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[feat] add rtp packet history module

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This commit is contained in:
dijunkun
2025-02-14 17:30:12 +08:00
parent 7b4bba4166
commit 1ef7c536f1
27 changed files with 365 additions and 1161 deletions
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57
src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
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@@ -17,15 +17,6 @@ RtpPacketizerH264::RtpPacketizerH264(uint32_t ssrc)
RtpPacketizerH264::~RtpPacketizerH264() {}
std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
uint32_t payload_size) {
if (payload_size <= MAX_NALU_LEN) {
return BuildNalu(payload, payload_size);
} else {
return BuildFua(payload, payload_size);
}
}
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
@@ -67,9 +58,18 @@ void RtpPacketizerH264::AddAbsSendTimeExtension(
rtp_packet_frame.push_back(abs_send_time & 0xFF);
}
std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
uint32_t payload_size) {
std::vector<RtpPacket> rtp_packets;
std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerH264::Build(
uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
if (payload_size <= MAX_NALU_LEN) {
return BuildNalu(payload, payload_size, use_rtp_packet_to_send);
} else {
return BuildFua(payload, payload_size, use_rtp_packet_to_send);
}
}
std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerH264::BuildNalu(
uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
std::vector<std::shared_ptr<RtpPacket>> rtp_packets;
version_ = kRtpVersion;
has_padding_ = false;
@@ -123,16 +123,23 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
rtp_packet_frame_.insert(rtp_packet_frame_.end(), payload,
payload + payload_size);
RtpPacket rtp_packet;
rtp_packet.Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
rtp_packets.emplace_back(rtp_packet);
if (use_rtp_packet_to_send) {
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet =
std::make_unique<webrtc::RtpPacketToSend>();
rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
rtp_packets.emplace_back(std::move(rtp_packet));
} else {
std::shared_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();
rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
rtp_packets.emplace_back(std::move(rtp_packet));
}
return rtp_packets;
}
std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
uint32_t payload_size) {
std::vector<RtpPacket> rtp_packets;
std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerH264::BuildFua(
uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
std::vector<std::shared_ptr<RtpPacket>> rtp_packets;
uint32_t last_packet_size = payload_size % MAX_NALU_LEN;
uint32_t packet_num =
@@ -214,10 +221,16 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
payload + index * MAX_NALU_LEN + MAX_NALU_LEN);
}
RtpPacket rtp_packet;
rtp_packet.Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
rtp_packets.emplace_back(rtp_packet);
if (use_rtp_packet_to_send) {
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet =
std::make_unique<webrtc::RtpPacketToSend>();
rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
rtp_packets.emplace_back(std::move(rtp_packet));
} else {
std::shared_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();
rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
rtp_packets.emplace_back(std::move(rtp_packet));
}
}
return rtp_packets;