[FFmpeg-devel] [PATCH 2/2] examples: demuxing: simplify audio output

[Stefano Sabatini]

On date Saturday 2013-07-13 15:53:20 +0200, wm4 encoded:
> There is no reason why this should copy the audio data in a very
> complicated way. Also, strictly write the first plane, instead of
> writing the whole buffer. This is more helpful in context of the
> example. This way a user can clearly confirm that it works by playing
> the written data as raw audio.
> ---
>  doc/examples/demuxing.c | 29 +++++++++--------------------
>  1 file changed, 9 insertions(+), 20 deletions(-)
> 
> diff --git a/doc/examples/demuxing.c b/doc/examples/demuxing.c
> index 1c0f1ff..089b5b0 100644
> --- a/doc/examples/demuxing.c
> +++ b/doc/examples/demuxing.c
> @@ -104,26 +104,15 @@ static int decode_packet(int *got_frame, int cached)
>                     audio_frame_count++, frame->nb_samples,
>                     av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
>  
> -            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, av_frame_get_channels(frame),
> -                                   frame->nb_samples, frame->format, 1);
> -            if (ret < 0) {
> -                fprintf(stderr, "Could not allocate audio buffer\n");
> -                return AVERROR(ENOMEM);
> -            }
> -
> -            /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
> -            audio_dst_bufsize =
> -                av_samples_get_buffer_size(NULL, av_frame_get_channels(frame),
> -                                           frame->nb_samples, frame->format, 1);
> -
> -            /* copy audio data to destination buffer:
> -             * this is required since rawaudio expects non aligned data */
> -            av_samples_copy(audio_dst_data, frame->data, 0, 0,
> -                            frame->nb_samples, av_frame_get_channels(frame), frame->format);
> -
> -            /* write to rawaudio file */
> -            fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
> -            av_freep(&audio_dst_data[0]);
> +            /* Write the raw audio data samples of the first plane. This works
> +             * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
> +             * most audio decoders output planar audio, which uses a separate
> +             * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
> +             * In other words, this code will write only the first audio channel
> +             * in these cases.
> +             * You should use libswresample or libavfilter to convert the frame
> +             * to packed data. */
> +            fwrite(frame->extended_data[0], 1, frame->linesize[0], audio_dst_file);
>          }
>      }

LGTM but you need to update the ffplay command which is output at the
end, adopting the logic:
av_sample_fmt_is_planar(sample_fmt) ? 1 : audio_dec_ctx->channels;
-- 
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