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[fix] use ntp timestamp ms as video rtp timestamp

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This commit is contained in:
dijunkun
2025-02-28 15:52:21 +08:00
parent 962d856946
commit 3bb12e3f60
19 changed files with 55 additions and 32 deletions
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2
src/channel/audio_channel_send.cpp
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@@ -48,7 +48,7 @@ void AudioChannelSend::Destroy() {
int AudioChannelSend::SendAudio(char *data, size_t size) {
if (rtp_audio_sender_ && rtp_packetizer_) {
std::vector<std::shared_ptr<RtpPacket>> rtp_packets =
rtp_packetizer_->Build((uint8_t *)data, (uint32_t)size, true);
rtp_packetizer_->Build((uint8_t *)data, (uint32_t)size, 0, true);
rtp_audio_sender_->Enqueue(rtp_packets);
}

2
src/channel/data_channel_send.cpp
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@@ -48,7 +48,7 @@ void DataChannelSend::Destroy() {
int DataChannelSend::SendData(const char *data, size_t size) {
if (rtp_data_sender_ && rtp_packetizer_) {
std::vector<std::shared_ptr<RtpPacket>> rtp_packets =
rtp_packetizer_->Build((uint8_t *)data, (uint32_t)size, true);
rtp_packetizer_->Build((uint8_t *)data, (uint32_t)size, 0, true);
rtp_data_sender_->Enqueue(rtp_packets);
}

6
src/channel/rtp_channel/rtp_video_sender.cpp
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@@ -45,7 +45,7 @@ RtpVideoSender::~RtpVideoSender() {
void RtpVideoSender::Enqueue(
std::vector<std::shared_ptr<RtpPacket>>& rtp_packets,
int64_t capture_timestamp) {
int64_t capture_timestamp_ms) {
if (!rtp_statistics_) {
rtp_statistics_ = std::make_unique<RtpStatistics>();
rtp_statistics_->Start();
@@ -55,7 +55,7 @@ void RtpVideoSender::Enqueue(
std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet_to_send =
std::dynamic_pointer_cast<webrtc::RtpPacketToSend>(rtp_packet);
rtp_packet_to_send->set_capture_time(
webrtc::Timestamp::Millis(capture_timestamp));
webrtc::Timestamp::Millis(capture_timestamp_ms));
rtp_packet_to_send->set_transport_sequence_number(transport_seq_++);
rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kVideo);
rtp_packet_queue_.push(std::move(rtp_packet_to_send));
@@ -93,7 +93,7 @@ int RtpVideoSender::SendRtpPacket(
#ifdef SAVE_RTP_SENT_STREAM
fwrite((unsigned char*)rtp_packet_to_send->Payload(), 1,
rtp_packet_to_send->PayloadSize(), file_rtp_sent_);
x rtp_packet_to_send->PayloadSize(), file_rtp_sent_);
#endif
last_send_bytes_ += (uint32_t)rtp_packet_to_send->Size();

2
src/channel/rtp_channel/rtp_video_sender.h
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@@ -23,7 +23,7 @@ class RtpVideoSender : public ThreadBase {
public:
void Enqueue(std::vector<std::shared_ptr<RtpPacket>> &rtp_packets,
int64_t capture_timestamp);
int64_t capture_timestamp_ms);
void SetSendDataFunc(std::function<int(const char *, size_t)> data_send_func);
void SetOnSentPacketFunc(
std::function<void(const webrtc::RtpPacketToSend &)> on_sent_packet_func);

3
src/channel/video_channel_send.cpp
View File

@@ -62,7 +62,8 @@ int VideoChannelSend::SendVideo(
if (rtp_video_sender_ && rtp_packetizer_) {
std::vector<std::shared_ptr<RtpPacket>> rtp_packets =
rtp_packetizer_->Build((uint8_t*)encoded_frame->Buffer(),
(uint32_t)encoded_frame->Size(), true);
(uint32_t)encoded_frame->Size(),
encoded_frame->CaptureTimestamp(), true);
rtp_video_sender_->Enqueue(rtp_packets, encoded_frame->CaptureTimestamp());
}