0b69826ffb
This prepares for making AudioSendStream use its own task queue. In the future more of the functionality that depends on running on the task queue is planned to be moved directly into RtpTransportControllerSend. They should instead get it from the transport controller. This affects the media transport tests which previously assumed that the transport controller could be missing. However, this is not something that is used in production, so this is an improvement of the tests as they will behave more like production code. Bug: webrtc:9883 Change-Id: Ie32f4c2f6433ec37ac16a08d531ceb690ea9c0b5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126000 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27010}
112 lines
3.4 KiB
C++
112 lines
3.4 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef VIDEO_VIDEO_SEND_STREAM_H_
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#define VIDEO_VIDEO_SEND_STREAM_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "api/fec_controller.h"
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#include "api/video/video_stream_encoder_interface.h"
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#include "call/bitrate_allocator.h"
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/event.h"
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#include "rtc_base/task_queue.h"
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#include "video/send_delay_stats.h"
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#include "video/send_statistics_proxy.h"
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namespace webrtc {
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namespace test {
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class VideoSendStreamPeer;
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} // namespace test
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class CallStats;
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class IvfFileWriter;
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class ProcessThread;
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class RateLimiter;
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class RtpRtcp;
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class RtpTransportControllerSendInterface;
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class RtcEventLog;
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namespace internal {
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class VideoSendStreamImpl;
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// VideoSendStream implements webrtc::VideoSendStream.
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// Internally, it delegates all public methods to VideoSendStreamImpl and / or
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// VideoStreamEncoder. VideoSendStreamInternal is created and deleted on
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// |worker_queue|.
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class VideoSendStream : public webrtc::VideoSendStream {
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public:
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using RtpStateMap = std::map<uint32_t, RtpState>;
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using RtpPayloadStateMap = std::map<uint32_t, RtpPayloadState>;
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VideoSendStream(
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Clock* clock,
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int num_cpu_cores,
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ProcessThread* module_process_thread,
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TaskQueueFactory* task_queue_factory,
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CallStats* call_stats,
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RtpTransportControllerSendInterface* transport,
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BitrateAllocatorInterface* bitrate_allocator,
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SendDelayStats* send_delay_stats,
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RtcEventLog* event_log,
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config,
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const std::map<uint32_t, RtpState>& suspended_ssrcs,
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const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
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std::unique_ptr<FecController> fec_controller);
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~VideoSendStream() override;
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void DeliverRtcp(const uint8_t* packet, size_t length);
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// webrtc::VideoSendStream implementation.
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void UpdateActiveSimulcastLayers(
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const std::vector<bool> active_layers) override;
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void Start() override;
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void Stop() override;
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void SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
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const DegradationPreference& degradation_preference) override;
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void ReconfigureVideoEncoder(VideoEncoderConfig) override;
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Stats GetStats() override;
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void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
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RtpPayloadStateMap* payload_state_map);
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private:
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friend class test::VideoSendStreamPeer;
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class ConstructionTask;
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absl::optional<float> GetPacingFactorOverride() const;
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rtc::ThreadChecker thread_checker_;
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rtc::TaskQueue* const worker_queue_;
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rtc::Event thread_sync_event_;
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SendStatisticsProxy stats_proxy_;
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const VideoSendStream::Config config_;
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const VideoEncoderConfig::ContentType content_type_;
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std::unique_ptr<VideoSendStreamImpl> send_stream_;
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std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_;
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};
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} // namespace internal
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} // namespace webrtc
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#endif // VIDEO_VIDEO_SEND_STREAM_H_
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