ffplay注解

/*
get_clock(&is->vidclk):
获取到的实际上是:最后一帧的pts 加上 从处理最后一帧开始到现在的时间,具体参考set_clock_at 和get_clock的代码
c->pts_drift=最后一帧的pts-从处理最后一帧时间
clock=c->pts_drift+现在的时候
get_clock(&is->vidclk) ==is->vidclk.pts, av_gettime_relative() / 1000000.0 -is->vidclk.last_updated  +is->vidclk.pts
*/
static double get_clock(Clock *c)
{
if (*c->queue_serial != c->serial)
return NAN;
if (c->paused) {
return c->pts;
} else {
double time = av_gettime_relative() / 1000000.0;
return c->pts_drift + time - (time - c->last_updated) * (1.0 - c->speed);
}
}

static void set_clock_at(Clock *c, double pts, int serial, double time)
{
c->pts = pts;
c->last_updated = time;
c->pts_drift = c->pts - time;
c->serial = serial;
}

static void set_clock(Clock *c, double pts, int serial)
{
double time = av_gettime_relative() / 1000000.0;
set_clock_at(c, pts, serial, time);
}

static void set_clock_speed(Clock *c, double speed)
{
set_clock(c, get_clock(c), c->serial);
c->speed = speed;
}

static void init_clock(Clock *c, int *queue_serial)
{
c->speed = 1.0;
c->paused = 0;
c->queue_serial = queue_serial;
set_clock(c, NAN, -1);
}

static void sync_clock_to_slave(Clock *c, Clock *slave)
{
double clock = get_clock(c);
double slave_clock = get_clock(slave);

if (!isnan(slave_clock) && (isnan(clock) || fabs(clock - slave_clock) > AV_NOSYNC_THRESHOLD))
set_clock(c, slave_clock, slave->serial);
}

/* get the current master clock value */
static double get_master_clock(VideoState *is)
{
double val;

switch (get_master_sync_type(is)) {
case AV_SYNC_VIDEO_MASTER:
val = get_clock(&is->vidclk);
break;
case AV_SYNC_AUDIO_MASTER:
val = get_clock(&is->audclk);
break;
default:
val = get_clock(&is->extclk);
break;
}
return val;
}

static void check_external_clock_speed(VideoState *is)
{
if (is->video_stream >= 0 && is->videoq.nb_packets <= MIN_FRAMES / 2 ||
is->audio_stream >= 0 && is->audioq.nb_packets <= MIN_FRAMES / 2) {
set_clock_speed(&is->extclk, FFMAX(EXTERNAL_CLOCK_SPEED_MIN, is->extclk.speed - EXTERNAL_CLOCK_SPEED_STEP));
} else if ((is->video_stream < 0 || is->videoq.nb_packets > MIN_FRAMES * 2) &&
(is->audio_stream < 0 || is->audioq.nb_packets > MIN_FRAMES * 2)) {
set_clock_speed(&is->extclk, FFMIN(EXTERNAL_CLOCK_SPEED_MAX, is->extclk.speed + EXTERNAL_CLOCK_SPEED_STEP));
} else {
double speed = is->extclk.speed;
if (speed != 1.0)
set_clock_speed(&is->extclk, speed + EXTERNAL_CLOCK_SPEED_STEP * (1.0 - speed) / fabs(1.0 - speed));
}
}

static double compute_target_delay(double delay, VideoState *is)
{
double sync_threshold, diff = 0;

/* update delay to follow master synchronisation source */
if (get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER) {
/* if video is slave, we try to correct big delays by
duplicating or deleting a frame */
diff = get_clock(&is->vidclk) - get_master_clock(is);

