[fix] fix h264 rtp packetization error

This commit is contained in:
dijunkun
2025-02-05 17:28:57 +08:00
parent 2d5749f93a
commit 794e33c325
11 changed files with 75 additions and 85 deletions

View File

@@ -4,7 +4,7 @@
#include "log.h"
#include "rtcp_sender.h"
#define SAVE_RTP_RECV_STREAM 1
// #define SAVE_RTP_RECV_STREAM
#define NV12_BUFFER_SIZE (1280 * 720 * 3 / 2)
#define RTCP_RR_INTERVAL 1000
@@ -73,13 +73,10 @@ void RtpVideoReceiver::InsertRtpPacket(RtpPacket& rtp_packet) {
rtp_statistics_->Start();
}
// #ifdef SAVE_RTP_RECV_STREAM
// // fwrite((unsigned char*)rtp_packet.Buffer().data(), 1,
// rtp_packet.Size(),
// // file_rtp_recv_);
// fwrite((unsigned char*)rtp_packet.Payload(), 1, rtp_packet.PayloadSize(),
// file_rtp_recv_);
// #endif
#ifdef SAVE_RTP_RECV_STREAM
fwrite((unsigned char*)rtp_packet.Payload(), 1, rtp_packet.PayloadSize(),
file_rtp_recv_);
#endif
webrtc::RtpPacketReceived rtp_packet_received;
rtp_packet_received.Build(rtp_packet.Buffer().data(), rtp_packet.Size());
@@ -312,6 +309,7 @@ bool RtpVideoReceiver::CheckIsH264FrameCompleted(
}
size_t complete_frame_size = 0;
int frame_fragment_count = 0;
for (uint16_t start = it->first;
start <= rtp_packet_h264.SequenceNumber(); start++) {
memcpy(nv12_data_ + complete_frame_size,
@@ -321,6 +319,7 @@ bool RtpVideoReceiver::CheckIsH264FrameCompleted(
complete_frame_size +=
incomplete_h264_frame_list_[start].PayloadSize();
incomplete_h264_frame_list_.erase(start);
frame_fragment_count++;
}
compelete_video_frame_queue_.push(
VideoFrame(nv12_data_, complete_frame_size));
@@ -462,10 +461,11 @@ bool RtpVideoReceiver::Process() {
// last_complete_frame_ts_ = now_complete_frame_ts;
on_receive_complete_frame_(video_frame);
#ifdef SAVE_RTP_RECV_STREAM
fwrite((unsigned char*)video_frame.Buffer(), 1, video_frame.Size(),
file_rtp_recv_);
#endif
// #ifdef SAVE_RTP_RECV_STREAM
// fwrite((unsigned char*)video_frame.Buffer(), 1,
// video_frame.Size(),
// file_rtp_recv_);
// #endif
}
}

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@@ -4,7 +4,7 @@
#include "log.h"
#define SAVE_RTP_SENT_STREAM 1
// #define SAVE_RTP_SENT_STREAM
#define RTCP_SR_INTERVAL 1000
@@ -63,8 +63,6 @@ int RtpVideoSender::SendRtpPacket(RtpPacket& rtp_packet) {
}
#ifdef SAVE_RTP_SENT_STREAM
// fwrite((unsigned char*)rtp_packet.Buffer().data(), 1, rtp_packet.Size(),
// file_rtp_sent_);
fwrite((unsigned char*)rtp_packet.Payload(), 1, rtp_packet.PayloadSize(),
file_rtp_sent_);
#endif

View File

@@ -4,7 +4,7 @@
#include "nvcodec_api.h"
// #define SAVE_DECODED_NV12_STREAM
#define SAVE_RECEIVED_H264_STREAM
// #define SAVE_RECEIVED_H264_STREAM
NvidiaVideoDecoder::NvidiaVideoDecoder() {}
NvidiaVideoDecoder::~NvidiaVideoDecoder() {

View File

@@ -7,7 +7,7 @@
#include "nvcodec_common.h"
// #define SAVE_RECEIVED_NV12_STREAM
#define SAVE_ENCODED_H264_STREAM
// #define SAVE_ENCODED_H264_STREAM
NvidiaVideoEncoder::NvidiaVideoEncoder() {}
NvidiaVideoEncoder::~NvidiaVideoEncoder() {

