Don't store RtpPacketInfo in the PacketBuffer.
Historically the PacketBuffer used a callback for assembled frames, and because of that RtpPacketInfos were piped through it even though they didn't have anything to do with the PacketBuffer. With this CL RtpPacketInfos are stored in the RtpVideoStreamReceiver(2) instead. Bug: webrtc:12579 Change-Id: Ia6285b59e135910eee7234b89b23425bb0fc0d2b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215320 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33980}
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@ -35,20 +35,13 @@ namespace webrtc {
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namespace video_coding {
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PacketBuffer::Packet::Packet(const RtpPacketReceived& rtp_packet,
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const RTPVideoHeader& video_header,
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Timestamp receive_time)
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const RTPVideoHeader& video_header)
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: marker_bit(rtp_packet.Marker()),
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payload_type(rtp_packet.PayloadType()),
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seq_num(rtp_packet.SequenceNumber()),
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timestamp(rtp_packet.Timestamp()),
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times_nacked(-1),
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video_header(video_header),
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packet_info(rtp_packet.Ssrc(),
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rtp_packet.Csrcs(),
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rtp_packet.Timestamp(),
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/*audio_level=*/absl::nullopt,
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rtp_packet.GetExtension<AbsoluteCaptureTimeExtension>(),
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receive_time) {}
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video_header(video_header) {}
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PacketBuffer::PacketBuffer(size_t start_buffer_size, size_t max_buffer_size)
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: max_size_(max_buffer_size),
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