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Add sample format converter to FFplay. 

Originally committed as revision 14508 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Peter Ross 2008-08-02 05:01:30 +00:00
parent aaef2bb345
commit 5a4476e229
1 changed files with 46 additions and 5 deletions
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51
ffplay.c
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@ -26,6 +26,7 @@
#include "libavformat/rtsp.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
#include "libavcodec/audioconvert.h"
#include "cmdutils.h"
@ -127,12 +128,16 @@ typedef struct VideoState {
int audio_hw_buf_size;
/* samples output by the codec. we reserve more space for avsync
compensation */
DECLARE_ALIGNED(16,uint8_t,audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]);
DECLARE_ALIGNED(16,uint8_t,audio_buf1[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]);
DECLARE_ALIGNED(16,uint8_t,audio_buf2[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]);
uint8_t *audio_buf;
unsigned int audio_buf_size; /* in bytes */
int audio_buf_index; /* in bytes */
AVPacket audio_pkt;
uint8_t *audio_pkt_data;
int audio_pkt_size;
enum SampleFormat audio_src_fmt;
AVAudioConvert *reformat_ctx;
int show_audio; /* if true, display audio samples */
int16_t sample_array[SAMPLE_ARRAY_SIZE];
@ -1568,7 +1573,7 @@ static int synchronize_audio(VideoState *is, short *samples,
}
/* decode one audio frame and returns its uncompressed size */
static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size, double *pts_ptr)
static int audio_decode_frame(VideoState *is, double *pts_ptr)
{
AVPacket *pkt = &is->audio_pkt;
AVCodecContext *dec= is->audio_st->codec;
@ -1578,9 +1583,9 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
for(;;) {
/* NOTE: the audio packet can contain several frames */
while (is->audio_pkt_size > 0) {
data_size = buf_size;
data_size = sizeof(is->audio_buf1);
len1 = avcodec_decode_audio2(dec,
(int16_t *)audio_buf, &data_size,
(int16_t *)is->audio_buf1, &data_size,
is->audio_pkt_data, is->audio_pkt_size);
if (len1 < 0) {
/* if error, we skip the frame */
@ -1592,6 +1597,39 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size,
is->audio_pkt_size -= len1;
if (data_size <= 0)
continue;
if (dec->sample_fmt != is->audio_src_fmt) {
if (is->reformat_ctx)
av_audio_convert_free(is->reformat_ctx);
is->reformat_ctx= av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
dec->sample_fmt, 1, NULL, 0);
if (!is->reformat_ctx) {
fprintf(stderr, "Cannot convert %s sample format to %s sample format\n",
avcodec_get_sample_fmt_name(dec->sample_fmt),
avcodec_get_sample_fmt_name(SAMPLE_FMT_S16));
break;
}
is->audio_src_fmt= dec->sample_fmt;
}
if (is->reformat_ctx) {
const void *ibuf[6]= {is->audio_buf1};
void *obuf[6]= {is->audio_buf2};
int istride[6]= {av_get_bits_per_sample_format(dec->sample_fmt)/8};
int ostride[6]= {2};
int len= data_size/istride[0];
if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
printf("av_audio_convert() failed\n");
break;
}
is->audio_buf= is->audio_buf2;
/* FIXME: existing code assume that data_size equals framesize*channels*2
remove this legacy cruft */
data_size= len*2;
}else{
is->audio_buf= is->audio_buf1;
}
/* if no pts, then compute it */
pts = is->audio_clock;
*pts_ptr = pts;
@ -1655,7 +1693,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
audio_size = audio_decode_frame(is, is->audio_buf, sizeof(is->audio_buf), &pts);
audio_size = audio_decode_frame(is, &pts);
if (audio_size < 0) {
/* if error, just output silence */
is->audio_buf_size = 1024;
@ -1731,6 +1769,7 @@ static int stream_component_open(VideoState *is, int stream_index)
return -1;
}
is->audio_hw_buf_size = spec.size;
is->audio_src_fmt= SAMPLE_FMT_S16;
}
if(thread_count>1)
@ -1797,6 +1836,8 @@ static void stream_component_close(VideoState *is, int stream_index)
SDL_CloseAudio();
packet_queue_end(&is->audioq);
if (is->reformat_ctx)
av_audio_convert_free(is->reformat_ctx);
break;
case CODEC_TYPE_VIDEO:
packet_queue_abort(&is->videoq);