qt5 开发的酷狗播放器源码，GUI完全仿酷狗，很值得学习的资源

#include "FFmpegPlayer.h" #include<QDebug> #include<windows.h> #include<QTime> #include<QImage> #include"Global_ValueGather.h" #define USE_MUTE 1 static bool isquit=false; //清空了 int VOL=80; // 包队列初始化 void packet_queue_init(PacketQueue* q) { q->last_pkt = NULL; q->first_pkt = NULL; #if USE_MUTE q->mutex = SDL_CreateMutex(); q->cond = SDL_CreateCond(); #endif } // 放入packet到队列中，不带头指针的队列 int packet_queue_put(PacketQueue*q, AVPacket *pkt) { AVPacketList *pktl; if (av_dup_packet(pkt) < 0) return -1; pktl = (AVPacketList*)av_malloc(sizeof(AVPacketList)); if (!pktl) return -1; pktl->pkt = *pkt; pktl->next = nullptr; #if USE_MUTE SDL_LockMutex(q->mutex); #endif if (!q->last_pkt) // 队列为空，新插入元素为第一个元素 q->first_pkt = pktl; else // 插入队尾 q->last_pkt->next = pktl; q->last_pkt = pktl; q->nb_packets++; q->size += pkt->size; #if USE_MUTE SDL_CondSignal(q->cond); SDL_UnlockMutex(q->mutex); #endif return 0; } // 从队列中取出packet int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) { AVPacketList *pkt1=NULL; int ret; #if USE_MUTE SDL_LockMutex(q->mutex); #endif for (;;) { if (isquit) return -1; pkt1 = q->first_pkt; if (pkt1) { q->first_pkt = pkt1->next; if (!q->first_pkt) { q->last_pkt = NULL; } q->nb_packets--; q->size -= pkt1->pkt.size; *pkt = pkt1->pkt; av_free(pkt1); ret = 1; break; } else if (!block) { ret = 0; break; } else { #if USE_MUTE SDL_CondWait(q->cond, q->mutex); #endif } } #if USE_MUTE SDL_UnlockMutex(q->mutex); #endif return ret; } void packet_queue_flush(PacketQueue *q) { #if USE_MUTE SDL_LockMutex(q->mutex); #endif AVPacketList *pkt=NULL, *pkt1=NULL; for(pkt = q->first_pkt; pkt != NULL; pkt = pkt1) { pkt1 = pkt->next; if(pkt1->pkt.data != (uint8_t *)"FLUSH") { } av_free_packet(&pkt->pkt); av_freep(&pkt); } q->last_pkt = NULL; q->first_pkt = NULL; q->nb_packets = 0; q->size = 0; #if USE_MUTE SDL_UnlockMutex(q->mutex); #endif } //////////////////////////////////////////////解码音频数据 int audio_decode_frame(mediaState* MS, uint8_t* audio_buf, int buf_size) { int len1; int data_size = 0; if (isquit) return -1; while (true) { while (MS->audio_pkt_size > 0) { int got_frame = 0; len1 = avcodec_decode_audio4(MS->acct, MS->frame, &got_frame, &MS->pkt); if (len1 < 0) // 出错，跳过 { MS->audio_pkt_size = 0; break; } MS->audio_pkt_data += len1; MS->audio_pkt_size -= len1; data_size = 0; if (got_frame) data_size = av_samples_get_buffer_size(nullptr, MS->acct->channels,MS-> frame->nb_samples, MS->acct->sample_fmt, 1); if (MS->frame->channels > 0 && MS->frame->channel_layout == 0) MS->frame->channel_layout = av_get_default_channel_layout(MS->frame->channels); else if (MS->frame->channels == 0 && MS->frame->channel_layout > 0) MS->frame->channels = av_get_channel_layout_nb_channels(MS->frame->channel_layout); if (MS->swr_ctx) { swr_free(&MS->swr_ctx); MS->swr_ctx = nullptr; } MS->swr_ctx = swr_alloc_set_opts(nullptr, MS->wanted_frame->channel_layout, (AVSampleFormat)MS->wanted_frame->format, MS->wanted_frame->sample_rate, MS->frame->channel_layout, (AVSampleFormat)MS->frame->format, MS->frame->sample_rate, 