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

[wm4]

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);
         }
     }
 
-- 
1.8.3.1