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ffplay: use libswresample instead of av_audio_convert

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Previously ffplay expected SDL_AudioOpen to provide the requested sample rate
and channel number. This is no longer a requirement because this patch replaces
the audio convert function with libswresample's swr_convert which is capable of
handling different sample formats, sample rates and different number of
channels and different channel layouts.

The patch also removes the hardcoded 16bit samples assumption and uses
av_get_bytes_per_sample almost everywhere. The only exceptions are
the update_sample_display and video_audio_display functions, it
seemed too much of a headache to make them generic.

We also fix a tiny bug in sdl_audio_callback, we ensure that the number of
bytes when we put silence in the buffer is a multiple of the frame size.
This commit is contained in:
Marton Balint 2011-08-17 01:28:23 +02:00
parent 60aaf044f3
commit 1dd3c473a2
1 changed files with 86 additions and 52 deletions
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138
ffplay.c
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@ -38,6 +38,7 @@
#include "libavcodec/audioconvert.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "libswresample/swresample.h"
#if CONFIG_AVFILTER
# include "libavfilter/avcodec.h"
@ -152,9 +153,9 @@ typedef struct VideoState {
PacketQueue audioq;
int audio_hw_buf_size;
/* samples output by the codec. we reserve more space for avsync
compensation */
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];
compensation, resampling and format conversion */
DECLARE_ALIGNED(16,uint8_t,audio_buf1)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4];
uint8_t *audio_buf;
unsigned int audio_buf_size; /* in bytes */
int audio_buf_index; /* in bytes */
@ -162,7 +163,14 @@ typedef struct VideoState {
AVPacket audio_pkt_temp;
AVPacket audio_pkt;
enum AVSampleFormat audio_src_fmt;
AVAudioConvert *reformat_ctx;
enum AVSampleFormat audio_tgt_fmt;
int audio_src_channels;
int audio_tgt_channels;
int64_t audio_src_channel_layout;
int64_t audio_tgt_channel_layout;
int audio_src_freq;
int audio_tgt_freq;
struct SwrContext *swr_ctx;
double audio_current_pts;
double audio_current_pts_drift;
@ -732,7 +740,7 @@ static void video_audio_display(VideoState *s)
nb_freq= 1<<(rdft_bits-1);
/* compute display index : center on currently output samples */
channels = s->audio_st->codec->channels;
channels = s->audio_tgt_channels;
nb_display_channels = channels;
if (!s->paused) {
int data_used= s->show_mode == SHOW_MODE_WAVES ? s->width : (2*nb_freq);
@ -744,7 +752,7 @@ static void video_audio_display(VideoState *s)
the last buffer computation */
if (audio_callback_time) {
time_diff = av_gettime() - audio_callback_time;
delay -= (time_diff * s->audio_st->codec->sample_rate) / 1000000;
delay -= (time_diff * s->audio_tgt_freq) / 1000000;
}
delay += 2*data_used;
@ -1922,7 +1930,7 @@ static int synchronize_audio(VideoState *is, short *samples,
int n, samples_size;
double ref_clock;
n = 2 * is->audio_st->codec->channels;
n = av_get_bytes_per_sample(is->audio_tgt_fmt) * is->audio_tgt_channels;
samples_size = samples_size1;
/* if not master, then we try to remove or add samples to correct the clock */
@ -1944,15 +1952,15 @@ static int synchronize_audio(VideoState *is, short *samples,
avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);
if (fabs(avg_diff) >= is->audio_diff_threshold) {
wanted_size = samples_size + ((int)(diff * is->audio_st->codec->sample_rate) * n);
wanted_size = samples_size + ((int)(diff * is->audio_tgt_freq) * n);
nb_samples = samples_size / n;
min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n;
if (wanted_size < min_size)
wanted_size = min_size;
else if (wanted_size > max_size)
wanted_size = max_size;
else if (wanted_size > FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2)))
wanted_size = FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2));
/* add or remove samples to correction the synchro */
if (wanted_size < samples_size) {
@ -1995,7 +2003,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
AVPacket *pkt_temp = &is->audio_pkt_temp;
AVPacket *pkt = &is->audio_pkt;
AVCodecContext *dec= is->audio_st->codec;
int n, len1, data_size;
int len1, len2, data_size, resampled_data_size;
int64_t dec_channel_layout;
double pts;
int new_packet = 0;
int flush_complete = 0;
@ -2026,44 +2035,54 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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(AV_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",
av_get_sample_fmt_name(dec->sample_fmt),
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16));
break;
}
is->audio_src_fmt= dec->sample_fmt;
}
dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels);
if (is->reformat_ctx) {
const void *ibuf[6]= {is->audio_buf1};
void *obuf[6]= {is->audio_buf2};
int istride[6]= {av_get_bytes_per_sample(dec->sample_fmt)};
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");