View File

@@ -287,7 +287,7 @@ TransportFeedbackAdapter::ProcessCongestionControlFeedback(
if (failed_lookups > 0) {
LOG_WARN(
"Failed to lookup send time for {} packet {}. Packets reordered or "
"Failed to lookup send time for {} packet{}. Packets reordered or "
"send time history too small?",
failed_lookups, (failed_lookups > 1 ? "s" : ""));
}

View File

@@ -2,13 +2,7 @@
#include <string>
static FILE *file_1_ = nullptr;
static FILE *file_2_ = nullptr;
RtpPacket::RtpPacket() {
if (file_1_ == nullptr) file_1_ = fopen("file_1_.h264", "w+b");
if (file_2_ == nullptr) file_2_ = fopen("file_2_.h264", "w+b");
}
RtpPacket::RtpPacket() {}
RtpPacket::RtpPacket(size_t size) : buffer_(size) {}
@@ -20,25 +14,14 @@ RtpPacket &RtpPacket::operator=(const RtpPacket &rtp_packet) = default;
RtpPacket &RtpPacket::operator=(RtpPacket &&rtp_packet) = default;
RtpPacket::~RtpPacket() {
// if (file_1_ != nullptr) {
// fclose(file_1_);
// file_1_ = nullptr;
// }
// if (file_2_ != nullptr) {
// fclose(file_2_);
// file_2_ = nullptr;
// }
}
RtpPacket::~RtpPacket() {}
bool RtpPacket::Build(const uint8_t *buffer, uint32_t size) {
fwrite((unsigned char *)buffer, 1, size, file_1_);
if (!Parse(buffer, size)) {
LOG_WARN("RtpPacket::Build: parse failed");
return false;
}
buffer_.SetData(buffer, size);
fwrite((unsigned char *)Payload(), 1, PayloadSize(), file_2_);
size_ = size;
return true;
}

View File

@@ -263,6 +263,10 @@ class RtpPacket {
CopyOnWriteBuffer Buffer() const { return buffer_; }
size_t Size() const { return size_; }
// Header
const uint8_t *Header() { return Buffer().data(); };
size_t HeaderSize() { return payload_offset_; }
// For webrtc module use
size_t headers_size() const { return payload_offset_; }
size_t payload_size() const { return payload_size_; }

View File

@@ -17,9 +17,6 @@ bool RtpPacketH264::GetFrameHeaderInfo() {
if (rtp::NAL_UNIT_TYPE::NALU == fu_indicator_.nal_unit_type) {
add_offset_to_payload(1);
LOG_ERROR("2 [{} {} {}]", (int)fu_indicator_.forbidden_bit,
(int)fu_indicator_.nal_reference_idc,
(int)fu_indicator_.nal_unit_type);
} else if (rtp::NAL_UNIT_TYPE::FU_A == fu_indicator_.nal_unit_type) {
fu_header_.start = (frame_buffer[1] >> 7) & 0x01;
fu_header_.end = (frame_buffer[1] >> 6) & 0x01;

View File

@@ -7,7 +7,7 @@ RtpPacketizerGeneric::RtpPacketizerGeneric()
csrc_count_(0),
marker_(false),
payload_type_(rtp::PAYLOAD_TYPE::DATA),
sequence_number_(1),
sequence_number_(0),
timestamp_(0),
ssrc_(0),
profile_(0),
@@ -67,28 +67,7 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
}
if (has_extension_) {
extension_profile_ = kOneByteExtensionProfileId;
extension_len_ = 5; // 2 bytes for profile, 2 bytes for length, 3 bytes
// for abs_send_time
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(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);
AddAbsSendTimeExtension(rtp_packet_frame_);
}
if (index == packet_num - 1 && last_packet_size > 0) {
@@ -107,3 +86,32 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
return rtp_packets;
}
void RtpPacketizerGeneric::AddAbsSendTimeExtension(
std::vector<uint8_t>& rtp_packet_frame) {
uint16_t extension_profile = 0xBEDE; // One-byte header extension
uint8_t sub_extension_id = 3; // ID for Absolute Send Time
uint8_t sub_extension_length =
2; // Length of the extension data in bytes minus 1
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; // Absolute Send Time is 24 bits
// Add extension profile
rtp_packet_frame.push_back((extension_profile >> 8) & 0xFF);
rtp_packet_frame.push_back(extension_profile & 0xFF);
// Add extension length (in 32-bit words, minus one)
rtp_packet_frame.push_back(
0x00); // Placeholder for length, will be updated later
rtp_packet_frame.push_back(0x01); // One 32-bit word
// Add Absolute Send Time extension
rtp_packet_frame.push_back((sub_extension_id << 4) | sub_extension_length);
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);
}

