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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00

examples/decode_audio: handle planar audio now produced by the MP2 decoder

This commit is contained in:
Anton Khirnov 2016-10-19 21:56:22 +02:00
parent 3d66717f7c
commit 45a1ce2ff7

View File

@ -29,6 +29,7 @@
#include <stdlib.h>
#include <string.h>
#include "libavutil/channel_layout.h"
#include "libavutil/frame.h"
#include "libavutil/mem.h"
@ -40,7 +41,8 @@
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
FILE *outfile)
{
int ret, data_size;
int16_t *interleave_buf;
int ret, data_size, i;
/* send the packet with the compressed data to the decoder */
ret = avcodec_send_packet(dec_ctx, pkt);
@ -59,10 +61,28 @@ static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
exit(1);
}
data_size = av_samples_get_buffer_size(NULL, dec_ctx->channels,
frame->nb_samples,
dec_ctx->sample_fmt, 1);
fwrite(frame->data[0], 1, data_size, outfile);
/* the stream parameters may change at any time, check that they are
* what we expect */
if (av_get_channel_layout_nb_channels(frame->channel_layout) != 2 ||
frame->format != AV_SAMPLE_FMT_S16P) {
fprintf(stderr, "Unsupported frame parameters\n");
exit(1);
}
/* The decoded data is signed 16-bit planar -- each channel in its own
* buffer. We interleave the two channels manually here, but using
* libavresample is recommended instead. */
data_size = sizeof(*interleave_buf) * 2 * frame->nb_samples;
interleave_buf = av_malloc(data_size);
if (!interleave_buf)
exit(1);
for (i = 0; i < frame->nb_samples; i++) {
interleave_buf[2 * i] = ((int16_t*)frame->data[0])[i];
interleave_buf[2 * i + 1] = ((int16_t*)frame->data[1])[i];
}
fwrite(interleave_buf, 1, data_size, outfile);
av_freep(&interleave_buf);
}
}