mirror of
https://github.com/kunkundi/crossdesk.git
synced 2025-10-26 12:15:34 +08:00
[feat] use packet sender to send all rtp packets
This commit is contained in:
@@ -8,8 +8,11 @@ AudioChannelSend::~AudioChannelSend() {}
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AudioChannelSend::AudioChannelSend(
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std::shared_ptr<IceAgent> ice_agent,
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std::shared_ptr<PacketSender> packet_sender,
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std::shared_ptr<IOStatistics> ice_io_statistics)
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: ice_agent_(ice_agent), ice_io_statistics_(ice_io_statistics) {}
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: packet_sender_(packet_sender),
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ice_agent_(ice_agent),
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ice_io_statistics_(ice_io_statistics) {}
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void AudioChannelSend::Initialize(rtp::PAYLOAD_TYPE payload_type) {
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rtp_audio_sender_ = std::make_unique<RtpAudioSender>(ice_io_statistics_);
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@@ -46,7 +49,7 @@ int AudioChannelSend::SendAudio(char *data, size_t size) {
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if (rtp_audio_sender_ && rtp_packetizer_) {
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std::vector<std::unique_ptr<RtpPacket>> rtp_packets =
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rtp_packetizer_->Build((uint8_t *)data, (uint32_t)size, 0, true);
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rtp_audio_sender_->Enqueue(std::move(rtp_packets));
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packet_sender_->EnqueueRtpPacket(rtp_packets, 0);
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}
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return 0;
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@@ -8,6 +8,7 @@
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#define _AUDIO_CHANNEL_SEND_H_
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#include "ice_agent.h"
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#include "packet_sender.h"
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#include "rtp_audio_sender.h"
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#include "rtp_packetizer.h"
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@@ -15,6 +16,7 @@ class AudioChannelSend {
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public:
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AudioChannelSend();
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AudioChannelSend(std::shared_ptr<IceAgent> ice_agent,
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std::shared_ptr<PacketSender> packet_sender,
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std::shared_ptr<IOStatistics> ice_io_statistics);
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~AudioChannelSend();
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@@ -34,6 +36,7 @@ class AudioChannelSend {
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void OnReceiverReport(const ReceiverReport& receiver_report) {}
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private:
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std::shared_ptr<PacketSender> packet_sender_ = nullptr;
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std::shared_ptr<IceAgent> ice_agent_ = nullptr;
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std::shared_ptr<IOStatistics> ice_io_statistics_ = nullptr;
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std::unique_ptr<RtpPacketizer> rtp_packetizer_ = nullptr;
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@@ -8,8 +8,11 @@ DataChannelSend::~DataChannelSend() {}
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DataChannelSend::DataChannelSend(
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std::shared_ptr<IceAgent> ice_agent,
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std::shared_ptr<PacketSender> packet_sender,
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std::shared_ptr<IOStatistics> ice_io_statistics)
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: ice_agent_(ice_agent), ice_io_statistics_(ice_io_statistics) {}
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: packet_sender_(packet_sender),
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ice_agent_(ice_agent),
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ice_io_statistics_(ice_io_statistics) {}
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void DataChannelSend::Initialize(rtp::PAYLOAD_TYPE payload_type) {
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rtp_data_sender_ = std::make_unique<RtpDataSender>(ice_io_statistics_);
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@@ -46,7 +49,7 @@ int DataChannelSend::SendData(const char *data, size_t size) {
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if (rtp_data_sender_ && rtp_packetizer_) {
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std::vector<std::unique_ptr<RtpPacket>> rtp_packets =
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rtp_packetizer_->Build((uint8_t *)data, (uint32_t)size, 0, true);
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rtp_data_sender_->Enqueue(std::move(rtp_packets));
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packet_sender_->EnqueueRtpPacket(rtp_packets, 0);
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}
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return 0;
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@@ -8,6 +8,7 @@
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#define _DATA_CHANNEL_SEND_H_
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#include "ice_agent.h"
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#include "packet_sender.h"
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#include "rtp_data_sender.h"
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#include "rtp_packetizer.h"
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@@ -15,6 +16,7 @@ class DataChannelSend {
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public:
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DataChannelSend();
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DataChannelSend(std::shared_ptr<IceAgent> ice_agent,
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std::shared_ptr<PacketSender> packet_sender,
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std::shared_ptr<IOStatistics> ice_io_statistics);
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~DataChannelSend();
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@@ -34,6 +36,7 @@ class DataChannelSend {
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void OnReceiverReport(const ReceiverReport& receiver_report) {}
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private:
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std::shared_ptr<PacketSender> packet_sender_ = nullptr;
