[FFmpeg-devel] [PATCH] examples/filtering_audio: get rid of AVABufferSinkParams

[Michael Niedermayer]

On Wed, Apr 17, 2013 at 11:17:57AM +0200, Nicolas George wrote:
> L'octidi 28 germinal, an CCXXI, pkoshevoy at gmail.com a écrit :
> > From: Pavel Koshevoy <pkoshevoy at gmail.com>
> > 
> > AVABufferSinkParams are ignored by avfilter_graph_create_filter,
> > therefore the example is misleading.  Use av_opt_set_int_list to
> > configure abuffersink directly.
> 
> Thanks for the patch. Comments follow below.
> 
> > 
> > Also, make the example a bit more interesting.
> 
> I believe the two changes belong in different patches.
> 
> > 
> > Signed-off-by: Pavel Koshevoy <pkoshevoy at gmail.com>
> > ---
> >  doc/examples/filtering_audio.c |   45 +++++++++++++++++++++++++++++++++-------
> >  1 file changed, 37 insertions(+), 8 deletions(-)
> > 
> > diff --git a/doc/examples/filtering_audio.c b/doc/examples/filtering_audio.c
> > index 67588aa..c891dfe 100644
> > --- a/doc/examples/filtering_audio.c
> > +++ b/doc/examples/filtering_audio.c
> > @@ -36,9 +36,19 @@
> >  #include <libavfilter/avcodec.h>
> >  #include <libavfilter/buffersink.h>
> >  #include <libavfilter/buffersrc.h>
> > +#include <libavutil/opt.h>
> >  
> > -const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
> > -const char *player       = "ffplay -f s16le -ar 8000 -ac 1 -";
> > +/*
> > + * Example of pitch-shifting effect:
> > + *
> > + * 1. use atempo filter at 48KHz and increase playback tempo
> > + * by a factor of 2 thus reducing number of samples per second in half.
> > + *
> 
> > + * 2. use ffplay to ingest raw audio at 24KHz thus increasing playback
> > + * duration by a factor of 2 and resulting in playback at a lower pitch.
> > +*/
> > +const char *filter_descr = "aresample=48000, atempo=2.0";
> > +const char *player       = "ffplay -f s16le -ar 24000 -ac 2 -";
> 
> I believe this kind of mismatch is most likely to confuse users, and
> therefore should be avoided in an example.
> 
> >  
> >  static AVFormatContext *fmt_ctx;
> >  static AVCodecContext *dec_ctx;
> > @@ -88,8 +98,9 @@ static int init_filters(const char *filters_descr)
> >      AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
> >      AVFilterInOut *outputs = avfilter_inout_alloc();
> >      AVFilterInOut *inputs  = avfilter_inout_alloc();
> > -    const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
> > -    AVABufferSinkParams *abuffersink_params;
> > +    const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
> > +    const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_STEREO, -1 };
> > +    const int out_sample_rates[] = { 48000, -1 };
> >      const AVFilterLink *outlink;
> >      AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
> >  
> > @@ -110,16 +121,34 @@ static int init_filters(const char *filters_descr)
> >      }
> >  
> >      /* buffer audio sink: to terminate the filter chain. */
> > -    abuffersink_params = av_abuffersink_params_alloc();
> > -    abuffersink_params->sample_fmts     = sample_fmts;
> >      ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
> > -                                       NULL, abuffersink_params, filter_graph);
> > -    av_free(abuffersink_params);
> > +                                       NULL, NULL, filter_graph);
> >      if (ret < 0) {
> >          av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
> >          return ret;
> >      }
> >  
> 
> > +    ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
> > +                              AV_OPT_SEARCH_CHILDREN);
> > +    if (ret < 0) {
> > +        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
> > +        return ret;
> > +    }
> > +
> > +    ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
> > +                              AV_OPT_SEARCH_CHILDREN);
> > +    if (ret < 0) {
> > +        av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
> > +        return ret;
> > +    }
> > +
> > +    ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
> > +                              AV_OPT_SEARCH_CHILDREN);
> > +    if (ret < 0) {
> > +        av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
> > +        return ret;
> > +    }
> > +
> 
> IMHO, since the lists are available as local arrays, av_opt_set_bin and
> sizeof are more elegant. Possibly, an example should show both
> possibilities.

oops, i applied before seeing the other replies ...
but seems ive fixed most issues raised here by not pushing the atempo
change

Whats left would be the "how to set the options", IMHO ubituxs
suggestion of using a format filter sounds like a good idea ...

[...]

-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Complexity theory is the science of finding the exact solution to an
approximation. Benchmarking OTOH is finding an approximation of the exact
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