14d5dbe5b3
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon. Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/. TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org BUG=webrtc:4741, webrtc:5603, chromium:609749 Review-Url: https://codereview.webrtc.org/2110113003 Cr-Commit-Position: refs/heads/master@{#13379}
71 lines
2.3 KiB
C++
71 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_RECEIVE_STREAM_H_
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#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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#include <memory>
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#include "webrtc/audio_receive_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/modules/rtp_rtcp/include/rtp_header_parser.h"
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namespace webrtc {
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class CongestionController;
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class RemoteBitrateEstimator;
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class RtcEventLog;
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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 AudioReceiveStream final : public webrtc::AudioReceiveStream {
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public:
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AudioReceiveStream(CongestionController* congestion_controller,
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const webrtc::AudioReceiveStream::Config& config,
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const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
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webrtc::RtcEventLog* event_log);
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~AudioReceiveStream() override;
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// webrtc::AudioReceiveStream implementation.
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void Start() override;
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void Stop() override;
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webrtc::AudioReceiveStream::Stats GetStats() const override;
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void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
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void SetGain(float gain) 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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bool DeliverRtp(const uint8_t* packet,
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size_t length,
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const PacketTime& packet_time);
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const webrtc::AudioReceiveStream::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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RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
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const webrtc::AudioReceiveStream::Config config_;
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rtc::scoped_refptr<webrtc::AudioState> audio_state_;
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std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
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std::unique_ptr<voe::ChannelProxy> channel_proxy_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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