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std::shared_ptr<IceAgent> ice_agent_ = nullptr;
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std::shared_ptr<IOStatistics> ice_io_statistics_ = nullptr;
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std::unique_ptr<RtpPacketizer> rtp_packetizer_ = nullptr;
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@@ -9,10 +9,12 @@ VideoChannelSend::~VideoChannelSend() {}
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VideoChannelSend::VideoChannelSend(
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std::shared_ptr<SystemClock> clock, std::shared_ptr<IceAgent> ice_agent,
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std::shared_ptr<PacketSender> packet_sender,
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std::shared_ptr<IOStatistics> ice_io_statistics,
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std::function<void(const webrtc::RtpPacketToSend& packet)>
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on_sent_packet_func)
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: ice_agent_(ice_agent),
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packet_sender_(packet_sender),
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ice_io_statistics_(ice_io_statistics),
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on_sent_packet_func_(on_sent_packet_func),
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clock_(clock){};
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@@ -76,7 +78,7 @@ int VideoChannelSend::SendVideo(
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rtp_packetizer_->Build((uint8_t*)encoded_frame->Buffer(),
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(uint32_t)encoded_frame->Size(),
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encoded_frame->CaptureTimestamp(), true);
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rtp_video_sender_->Enqueue(std::move(rtp_packets),
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packet_sender_->EnqueueRtpPacket(std::move(rtp_packets),
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encoded_frame->CaptureTimestamp());
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}
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@@ -13,6 +13,7 @@
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#include "congestion_control.h"
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#include "congestion_control_feedback.h"
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#include "ice_agent.h"
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#include "packet_sender.h"
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#include "rtp_packetizer.h"
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#include "rtp_video_sender.h"
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#include "transport_feedback_adapter.h"
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@@ -23,6 +24,7 @@ class VideoChannelSend {
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VideoChannelSend();
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VideoChannelSend(std::shared_ptr<SystemClock> clock,
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std::shared_ptr<IceAgent> ice_agent,
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std::shared_ptr<PacketSender> packet_sender,
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std::shared_ptr<IOStatistics> ice_io_statistics,
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std::function<void(const webrtc::RtpPacketToSend& packet)>
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on_sent_packet_func_);
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@@ -71,6 +73,7 @@ class VideoChannelSend {
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void UpdateCongestedState();
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private:
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std::shared_ptr<PacketSender> packet_sender_ = nullptr;
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std::shared_ptr<IceAgent> ice_agent_ = nullptr;
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std::shared_ptr<IOStatistics> ice_io_statistics_ = nullptr;
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std::unique_ptr<RtpPacketizer> rtp_packetizer_ = nullptr;
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@@ -53,7 +53,7 @@ void IceTransportController::Create(
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CreateAudioCodec();
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controller_ = std::make_unique<CongestionControl>();
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packet_sender_ = std::make_unique<PacketSenderImp>(ice_agent, webrtc_clock_);
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packet_sender_ = std::make_shared<PacketSenderImp>(ice_agent, webrtc_clock_);
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packet_sender_->SetPacingRates(DataRate::BitsPerSec(300000),
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DataRate::Zero());
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packet_sender_->SetOnSentPacketFunc(
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@@ -67,7 +67,7 @@ void IceTransportController::Create(
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resolution_adapter_ = std::make_unique<ResolutionAdapter>();
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video_channel_send_ = std::make_unique<VideoChannelSend>(
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clock_, ice_agent, ice_io_statistics,
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clock_, ice_agent, packet_sender_, ice_io_statistics,
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[this](const webrtc::RtpPacketToSend& packet) {
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OnSentRtpPacket(packet);
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});
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@@ -78,10 +78,10 @@ void IceTransportController::Create(
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return video_channel_send_->GeneratePadding(size, capture_timestamp_ms);
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});
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audio_channel_send_ =
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std::make_unique<AudioChannelSend>(ice_agent, ice_io_statistics);
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data_channel_send_ =
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std::make_unique<DataChannelSend>(ice_agent, ice_io_statistics);
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audio_channel_send_ = std::make_unique<AudioChannelSend>(
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ice_agent, packet_sender_, ice_io_statistics);
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data_channel_send_ = std::make_unique<DataChannelSend>(
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ice_agent, packet_sender_, ice_io_statistics);
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video_channel_send_->Initialize(video_codec_payload_type);
