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[fix] fix h264 rtp packet packetization and depacketization

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This commit is contained in:
dijunkun
2025-01-23 17:28:17 +08:00
parent cd349cd98d
commit 7b839ab773
50 changed files with 871 additions and 422 deletions
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46
src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
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@@ -6,7 +6,7 @@ RtpPacketizerH264::RtpPacketizerH264()
has_extension_(true),
csrc_count_(0),
marker_(false),
payload_type_(RtpPacket::PAYLOAD_TYPE::H264),
payload_type_(rtp::PAYLOAD_TYPE::H264),
sequence_number_(1),
timestamp_(0),
ssrc_(0),
@@ -28,13 +28,14 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
std::chrono::system_clock::now().time_since_epoch())
.count();
std::vector<RtpPacket> rtp_packets;
for (uint32_t index = 0; index < packet_num; index++) {
version_ = kRtpVersion;
has_padding_ = false;
has_extension_ = true;
csrc_count_ = 0;
marker_ = index == packet_num - 1 ? 1 : 0;
payload_type_ = RtpPacket::PAYLOAD_TYPE(payload_type_);
payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
sequence_number_++;
timestamp_ = timestamp_;
ssrc_ = ssrc_;
@@ -43,16 +44,16 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
csrcs_ = csrcs_;
}
RtpPacket::FU_INDICATOR fu_indicator;
rtp::FU_INDICATOR fu_indicator;
fu_indicator.forbidden_bit = 0;
fu_indicator.nal_reference_idc = 0;
fu_indicator.nal_unit_type = FU_A;
fu_indicator.nal_unit_type = rtp::NAL_UNIT_TYPE::FU_A;
RtpPacket::FU_HEADER fu_header;
rtp::FU_HEADER fu_header;
fu_header.start = index == 0 ? 1 : 0;
fu_header.end = index == packet_num - 1 ? 1 : 0;
fu_header.remain_bit = 0;
fu_header.nal_unit_type = FU_A;
fu_header.nal_unit_type = rtp::NAL_UNIT_TYPE::FU_A;
rtp_packet_frame_.clear();
rtp_packet_frame_.push_back((version_ << 6) | (has_padding_ << 5) |
@@ -77,20 +78,38 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
}
if (has_extension_) {
// 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
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | L=2 | Absolute Send Time |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// ID (4 bits): The identifier of the extension header field. In WebRTC,
// the ID for Absolute Send Time is typically 3.
// L (4 bits): The length of the extension data in bytes minus 1. For
// Absolute Send Time: the length is 2 (indicating 3 bytes of data).
// Absolute Send Time (24 bits): The absolute send time, with a unit of
// 1/65536 seconds (approximately 15.258 microseconds).
extension_profile_ = kOneByteExtensionProfileId;
extension_len_ = 5;
// 2 bytes for profile, 2 bytes for length, 3 bytes for abs_send_time, 1
// byte for id and sub extension length
extension_len_ = 8;
uint32_t abs_send_time =
std::chrono::duration_cast<std::chrono::microseconds>(
std::chrono::system_clock::now().time_since_epoch())
.count();
abs_send_time &= 0x00FFFFFF;
uint8_t sub_extension_id = 0;
uint8_t sub_extension_len = 2;
rtp_packet_frame_.push_back(extension_profile_ >> 8);
rtp_packet_frame_.push_back(extension_profile_ & 0xff);
rtp_packet_frame_.push_back((extension_len_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(extension_len_ & 0xFF);
rtp_packet_frame_.push_back(0x00);
rtp_packet_frame_.push_back(0x02);
rtp_packet_frame_.push_back(sub_extension_id << 4 | sub_extension_len);
rtp_packet_frame_.push_back((abs_send_time >> 16) & 0xFF);
rtp_packet_frame_.push_back((abs_send_time >> 8) & 0xFF);
rtp_packet_frame_.push_back(abs_send_time & 0xFF);
@@ -115,8 +134,9 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
RtpPacket rtp_packet;
rtp_packet.Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
packets.emplace_back(rtp_packet);
rtp_packets.emplace_back(rtp_packet);
}
return rtp_packets;
}
// bool BuildFec(uint8_t* payload, uint32_t payload_size) {
@@ -143,7 +163,7 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
// rtp_packet.SetHasPadding(false);
// rtp_packet.SetHasExtension(has_extension_);
// rtp_packet.SetMarker(index == num_of_source_packets - 1 ? 1 : 0);
// rtp_packet.SetPayloadType(RtpPacket::PAYLOAD_TYPE::H264_FEC_SOURCE);
// rtp_packet.SetPayloadType(rtp::PAYLOAD_TYPE::H264_FEC_SOURCE);
// rtp_packet.SetSequenceNumber(sequence_number_++);
// rtp_packet.SetTimestamp(timestamp_);
// rtp_packet.SetSsrc(ssrc_);
@@ -192,7 +212,7 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
// rtp_packet.SetHasPadding(false);
// rtp_packet.SetHasExtension(has_extension_);
// rtp_packet.SetMarker(index == num_of_total_packets - 1 ? 1 : 0);
// rtp_packet.SetPayloadType(RtpPacket::PAYLOAD_TYPE::H264_FEC_REPAIR);
// rtp_packet.SetPayloadType(rtp::PAYLOAD_TYPE::H264_FEC_REPAIR);
// rtp_packet.SetSequenceNumber(sequence_number_++);
// rtp_packet.SetTimestamp(timestamp_);
// rtp_packet.SetSsrc(ssrc_);
@@ -234,7 +254,7 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
// rtp_packet.SetHasExtension(has_extension_);
// rtp_packet.SetMarker(1);
// rtp_packet.SetPayloadType(RtpPacket::PAYLOAD_TYPE(payload_type_));
// rtp_packet.SetPayloadType(rtp::PAYLOAD_TYPE(payload_type_));
// rtp_packet.SetSequenceNumber(sequence_number_++);
// timestamp_ = std::chrono::duration_cast<std::chrono::microseconds>(