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AI automatically translates all comments in the code into English (#3917)

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alex
2024-09-19 14:53:50 +08:00
committed by GitHub
parent 046de691cb
commit 4152dcd409
279 changed files with 10602 additions and 3038 deletions
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12
webrtc/WebRtcPlayer.h
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@@ -34,15 +34,19 @@ private:
void sendConfigFrames(uint32_t before_seq, uint32_t sample_rate, uint32_t timestamp, uint64_t ntp_timestamp);
private:
//媒体相关元数据
// 媒体相关元数据 [AUTO-TRANSLATED:f4cf8045]
// Media related metadata
MediaInfo _media_info;
//播放的rtsp源
// 播放的rtsp源 [AUTO-TRANSLATED:9963eed1]
// Playing rtsp source
std::weak_ptr<RtspMediaSource> _play_src;
// rtp 直接转发情况下通常会缺少 sps/pps, 在转发 rtp 前, 先发送一次相关帧信息, 部分情况下是可以播放的
// rtp 直接转发情况下通常会缺少 sps/pps, 在转发 rtp 前, 先发送一次相关帧信息, 部分情况下是可以播放的 [AUTO-TRANSLATED:65fdf16a]
// In the case of direct RTP forwarding, sps/pps is usually missing. Before forwarding RTP, send the relevant frame information once. In some cases, it can be played.
bool _send_config_frames_once { false };
//播放rtsp源的reader对象
// 播放rtsp源的reader对象 [AUTO-TRANSLATED:7b305055]
// Reader object for playing rtsp source
RtspMediaSource::RingType::RingReader::Ptr _reader;
};