1/*
2 *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
12#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
13
14#include "webrtc/audio_send_stream.h"
15#include "webrtc/audio_state.h"
16#include "webrtc/base/thread_checker.h"
17#include "webrtc/base/scoped_ptr.h"
18
19namespace webrtc {
20class CongestionController;
21class VoiceEngine;
22
23namespace voe {
24class ChannelProxy;
25}  // namespace voe
26
27namespace internal {
28class AudioSendStream final : public webrtc::AudioSendStream {
29 public:
30  AudioSendStream(const webrtc::AudioSendStream::Config& config,
31                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
32                  CongestionController* congestion_controller);
33  ~AudioSendStream() override;
34
35  // webrtc::SendStream implementation.
36  void Start() override;
37  void Stop() override;
38  void SignalNetworkState(NetworkState state) override;
39  bool DeliverRtcp(const uint8_t* packet, size_t length) override;
40
41  // webrtc::AudioSendStream implementation.
42  bool SendTelephoneEvent(int payload_type, uint8_t event,
43                          uint32_t duration_ms) override;
44  webrtc::AudioSendStream::Stats GetStats() const override;
45
46  const webrtc::AudioSendStream::Config& config() const;
47
48 private:
49  VoiceEngine* voice_engine() const;
50
51  rtc::ThreadChecker thread_checker_;
52  const webrtc::AudioSendStream::Config config_;
53  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
54  rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
55
56  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
57};
58}  // namespace internal
59}  // namespace webrtc
60
61#endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
62