/
audio_receive_stream.h
131 lines (111 loc) · 4.88 KB
/
audio_receive_stream.h
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
/*
* 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 AUDIO_AUDIO_RECEIVE_STREAM_H_
#define AUDIO_AUDIO_RECEIVE_STREAM_H_
#include <memory>
#include <vector>
#include "api/audio/audio_mixer.h"
#include "api/neteq/neteq_factory.h"
#include "api/rtp_headers.h"
#include "audio/audio_state.h"
#include "call/audio_receive_stream.h"
#include "call/syncable.h"
#include "modules/rtp_rtcp/source/source_tracker.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/thread_checker.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class PacketRouter;
class ProcessThread;
class RtcEventLog;
class RtpPacketReceived;
class RtpStreamReceiverControllerInterface;
class RtpStreamReceiverInterface;
namespace voe {
class ChannelReceiveInterface;
} // namespace voe
namespace internal {
class AudioSendStream;
class AudioReceiveStream final : public webrtc::AudioReceiveStream,
public AudioMixer::Source,
public Syncable {
public:
AudioReceiveStream(Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
ProcessThread* module_process_thread,
NetEqFactory* neteq_factory,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log);
// For unit tests, which need to supply a mock channel receive.
AudioReceiveStream(
Clock* clock,
RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
~AudioReceiveStream() override;
// webrtc::AudioReceiveStream implementation.
void Reconfigure(const webrtc::AudioReceiveStream::Config& config) override;
void Start() override;
void Stop() override;
webrtc::AudioReceiveStream::Stats GetStats() const override;
void SetSink(AudioSinkInterface* sink) override;
void SetGain(float gain) override;
bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
int GetBaseMinimumPlayoutDelayMs() const override;
std::vector<webrtc::RtpSource> GetSources() const override;
#ifndef DISABLE_RECORDER
void InjectRecorder(Recorder* recorder) override;
#endif
// TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
// method shouldn't be needed. But it's currently used by the
// AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
// shuld be refactored or deleted, and then delete this method.
void OnRtpPacket(const RtpPacketReceived& packet);
// AudioMixer::Source
AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
AudioFrame* audio_frame) override;
int Ssrc() const override;
int PreferredSampleRate() const override;
// Syncable
int id() const override;
absl::optional<Syncable::Info> GetInfo() const override;
bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const override;
void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
int64_t time_ms) override;
void SetMinimumPlayoutDelay(int delay_ms) override;
void AssociateSendStream(AudioSendStream* send_stream);
void DeliverRtcp(const uint8_t* packet, size_t length);
const webrtc::AudioReceiveStream::Config& config() const;
const AudioSendStream* GetAssociatedSendStreamForTesting() const;
private:
static void ConfigureStream(AudioReceiveStream* stream,
const Config& new_config,
bool first_time);
AudioState* audio_state() const;
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker module_process_thread_checker_;
webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
SourceTracker source_tracker_;
AudioSendStream* associated_send_stream_ = nullptr;
bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_RECEIVE_STREAM_H_