79e05888e8
BUG=webrtc:6557 Review-Url: https://codereview.webrtc.org/2437503004 Cr-Commit-Position: refs/heads/master@{#14968}
81 lines
2.6 KiB
C++
81 lines
2.6 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
|
|
#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
|
|
|
|
#include <memory>
|
|
|
|
#include "webrtc/api/call/audio_send_stream.h"
|
|
#include "webrtc/api/call/audio_state.h"
|
|
#include "webrtc/base/constructormagic.h"
|
|
#include "webrtc/base/thread_checker.h"
|
|
#include "webrtc/call/bitrate_allocator.h"
|
|
|
|
namespace webrtc {
|
|
class CongestionController;
|
|
class VoiceEngine;
|
|
class RtcEventLog;
|
|
|
|
namespace voe {
|
|
class ChannelProxy;
|
|
} // namespace voe
|
|
|
|
namespace internal {
|
|
class AudioSendStream final : public webrtc::AudioSendStream,
|
|
public webrtc::BitrateAllocatorObserver {
|
|
public:
|
|
AudioSendStream(const webrtc::AudioSendStream::Config& config,
|
|
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
|
|
rtc::TaskQueue* worker_queue,
|
|
CongestionController* congestion_controller,
|
|
BitrateAllocator* bitrate_allocator,
|
|
RtcEventLog* event_log);
|
|
~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);
|
|
|
|
// Implements BitrateAllocatorObserver.
|
|
uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
|
|
uint8_t fraction_loss,
|
|
int64_t rtt) override;
|
|
|
|
const webrtc::AudioSendStream::Config& config() const;
|
|
void SetTransportOverhead(int transport_overhead_per_packet);
|
|
|
|
private:
|
|
VoiceEngine* voice_engine() const;
|
|
|
|
bool SetupSendCodec();
|
|
|
|
rtc::ThreadChecker thread_checker_;
|
|
rtc::TaskQueue* worker_queue_;
|
|
const webrtc::AudioSendStream::Config config_;
|
|
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
|
|
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
|
|
|
|
BitrateAllocator* const bitrate_allocator_;
|
|
|
|
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
|
|
};
|
|
} // namespace internal
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
|