/* skip or repeat frame. We take into account the
delay to compute the threshold. I still don't know
if it is the best guess */
sync_threshold = FFMAX(AV_SYNC_THRESHOLD_MIN, FFMIN(AV_SYNC_THRESHOLD_MAX, delay));
if (!isnan(diff) && fabs(diff) < is->max_frame_duration) {
if (diff <= -sync_threshold)  )/*当前视频帧落后于主时钟源,减小delay*/
delay = FFMAX(0, delay + diff);
else if (diff >= sync_threshold && delay > AV_SYNC_FRAMEDUP_THRESHOLD)
delay = delay + diff; /*大概意思是:本来当视频帧超前的时候,
我们应该要选择重复该帧或者下面的2倍延时(即加重延时的策略),
但因为该帧的显示时间大于显示更新门槛,
所以这个时候不应该以该帧做同步*/
else if (diff >= sync_threshold)
delay = 2 * delay;    /*采取加倍延时*/
}
}

av_log(NULL, AV_LOG_TRACE, "video: delay=%0.3f A-V=%f\n",
delay, -diff);

return delay;
}

static double vp_duration(VideoState *is, Frame *vp, Frame *nextvp)
{
if (vp->serial == nextvp->serial) {
double duration = nextvp->pts - vp->pts;
if (isnan(duration) || duration <= 0 || duration > is->max_frame_duration)
return vp->duration;
else
return duration;
} else {
return 0.0;
}
}

static void update_video_pts(VideoState *is, double pts, int64_t pos, int serial)
{
/* update current video pts */
set_clock(&is->vidclk, pts, serial);
sync_clock_to_slave(&is->extclk, &is->vidclk);
}

/* called to display each frame */
static void video_refresh(void *opaque, double *remaining_time)
{
VideoState *is = opaque;
double time;

if (!is->paused && get_master_sync_type(is) == AV_SYNC_EXTERNAL_CLOCK && is->realtime)
check_external_clock_speed(is);

if (is->video_st) {
int redisplay = 0;
if (is->force_refresh)
redisplay = frame_queue_prev(&is->pictq);
retry:
if (frame_queue_nb_remaining(&is->pictq) == 0) {
// nothing to do, no picture to display in the queue
} else {
double last_duration, duration, delay;
Frame *vp, *lastvp;

/* dequeue the picture */
lastvp = frame_queue_peek_last(&is->pictq);
vp = frame_queue_peek(&is->pictq);

if (vp->serial != is->videoq.serial) {
frame_queue_next(&is->pictq);
redisplay = 0;
goto retry;
}

if (lastvp->serial != vp->serial && !redisplay)  //lastvp->serial != vp->serial 说明SEEK过,重新调整frame_timer
is->frame_timer = av_gettime_relative() / 1000000.0;

if (is->paused)
goto display;

/*通过pts计算duration，duration是上一帧videoframe的持续时间，当前帧的pts减去上一帧的pts*/
/* compute nominal last_duration */
last_duration = vp_duration(is, lastvp, vp);
if (redisplay)
delay = 0.0;
else
delay = compute_target_delay(last_duration, is);

time= av_gettime_relative()/1000000.0;
/*frame_timer实际上就是上一帧的播放时间，而 frame_timer + delay 实际上就是当前这一帧的播放时间*/
if (time < is->frame_timer + delay && !redisplay) {
/*remaining 就是在refresh_loop_wait_event 中还需要睡眠的时间，其实就是现在还没到这一帧的播放时间，我们需要睡眠等待*/
*remaining_time = FFMIN(is->frame_timer + delay - time, *remaining_time);
return;
}

is->frame_timer += delay;
/*如果当前这一帧播放时间已经过了，并且其和当前系统时间的差值超过AV_SYNC_THRESHOLD_MAX，
则将当前这一帧的播放时间改为当前系统时间，并在后续判断是否需要丢帧，
其目的是  为后面帧的播放时间重新调整frame_timer */
if (delay > 0 && time - is->frame_timer > AV_SYNC_THRESHOLD_MAX)
is->frame_timer = time;

