mirror of
https://github.com/mollyim/webrtc.git
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b9b146c9fe
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
116 lines
4.3 KiB
C++
116 lines
4.3 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 AUDIO_AUDIO_RECEIVE_STREAM_H_
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#define AUDIO_AUDIO_RECEIVE_STREAM_H_
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#include <memory>
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "api/rtp_headers.h"
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#include "audio/audio_state.h"
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#include "call/audio_receive_stream.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "call/syncable.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/thread_checker.h"
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namespace webrtc {
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class PacketRouter;
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class ProcessThread;
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class RtcEventLog;
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class RtpPacketReceived;
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class RtpStreamReceiverControllerInterface;
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class RtpStreamReceiverInterface;
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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;
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class AudioReceiveStream final : public webrtc::AudioReceiveStream,
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public AudioMixer::Source,
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public Syncable {
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public:
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AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
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PacketRouter* packet_router,
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ProcessThread* module_process_thread,
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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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// For unit tests, which need to supply a mock channel proxy.
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AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
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PacketRouter* packet_router,
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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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std::unique_ptr<voe::ChannelProxy> channel_proxy);
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~AudioReceiveStream() override;
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// webrtc::AudioReceiveStream implementation.
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void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override;
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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(AudioSinkInterface* sink) override;
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void SetGain(float gain) override;
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std::vector<webrtc::RtpSource> GetSources() const override;
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// TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
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// method shouldn't be needed. But it's currently used by the
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// AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
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// shuld be refactored or deleted, and then delete this method.
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void OnRtpPacket(const RtpPacketReceived& packet);
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// AudioMixer::Source
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AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
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AudioFrame* audio_frame) override;
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int Ssrc() const override;
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int PreferredSampleRate() const override;
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// Syncable
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int id() const override;
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absl::optional<Syncable::Info> GetInfo() const override;
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uint32_t GetPlayoutTimestamp() const override;
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void SetMinimumPlayoutDelay(int delay_ms) override;
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void AssociateSendStream(AudioSendStream* send_stream);
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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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const webrtc::AudioReceiveStream::Config& config() const;
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const AudioSendStream* GetAssociatedSendStreamForTesting() const;
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private:
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static void ConfigureStream(AudioReceiveStream* stream,
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const Config& new_config,
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bool first_time);
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AudioState* audio_state() const;
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rtc::ThreadChecker worker_thread_checker_;
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rtc::ThreadChecker module_process_thread_checker_;
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webrtc::AudioReceiveStream::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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AudioSendStream* associated_send_stream_ = nullptr;
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bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
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std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
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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 // AUDIO_AUDIO_RECEIVE_STREAM_H_
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