Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
index 6d24ee1376bb02f2eef6f190ee0c60349b515c91..f1334174287b9539bd6b0de0a536ab1a6524a0c8 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -17,8 +17,6 @@ |
#include "webrtc/base/thread_annotations.h" |
#include "webrtc/common_types.h" |
-#include "webrtc/modules/pacing/include/paced_sender.h" |
-#include "webrtc/modules/pacing/include/packet_router.h" |
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
#include "webrtc/modules/rtp_rtcp/source/bitrate.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
@@ -71,10 +69,12 @@ class RTPSenderInterface { |
virtual uint16_t PacketOverHead() const = 0; |
virtual uint16_t ActualSendBitrateKbit() const = 0; |
- virtual int32_t SendToNetwork( |
- uint8_t *data_buffer, size_t payload_length, size_t rtp_header_length, |
- int64_t capture_time_ms, StorageType storage, |
- PacedSender::Priority priority) = 0; |
+ virtual int32_t SendToNetwork(uint8_t* data_buffer, |
+ size_t payload_length, |
+ size_t rtp_header_length, |
+ int64_t capture_time_ms, |
+ StorageType storage, |
+ RtpPacketSender::Priority priority) = 0; |
virtual bool UpdateVideoRotation(uint8_t* rtp_packet, |
size_t rtp_packet_length, |
@@ -90,8 +90,8 @@ class RTPSender : public RTPSenderInterface { |
Clock* clock, |
Transport* transport, |
RtpAudioFeedback* audio_feedback, |
- PacedSender* paced_sender, |
- PacketRouter* packet_router, |
+ RtpPacketSender* paced_sender, |
+ TransportSequenceNumberAllocator* sequence_number_allocator, |
TransportFeedbackObserver* transport_feedback_callback, |
BitrateStatisticsObserver* bitrate_callback, |
FrameCountObserver* frame_count_observer, |
@@ -257,7 +257,7 @@ class RTPSender : public RTPSenderInterface { |
size_t rtp_header_length, |
int64_t capture_time_ms, |
StorageType storage, |
- PacedSender::Priority priority) override; |
+ RtpPacketSender::Priority priority) override; |
// Audio. |
@@ -370,8 +370,8 @@ class RTPSender : public RTPSenderInterface { |
const RTPHeader& rtp_header, |
int64_t now_ms) const; |
// Update the transport sequence number of the packet using a new sequence |
- // number allocated by PacketRouter. Returns the assigned sequence number, |
- // or 0 if extension could not be updated. |
+ // number allocated by SequenceNumberAllocator. Returns the assigned sequence |
+ // number, or 0 if extension could not be updated. |
uint16_t UpdateTransportSequenceNumber(uint8_t* rtp_packet, |
size_t rtp_packet_length, |
const RTPHeader& rtp_header) const; |
@@ -393,8 +393,8 @@ class RTPSender : public RTPSenderInterface { |
rtc::scoped_ptr<RTPSenderAudio> audio_; |
rtc::scoped_ptr<RTPSenderVideo> video_; |
- PacedSender* const paced_sender_; |
- PacketRouter* const packet_router_; |
+ RtpPacketSender* const paced_sender_; |
+ TransportSequenceNumberAllocator* const transport_sequence_number_allocator_; |
TransportFeedbackObserver* const transport_feedback_observer_; |
int64_t last_capture_time_ms_sent_; |
rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_; |