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Merge remote-tracking branch 'cus/stable'

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* cus/stable:
  ffplay: use libswresample instead of av_audio_convert
  audioconvert: add av_get_default_channel_layout public function
  ffplay: use avctx->channels and avctx->freq before avcodec_open2 consistently
  ffplay: remove now unnecessary request_channels, we set it now with options
  ffplay: set request_channels to 2

Merged-by: Michael Niedermayer <michaelni@gmx.at>
pull/2/head
Michael Niedermayer 13 years ago
parent 3ebab62fc6 1dd3c473a2
commit 57fa2fc69f
3 changed files with 104 additions and 61 deletions
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  1. 152
      ffplay.c
  2. 8
      libavutil/audioconvert.c
  3. 5
      libavutil/audioconvert.h

152
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",
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 (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),
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16));
break;
dec->channels,
is->audio_tgt_freq,
av_get_sample_fmt_name(is->audio_tgt_fmt),
is->audio_tgt_channels);
break;
}
is->audio_src_fmt= dec->sample_fmt;
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;
}
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");
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;
}
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{
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;
@ -2160,15 +2179,6 @@ static int stream_component_open(VideoState *is, int stream_index)
opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index]);
/* prepare audio output */
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (avctx->channels > 0) {
avctx->request_channels = FFMIN(2, avctx->channels);
} else {
avctx->request_channels = 2;
}
}
codec = avcodec_find_decoder(avctx->codec_id);
switch(avctx->codec_type){
case AVMEDIA_TYPE_AUDIO : if(audio_codec_name ) codec= avcodec_find_decoder_by_name( audio_codec_name); break;
@ -2192,8 +2202,17 @@ static int stream_component_open(VideoState *is, int stream_index)
if(codec->capabilities & CODEC_CAP_DR1)
avctx->flags |= CODEC_FLAG_EMU_EDGE;
wanted_spec.freq = avctx->sample_rate;
wanted_spec.channels = avctx->channels;
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
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;
}
}
if (!codec ||
avcodec_open2(avctx, codec, &opts) < 0)
return -1;
@ -2204,10 +2223,6 @@ static int stream_component_open(VideoState *is, int stream_index)
/* prepare audio output */
if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if(avctx->sample_rate <= 0 || avctx->channels <= 0){
fprintf(stderr, "Invalid sample rate or channel count\n");
return -1;
}
wanted_spec.format = AUDIO_S16SYS;
wanted_spec.silence = 0;
wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE;
@ -2218,7 +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;
@ -2234,7 +2263,7 @@ static int stream_component_open(VideoState *is, int stream_index)
is->audio_diff_avg_count = 0;
/* since we do not have a precise anough audio fifo fullness,
we correct audio sync only if larger than this threshold */
is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / avctx->sample_rate;
is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / wanted_spec.freq;
memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
packet_queue_init(&is->audioq);
@ -2276,9 +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);
@ -2379,6 +2407,8 @@ static int read_thread(void *arg)
if(genpts)
ic->flags |= AVFMT_FLAG_GENPTS;
av_dict_set(&codec_opts, "request_channels", "2", 0);
opts = setup_find_stream_info_opts(ic, codec_opts);
orig_nb_streams = ic->nb_streams;

8
libavutil/audioconvert.c
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@ -131,3 +131,11 @@ int av_get_channel_layout_nb_channels(int64_t channel_layout)
x &= x-1; // unset lowest set bit
return count;
}
int av_get_default_channel_layout(int nb_channels) {
int i;
for (i = 0; channel_layout_map[i].name; i++)
if (nb_channels == channel_layout_map[i].nb_channels)
return channel_layout_map[i].layout;
return 0;
}

5
libavutil/audioconvert.h
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@ -92,4 +92,9 @@ void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, int6
*/
int av_get_channel_layout_nb_channels(int64_t channel_layout);
/**
* Return default channel layout for a given number of channels.
*/
int av_get_default_channel_layout(int nb_channels);
#endif /* AVUTIL_AUDIOCONVERT_H */

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