RTC: audio packet jitter buffer. #4295
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| // If packet is beyond window end, stop processing | ||
| srs_warn("Audio packet beyond window end, seq=%u, window_end=%u", next_seq, window_end); | ||
| break; | ||
| } else if (srs_rtp_seq_distance(last_audio_seq_num_, next_seq) > 1) {// If there's a gap and we haven't exceeded wait time, wait for missing packets |
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// If there's a gap and we haven't exceeded wait time, wait for missing packets
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duiniuluantanqin
approved these changes
Feb 27, 2025
winlinvip
reviewed
Jun 4, 2025
| @@ -335,6 +342,13 @@ class SrsRtcFrameBuilder | |||
| uint16_t header_sn_; | |||
| uint16_t lost_sn_; | |||
| int64_t rtp_key_frame_ts_; | |||
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| // Audio jitter buffer, map sequence number to packet | |||
| std::map<uint16_t, SrsRtpPacket*> audio_buffer_; | |||
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Obviously, the audio jitter buffer should be a class with its own fields and behaviors, rather than being directly exposed and mixed with the SrsRtcFrameBuilder code.
// Audio jitter buffer, map sequence number to packet
std::map<uint16_t, SrsRtpPacket*> audio_buffer_;
// Last processed sequence number
uint16_t last_audio_seq_num_;
// Last time we processed the jitter buffer
int64_t last_audio_process_time_ms_;
winlinvip
reviewed
Jun 4, 2025
| @@ -335,6 +342,13 @@ class SrsRtcFrameBuilder | |||
| uint16_t header_sn_; | |||
| uint16_t lost_sn_; | |||
| int64_t rtp_key_frame_ts_; | |||
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| // Audio jitter buffer, map sequence number to packet | |||
| std::map<uint16_t, SrsRtpPacket*> audio_buffer_; | |||
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It seems that the audio buffer is similar to SrsRtpRingBuffer; please consider whether you can use it directly.
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Rtp packets may be retransmitted, disordered, jittery, delayed, etc.There may be abnormalities when converting to rtmp.