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@@ -186,7 +186,7 @@ void VideoPlayer::decodeAudioThread()
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seek_flag_audio = 0;
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}
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}
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-
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+#ifdef USE_PCM_PLAYER
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///直播流为了保证降低延迟，当音频播放队列过大时，会提高播放的采样率
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///因此，非直播流，不能一次性把所有音频塞入队列，需要限制播放队列中的数据量
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if (!m_is_live_mode)
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@@ -196,7 +196,7 @@ void VideoPlayer::decodeAudioThread()
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mSleep(5);
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}
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}
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-
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+#endif
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//解码AVPacket->AVFrame
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if (int ret = avcodec_send_packet(aCodecCtx, packet) && ret != 0)
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{
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@@ -294,7 +294,7 @@ void VideoPlayer::decodeAudioThread()
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PCMFramePtr pcmFramePtr = std::make_shared<PCMFrame>();
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pcmFramePtr->setFrameBuffer(audio_buf, resampled_data_size);
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pcmFramePtr->setFrameInfo(m_out_sample_rate, audio_tgt_channels, pts_ms);
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-
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+#ifdef USE_PCM_PLAYER
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int audio_frame_size = m_pcm_player->inputPCMFrame(pcmFramePtr);
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if (audio_frame_size > 0) //大于0 表示音频播放设备处于打开状态
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@@ -302,7 +302,7 @@ void VideoPlayer::decodeAudioThread()
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// std::lock_guard<std::mutex> lck(m_mutex_audio_clk);
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audio_clock = m_pcm_player->getCurrentPts();
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}
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-
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+#endif
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// std::cout<<resampled_data_size<<" "<<audio_frame_size<<" "<<audio_clock<<" "<<pts_ms<<std::endl;
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}
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}
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@@ -310,205 +310,13 @@ void VideoPlayer::decodeAudioThread()
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av_packet_unref(packet);
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}
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-
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+#ifdef USE_PCM_PLAYER
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///等待播放完成
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while (m_pcm_player->getPcmFrameSize() > 0)
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{
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mSleep(5);
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}
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-
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+#endif
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mIsAudioThreadFinished = true;
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fprintf(stderr, "%s finished \n", __FUNCTION__);
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}
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-
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-
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-// int VideoPlayer::decodeAudioFrame(bool isBlock)
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-// {
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-// int audioBufferSize = 0;
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-// float pts_s = 0.0f; //时间戳(秒)
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-
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-// while(1)
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-// {
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-// if (mIsQuit)
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-// {
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-// mIsAudioThreadFinished = true;
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-// clearAudioQuene(); //清空队列
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-// break;
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-// }
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-
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-// if (mIsPause == true) //判断暂停
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-// {
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-// break;
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-// }
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-
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-// mConditon_Audio->Lock();
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-// int audio_list_size = mAudioPacktList.size();
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-// //fprintf(stderr, "mAudioPacktList.size()=%d \n", audio_list_size);
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-// // if (mAudioPacktList.size() <= 0)
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-// // {
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-// // if (isBlock)
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-// // {
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-// // mConditon_Audio->Wait();
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-// // }
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-// // else
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-// // {
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-// // mConditon_Audio->Unlock();
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-// // break;
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-// // }
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-// // }
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-
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-// AVPacket packet = mAudioPacktList.front();
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-// mAudioPacktList.pop_front();
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-// //qDebug()<<__FUNCTION__<<mAudioPacktList.size();
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-// mConditon_Audio->Unlock();
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-
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-// AVPacket *pkt = &packet;
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-
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-// /* if update, update the audio clock w/pts */
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-// if (packet->pts != AV_NOPTS_VALUE)
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-// {
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-// pts_s = av_q2d(mAudioStream->time_base) * packet->pts;
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-// }
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-
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-// //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
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-// if(strcmp((char*)packet->data,FLUSH_DATA) == 0)
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-// {
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-// avcodec_flush_buffers(aCodecCtx);
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-// av_packet_unref(pkt);
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-// continue;
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-// }
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-
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-// if (seek_flag_audio)
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-// {
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-// //发生了跳转 则跳过关键帧到目的时间的这几帧
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-// if (pts_s < seek_time)
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-// {
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-// continue;
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-// }
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-// else
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-// {
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-// seek_flag_audio = 0;
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-// }
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-// }
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-
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-// int got_frame = 0;
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-
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-// //解码AVPacket->AVFrame
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-// if (int ret = avcodec_send_packet(aCodecCtx, &packet) && ret != 0)
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-// {
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-// char buffer[1024] = {0};
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-// av_strerror(ret, buffer, 1024);
