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[feat] add task queue module

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This commit is contained in:
dijunkun
2025-03-14 18:29:07 +08:00
parent d2b45b91e7
commit bd65d87137
8 changed files with 270 additions and 63 deletions
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96
src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
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@@ -246,55 +246,63 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildPadding(
bool use_rtp_packet_to_send) {
std::vector<std::unique_ptr<RtpPacket>> rtp_packets;
version_ = kRtpVersion;
has_padding_ = true;
has_extension_ = true;
csrc_count_ = 0;
marker_ = 0;
uint8_t payload_type = rtp::PAYLOAD_TYPE(payload_type_ - 1);
sequence_number_++;
timestamp_ = kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_ms);
uint32_t remaining_size = payload_size;
while (remaining_size > 0) {
uint32_t current_payload_size =
std::min<uint32_t>(remaining_size, MAX_NALU_LEN);
rtp_packet_frame_.clear();
rtp_packet_frame_.push_back((version_ << 6) | (has_padding_ << 5) |
(has_extension_ << 4) | csrc_count_);
rtp_packet_frame_.push_back((marker_ << 7) | payload_type);
rtp_packet_frame_.push_back((sequence_number_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(sequence_number_ & 0xFF);
rtp_packet_frame_.push_back((timestamp_ >> 24) & 0xFF);
rtp_packet_frame_.push_back((timestamp_ >> 16) & 0xFF);
rtp_packet_frame_.push_back((timestamp_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(timestamp_ & 0xFF);
rtp_packet_frame_.push_back((ssrc_ >> 24) & 0xFF);
rtp_packet_frame_.push_back((ssrc_ >> 16) & 0xFF);
rtp_packet_frame_.push_back((ssrc_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(ssrc_ & 0xFF);
version_ = kRtpVersion;
has_padding_ = true;
has_extension_ = true;
csrc_count_ = 0;
marker_ = 0;
uint8_t payload_type = rtp::PAYLOAD_TYPE(payload_type_ - 1);
sequence_number_++;
timestamp_ =
kMsToRtpTimestamp * static_cast<uint32_t>(capture_timestamp_ms);
for (uint32_t index = 0; index < csrc_count_ && !csrcs_.empty(); index++) {
rtp_packet_frame_.push_back((csrcs_[index] >> 24) & 0xFF);
rtp_packet_frame_.push_back((csrcs_[index] >> 16) & 0xFF);
rtp_packet_frame_.push_back((csrcs_[index] >> 8) & 0xFF);
rtp_packet_frame_.push_back(csrcs_[index] & 0xFF);
}
rtp_packet_frame_.clear();
rtp_packet_frame_.push_back((version_ << 6) | (has_padding_ << 5) |
(has_extension_ << 4) | csrc_count_);
rtp_packet_frame_.push_back((marker_ << 7) | payload_type);
rtp_packet_frame_.push_back((sequence_number_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(sequence_number_ & 0xFF);
rtp_packet_frame_.push_back((timestamp_ >> 24) & 0xFF);
rtp_packet_frame_.push_back((timestamp_ >> 16) & 0xFF);
rtp_packet_frame_.push_back((timestamp_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(timestamp_ & 0xFF);
rtp_packet_frame_.push_back((ssrc_ >> 24) & 0xFF);
rtp_packet_frame_.push_back((ssrc_ >> 16) & 0xFF);
rtp_packet_frame_.push_back((ssrc_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(ssrc_ & 0xFF);
if (has_extension_) {
AddAbsSendTimeExtension(rtp_packet_frame_);
}
for (uint32_t index = 0; index < csrc_count_ && !csrcs_.empty(); index++) {
rtp_packet_frame_.push_back((csrcs_[index] >> 24) & 0xFF);
rtp_packet_frame_.push_back((csrcs_[index] >> 16) & 0xFF);
rtp_packet_frame_.push_back((csrcs_[index] >> 8) & 0xFF);
rtp_packet_frame_.push_back(csrcs_[index] & 0xFF);
}
// Add padding bytes
uint32_t padding_size = payload_size;
rtp_packet_frame_.insert(rtp_packet_frame_.end(), padding_size - 1, 0);
rtp_packet_frame_.push_back(padding_size);
if (has_extension_) {
AddAbsSendTimeExtension(rtp_packet_frame_);
}
if (use_rtp_packet_to_send) {
std::unique_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::unique_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));
// Add padding bytes
uint32_t padding_size = current_payload_size;
rtp_packet_frame_.insert(rtp_packet_frame_.end(), padding_size - 1, 0);
if (use_rtp_packet_to_send) {
std::unique_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::unique_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));
}
remaining_size -= current_payload_size;
}
return rtp_packets;