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 44ac96541325af59a565b9d76281ece1b4c4ab8f..72e6f7a8818327bfe6a43b21fc5a6c33a2ff1ef9 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc |
@@ -101,8 +101,8 @@ RTPSender::RTPSender(bool audio, |
Clock* clock, |
Transport* transport, |
RtpAudioFeedback* audio_feedback, |
- PacedSender* paced_sender, |
- PacketRouter* packet_router, |
+ RtpPacketSender* paced_sender, |
+ SequenceNumberAllocator* sequence_number_allocator, |
TransportFeedbackObserver* transport_feedback_observer, |
BitrateStatisticsObserver* bitrate_callback, |
FrameCountObserver* frame_count_observer, |
@@ -118,7 +118,7 @@ RTPSender::RTPSender(bool audio, |
audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr), |
video_(audio ? nullptr : new RTPSenderVideo(clock, this)), |
paced_sender_(paced_sender), |
- packet_router_(packet_router), |
+ sequence_number_allocator_(sequence_number_allocator), |
transport_feedback_observer_(transport_feedback_observer), |
last_capture_time_ms_sent_(0), |
send_critsect_(CriticalSectionWrapper::CreateCriticalSection()), |
@@ -586,7 +586,8 @@ size_t RTPSender::SendPadData(size_t bytes, |
bool timestamp_provided, |
uint32_t timestamp, |
int64_t capture_time_ms) { |
- // Always send full padding packets. This is accounted for by the PacedSender, |
+ // Always send full padding packets. This is accounted for by the |
+ // 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 = |
@@ -594,7 +595,7 @@ size_t RTPSender::SendPadData(size_t bytes, |
size_t bytes_sent = 0; |
bool using_transport_seq = rtp_header_extension_map_.IsRegistered( |
kRtpExtensionTransportSequenceNumber) && |
- packet_router_; |
+ sequence_number_allocator_; |
for (; bytes > 0; bytes -= padding_bytes_in_packet) { |
if (bytes < padding_bytes_in_packet) |
bytes = padding_bytes_in_packet; |
@@ -711,7 +712,7 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) { |
// TickTime. |
int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_; |
if (!paced_sender_->SendPacket( |
- PacedSender::kHighPriority, header.ssrc, header.sequenceNumber, |
+ RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber, |
corrected_capture_tims_ms, length - header.headerLength, true)) { |
// We can't send the packet right now. |
// We will be called when it is time. |
@@ -917,7 +918,7 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, |
// TODO(sprang): Potentially too much overhead in IsRegistered()? |
bool using_transport_seq = rtp_header_extension_map_.IsRegistered( |
kRtpExtensionTransportSequenceNumber) && |
- packet_router_ && !is_retransmit; |
+ sequence_number_allocator_ && !is_retransmit; |
if (using_transport_seq) { |
transport_seq = |
UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header); |
@@ -1000,10 +1001,12 @@ size_t RTPSender::TimeToSendPadding(size_t bytes) { |
} |
// TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again. |
-int32_t RTPSender::SendToNetwork( |
- uint8_t *buffer, size_t payload_length, size_t rtp_header_length, |
- int64_t capture_time_ms, StorageType storage, |
- PacedSender::Priority priority) { |
+int32_t RTPSender::SendToNetwork(uint8_t* buffer, |
+ size_t payload_length, |
+ size_t rtp_header_length, |
+ int64_t capture_time_ms, |
+ StorageType storage, |
+ RtpPacketSender::Priority priority) { |
RtpUtility::RtpHeaderParser rtp_parser(buffer, |
payload_length + rtp_header_length); |
RTPHeader rtp_header; |
@@ -1615,7 +1618,7 @@ uint16_t RTPSender::UpdateTransportSequenceNumber( |
RTC_NOTREACHED(); |
} |
- uint16_t seq = packet_router_->AllocateSequenceNumber(); |
+ uint16_t seq = sequence_number_allocator_->AllocateSequenceNumber(); |
BuildTransportSequenceNumberExtension(rtp_packet + offset, seq); |
return seq; |
} |