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-// fprintf(stderr, "input AVPacket to decoder failed! ret = %d %s\n", ret, buffer);
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-// }
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-// else
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-// {
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-// while(1)
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-// {
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-// int ret = avcodec_receive_frame(aCodecCtx, aFrame);
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-// if (ret != 0)
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-// {
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-// // char buffer[1024] = {0};
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-// // av_strerror(ret, buffer, 1024);
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-// // fprintf(stderr, "avcodec_receive_frame = %d %s\n", ret, buffer);
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-// break;
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-// }
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-
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-// /* if update, update the audio clock w/pts */
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-// if (packet.pts != AV_NOPTS_VALUE)
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-// {
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-// audio_clock = 1000 * av_q2d(mAudioStream->time_base) * packet.pts;
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-// }
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-
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-// int out_sample_rate = m_out_sample_rate;
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-
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-// /// ffmpeg解码之后得到的音频数据不是SDL想要的，
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-// /// 因此这里需要重采样成44100 双声道 AV_SAMPLE_FMT_S16
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-
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-// /// 需要保证重采样后音频的时间是相同的，不同采样率下同样时间采集的数据采样点个数肯定不一样。
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-// /// 因此就需要重新计算采样点个数（使用下面的函数）
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-// /// 将in_sample_rate的采样次数换算成out_sample_rate对应的采样次数
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-// int nb_samples = av_rescale_rnd(swr_get_delay(swrCtx, out_sample_rate) + aFrame->nb_samples, out_sample_rate, m_in_sample_rate, AV_ROUND_UP);
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-// //qDebug()<<swr_get_delay(swrCtx, out_sample_rate) + aFrame->nb_samples<<aFrame->nb_samples<<nb_samples;
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-// // int nb_samples = av_rescale_rnd(aFrame->nb_samples, out_sample_rate, m_in_sample_rate, AV_ROUND_INF);
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-// if (aFrame_ReSample != nullptr)
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-// {
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-// if (aFrame_ReSample->nb_samples != nb_samples || aFrame_ReSample->sample_rate != out_sample_rate)
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-// {
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-// av_frame_free(&aFrame_ReSample);
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-// aFrame_ReSample = nullptr;
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-// }
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-// }
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-
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-// ///解码一帧后才能获取到采样率等信息，因此将初始化放到这里
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-// if (aFrame_ReSample == nullptr)
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-// {
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-// aFrame_ReSample = av_frame_alloc();
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-
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-// aFrame_ReSample->format = out_sample_fmt;
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-// aFrame_ReSample->channel_layout = out_ch_layout;
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-// aFrame_ReSample->sample_rate = out_sample_rate;
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-// aFrame_ReSample->nb_samples = nb_samples;
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-
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-// int ret = av_samples_fill_arrays(aFrame_ReSample->data, aFrame_ReSample->linesize, audio_buf, audio_tgt_channels, aFrame_ReSample->nb_samples, out_sample_fmt, 0);
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-// // int ret = av_frame_get_buffer(aFrame_ReSample, 0);
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-// if (ret < 0)
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-// {
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-// fprintf(stderr, "Error allocating an audio buffer\n");
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-// // exit(1);
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-// }
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-// }
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-
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-// ///2024-09-16
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-// //当重采样的采样率不一致的时候，原先的方法会存在杂音问题，解决方法如下：
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-// // 1) 如果没有提供足够的空间用于保存输出数据，采样数据会缓存在swr中。可以通过 swr_get_out_samples()来获取下一次调用swr_convert在给定输入样本数量下输出样本数量的上限，来提供足够的空间。
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-// // 2)如果是采样频率转换，转换完成后采样数据可能会缓存在swr中，它希望你能给它更多的输入数据。
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-// // 3)如果实际上并不需要更多输入数据，通过调用swr_convert()，其中参数in_count设置为0来获取缓存在swr中的数据。
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-// // 4)转换结束之后需要冲刷swr_context的缓冲区，通过调用swr_convert()，其中参数in设置为NULL，参数in_count设置为0。
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-// for(int i=0;i<2;i++)
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-// {
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-// ///执行重采样
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-// int len2;
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-
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-// if (i == 0)
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-// {
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-// len2 = swr_convert(swrCtx, aFrame_ReSample->data, aFrame_ReSample->nb_samples, (const uint8_t**)aFrame->data, aFrame->nb_samples);
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-// }
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-// else
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-// {
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-// len2 = swr_convert(swrCtx, aFrame_ReSample->data, aFrame_ReSample->nb_samples, NULL, 0);
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-// }
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-
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-// if (len2 <= 0)
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-// {
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-// break;
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-// }
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-
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-// ///必须根据swr_convert实际返回的值计算数据大小
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-// int resampled_data_size = len2 * audio_tgt_channels * av_get_bytes_per_sample(out_sample_fmt);
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-// int OneChannelDataSize = resampled_data_size / audio_tgt_channels;
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-
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-// audioBufferSize = resampled_data_size;
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-
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-// PCMFramePtr pcmFramePtr = std::make_shared<PCMFrame>();
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-// pcmFramePtr->setFrameBuffer(audio_buf, resampled_data_size);
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-// pcmFramePtr->setFrameInfo(m_out_sample_rate, audio_tgt_channels, pts_s*1000);
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-
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-// int audio_frame_size = m_pcm_player->inputPCMFrame(pcmFramePtr);
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-// //qDebug()<<resampled_data_size<<audio_frame_size;
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-// audio_clock = m_pcm_player->getCurrentPts() / 1000.0;
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-// }
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-
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-// got_frame = 1;
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-// }
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-// }
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-
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-// av_packet_unref(&packet);
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-
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-// if (got_frame)
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-// {
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-// break;
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-// }
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-// }
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-
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-// return audioBufferSize;
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-// }
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