Allan Hsu | 1 Feb 2007 01:28
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Re: [BUG] slowness and/or memory leak with adpcm_ima_qt .mov files


On Jan 31, 2007, at 3:49 PM, Michael Niedermayer wrote:

[...]
>> I've run across what looks like a major memory leak caused by
>> adpcm_ima_qt audio track decoding. ffmpeg output doesn't look out of
>> the ordinary, but it is much slower than it should be (kind of slow
>> on a PPC OS X machine, but *super slow* on a x86_64 linux machine).
>> I've uploaded a sample to the FTP site as "shuffle-ima41.mov". Here
>> is the ffmpeg output, killed because it was taking up too much  
>> memory:
>>
>> ./ffmpeg -i ~/Desktop/shuffle-ima41.mov -acodec pcm_s16le -vcodec
>> mpeg4 -b 768k ~/Desktop/shuffle.avi
>
> does that also happen if you dont encode the video? -vn

Interesting results. The following command line does not seem to leak  
memory, but it doesn't ever seem to stop transcoding either:

./ffmpeg -i ~/Desktop/shuffle-ima41.mov -vn -b 768k -acodec pcm_s16le  
~/Desktop/shuffle.avi

I stopped ffmpeg by hitting 'q' after it had written over 1.5GB of  
data to shuffle.avi (the source movie is only 30 seconds long). I  
tried to play back this file in VLC and the audio doesn't seem to  
have survived. All I can hear is a periodic quiet clicking noise.

> and what if you just copy the audio -acodec copy?

(Continue reading)

Michael Niedermayer | 1 Feb 2007 01:31
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Re: [BUG] decoding pcm 24 bit

Hi

On Wed, Jan 31, 2007 at 11:00:01AM +0100, Baptiste Coudurier wrote:
> Hi
> 
> Here is a bug when decoding pcm 24 bit.

fixed

[...]
--

-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Everything should be made as simple as possible, but not simpler.
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Michael Niedermayer | 1 Feb 2007 01:39
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Re: [PATCH] Fix to loss of sync in ffplay when paused

Hi

On Wed, Jan 31, 2007 at 10:39:08AM +0000, Michael McConnell wrote:
> When a video in ffplay is paused, upon release it rushes ahead leaving 
> the audio and video way out of sync.
> 
> This patch corrects that by updating the frame timer when pause is released.

patch ok if it works with all the -sync variants

[...]
--

-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Why not whip the teacher when the pupil misbehaves? -- Diogenes of Sinope
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Michael Niedermayer | 1 Feb 2007 01:46
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Re: [BUG] slowness and/or memory leak with adpcm_ima_qt .mov files

Hi

On Wed, Jan 31, 2007 at 04:28:12PM -0800, Allan Hsu wrote:
> 
> On Jan 31, 2007, at 3:49 PM, Michael Niedermayer wrote:
> 
> [...]
> >>I've run across what looks like a major memory leak caused by
> >>adpcm_ima_qt audio track decoding. ffmpeg output doesn't look out of
> >>the ordinary, but it is much slower than it should be (kind of slow
> >>on a PPC OS X machine, but *super slow* on a x86_64 linux machine).
> >>I've uploaded a sample to the FTP site as "shuffle-ima41.mov". Here
> >>is the ffmpeg output, killed because it was taking up too much  
> >>memory:
> >>
> >>./ffmpeg -i ~/Desktop/shuffle-ima41.mov -acodec pcm_s16le -vcodec
> >>mpeg4 -b 768k ~/Desktop/shuffle.avi
> >
> >does that also happen if you dont encode the video? -vn
> 
> Interesting results. The following command line does not seem to leak  
> memory, but it doesn't ever seem to stop transcoding either:
> 
> ./ffmpeg -i ~/Desktop/shuffle-ima41.mov -vn -b 768k -acodec pcm_s16le  
> ~/Desktop/shuffle.avi
> 
> I stopped ffmpeg by hitting 'q' after it had written over 1.5GB of  
> data to shuffle.avi (the source movie is only 30 seconds long). I  
> tried to play back this file in VLC and the audio doesn't seem to  
> have survived. All I can hear is a periodic quiet clicking noise.
(Continue reading)

Diego Biurrun | 1 Feb 2007 01:53
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Re: [PATCH] configure - Simplify disabled gpl block

On Wed, Jan 31, 2007 at 05:45:07PM -0200, ramiro <at> lisha.ufsc.br wrote:
> $subj

You beat me to it...

Thanks, applied.

Diego
Roman Shaposhnik | 1 Feb 2007 05:10
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Interleaving audio and video

While implementing an idea suggested by Michael for restructuring DV 
audio muxer I've come to realize that DV muxer currently expects a 
caller to take care of interleaving audio and video properly based
on PTS valued. Now we know that ffmpeg.c does it, but is it fair
to require that kind of behaviour in general of all users of 
libavformat ?