SDL_LockMutex(is->pictq.mutex);
if (!redisplay && !isnan(vp->pts))
/*更新视频的clock，将当前帧的pts和当前系统的时间保存起来，这2个数据将和audio  clock的pts 和系统时间一起计算delay*/
update_video_pts(is, vp->pts, vp->pos, vp->serial);
SDL_UnlockMutex(is->pictq.mutex);
/*frame_timer+duration 当前帧的播放时间+当前帧的持续时间=下一帧的播放时间
time > is->frame_timer + duration  当前时间>下一帧的播放时间,来不及播放本帧,下一帧的播放时间已经到了,说明当前帧可以丢弃了*/
if (frame_queue_nb_remaining(&is->pictq) > 1) {
Frame *nextvp = frame_queue_peek_next(&is->pictq);
duration = vp_duration(is, vp, nextvp); //当前帧videoframe的持续时间
/*如果延迟时间超过一帧，并且允许丢帧，则进行丢帧处理*/
if(!is->step && (redisplay || framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) && time > is->frame_timer + duration) {
if (!redisplay)
is->frame_drops_late++;
/*丢掉延迟的帧，取下一帧*/
frame_queue_next(&is->pictq);
redisplay = 0;
goto retry;
}
}
display:
/* display picture */
video_display(is);

frame_queue_next(&is->pictq);

if (is->step && !is->paused)
stream_toggle_pause(is);
}
}
is->force_refresh = 0;
}

static int audio_thread(void *arg)
{
VideoState *is = arg;
AVFrame *frame = av_frame_alloc();
Frame *af;

int got_frame = 0;
AVRational tb;
int ret = 0;

do {
if ((got_frame = decoder_decode_frame(&is->auddec, frame, NULL)) < 0)
goto the_end;

if (got_frame) {
tb = (AVRational) {
1, frame->sample_rate
};

af = frame_queue_peek_writable(&is->sampq);

af->pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);
af->pos = av_frame_get_pkt_pos(frame);
af->serial = is->auddec.pkt_serial;
af->duration = av_q2d((AVRational) {
frame->nb_samples, frame->sample_rate
});

av_frame_move_ref(af->frame, frame);
frame_queue_push(&is->sampq);

}
} while (ret >= 0 || ret == AVERROR(EAGAIN) || ret == AVERROR_EOF);

the_end:
return ret;
}

static int video_thread(void *arg)
{
VideoState *is = arg;
AVFrame *frame = av_frame_alloc();
double pts;
double duration;
int ret;
AVRational tb = is->video_st->time_base;
AVRational frame_rate = av_guess_frame_rate(is->ic, is->video_st, NULL);

for (;;) {
ret = get_video_frame(is, frame);

if (ret < 0)
goto the_end;

if (!ret)
continue;

duration = (frame_rate.num && frame_rate.den ? av_q2d((AVRational) {
frame_rate.den, frame_rate.num
}) : 0);

pts = (frame->pts == AV_NOPTS_VALUE) ? NAN : frame->pts * av_q2d(tb);

queue_picture(is, frame, pts, duration, av_frame_get_pkt_pos(frame), is->viddec.pkt_serial);
}

the_end:
return 0;
}

/* return the wanted number of samples to get better sync if sync_type is video
* or external master clock */
static int synchronize_audio(VideoState *is, int nb_samples)
{
int wanted_nb_samples = nb_samples;

/* if not master, then we try to remove or add samples to correct the clock */
if (get_master_sync_type(is) != AV_SYNC_AUDIO_MASTER) {
double diff, avg_diff;
int min_nb_samples, max_nb_samples;

diff = get_clock(&is->audclk) - get_master_clock(is);

if (!isnan(diff) && fabs(diff) < AV_NOSYNC_THRESHOLD) {
is->audio_diff_cum = diff + is->audio_diff_avg_coef * is->audio_diff_cum;
if (is->audio_diff_avg_count < AUDIO_DIFF_AVG_NB) {
/* not enough measures to have a correct estimate */
is->audio_diff_avg_count++;
} else {
/* estimate the A-V difference */
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);
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 = av_clip(wanted_nb_samples, min_nb_samples, max_nb_samples);
}
av_log(NULL, AV_LOG_TRACE, "diff=%f adiff=%f sample_diff=%d apts=%0.3f %f\n",
diff, avg_diff, wanted_nb_samples - nb_samples,
is->audio_clock, is->audio_diff_threshold);
}
} else {
/* too big difference : may be initial PTS errors, so
reset A-V filter */
is->audio_diff_avg_count = 0;
is->audio_diff_cum       = 0;
}
}

