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 a55ccc37b9b189dd7bb20bec06a57916e3672d27..ae491209b7f22edf1ff3d063a66ded37dfeff200 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -92,8 +92,8 @@ RTPSender::RTPSender( |
transport_feedback_observer_(transport_feedback_observer), |
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. |
+ sending_media_(true), // Default to sending media. |
+ 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. |
- - video_->FecPacketOverhead() // FEC/ULP/RED overhead. |
- - (RtxStatus() ? kRtxHeaderSize : 0); // RTX 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(); |
@@ -974,7 +973,7 @@ 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. |
@@ -1253,31 +1252,13 @@ 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_) { |
@@ -1289,15 +1270,14 @@ 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); |
} |