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audio_channel_send_->Initialize(rtp::PAYLOAD_TYPE::OPUS);
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@@ -21,6 +21,7 @@
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#include "data_channel_receive.h"
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#include "data_channel_send.h"
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#include "ice_agent.h"
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#include "packet_sender.h"
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#include "packet_sender_imp.h"
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#include "resolution_adapter.h"
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#include "transport_feedback_adapter.h"
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@@ -107,7 +108,7 @@ class IceTransportController
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std::shared_ptr<IceAgent> ice_agent_ = nullptr;
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std::shared_ptr<IOStatistics> ice_io_statistics_ = nullptr;
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std::unique_ptr<RtpPacketizer> rtp_packetizer_ = nullptr;
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std::unique_ptr<PacketSenderImp> packet_sender_ = nullptr;
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std::shared_ptr<PacketSenderImp> packet_sender_ = nullptr;
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std::string remote_user_id_;
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void *user_data_ = nullptr;
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@@ -18,8 +18,9 @@ class PacketSender {
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virtual ~PacketSender() {}
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virtual int Send() = 0;
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virtual int InsertRtpPacket(
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std::vector<std::unique_ptr<RtpPacket>> &rtp_packets) = 0;
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virtual int EnqueueRtpPacket(
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std::vector<std::unique_ptr<RtpPacket>> &rtp_packets,
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int64_t capture_timestamp_ms) = 0;
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};
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#endif
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@@ -251,3 +251,46 @@ void PacketSenderImp::UpdateStats() {
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PacketSenderImp::Stats PacketSenderImp::GetStats() const {
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return current_stats_;
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}
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/*----------------------------------------------------------------------------*/
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int PacketSenderImp::EnqueueRtpPacket(
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std::vector<std::unique_ptr<RtpPacket>> &rtp_packets,
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int64_t capture_timestamp_ms) {
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std::vector<std::unique_ptr<webrtc::RtpPacketToSend>> to_send_rtp_packets;
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for (auto &rtp_packet : rtp_packets) {
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std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send(
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static_cast<webrtc::RtpPacketToSend *>(rtp_packet.release()));
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rtp_packet_to_send->set_capture_time(
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webrtc::Timestamp::Millis(capture_timestamp_ms));
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rtp_packet_to_send->set_transport_sequence_number(transport_seq_++);
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switch (rtp_packet_to_send->PayloadType()) {
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case rtp::PAYLOAD_TYPE::H264:
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rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kVideo);
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break;
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case rtp::PAYLOAD_TYPE::AV1:
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rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kVideo);
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break;
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case rtp::PAYLOAD_TYPE::H264_FEC_SOURCE:
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rtp_packet_to_send->set_packet_type(
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webrtc::RtpPacketMediaType::kForwardErrorCorrection);
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break;
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case rtp::PAYLOAD_TYPE::H264_FEC_REPAIR:
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rtp_packet_to_send->set_packet_type(
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webrtc::RtpPacketMediaType::kForwardErrorCorrection);
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break;
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case rtp::PAYLOAD_TYPE::OPUS:
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rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kAudio);
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break;
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default:
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rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kVideo);
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break;
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}
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to_send_rtp_packets.push_back(std::move(rtp_packet_to_send));
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}
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EnqueuePackets(std::move(to_send_rtp_packets));
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return 0;
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}
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@@ -37,9 +37,8 @@ class PacketSenderImp : public PacketSender,
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public:
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int Send() { return 0; }
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int InsertRtpPacket(std::vector<std::unique_ptr<RtpPacket>>& rtp_packets) {
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return 0;
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}
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int EnqueueRtpPacket(std::vector<std::unique_ptr<RtpPacket>>& rtp_packets,
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int64_t capture_timestamp_ms);
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void SetOnSentPacketFunc(
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std::function<void(const webrtc::RtpPacketToSend&)> on_sent_packet_func) {
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