return wanted_nb_samples;
}
/**
* Decode one audio frame and return its uncompressed size.
*
* The processed audio frame is decoded, converted if required, and
* stored in is->audio_buf, with size in bytes given by the return
* value.
*/
static int audio_decode_frame(VideoState *is)
{
int data_size, resampled_data_size;
int64_t dec_channel_layout;
av_unused double audio_clock0;
int wanted_nb_samples;
Frame *af;

if (is->paused)
return -1;

do {
af = frame_queue_peek_readable(&is->sampq);
frame_queue_next(&is->sampq);
} while (af->serial != is->audioq.serial);

data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame), af->frame->nb_samples, af->frame->format, 1);

dec_channel_layout =
(af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame));

wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);

if (af->frame->format    != is->audio_src.fmt            		 ||
dec_channel_layout       != is->audio_src.channel_layout ||
af->frame->sample_rate   != is->audio_src.freq           ||
(wanted_nb_samples       != af->frame->nb_samples && !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,
dec_channel_layout,  af->frame->format, af->frame->sample_rate, 0, NULL);

swr_init(is->swr_ctx);

is->audio_src.channel_layout = dec_channel_layout;
is->audio_src.channels       = av_frame_get_channels(af->frame);
is->audio_src.freq = af->frame->sample_rate;
is->audio_src.fmt = af->frame->format;
}

if (is->swr_ctx) {
const uint8_t **in = (const uint8_t **)af->frame->extended_data;
uint8_t **out = &is->audio_buf1;
int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
int len2;

if (wanted_nb_samples != af->frame->nb_samples) {
if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
return -1;
}
}
av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);

len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);

is->audio_buf = is->audio_buf1;
//每声道采样数 x 声道数 x 每个采样字节数
resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
} else {
is->audio_buf = af->frame->data[0];
resampled_data_size = data_size;
}

audio_clock0 = is->audio_clock;
/* update the audio clock with the pts */
//  1/af->frame->sample_rate=采样一个样本点所需要的时候
if (!isnan(af->pts))
is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
else
is->audio_clock = NAN;
is->audio_clock_serial = af->serial;

#ifdef DEBUG
{
static double last_clock;
printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",
is->audio_clock - last_clock,
is->audio_clock, audio_clock0);
last_clock = is->audio_clock;
}
#endif
return resampled_data_size;
}

/* prepare a new audio buffer */
static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
{
VideoState *is = opaque;
int audio_size, len1;

audio_callback_time = av_gettime_relative();

while (len > 0) {
if (is->audio_buf_index >= is->audio_buf_size) {
audio_size = audio_decode_frame(is);
if (audio_size < 0) {
/* if error, just output silence */
is->audio_buf      = is->silence_buf;
is->audio_buf_size = sizeof(is->silence_buf) / is->audio_tgt.frame_size * is->audio_tgt.frame_size;
} else {
is->audio_buf_size = audio_size;
}
is->audio_buf_index = 0;
}
len1 = is->audio_buf_size - is->audio_buf_index;
if (len1 > len)
len1 = len;
memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1);
len -= len1;
stream += len1;
is->audio_buf_index += len1;
}
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. */
if (!isnan(is->audio_clock)) {
/*set_clock_at第二个参数是计算音频已经播放的时间，相当于video中的上一帧的播放时间，如果不同过SDL，例如直接使用linux下的dsp设备进行播放，那么我们可以通过ioctl接口获取到驱动的audiobuffer中还有多少数据没播放，这样，我们通过音频的采样率和位深，可以很精确的算出音频播放到哪个点了，但是此处的计算方法有点让人看不懂*/
set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
sync_clock_to_slave(&is->extclk, &is->audclk);
}
}

