Refactoring PayloadRouter.
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index c3a56ad..e954b02 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -13,11 +13,16 @@
#include <stddef.h>
#include <stdint.h>
+#include <map>
#include <string>
+#include <vector>
#include "absl/types/optional.h"
#include "api/bitrate_constraints.h"
#include "api/transport/bitrate_settings.h"
+#include "call/rtp_config.h"
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace rtc {
struct SentPacket;
@@ -26,18 +31,36 @@
} // namespace rtc
namespace webrtc {
+class CallStats;
class CallStatsObserver;
class TargetTransferRateObserver;
+class Transport;
class Module;
class PacedSender;
class PacketFeedbackObserver;
class PacketRouter;
+class VideoRtpSenderInterface;
class RateLimiter;
class RtcpBandwidthObserver;
class RtpPacketSender;
struct RtpKeepAliveConfig;
+class SendDelayStats;
+class SendStatisticsProxy;
class TransportFeedbackObserver;
+struct RtpSenderObservers {
+ RtcpRttStats* rtcp_rtt_stats;
+ RtcpIntraFrameObserver* intra_frame_callback;
+ RtcpStatisticsCallback* rtcp_stats;
+ StreamDataCountersCallback* rtp_stats;
+ BitrateStatisticsObserver* bitrate_observer;
+ FrameCountObserver* frame_count_observer;
+ RtcpPacketTypeCounterObserver* rtcp_type_observer;
+ SendSideDelayObserver* send_delay_observer;
+ SendPacketObserver* send_packet_observer;
+ OverheadObserver* overhead_observer;
+};
+
// An RtpTransportController should own everything related to the RTP
// transport to/from a remote endpoint. We should have separate
// interfaces for send and receive side, even if they are implemented
@@ -66,6 +89,18 @@
virtual ~RtpTransportControllerSendInterface() {}
virtual rtc::TaskQueue* GetWorkerQueue() = 0;
virtual PacketRouter* packet_router() = 0;
+
+ virtual VideoRtpSenderInterface* CreateVideoRtpSender(
+ const std::vector<uint32_t>& ssrcs,
+ std::map<uint32_t, RtpState> suspended_ssrcs,
+ // TODO(holmer): Move states into RtpTransportControllerSend.
+ const std::map<uint32_t, RtpPayloadState>& states,
+ const RtpConfig& rtp_config,
+ const RtcpConfig& rtcp_config,
+ Transport* send_transport,
+ const RtpSenderObservers& observers,
+ RtcEventLog* event_log) = 0;
+
virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
virtual RtpPacketSender* packet_sender() = 0;