| 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..819ca7c1fbcde185815f267203e530a27caa2816 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,
|
| + SequenceNumberAllocator* 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_;
|
| + SequenceNumberAllocator* const sequence_number_allocator_;
|
| TransportFeedbackObserver* const transport_feedback_observer_;
|
| int64_t last_capture_time_ms_sent_;
|
| rtc::scoped_ptr<CriticalSectionWrapper> send_critsect_;
|
|
|