[FFmpeg-devel] [PATCH] lavfi/buffersink: implement av_buffersink_get_samples().

[Stefano Sabatini]

On date Sunday 2013-03-10 17:16:36 +0100, Nicolas George encoded:
> Note: the implementation could be more efficient, but at
> the cost of more diff.
> 
> Signed-off-by: Nicolas George <nicolas.george at normalesup.org>
> ---
>  libavfilter/buffersink.c |   72 ++++++++++++++++++++++++++++++++++++++++++++--
>  1 file changed, 70 insertions(+), 2 deletions(-)
> 
> 
> With this patch (and a lot of hacks to make lavc and lavf compile with a
> major bump), I could transcode audio using avconv dynamically linked to
> ffmpeg's libs, without valgrind errors. I believe this is good enough.
> 
> Note: during this patch, I added minor variable renames to the sink_buffer.c
> cleanup. I do not think it matters enough to submit it for approval.

What's the plan then? What's the new API we're supposed to adopt?
av_buffersink_get_frame/samples(), or you plan to add new _flags()
variants supporting our AVFilterBufferRef API API?
  
> diff --git a/libavfilter/buffersink.c b/libavfilter/buffersink.c
> index 9f92051..3560785 100644
> --- a/libavfilter/buffersink.c
> +++ b/libavfilter/buffersink.c
> @@ -23,7 +23,7 @@
>   * buffer sink
>   */
>  
> -#include "libavutil/fifo.h"
> +#include "libavutil/audio_fifo.h"
>  #include "libavutil/avassert.h"
>  #include "libavutil/channel_layout.h"
>  #include "libavutil/common.h"
> @@ -46,6 +46,10 @@ typedef struct {
>      int64_t *channel_layouts;               ///< list of accepted channel layouts, terminated by -1
>      int all_channel_counts;
>      int *sample_rates;                      ///< list of accepted sample rates, terminated by -1
> +
> +    /* only used for compat API */
> +    AVAudioFifo  *audio_fifo;    ///< FIFO for audio samples
> +    int64_t next_pts;            ///< interpolating audio pts
>  } BufferSinkContext;
>  
>  static av_cold void uninit(AVFilterContext *ctx)
> @@ -53,6 +57,9 @@ static av_cold void uninit(AVFilterContext *ctx)
>      BufferSinkContext *sink = ctx->priv;
>      AVFrame *frame;
>  
> +    if (sink->audio_fifo)
> +        av_audio_fifo_free(sink->audio_fifo);
> +
>      if (sink->fifo) {
>          while (av_fifo_size(sink->fifo) >= sizeof(AVFilterBufferRef *)) {
>              av_fifo_generic_read(sink->fifo, &frame, sizeof(frame), NULL);
> @@ -140,9 +147,70 @@ int av_buffersink_get_frame_flags(AVFilterContext *ctx, AVFrame *frame, int flag
>      return 0;
>  }
>  

> +static int read_from_fifo(AVFilterContext *ctx, AVFrame *frame,
> +                          int nb_samples)
> +{
> +    BufferSinkContext *s = ctx->priv;
> +    AVFilterLink   *link = ctx->inputs[0];
> +    AVFrame *tmp;
> +
> +    if (!(tmp = ff_get_audio_buffer(link, nb_samples)))
> +        return AVERROR(ENOMEM);
> +    av_audio_fifo_read(s->audio_fifo, (void**)tmp->extended_data, nb_samples);
> +
> +    tmp->pts = s->next_pts;
> +    s->next_pts += av_rescale_q(nb_samples, (AVRational){1, link->sample_rate},
> +                                link->time_base);
> +
> +    av_frame_move_ref(frame, tmp);
> +    av_frame_free(&tmp);
> +
> +    return 0;
> +
> +}
> +
>  int av_buffersink_get_samples(AVFilterContext *ctx, AVFrame *frame, int nb_samples)
>  {
> -    av_assert0(!"TODO");
> +    BufferSinkContext *s = ctx->priv;
> +    AVFilterLink   *link = ctx->inputs[0];
> +    AVFrame *cur_frame;
> +    int ret = 0;
> +
> +    if (!s->audio_fifo) {
> +        int nb_channels = link->channels;
> +        if (!(s->audio_fifo = av_audio_fifo_alloc(link->format, nb_channels, nb_samples)))
> +            return AVERROR(ENOMEM);
> +    }
> +
> +    while (ret >= 0) {
> +        if (av_audio_fifo_size(s->audio_fifo) >= nb_samples)
> +            return read_from_fifo(ctx, frame, nb_samples);
> +
> +        if (!(cur_frame = av_frame_alloc()))
> +            return AVERROR(ENOMEM);
> +        ret = av_buffersink_get_frame_flags(ctx, cur_frame, 0);
> +        if (ret == AVERROR_EOF && av_audio_fifo_size(s->audio_fifo)) {
> +            av_frame_free(&cur_frame);
> +            return read_from_fifo(ctx, frame, av_audio_fifo_size(s->audio_fifo));
> +        } else if (ret < 0) {
> +            av_frame_free(&cur_frame);
> +            return ret;
> +        }
> +
> +        if (cur_frame->pts != AV_NOPTS_VALUE) {
> +            s->next_pts = cur_frame->pts -
> +                          av_rescale_q(av_audio_fifo_size(s->audio_fifo),
> +                                       (AVRational){ 1, link->sample_rate },
> +                                       link->time_base);
> +        }
> +
> +        ret = av_audio_fifo_write(s->audio_fifo, (void**)cur_frame->extended_data,
> +                                  cur_frame->nb_samples);
> +        av_frame_free(&cur_frame);
> +    }
> +
> +    return ret;
> +
>  }

Most of this code is from libav. Mention the original author and
please states the few differences in the commit log.

Should be fine otherwise, thanks.
-- 
FFmpeg = Fascinating & Foolish Mysterious Picky Earthshaking Gladiator