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ffplay: put audio parameters to their own struct

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Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint 2012-04-14 17:53:45 +02:00
parent 03095d73a3
commit e1248f5c52
1 changed files with 34 additions and 33 deletions
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67
ffplay.c
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@ -117,6 +117,13 @@ typedef struct SubPicture {
AVSubtitle sub;
} SubPicture;
typedef struct AudioParams {
int freq;
int channels;
int channel_layout;
enum AVSampleFormat fmt;
} AudioParams;
enum {
AV_SYNC_AUDIO_MASTER, /* default choice */
AV_SYNC_VIDEO_MASTER,
@ -163,14 +170,8 @@ typedef struct VideoState {
int audio_write_buf_size;
AVPacket audio_pkt_temp;
AVPacket audio_pkt;
enum AVSampleFormat audio_src_fmt;
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 AudioParams audio_src;
struct AudioParams audio_tgt;
struct SwrContext *swr_ctx;
double audio_current_pts;
double audio_current_pts_drift;
@ -759,7 +760,7 @@ static void video_audio_display(VideoState *s)
nb_freq = 1 << (rdft_bits - 1);
/* compute display index : center on currently output samples */
channels = s->audio_tgt_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);
@ -771,7 +772,7 @@ 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_tgt_freq) / 1000000;
delay -= (time_diff * s->audio_tgt.freq) / 1000000;
}
delay += 2 * data_used;
@ -2032,7 +2033,7 @@ static int synchronize_audio(VideoState *is, int nb_samples)
avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef);
if (fabs(avg_diff) >= is->audio_diff_threshold) {
wanted_nb_samples = nb_samples + (int)(diff * is->audio_src_freq);
wanted_nb_samples = nb_samples + (int)(diff * is->audio_src.freq);
min_nb_samples = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX) / 100));
max_nb_samples = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX) / 100));
wanted_nb_samples = FFMIN(FFMAX(wanted_nb_samples, min_nb_samples), max_nb_samples);
@ -2104,14 +2105,14 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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);
wanted_nb_samples = synchronize_audio(is, is->frame->nb_samples);
if (dec->sample_fmt != is->audio_src_fmt ||
dec_channel_layout != is->audio_src_channel_layout ||
dec->sample_rate != is->audio_src_freq ||
if (dec->sample_fmt != is->audio_src.fmt ||
dec_channel_layout != is->audio_src.channel_layout ||
dec->sample_rate != is->audio_src.freq ||
(wanted_nb_samples != is->frame->nb_samples && !is->swr_ctx)) {
if (is->swr_ctx)
swr_free(&is->swr_ctx);
is->swr_ctx = swr_alloc_set_opts(NULL,
is->audio_tgt_channel_layout, is->audio_tgt_fmt, is->audio_tgt_freq,
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) {
@ -2119,15 +2120,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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);
is->audio_tgt.freq,
av_get_sample_fmt_name(is->audio_tgt.fmt),
is->audio_tgt.channels);
break;
}
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;
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;
@ -2135,24 +2136,24 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
const uint8_t *in[] = { is->frame->data[0] };
uint8_t *out[] = {is->audio_buf2};
if (wanted_nb_samples != is->frame->nb_samples) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt_freq / dec->sample_rate,
wanted_nb_samples * is->audio_tgt_freq / dec->sample_rate) < 0) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - is->frame->nb_samples) * is->audio_tgt.freq / dec->sample_rate,
wanted_nb_samples * is->audio_tgt.freq / dec->sample_rate) < 0) {
fprintf(stderr, "swr_set_compensation() failed\n");
break;
}
}
len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt),
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, is->frame->nb_samples);
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)) {
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);
resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
} else {
is->audio_buf = is->frame->data[0];
}
@ -2207,7 +2208,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
VideoState *is = opaque;
int audio_size, len1;
int bytes_per_sec;
int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt_channels, 1, is->audio_tgt_fmt, 1);
int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, 1, is->audio_tgt.fmt, 1);
double pts;
audio_callback_time = av_gettime();
@ -2234,7 +2235,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
stream += len1;
is->audio_buf_index += len1;
}
bytes_per_sec = is->audio_tgt_freq * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt);
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;
@ -2289,10 +2290,10 @@ static int audio_open(VideoState *is, int64_t channel_layout, int channels, int
}
is->audio_hw_buf_size = spec.size;
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;
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;
return 0;
}