Chromium Code Reviews| 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); |
|
stefan-webrtc
2016/12/20 13:05:01
This should now be renamed OnRtpOverheadChanged()
nisse-webrtc
2016/12/20 13:38:13
And the abstract-looking OverheadObserver interfac
stefan-webrtc
2016/12/20 14:18:23
True. Maybe it's clear enough since it's only poss
|
| } |