Or to put it differently, do I have to care about caller giving 
me the following pattern of DV audio/video packets:

   video1, video2, audio1, audio2

or even worse:

   video1, audio2, video2, audio1

Thanks,
Roman.
Zuxy Meng | 1 Feb 2007 05:55
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Re: Bug when decoding WMV HD file. The aspect ratio is not preserved

Hi,

2007/2/1, Michael Niedermayer <michaelni <at> gmx.at>:
> Hi
>
> On Sat, Jan 27, 2007 at 07:44:59AM +0200, Kostya wrote:
> > On Sat, Jan 27, 2007 at 02:01:00AM +0100, Michael Niedermayer wrote:
> > > Hi
> > >
> > > On Wed, Jan 24, 2007 at 07:45:24PM +0100, Thomas Parmegiani wrote:
> > > > Hi all!
> > > >
> > > > I downloaded some WMV HD files on :
> > > >
> > > > http://www.microsoft.com/windows/windowsmedia/musicandvideo/hdvideo/contents
> > > > howcase.aspx
> > > >
> > > > I wanted to convert it to MPEG2 without audio (Audio is WMAPro which is not
> > > > currently supported by ffmpeg).
> > > > For 720p files, converted MPEG2 files preserve aspect ratio (16/9).
> > > > But for 1080p files (which have a resolution of 1440 x 1080), converted
> > > > aspect ratio is 4/3.
> > > >
> > > > The same problem appears if I try to use VLC to play WMV file
> > > >
> > > > Problem seen with files :
> > > >
> > > > Amazing caves :
> > > > http://download.microsoft.com/download/0/9/d/09d051c4-decc-4d39-9c57-f520187
> > > > 213a1/Amazing_Caves_1080.exe
(Continue reading)

Eric Dantan Rzewnicki | 1 Feb 2007 06:40

Re: FFmpeg Theora encoding patch

On Sat, Jan 06, 2007 at 03:51:28PM +0000, Paul Richards wrote:
> Hi,
> ===================================================================
> +static int encode_init(AVCodecContext* avc_context)
> +{
> +    theora_info t_info;
> +    theora_comment t_comment;
> +    ogg_packet o_packet;
> +    unsigned int offset;
> +    TheoraContext *h = avc_context->priv_data;
> +
> +    /** Check pixel format contraints */
> +    if (check_constraints(avc_context) != 0) {
> +        return -1;
> +    }
> +    
> +    /** Set up the theora_info struct */
> +    theora_info_init( &t_info );
> +    t_info.width = avc_context->width;
> +    t_info.height = avc_context->height;
> +    t_info.frame_width = avc_context->width;
> +    t_info.frame_height = avc_context->height;
> +    t_info.offset_x = 0;
> +    t_info.offset_y = 0;
> +    t_info.fps_numerator = avc_context->time_base.den;
> +    t_info.fps_denominator = avc_context->time_base.num;

This is a naive and probably dumb question ... but why do you flip the
numerator and denominator here?

(Continue reading)

Scott Harper | 1 Feb 2007 08:24
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Trouble Decoding Audio

Hello,

I hope this is the proper mailing list for this sort of thing.

I'm using the ffplay source to try to help me with decoding and  
playing back audio using the libavcodec and its associated libraries.

I'm using C++, and I have a class which maintains the ffmpeg  
interface-type code (AVCodecContext, etc...) and runs a thread to  
grab the next frame whilst the current one draws.  The video fetching  
code works fine, but I'll post it anyway for completeness' sake.   
Here's the threaded function:

int VE_Media::thread_func() {
	mMutex = SDL_CreateMutex();
	
	while( mThreadLives ) {
		if( needNextFrame ) {
			int frameFinished;
			SDL_mutexP(mMutex);	//lock
			av_read_frame(pFormatCtx, &packet);// >= 0) {
				if(packet.stream_index == videoStream ) {
					// Decode video frame
					avcodec_decode_video(pCodecCtx, pFrameYUV, &frameFinished,
											packet.data, packet.size);
					
					if(frameFinished) {
						sws_scale( pSwsCtx, pFrameYUV->data, pFrameYUV->linesize,
									0, pCodecCtx->height, pFrameRGB->data, pFrameRGB->linesize );
					}
(Continue reading)

Michael Niedermayer | 1 Feb 2007 09:42
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Re: Interleaving audio and video

Hi

On Wed, Jan 31, 2007 at 08:10:28PM -0800, Roman Shaposhnik wrote:
> While implementing an idea suggested by Michael for restructuring DV 
> audio muxer I've come to realize that DV muxer currently expects a 
> caller to take care of interleaving audio and video properly based
> on PTS valued. Now we know that ffmpeg.c does it, but is it fair
> to require that kind of behaviour in general of all users of 
> libavformat ?
> 
> Or to put it differently, do I have to care about caller giving 
> me the following pattern of DV audio/video packets:

see av_interleaved_write_frame() (the docs above it)
and AVOutputFormat.interleave_packet()

[...]
--

-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

The educated differ from the uneducated as much as the living from the
dead. -- Aristotle 
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