| Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| index 4a285ca2198e04ec4de86e742c34fa118ededd42..89bedc89ff987a81db655e3c402624b1bbd015fb 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -93,7 +93,7 @@ RTPSender::RTPSender(
|
| last_capture_time_ms_sent_(0),
|
| transport_(transport),
|
| sending_media_(true), // Default to sending media.
|
| - max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
|
| + max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
|
| payload_type_(-1),
|
| payload_type_map_(),
|
| rtp_header_extension_map_(),
|
| @@ -121,7 +121,6 @@ RTPSender::RTPSender(
|
| last_packet_marker_bit_(false),
|
| csrcs_(),
|
| rtx_(kRtxOff),
|
| - transport_overhead_bytes_per_packet_(0),
|
| rtp_overhead_bytes_per_packet_(0),
|
| retransmission_rate_limiter_(retransmission_rate_limiter),
|
| overhead_observer_(overhead_observer) {
|
| @@ -297,26 +296,26 @@ int8_t RTPSender::SendPayloadType() const {
|
| return payload_type_;
|
| }
|
|
|
| -void RTPSender::SetMaxPayloadLength(size_t max_payload_length) {
|
| +void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
|
| // Sanity check.
|
| - RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
|
| - << "Invalid max payload length: " << max_payload_length;
|
| + RTC_DCHECK(max_packet_size >= 100 && max_packet_size <= IP_PACKET_SIZE)
|
| + << "Invalid max payload length: " << max_packet_size;
|
| rtc::CritScope lock(&send_critsect_);
|
| - max_payload_length_ = max_payload_length;
|
| + max_packet_size_ = max_packet_size;
|
| }
|
|
|
| -size_t RTPSender::MaxDataPayloadLength() const {
|
| +size_t RTPSender::MaxPayloadSize() const {
|
| if (audio_configured_) {
|
| - return max_payload_length_ - RtpHeaderLength();
|
| + return max_packet_size_ - RtpHeaderLength();
|
| } else {
|
| - return max_payload_length_ - RtpHeaderLength() // RTP overhead.
|
| + return max_packet_size_ - RtpHeaderLength() // RTP overhead.
|
| - video_->FecPacketOverhead() // FEC/ULP/RED overhead.
|
| - (RtxStatus() ? kRtxHeaderSize : 0); // RTX overhead.
|
| }
|
| }
|
|
|
| -size_t RTPSender::MaxPayloadLength() const {
|
| - return max_payload_length_;
|
| +size_t RTPSender::MaxRtpPacketSize() const {
|
| + return max_packet_size_;
|
| }
|
|
|
| void RTPSender::SetRtxStatus(int mode) {
|
| @@ -483,7 +482,7 @@ size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
|
| // RtpPacketSender, which will make sure we don't send too much padding even
|
| // if a single packet is larger than requested.
|
| size_t padding_bytes_in_packet =
|
| - std::min(MaxDataPayloadLength(), kMaxPaddingLength);
|
| + std::min(MaxPayloadSize(), kMaxPaddingLength);
|
| size_t bytes_sent = 0;
|
| while (bytes_sent < bytes) {
|
| int64_t now_ms = clock_->TimeInMilliseconds();
|
| @@ -971,7 +970,7 @@ void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
|
| std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
|
| rtc::CritScope lock(&send_critsect_);
|
| std::unique_ptr<RtpPacketToSend> packet(
|
| - new RtpPacketToSend(&rtp_header_extension_map_, max_payload_length_));
|
| + new RtpPacketToSend(&rtp_header_extension_map_, max_packet_size_));
|
| packet->SetSsrc(ssrc_);
|
| packet->SetCsrcs(csrcs_);
|
| // Reserve extensions, if registered, RtpSender set in SendToNetwork.
|
| @@ -1250,31 +1249,13 @@ RtpState RTPSender::GetRtxRtpState() const {
|
| return state;
|
| }
|
|
|
| -void RTPSender::SetTransportOverhead(int transport_overhead) {
|
| - if (!overhead_observer_)
|
| - return;
|
| - size_t overhead_bytes_per_packet = 0;
|
| - {
|
| - rtc::CritScope lock(&send_critsect_);
|
| - if (transport_overhead_bytes_per_packet_ ==
|
| - static_cast<size_t>(transport_overhead)) {
|
| - return;
|
| - }
|
| - transport_overhead_bytes_per_packet_ = transport_overhead;
|
| - overhead_bytes_per_packet =
|
| - rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
|
| - }
|
| - overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
|
| -}
|
| -
|
| void RTPSender::AddPacketToTransportFeedback(uint16_t packet_id,
|
| const RtpPacketToSend& packet,
|
| int probe_cluster_id) {
|
| size_t packet_size = packet.payload_size() + packet.padding_size();
|
| if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") ==
|
| "Enabled") {
|
| - rtc::CritScope lock(&send_critsect_);
|
| - packet_size = packet.size() + transport_overhead_bytes_per_packet_;
|
| + packet_size = packet.size();
|
| }
|
|
|
| if (transport_feedback_observer_) {
|
| @@ -1286,15 +1267,14 @@ void RTPSender::AddPacketToTransportFeedback(uint16_t packet_id,
|
| void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
|
| if (!overhead_observer_)
|
| return;
|
| - size_t overhead_bytes_per_packet = 0;
|
| + size_t overhead_bytes_per_packet;
|
| {
|
| rtc::CritScope lock(&send_critsect_);
|
| if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
|
| return;
|
| }
|
| rtp_overhead_bytes_per_packet_ = packet.headers_size();
|
| - overhead_bytes_per_packet =
|
| - rtp_overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_;
|
| + overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
|
| }
|
| overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
|
| }
|
|
|