0, nullptr); if (!MS->swr_ctx || swr_init(MS->swr_ctx) < 0) { qDebug() << "swr_init failed:" << endl; break; } int dst_nb_samples = av_rescale_rnd(swr_get_delay(MS->swr_ctx, MS->frame->sample_rate) + MS->frame->nb_samples, MS->frame->sample_rate, MS->frame->sample_rate, AVRounding(1)); int len2 = swr_convert(MS->swr_ctx, &audio_buf, dst_nb_samples,(const uint8_t**)MS->frame->data, MS->frame->nb_samples);//这个才是最重要的~前面所做的工作都是为这个 if (len2 < 0) { qDebug() << "swr_convert failed\n"; break; } //[][]相当重要的一步，转换成时间 int resampled_data_size = len2 * MS->wanted_frame->channels* av_get_bytes_per_sample((AVSampleFormat)MS->wanted_frame->format); int n = 2 * MS->audio_st->codec->channels; MS->audio_clock += (double)resampled_data_size/(double)(n * MS->audio_st->codec->sample_rate); //[][] return MS->wanted_frame->channels * len2 * av_get_bytes_per_sample((AVSampleFormat)MS->wanted_frame->format); } //end while if (MS->pkt.buf) av_free_packet(&MS->pkt); //删除包 if (packet_queue_get(&MS->audioq,&MS->pkt,0)<=0) //重新从队列中获取包 { return -1; } //收到这个数据 说明刚刚执行过跳转 现在需要把解码器的数据 清除一下 if(strcmp((char*)MS->pkt.data,FLUSH_DATA) == 0) { avcodec_flush_buffers(MS->audio_st->codec); av_free_packet(&MS->pkt); continue; } if (MS->pkt.pts != AV_NOPTS_VALUE) { MS->audio_clock = (double)av_q2d(MS->audio_st->time_base) * (double)MS->pkt.pts; } MS->audio_pkt_data =MS->pkt.data; MS->audio_pkt_size = MS->pkt.size; } } // 解码后的回调函数 void audio_callback(void* userdata, Uint8* stream, int len) { mediaState* MS = (mediaState*)userdata; int len1, audio_size; SDL_memset(stream, 0, len); if (isquit) return; while (len > 0) { uint8_t audio_buff[MAX_AUDIO_FRAME_SIZE*2]; if (MS->audio_buf_index >= MS->audio_buf_size) { audio_size = audio_decode_frame(MS, audio_buff, sizeof(audio_buff)); if (isquit) return; if (audio_size < 0) { MS->audio_buf_size = 1024; SDL_memset(audio_buff, 0, MS->audio_buf_size); } else MS->audio_buf_size = audio_size; MS->audio_buf_index = 0; } len1 = MS->audio_buf_size - MS->audio_buf_index; if (len1 > len) len1 = len; SDL_MixAudio(stream, audio_buff + MS->audio_buf_index, len, VOL); len -= len1; stream += len1; MS->audio_buf_index += len1; } } static double synchronize_video(mediaState *MS, AVFrame *src_frame, double pts) { double frame_delay; if (pts != 0) { /* if we have pts, set video clock to it */ MS->video_clock = pts; } else { /* if we aren't given a pts, set it to the clock */ pts = MS->video_clock; } /* update the video clock */ frame_delay = av_q2d(MS->video_st->codec->time_base); /* if we are repeating a frame, adjust clock accordingly */ frame_delay += src_frame->repeat_pict * (frame_delay * 0.5); MS->video_clock += frame_delay; return pts; } int video_thread(void *arg) { mediaState *is = (mediaState *) arg; AVPacket pkt1, *packet = &pkt1; int ret, got_picture, numBytes; double video_pts = 0; //当前视频的pts double audio_pts = 0; //音频pts ///