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RTC: Refactor source to accept RTP audio packet.
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5 changed files with 44 additions and 30 deletions
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@ -2174,7 +2174,7 @@ srs_error_t SrsRtcPublisher::on_audio_frame(SrsRtpPacket2* frame)
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msg.set_extra_payloads(&sample, 1);
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// TODO: FIXME: Check error.
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source->on_rtc_audio(&msg);
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source->on_audio_imp(&msg);
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if (nn_audio_frames++ == 0) {
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SrsRtpHeader* h = &frame->rtp_header;
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