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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender.h

Issue 1350163005: Avoid circular dependency rtp_rtcp <-> paced_sender (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years, 3 months ago
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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_;
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