8eb37a39e7
Reason for revert: Failed on Win 10 Chrome FYI. https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3847/steps/content_browsertests/logs/stdio # # Fatal error in e:\b\c\b\win_builder\src\third_party\webrtc\base\task_queue_win.cc, line 138 # last system error: 87 # Check failed: ((DWORD)0xFFFFFFFF) != result (4294967295 vs. 4294967295) # WebRtcBrowserTest # Original issue's description: > - Add task queue to Call with the intent of replacing the use of one of the process threads. > > - Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue. > > - BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread. > > - VideoEncoderConfig and VideoSendStream::Config support move semantics. > > - The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames. > > BUG=webrtc:5687 > > Committed: https://crrev.com/cc168360f41322332860cb075edeb1cde21aa473 > Cr-Commit-Position: refs/heads/master@{#13767} TBR=tommi@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org,sprang@webrtc.org,pbos@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5687 Review-Url: https://codereview.webrtc.org/2248713003 Cr-Commit-Position: refs/heads/master@{#13774}
74 lines
2.3 KiB
C++
74 lines
2.3 KiB
C++
/*
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* Copyright (c) 2015 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 WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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#include <memory>
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#include "webrtc/audio_send_stream.h"
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#include "webrtc/audio_state.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/call/bitrate_allocator.h"
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namespace webrtc {
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class CongestionController;
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class VoiceEngine;
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namespace voe {
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class ChannelProxy;
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} // namespace voe
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namespace internal {
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class AudioSendStream final : public webrtc::AudioSendStream,
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public webrtc::BitrateAllocatorObserver {
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public:
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AudioSendStream(const webrtc::AudioSendStream::Config& config,
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const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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CongestionController* congestion_controller,
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BitrateAllocator* bitrate_allocator);
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~AudioSendStream() override;
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// webrtc::AudioSendStream implementation.
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void Start() override;
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void Stop() override;
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bool SendTelephoneEvent(int payload_type, int event,
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int duration_ms) override;
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void SetMuted(bool muted) override;
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webrtc::AudioSendStream::Stats GetStats() const 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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// Implements BitrateAllocatorObserver.
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uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
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uint8_t fraction_loss,
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int64_t rtt) override;
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const webrtc::AudioSendStream::Config& config() const;
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private:
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VoiceEngine* voice_engine() const;
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rtc::ThreadChecker thread_checker_;
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const webrtc::AudioSendStream::Config config_;
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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std::unique_ptr<voe::ChannelProxy> channel_proxy_;
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BitrateAllocator* const bitrate_allocator_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
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