a45c8da852
GetPacingFactor exposed internal details that should not be relied upon. In a later CL theese won't be available any more, this CL is in preparation for that change. The only usage was in video send stream tests. To keep the tests working, they now access the internal video send stream directly. The test code retrieves an optional that indicates whether the send stream has overridden the pacing factor. This means the implementation dependency between video send stream and video send stream tests is increased. This is an improvement compared to depending on the paced sender implementation. Bug: webrtc:8415 Change-Id: Id357553692b3ff3283fa3b64da1b1ebb3c97f04d Reviewed-on: https://webrtc-review.googlesource.com/39265 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21675}
123 lines
4.0 KiB
C++
123 lines
4.0 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 "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 "common_video/libyuv/include/webrtc_libyuv.h"
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#include "modules/video_coding/protection_bitrate_calculator.h"
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#include "rtc_base/criticalsection.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/encoder_rtcp_feedback.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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#include "video/video_stream_encoder.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 SendSideCongestionController;
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class IvfFileWriter;
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class ProcessThread;
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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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VideoSendStream(
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int num_cpu_cores,
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ProcessThread* module_process_thread,
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rtc::TaskQueue* worker_queue,
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CallStats* call_stats,
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RtpTransportControllerSendInterface* transport,
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BitrateAllocator* 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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~VideoSendStream() override;
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void SignalNetworkState(NetworkState state);
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bool DeliverRtcp(const uint8_t* packet, size_t length);
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// webrtc::VideoSendStream implementation.
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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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typedef std::map<uint32_t, RtpState> RtpStateMap;
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typedef std::map<uint32_t, RtpPayloadState> RtpPayloadStateMap;
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// Takes ownership of each file, is responsible for closing them later.
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// Calling this method will close and finalize any current logs.
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// Giving rtc::kInvalidPlatformFileValue in any position disables logging
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// for the corresponding stream.
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// If a frame to be written would make the log too large the write fails and
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// the log is closed and finalized. A |byte_limit| of 0 means no limit.
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void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
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size_t byte_limit) override;
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void StopPermanentlyAndGetRtpStates(RtpStateMap* rtp_state_map,
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RtpPayloadStateMap* payload_state_map);
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void SetTransportOverhead(size_t transport_overhead_per_packet);
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private:
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friend class test::VideoSendStreamPeer;
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class ConstructionTask;
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class DestructAndGetRtpStateTask;
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rtc::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<VideoStreamEncoder> 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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