View File

@@ -18,6 +18,9 @@ class RtpPacketizerGeneric : public RtpPacketizer {
std::vector<RtpPacket> Build(uint8_t* payload,
uint32_t payload_size) override;
private:
void AddAbsSendTimeExtension(std::vector<uint8_t>& rtp_packet_frame);
private:
uint8_t version_;
bool has_padding_;

View File

@@ -7,7 +7,7 @@ RtpPacketizerH264::RtpPacketizerH264()
csrc_count_(0),
marker_(false),
payload_type_(rtp::PAYLOAD_TYPE::H264),
sequence_number_(1),
sequence_number_(0),
timestamp_(0),
ssrc_(0),
profile_(0),
@@ -23,7 +23,6 @@ std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
return BuildNalu(payload, payload_size);
} else {
return BuildFua(payload, payload_size);
// return std::vector<RtpPacket>();
}
}
@@ -70,7 +69,6 @@ void RtpPacketizerH264::AddAbsSendTimeExtension(
std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
uint32_t payload_size) {
LOG_ERROR("payload_size_ = {}", payload_size);
std::vector<RtpPacket> rtp_packets;
version_ = kRtpVersion;
@@ -120,20 +118,15 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
AddAbsSendTimeExtension(rtp_packet_frame_);
}
rtp_packet_frame_.push_back(fu_indicator.forbidden_bit << 7 |
fu_indicator.nal_reference_idc << 6 |
rtp_packet_frame_.push_back((fu_indicator.forbidden_bit << 7) |
(fu_indicator.nal_reference_idc << 5) |
fu_indicator.nal_unit_type);
LOG_ERROR("1 [{} {} {}]", (int)fu_indicator.forbidden_bit,
(int)fu_indicator.nal_reference_idc,
(int)fu_indicator.nal_unit_type);
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);
return rtp_packets;
@@ -157,7 +150,7 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
has_padding_ = false;
has_extension_ = true;
csrc_count_ = 0;
marker_ = index == packet_num - 1 ? 1 : 0;
marker_ = (index == (packet_num - 1)) ? 1 : 0;
payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
sequence_number_++;
timestamp_ = timestamp_;
@@ -193,11 +186,12 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
rtp_packet_frame_.push_back((ssrc_ >> 8) & 0xFF);
rtp_packet_frame_.push_back(ssrc_ & 0xFF);
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);
for (uint32_t csrc_index = 0; csrc_index < csrc_count_ && !csrcs_.empty();
csrc_index++) {
rtp_packet_frame_.push_back((csrcs_[csrc_index] >> 24) & 0xFF);
rtp_packet_frame_.push_back((csrcs_[csrc_index] >> 16) & 0xFF);
rtp_packet_frame_.push_back((csrcs_[csrc_index] >> 8) & 0xFF);
rtp_packet_frame_.push_back(csrcs_[csrc_index] & 0xFF);
}
if (has_extension_) {
@@ -205,19 +199,21 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
}
rtp_packet_frame_.push_back(fu_indicator.forbidden_bit << 7 |
fu_indicator.nal_reference_idc << 6 |
fu_indicator.nal_reference_idc << 5 |
fu_indicator.nal_unit_type);
rtp_packet_frame_.push_back(fu_header.start << 7 | fu_header.end << 6 |
fu_header.remain_bit << 1 |
fu_header.remain_bit << 5 |
fu_header.nal_unit_type);
if (index == packet_num - 1 && last_packet_size > 0) {
rtp_packet_frame_.insert(rtp_packet_frame_.end(), payload,
payload + last_packet_size);
rtp_packet_frame_.insert(
rtp_packet_frame_.end(), payload + index * MAX_NALU_LEN,
payload + index * MAX_NALU_LEN + last_packet_size);
} else {
rtp_packet_frame_.insert(rtp_packet_frame_.end(), payload,
payload + MAX_NALU_LEN);
rtp_packet_frame_.insert(rtp_packet_frame_.end(),
payload + index * MAX_NALU_LEN,
payload + index * MAX_NALU_LEN + MAX_NALU_LEN);
}
RtpPacket rtp_packet;
@@ -225,6 +221,7 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
rtp_packets.emplace_back(rtp_packet);
}
return rtp_packets;
}