examples: demuxing: simplify audio output

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

@@ -47,10 +47,6 @@ static uint8_t *video_dst_data[4] = {NULL};
 static int      video_dst_linesize[4];
 static int video_dst_bufsize;
 
-static uint8_t **audio_dst_data = NULL;
-static int       audio_dst_linesize;
-static int audio_dst_bufsize;
-
 static int video_stream_idx = -1, audio_stream_idx = -1;
 static AVFrame *frame = NULL;
 static AVPacket pkt;
@@ -99,31 +95,21 @@ static int decode_packet(int *got_frame, int cached)
         decoded = FFMIN(ret, pkt.size);
 
         if (*got_frame) {
+            size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
             printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
                    cached ? "(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, unpadded_linesize, audio_dst_file);
         }
     }
 
@@ -250,8 +236,6 @@ int main (int argc, char **argv)
     }
 
     if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
-        int nb_planes;
-
         audio_stream = fmt_ctx->streams[audio_stream_idx];
         audio_dec_ctx = audio_stream->codec;
         audio_dst_file = fopen(audio_dst_filename, "wb");
@@ -260,15 +244,6 @@ int main (int argc, char **argv)
             ret = 1;
             goto end;
         }
-
-        nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
-            audio_dec_ctx->channels : 1;
-        audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
-        if (!audio_dst_data) {
-            fprintf(stderr, "Could not allocate audio data buffers\n");
-            ret = AVERROR(ENOMEM);
-            goto end;
-        }
     }
 
     /* dump input information to stderr */
@@ -349,7 +324,6 @@ end:
         fclose(audio_dst_file);
     av_free(frame);
     av_free(video_dst_data[0]);
-    av_free(audio_dst_data);
 
     return ret < 0;
 }