blob: 264f9b3a69f53d9c71e732919e10619e51edb26c [file] [log] [blame]
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <memory>
#include "webrtc/audio_send_stream.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_checker.h"
namespace webrtc {
class CongestionController;
class VoiceEngine;
namespace voe {
class ChannelProxy;
} // namespace voe
namespace internal {
class AudioSendStream final : public webrtc::AudioSendStream {
AudioSendStream(const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
CongestionController* congestion_controller);
~AudioSendStream() override;
// webrtc::AudioSendStream implementation.
void Start() override;
void Stop() override;
bool SendTelephoneEvent(int payload_type, int event,
int duration_ms) override;
void SetMuted(bool muted) override;
webrtc::AudioSendStream::Stats GetStats() const override;
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
const webrtc::AudioSendStream::Config& config() const;
VoiceEngine* voice_engine() const;
rtc::ThreadChecker thread_checker_;
const webrtc::AudioSendStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
} // namespace internal
} // namespace webrtc