[FFmpeg-cvslog] r14692 - in trunk: libavcodec/pcm.c tests/regression.sh

[Baptiste Coudurier]

Daniel Serpell wrote:
> Hi!
> 
> On Wed, Aug 13, 2008 at 3:43 AM,  <pross at xvid.org> wrote:
>> On Tue, Aug 12, 2008 at 08:45:23PM -0400, Daniel Serpell wrote:
>>> On Tue, Aug 12, 2008 at 8:40 PM, Daniel Serpell
>>> <daniel.serpell at gmail.com> wrote:
>>>> Hi!
>>>>
>>>> On Mon, Aug 11, 2008 at 5:52 AM, pross <subversion at mplayerhq.hu> wrote:
>>>>> Author: pross
>>>>> Date: Mon Aug 11 11:52:17 2008
>>>>> New Revision: 14692
>>>>>
>>>>> Log:
>>>>> Apply PCM ENCODE/DECODE() macros to the S/U,8/24/32,LE/BE PCM codecs.
>>>>>
>>>>>
>>>>> Modified:
>>>>>   trunk/libavcodec/pcm.c
>>>>>   trunk/tests/regression.sh
>>>>>
>>>> This commit also broke transcoding from PCM from 8 bit to 16 bit, I
>>>> uploaded a sample to
>>>>   ftp://upload.mplayerhq.hu:/MPlayer/incoming/pcm-audio-11024
>>>>
>>>> The bug can be heard in output.avi from the command line:
>>> Sorry, the correct command line is:
>>>
>>> ffmpeg -y -i pcm-audio-bug-r14692.avi -acodec pcm_s16le -ar 48000
>>> -vcodec copy output.avi
>>>
>>> The bug is not present with only 8-16 bit conversion, you need to resample audio
>>> also.
>> The resampler only supports SAMPLE_FMT_S16, and is performed by conversion
>> to the target format. Hence why transcoding U8->S16 fails.
>>
>> I guess the next step is to make resample.c handle different foramts.
>>
> 
> I think is better to resample *after* conversion to S16.
> 
> This set of patches fixes my issue, first one exits ffmpeg if the resample is
> called on any sample format different of S16.
> 
> The second patch resamples after sample format conversion, allowing to resample
> from U8 to S16.
> 
> Please, consider applying.
> 
>    Daniel.
> 
> 
> ------------------------------------------------------------------------
> 
> Index: ffmpeg.c
> ===================================================================
> --- ffmpeg.c	(revision 14745)
> +++ ffmpeg.c	(working copy)
> @@ -534,6 +534,11 @@
>          ost->audio_resample = 1;
>  
>      if (ost->audio_resample && !ost->resample) {
> +        if (dec->sample_fmt != SAMPLE_FMT_S16)
> +        {
> +            fprintf(stderr, "Resampler only works with 16 bits per sample\n");
> +            av_exit(1);
> +        }
>          ost->resample = audio_resample_init(enc->channels,    dec->channels,
>                                              enc->sample_rate, dec->sample_rate);
>          if (!ost->resample) {
> 
> 
> ------------------------------------------------------------------------
> 
> --- ffmpeg.ab.c	2008-08-13 21:36:23.000000000 -0400
> +++ ffmpeg.c	2008-08-13 21:36:47.000000000 -0400
> @@ -534,7 +534,7 @@
>          ost->audio_resample = 1;
>  
>      if (ost->audio_resample && !ost->resample) {
> -        if (dec->sample_fmt != SAMPLE_FMT_S16)
> +        if (enc->sample_fmt != SAMPLE_FMT_S16)
>          {
>              fprintf(stderr, "Resampler only works with 16 bits per sample\n");
>              av_exit(1);
> @@ -616,23 +616,12 @@
>          ost->sync_opts= lrintf(get_sync_ipts(ost) * enc->sample_rate)
>                          - av_fifo_size(&ost->fifo)/(ost->st->codec->channels * 2); //FIXME wrong
>  
> -    if (ost->audio_resample) {
> -        buftmp = audio_buf;
> -        size_out = audio_resample(ost->resample,
> -                                  (short *)buftmp, (short *)buf,
> -                                  size / (ist->st->codec->channels * 2));
> -        size_out = size_out * enc->channels * 2;
> -    } else {
> -        buftmp = buf;
> -        size_out = size;
> -    }
> -
>      if (dec->sample_fmt!=enc->sample_fmt) {
> -        const void *ibuf[6]= {buftmp};
> +        const void *ibuf[6]= {buf};
>          void *obuf[6]= {audio_out2};
>          int istride[6]= {av_get_bits_per_sample_format(dec->sample_fmt)/8};
>          int ostride[6]= {av_get_bits_per_sample_format(enc->sample_fmt)/8};
> -        int len= size_out/istride[0];
> +        int len= size/istride[0];
>          if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
>              printf("av_audio_convert() failed\n");
>              return;
> @@ -641,6 +630,17 @@
>          /* FIXME: existing code assume that size_out equals framesize*channels*2
>                    remove this legacy cruft */
>          size_out = len*2;
> +    } else {
> +        buftmp = buf;
> +        size_out = size;
> +    }
> +
> +    if (ost->audio_resample) {
> +        size_out = audio_resample(ost->resample,
> +                                  (short *)audio_buf, (short *)buftmp,
> +                                  size / (ist->st->codec->channels * 2));
> +        size_out = size_out * enc->channels * 2;
> +        buftmp = audio_buf;
>      }
>  
>      /* now encode as many frames as possible */
> 

Ping ? This is related to roundup issue #582.

-- 
Baptiste COUDURIER                              GnuPG Key Id: 0x5C1ABAAA
Smartjog USA Inc.                                http://www.smartjog.com
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