/* this thread gets the stream from the disk or the network */
static int read_thread(void *arg)
{
VideoState *is = arg;
AVFormatContext *ic = NULL;
int err, i, ret;
AVPacket pkt1, *pkt = &pkt1;
int64_t stream_start_time;
int pkt_in_play_range = 0;
AVDictionaryEntry *t;
AVDictionary **opts;
int orig_nb_streams;
SDL_mutex *wait_mutex = SDL_CreateMutex();
int scan_all_pmts_set = 0;
int64_t pkt_ts;
int video_index = -1;
int audio_index = -1;

is->last_video_stream = is->video_stream = -1;
is->last_audio_stream = is->audio_stream = -1;
is->eof = 0;

ic = avformat_alloc_context();

avformat_open_input(&ic, is->filename, is->iformat, &format_opts);
is->ic = ic;

is->max_frame_duration = (ic->iformat->flags & AVFMT_TS_DISCONT) ? 10.0 : 3600.0;

is->realtime = is_realtime(ic);

for(i=0; i<ic->nb_streams; i++) {
if(ic->streams[i]->codec->codec_type==AVMEDIA_TYPE_VIDEO &&
video_index < 0) {
video_index=i;
}
if(ic->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO &&
audio_index < 0) {
audio_index=i;
}
}

stream_component_open(is, audio_index);

stream_component_open(is, video_index);

for (;;) {

if (is->abort_request)
break;

if (is->paused != is->last_paused) {
is->last_paused = is->paused;
if (is->paused)
is->read_pause_return = av_read_pause(ic);
else
av_read_play(ic);
}

if (is->seek_req) {
int64_t seek_target = is->seek_pos;
int64_t seek_min    = is->seek_rel > 0 ? seek_target - is->seek_rel + 2: INT64_MIN;
int64_t seek_max    = is->seek_rel < 0 ? seek_target - is->seek_rel - 2: INT64_MAX;
// FIXME the +-2 is due to rounding being not done in the correct direction in generation
//      of the seek_pos/seek_rel variables

ret = avformat_seek_file(is->ic, -1, seek_min, seek_target, seek_max, is->seek_flags);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR,
"%s: error while seeking\n", is->ic->filename);
} else {
if (is->audio_stream >= 0) {
packet_queue_flush(&is->audioq);
packet_queue_put(&is->audioq, &flush_pkt);
}
if (is->video_stream >= 0) {
packet_queue_flush(&is->videoq);
packet_queue_put(&is->videoq, &flush_pkt);
}
if (is->seek_flags & AVSEEK_FLAG_BYTE) {
set_clock(&is->extclk, NAN, 0);
} else {
set_clock(&is->extclk, seek_target / (double)AV_TIME_BASE, 0);
}
}
is->seek_req = 0;
is->queue_attachments_req = 1;
is->eof = 0;
if (is->paused)
step_to_next_frame(is);
}
if (is->queue_attachments_req) {
if (is->video_st && is->video_st->disposition & AV_DISPOSITION_ATTACHED_PIC) {
AVPacket copy;
if ((ret = av_copy_packet(©, &is->video_st->attached_pic)) < 0)
goto fail;
packet_queue_put(&is->videoq, ©);
packet_queue_put_nullpacket(&is->videoq, is->video_stream);
}
is->queue_attachments_req = 0;
}

if (!is->paused &&
(!is->audio_st || (is->auddec.finished == is->audioq.serial && frame_queue_nb_remaining(&is->sampq) == 0)) &&
(!is->video_st || (is->viddec.finished == is->videoq.serial && frame_queue_nb_remaining(&is->pictq) == 0))) {
if (loop != 1 && (!loop || --loop)) {
stream_seek(is, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0);
} else if (autoexit) {
ret = AVERROR_EOF;
goto fail;
}
}
ret = av_read_frame(ic, pkt);
if (ret < 0) {
if ((ret == AVERROR_EOF || avio_feof(ic->pb)) && !is->eof) {
if (is->video_stream >= 0)
packet_queue_put_nullpacket(&is->videoq, is->video_stream);
if (is->audio_stream >= 0)
packet_queue_put_nullpacket(&is->audioq, is->audio_stream);
is->eof = 1;
}
if (ic->pb && ic->pb->error)
break;
SDL_LockMutex(wait_mutex);
SDL_CondWaitTimeout(is->continue_read_thread, wait_mutex, 10);
SDL_UnlockMutex(wait_mutex);
continue;
} else {
is->eof = 0;
}
/* check if packet is in play range specified by user, then queue, otherwise discard */
stream_start_time = ic->streams[pkt->stream_index]->start_time;
pkt_ts = pkt->pts == AV_NOPTS_VALUE ? pkt->dts : pkt->pts;

