mirror of
https://github.com/kunkundi/crossdesk.git
synced 2025-10-26 20:25:34 +08:00
[fix] fix padding packets building
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@@ -412,7 +412,6 @@ void PacingController::ProcessPackets() {
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PacedPacketInfo pacing_info;
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PacedPacketInfo pacing_info;
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DataSize recommended_probe_size = DataSize::Zero();
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DataSize recommended_probe_size = DataSize::Zero();
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bool is_probing = prober_.is_probing();
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bool is_probing = prober_.is_probing();
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LOG_WARN("is probing");
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if (is_probing) {
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if (is_probing) {
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// Probe timing is sensitive, and handled explicitly by BitrateProber, so
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// Probe timing is sensitive, and handled explicitly by BitrateProber, so
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// use actual send time rather than target.
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// use actual send time rather than target.
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@@ -474,7 +473,7 @@ void PacingController::ProcessPackets() {
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transport_overhead_per_packet_;
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transport_overhead_per_packet_;
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}
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}
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LOG_ERROR("Send packet_size {}", rtp_packet->Size());
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// LOG_ERROR("Send packet_size {}", rtp_packet->Size());
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packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
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packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
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for (auto& packet : packet_sender_->FetchFec()) {
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for (auto& packet : packet_sender_->FetchFec()) {
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EnqueuePacket(std::move(packet));
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EnqueuePacket(std::move(packet));
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@@ -289,7 +289,8 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildPadding(
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// Add padding bytes
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// Add padding bytes
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uint32_t padding_size = current_payload_size;
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uint32_t padding_size = current_payload_size;
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rtp_packet_frame_.insert(rtp_packet_frame_.end(), padding_size - 1, 0);
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rtp_packet_frame_.insert(rtp_packet_frame_.end(), padding_size, 0);
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rtp_packet_frame_.push_back(padding_size);
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if (use_rtp_packet_to_send) {
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if (use_rtp_packet_to_send) {
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std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet =
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std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet =
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@@ -183,7 +183,6 @@ int IceTransportController::SendVideo(const XVideoFrame* video_frame) {
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[this](std::shared_ptr<VideoFrameWrapper> encoded_frame) -> int {
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[this](std::shared_ptr<VideoFrameWrapper> encoded_frame) -> int {
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if (video_channel_send_) {
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if (video_channel_send_) {
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video_channel_send_->SendVideo(encoded_frame);
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video_channel_send_->SendVideo(encoded_frame);
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LOG_WARN("SendVideo rtp packets");
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}
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}
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return 0;
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return 0;
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@@ -72,17 +72,21 @@ void PacketSender::SetPacingRates(webrtc::DataRate pacing_rate,
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void PacketSender::EnqueuePackets(
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void PacketSender::EnqueuePackets(
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std::vector<std::unique_ptr<webrtc::RtpPacketToSend>> packets) {
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std::vector<std::unique_ptr<webrtc::RtpPacketToSend>> packets) {
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task_queue_.PostTask([this, packets = std::move(packets)]() mutable {
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webrtc::PacedPacketInfo cluster_info;
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for (auto &packet : packets) {
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for (auto &packet : packets) {
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size_t packet_size = packet->payload_size() + packet->padding_size();
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SendPacket(std::move(packet), cluster_info);
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if (include_overhead_) {
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}
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packet_size += packet->headers_size();
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// task_queue_.PostTask([this, packets = std::move(packets)]() mutable {
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}
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// for (auto &packet : packets) {
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packet_size_.Apply(1, packet_size);
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// size_t packet_size = packet->payload_size() + packet->padding_size();
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pacing_controller_.EnqueuePacket(std::move(packet));
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// if (include_overhead_) {
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}
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// packet_size += packet->headers_size();
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MaybeProcessPackets(webrtc::Timestamp::MinusInfinity());
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// }
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});
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// packet_size_.Apply(1, packet_size);
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// pacing_controller_.EnqueuePacket(std::move(packet));
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// }
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// MaybeProcessPackets(webrtc::Timestamp::MinusInfinity());
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// });
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}
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}
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void PacketSender::RemovePacketsForSsrc(uint32_t ssrc) {
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void PacketSender::RemovePacketsForSsrc(uint32_t ssrc) {
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