int)err);
        
        NSLog(@&quot;Process&quot;);
    
    }
    
    void init(MTAudioProcessingTapRef tap, void *clientInfo, void **tapStorageOut)
    {
        NSLog(@&quot;Initialising the Audio Tap Processor&quot;);
        *tapStorageOut = clientInfo;
    }
    
    void finalize(MTAudioProcessingTapRef tap)
    {
        NSLog(@&quot;Finalizing the Audio Tap Processor&quot;);
    }
    
    void prepare(MTAudioProcessingTapRef tap, CMItemCount maxFrames, const AudioStreamBasicDescription *processingFormat)
    {
        NSLog(@&quot;Preparing the Audio Tap Processor&quot;);
    }
    
    void unprepare(MTAudioProcessingTapRef tap)
    {
        NSLog(@&quot;Unpreparing the Audio Tap Processor&quot;);
    }

`void init` is called `void prepare` and `process` has to be called as well. 

how can I do this? 



 


Content Type	Original Author	Original Content on Stackoverflow
Question	David van Geest	View Question on Stackoverflow
Solution 1 - Ffmpeg	Chris Haas	View Answer on Stackoverflow
Solution 2 - Ffmpeg	Jayhello	View Answer on Stackoverflow

Can ffmpeg convert audio to raw PCM? If so, how?

Ffmpeg Problem Overview

Ffmpeg Solutions

Solution 1 - Ffmpeg

Solution 2 - Ffmpeg

Perm space vs Heap space

Get the indices of an array after sorting?

Attributions