pkt_in_play_range = duration == AV_NOPTS_VALUE ||
(pkt_ts - (stream_start_time != AV_NOPTS_VALUE ? stream_start_time : 0)) *
av_q2d(ic->streams[pkt->stream_index]->time_base) -
(double)(start_time != AV_NOPTS_VALUE ? start_time : 0) / 1000000 <= ((double)duration / 1000000);

if (pkt->stream_index == is->audio_stream && pkt_in_play_range) {
packet_queue_put(&is->audioq, pkt);
} else if (pkt->stream_index == is->video_stream && pkt_in_play_range
&& !(is->video_st->disposition & AV_DISPOSITION_ATTACHED_PIC)) {
packet_queue_put(&is->videoq, pkt);
} else {
av_free_packet(pkt);
}
}

fail:
return 0;
}

/* seek in the stream */
static void stream_seek(VideoState *is, int64_t pos, int64_t rel, int seek_by_bytes)
{
if (!is->seek_req) {
is->seek_pos = pos;
is->seek_rel = rel;
is->seek_flags &= ~AVSEEK_FLAG_BYTE;
if (seek_by_bytes)
is->seek_flags |= AVSEEK_FLAG_BYTE;
is->seek_req = 1;
SDL_CondSignal(is->continue_read_thread);
}
}

/* pause or resume the video */
static void stream_toggle_pause(VideoState *is)
{
if (is->paused) {
// last_updated 记录了上一帧视频图像显示时的系统时钟， av_gettime() - last_updated得到的结果刚好是pause这段时间间隔，
//通过这种方式保证了frame_timer永远记录的是ffplay启动后到当前时间点的时间间隔
is->frame_timer += av_gettime_relative() / 1000000.0 - is->vidclk.last_updated;
if (is->read_pause_return != AVERROR(ENOSYS)) {
is->vidclk.paused = 0;
}
set_clock(&is->vidclk, get_clock(&is->vidclk), is->vidclk.serial);
}
set_clock(&is->extclk, get_clock(&is->extclk), is->extclk.serial);
is->paused = is->audclk.paused = is->vidclk.paused = is->extclk.paused = !is->paused;
}

static void toggle_pause(VideoState *is)
{
stream_toggle_pause(is);
is->step = 0;
}

/*

44100是每秒采样次数

一般pcm如果是双通道16位的话(32bit)，每个样本是4Byte

所以 一秒的数据量是4410*4bytes  实践中ACC是1024（4096字节）个样本一个avframe---MP3是1152

也就是一秒内有44100/1024个avframe被打上了PTS,一秒内约 43或44个avframe来包含44100*4/4096的数据

用32位表示其实是用32位空间来存储

就是4字节才能把2个声道的信息全部存下来进行编码

*/

/*

serial这个变量主要是维护数据的一致性

PacketQueue队列自己有一个serail变量

他管理的链表每个包有一个serail变量

当插入flush_pkt包的时候,队列的serail变量会++,说明又是一个新开始

FrameQueue队列的每一帧图像都有一个serail变量(解码之前的包serail的值),

显示之前,他先和PacketQueue队列的serail变量比较,,,,

Clock 结结中也有这个变量

目前主要是seek的时候,插入flush_pkt包 ,这个时候PacketQueue被分成两部分

flush_pkt之前的部分,和flush_pkt之后的部分,之前的serial=1,测试之后的为serial=2;

而FrameQueue队列只能依据serail变量来区分seek之前的包,和之后的包,之前的肯定就不会再显示了

Clock结构中的这个变量,也是这个作用

*/