ffplay注解

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/*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;elsereturn 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_timeris->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;elsedelay = 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;elseis->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;}#endifreturn 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);elseav_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 variablesret = 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(&copy, &is->video_st->attached_pic)) < 0)goto fail;packet_queue_put(&is->videoq, &copy);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
*/

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