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examples: demuxing: simplify audio output

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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.
This commit is contained in:
wm4
2013-07-18 15:45:36 +02:00
committed by Stefano Sabatini
parent 06b269dacb
commit 9f31c1608c
1 changed files with 10 additions and 36 deletions
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46
doc/examples/demuxing.c
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@@ -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;
}