if (dec->sample_fmt != is->audio_src_fmt || dec_channel_layout != is->audio_src_channel_layout || dec->sample_rate != is->audio_src_freq) {
if (is->swr_ctx)
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc2(NULL, is->audio_tgt_channel_layout, is->audio_tgt_fmt, is->audio_tgt_freq,
dec_channel_layout, dec->sample_fmt, dec->sample_rate,
0, NULL);
if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
fprintf(stderr, "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
dec->sample_rate,
av_get_sample_fmt_name(dec->sample_fmt),
dec->channels,
is->audio_tgt_freq,
av_get_sample_fmt_name(is->audio_tgt_fmt),
is->audio_tgt_channels);
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_src_channel_layout = dec_channel_layout;
is->audio_src_channels = dec->channels;
is->audio_src_freq = dec->sample_rate;
is->audio_src_fmt = dec->sample_fmt;
}
resampled_data_size = data_size;
if (is->swr_ctx) {
const uint8_t *in[] = {is->audio_buf1};
uint8_t *out[] = {is->audio_buf2};
len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt),
in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt));
if (len2 < 0) {
fprintf(stderr, "audio_resample() failed\n");
break;
}
if (len2 == sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt)) {
fprintf(stderr, "warning: audio buffer is probably too small\n");
swr_init(is->swr_ctx);
}
is->audio_buf = is->audio_buf2;
resampled_data_size = len2 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
} else {
is->audio_buf= is->audio_buf1;
}
/* if no pts, then compute it */
pts = is->audio_clock;
*pts_ptr = pts;
n = 2 * dec->channels;
is->audio_clock += (double)data_size /
(double)(n * dec->sample_rate);
is->audio_clock += (double)data_size / (dec->channels * dec->sample_rate * av_get_bytes_per_sample(dec->sample_fmt));
#ifdef DEBUG
{
static double last_clock;
@ -2073,7 +2092,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
last_clock = is->audio_clock;
}
#endif
return data_size;
return resampled_data_size;
}
/* free the current packet */
@ -2117,7 +2136,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
if (audio_size < 0) {
/* if error, just output silence */
is->audio_buf = is->audio_buf1;
is->audio_buf_size = 1024;
is->audio_buf_size = 256 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
memset(is->audio_buf, 0, is->audio_buf_size);
} else {
if (is->show_mode != SHOW_MODE_VIDEO)
@ -2136,8 +2155,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
stream += len1;
is->audio_buf_index += len1;
}
bytes_per_sec = is->audio_st->codec->sample_rate *
2 * is->audio_st->codec->channels;
bytes_per_sec = is->audio_tgt_freq * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
/* Let's assume the audio driver that is used by SDL has two periods. */
is->audio_current_pts = is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec;
@ -2153,6 +2171,7 @@ static int stream_component_open(VideoState *is, int stream_index)
SDL_AudioSpec wanted_spec, spec;
AVDictionary *opts;
AVDictionaryEntry *t = NULL;
int64_t wanted_channel_layout = 0;
if (stream_index < 0 || stream_index >= ic->nb_streams)
return -1;
@ -2184,12 +2203,14 @@ static int stream_component_open(VideoState *is, int stream_index)
avctx->flags |= CODEC_FLAG_EMU_EDGE;
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if(avctx->sample_rate <= 0 || avctx->channels <= 0){
fprintf(stderr, "Invalid sample rate or channel count\n");
wanted_channel_layout = (avctx->channel_layout && avctx->channels == av_get_channel_layout_nb_channels(avctx->channels)) ? avctx->channel_layout : av_get_default_channel_layout(avctx->channels);
wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX;
wanted_spec.channels = av_get_channel_layout_nb_channels(wanted_channel_layout);
wanted_spec.freq = avctx->sample_rate;
if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) {
fprintf(stderr, "Invalid sample rate or channel count!\n");
return -1;
}
wanted_spec.freq = avctx->sample_rate;
wanted_spec.channels = avctx->channels;
}
if (!codec ||
@ -2212,7 +2233,21 @@ static int stream_component_open(VideoState *is, int stream_index)
return -1;
}
is->audio_hw_buf_size = spec.size;
is->audio_src_fmt= AV_SAMPLE_FMT_S16;
if (spec.format != AUDIO_S16SYS) {
fprintf(stderr, "SDL advised audio format %d is not supported!\n", spec.format);
return -1;
}
if (spec.channels != wanted_spec.channels) {
wanted_channel_layout = av_get_default_channel_layout(spec.channels);
if (!wanted_channel_layout) {
fprintf(stderr, "SDL advised channel count %d is not supported!\n", spec.channels);
return -1;
}
}
is->audio_src_fmt = is->audio_tgt_fmt = AV_SAMPLE_FMT_S16;
is->audio_src_freq = is->audio_tgt_freq = spec.freq;
is->audio_src_channel_layout = is->audio_tgt_channel_layout = wanted_channel_layout;
is->audio_src_channels = is->audio_tgt_channels = spec.channels;
}
ic->streams[stream_index]->discard = AVDISCARD_DEFAULT;
@ -2270,9 +2305,8 @@ 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);
is->reformat_ctx = NULL;
if (is->swr_ctx)
swr_free(&is->swr_ctx);
break;
case AVMEDIA_TYPE_VIDEO:
packet_queue_abort(&is->videoq);