/*
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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 "webrtc/audio_send_stream.h"
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#include "webrtc/audio_state.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/base/scoped_ptr.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:
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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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~AudioSendStream() override;
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// webrtc::SendStream implementation.
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void Start() override;
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void Stop() override;
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void SignalNetworkState(NetworkState state) override;
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bool DeliverRtcp(const uint8_t* packet, size_t length) override;
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// webrtc::AudioSendStream implementation.
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bool SendTelephoneEvent(int payload_type, uint8_t event,
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uint32_t duration_ms) override;
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webrtc::AudioSendStream::Stats GetStats() const 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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rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
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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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