Rename PayloadRouter to RtpVideoSender.
Bug: webrtc:9517
Change-Id: I18397a28067dbe5029fc80fe2eef360869abb339
Reviewed-on: https://webrtc-review.googlesource.com/89380
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24039}diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
new file mode 100644
index 0000000..5c56753
--- /dev/null
+++ b/call/rtp_video_sender.h
@@ -0,0 +1,128 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef CALL_RTP_VIDEO_SENDER_H_
+#define CALL_RTP_VIDEO_SENDER_H_
+
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/call/transport.h"
+#include "api/video_codecs/video_encoder.h"
+#include "call/rtp_config.h"
+#include "call/rtp_payload_params.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "call/rtp_video_sender_interface.h"
+#include "common_types.h" // NOLINT(build/include)
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "modules/rtp_rtcp/include/flexfec_sender.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/utility/include/process_thread.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/thread_checker.h"
+
+namespace webrtc {
+
+class RTPFragmentationHeader;
+class RtpRtcp;
+class RtpTransportControllerSendInterface;
+
+// RtpVideoSender routes outgoing data to the correct sending RTP module, based
+// on the simulcast layer in RTPVideoHeader.
+class RtpVideoSender : public RtpVideoSenderInterface {
+ public:
+ // Rtp modules are assumed to be sorted in simulcast index order.
+ RtpVideoSender(
+ const std::vector<uint32_t>& ssrcs,
+ std::map<uint32_t, RtpState> suspended_ssrcs,
+ const std::map<uint32_t, RtpPayloadState>& states,
+ const RtpConfig& rtp_config,
+ const RtcpConfig& rtcp_config,
+ Transport* send_transport,
+ const RtpSenderObservers& observers,
+ RtpTransportControllerSendInterface* transport,
+ RtcEventLog* event_log,
+ RateLimiter* retransmission_limiter); // move inside RtpTransport
+ ~RtpVideoSender() override;
+
+ // RegisterProcessThread register |module_process_thread| with those objects
+ // that use it. Registration has to happen on the thread were
+ // |module_process_thread| was created (libjingle's worker thread).
+ // TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
+ // maybe |worker_queue|.
+ void RegisterProcessThread(ProcessThread* module_process_thread) override;
+ void DeRegisterProcessThread() override;
+
+ // RtpVideoSender will only route packets if being active, all packets will be
+ // dropped otherwise.
+ void SetActive(bool active) override;
+ // Sets the sending status of the rtp modules and appropriately sets the
+ // payload router to active if any rtp modules are active.
+ void SetActiveModules(const std::vector<bool> active_modules) override;
+ bool IsActive() override;
+
+ void OnNetworkAvailability(bool network_available) override;
+ std::map<uint32_t, RtpState> GetRtpStates() const override;
+ std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const override;
+
+ bool FecEnabled() const override;
+
+ bool NackEnabled() const override;
+
+ void DeliverRtcp(const uint8_t* packet, size_t length) override;
+
+ void ProtectionRequest(const FecProtectionParams* delta_params,
+ const FecProtectionParams* key_params,
+ uint32_t* sent_video_rate_bps,
+ uint32_t* sent_nack_rate_bps,
+ uint32_t* sent_fec_rate_bps) override;
+
+ void SetMaxRtpPacketSize(size_t max_rtp_packet_size) override;
+
+ // Implements EncodedImageCallback.
+ // Returns 0 if the packet was routed / sent, -1 otherwise.
+ EncodedImageCallback::Result OnEncodedImage(
+ const EncodedImage& encoded_image,
+ const CodecSpecificInfo* codec_specific_info,
+ const RTPFragmentationHeader* fragmentation) override;
+
+ void OnBitrateAllocationUpdated(
+ const VideoBitrateAllocation& bitrate) override;
+
+ private:
+ void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
+ void ConfigureProtection(const RtpConfig& rtp_config);
+ void ConfigureSsrcs(const RtpConfig& rtp_config);
+
+ rtc::CriticalSection crit_;
+ bool active_ RTC_GUARDED_BY(crit_);
+
+ ProcessThread* module_process_thread_;
+ rtc::ThreadChecker module_process_thread_checker_;
+ std::map<uint32_t, RtpState> suspended_ssrcs_;
+
+ std::unique_ptr<FlexfecSender> flexfec_sender_;
+ // Rtp modules are assumed to be sorted in simulcast index order. Not owned.
+ const std::vector<std::unique_ptr<RtpRtcp>> rtp_modules_;
+ const RtpConfig rtp_config_;
+ RtpTransportControllerSendInterface* const transport_;
+
+ std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(crit_);
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(RtpVideoSender);
+};
+
+} // namespace webrtc
+
+#endif // CALL_RTP_VIDEO_SENDER_H_