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improve timestamp type: int -> uint.
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parent
7951bf3bd6
commit
00d14daa16
6 changed files with 15 additions and 15 deletions
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@ -1489,7 +1489,7 @@ srs_error_t SrsRtcFrameBuilder::transcode_audio(SrsRtpPacket *pkt)
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srs_error_t err = srs_success;
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// to common message.
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uint32_t ts = pkt->get_avsync_time();
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uint64_t ts = pkt->get_avsync_time();
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if (is_first_audio_) {
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int header_len = 0;
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uint8_t* header = NULL;
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@ -1543,7 +1543,7 @@ srs_error_t SrsRtcFrameBuilder::transcode_audio(SrsRtpPacket *pkt)
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return err;
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}
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void SrsRtcFrameBuilder::packet_aac(SrsCommonMessage* audio, char* data, int len, uint32_t pts, bool is_header)
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void SrsRtcFrameBuilder::packet_aac(SrsCommonMessage* audio, char* data, int len, uint64_t pts, bool is_header)
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{
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int rtmp_len = len + 2;
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audio->header.initialize_audio(rtmp_len, pts, 1);
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@ -352,7 +352,7 @@ public:
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virtual srs_error_t on_rtp(SrsRtpPacket *pkt);
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private:
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srs_error_t transcode_audio(SrsRtpPacket *pkt);
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void packet_aac(SrsCommonMessage* audio, char* data, int len, uint32_t pts, bool is_header);
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void packet_aac(SrsCommonMessage* audio, char* data, int len, uint64_t pts, bool is_header);
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private:
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srs_error_t packet_video(SrsRtpPacket* pkt);
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srs_error_t packet_video_key_frame(SrsRtpPacket* pkt);
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@ -1259,7 +1259,7 @@ class SrsFrame
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{
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public:
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// The DTS/PTS in milliseconds, which is TBN=1000.
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int64_t dts;
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uint64_t dts;
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// PTS = DTS + CTS.
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int32_t cts;
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public:
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@ -109,31 +109,31 @@ void SrsMessageHeader::initialize_amf0_script(int size, int stream)
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message_type = RTMP_MSG_AMF0DataMessage;
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payload_length = (int32_t)size;
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timestamp_delta = (int32_t)0;
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timestamp = (int64_t)0;
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timestamp = 0;
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stream_id = (int32_t)stream;
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// amf0 script use connection2 chunk-id
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prefer_cid = RTMP_CID_OverConnection2;
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}
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void SrsMessageHeader::initialize_audio(int size, uint32_t time, int stream)
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void SrsMessageHeader::initialize_audio(int size, uint64_t time, int stream)
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{
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message_type = RTMP_MSG_AudioMessage;
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payload_length = (int32_t)size;
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timestamp_delta = (int32_t)time;
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timestamp = (int64_t)time;
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timestamp = time;
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stream_id = (int32_t)stream;
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// audio chunk-id
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prefer_cid = RTMP_CID_Audio;
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}
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void SrsMessageHeader::initialize_video(int size, uint32_t time, int stream)
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void SrsMessageHeader::initialize_video(int size, uint64_t time, int stream)
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{
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message_type = RTMP_MSG_VideoMessage;
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payload_length = (int32_t)size;
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timestamp_delta = (int32_t)time;
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timestamp = (int64_t)time;
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timestamp = time;
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stream_id = (int32_t)stream;
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// video chunk-id
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@ -151,7 +151,7 @@ public:
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// The 4 bytes are packed in the big-endian order.
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// @remark, used as calc timestamp when decode and encode time.
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// @remark, we use 64bits for large time for jitter detect and hls.
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int64_t timestamp;
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uint64_t timestamp;
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public:
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// Get the prefered cid(chunk stream id) which sendout over.
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// set at decoding, and canbe used for directly send message,
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@ -177,9 +177,9 @@ public:
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// Create a amf0 script header, set the size and stream_id.
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void initialize_amf0_script(int size, int stream);
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// Create a audio header, set the size, timestamp and stream_id.
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void initialize_audio(int size, uint32_t time, int stream);
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void initialize_audio(int size, uint64_t time, int stream);
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// Create a video header, set the size, timestamp and stream_id.
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void initialize_video(int size, uint32_t time, int stream);
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void initialize_video(int size, uint64_t time, int stream);
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};
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// The message is raw data RTMP message, bytes oriented,
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@ -299,7 +299,7 @@ private:
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// The helper handler for decoder, use RAW payload if NULL.
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ISrsRtspPacketDecodeHandler* decode_handler;
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private:
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int64_t avsync_time_;
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uint64_t avsync_time_;
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public:
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SrsRtpPacket();
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virtual ~SrsRtpPacket();
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@ -336,8 +336,8 @@ public:
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public:
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bool is_keyframe();
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// Get and set the packet sync time in milliseconds.
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void set_avsync_time(int64_t avsync_time) { avsync_time_ = avsync_time; }
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int64_t get_avsync_time() const { return avsync_time_; }
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void set_avsync_time(uint64_t avsync_time) { avsync_time_ = avsync_time; }
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uint64_t get_avsync_time() const { return avsync_time_; }
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};
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// Single payload data.
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