| 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..fcaf260d3313734ecf6da373dbad27ae2bb5418e 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
| @@ -97,16 +97,17 @@ class BitrateAggregator {
|
| uint32_t ssrc_;
|
| };
|
|
|
| -RTPSender::RTPSender(bool audio,
|
| - Clock* clock,
|
| - Transport* transport,
|
| - RtpAudioFeedback* audio_feedback,
|
| - PacedSender* paced_sender,
|
| - PacketRouter* packet_router,
|
| - TransportFeedbackObserver* transport_feedback_observer,
|
| - BitrateStatisticsObserver* bitrate_callback,
|
| - FrameCountObserver* frame_count_observer,
|
| - SendSideDelayObserver* send_side_delay_observer)
|
| +RTPSender::RTPSender(
|
| + bool audio,
|
| + Clock* clock,
|
| + Transport* transport,
|
| + RtpAudioFeedback* audio_feedback,
|
| + RtpPacketSender* paced_sender,
|
| + TransportSequenceNumberAllocator* sequence_number_allocator,
|
| + TransportFeedbackObserver* transport_feedback_observer,
|
| + BitrateStatisticsObserver* bitrate_callback,
|
| + FrameCountObserver* frame_count_observer,
|
| + SendSideDelayObserver* send_side_delay_observer)
|
| : clock_(clock),
|
| // TODO(holmer): Remove this conversion when we remove the use of
|
| // TickTime.
|
| @@ -118,7 +119,7 @@ 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),
|
| + transport_sequence_number_allocator_(sequence_number_allocator),
|
| transport_feedback_observer_(transport_feedback_observer),
|
| last_capture_time_ms_sent_(0),
|
| send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
|
| @@ -586,7 +587,8 @@ 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 +596,7 @@ size_t RTPSender::SendPadData(size_t bytes,
|
| size_t bytes_sent = 0;
|
| bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
|
| kRtpExtensionTransportSequenceNumber) &&
|
| - packet_router_;
|
| + transport_sequence_number_allocator_;
|
| for (; bytes > 0; bytes -= padding_bytes_in_packet) {
|
| if (bytes < padding_bytes_in_packet)
|
| bytes = padding_bytes_in_packet;
|
| @@ -711,7 +713,7 @@ 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 +919,8 @@ 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;
|
| + transport_sequence_number_allocator_ &&
|
| + !is_retransmit;
|
| if (using_transport_seq) {
|
| transport_seq =
|
| UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
|
| @@ -1000,10 +1003,12 @@ 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 +1620,7 @@ uint16_t RTPSender::UpdateTransportSequenceNumber(
|
| RTC_NOTREACHED();
|
| }
|
|
|
| - uint16_t seq = packet_router_->AllocateSequenceNumber();
|
| + uint16_t seq = transport_sequence_number_allocator_->AllocateSequenceNumber();
|
| BuildTransportSequenceNumberExtension(rtp_packet + offset, seq);
|
| return seq;
|
| }
|
|
|