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
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
|
/*
* 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 <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "api/audio/audio_mixer.h"
#include "api/neteq/neteq_factory.h"
#include "api/rtp_headers.h"
#include "api/sequence_checker.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/system/no_unique_address.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class PacketRouter;
class RtcEventLog;
class RtpStreamReceiverControllerInterface;
class RtpStreamReceiverInterface;
namespace voe {
class ChannelReceiveInterface;
} // namespace voe
namespace internal {
class AudioSendStream;
} // namespace internal
class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface,
public AudioMixer::Source,
public Syncable {
public:
AudioReceiveStreamImpl(
Clock* clock,
PacketRouter* packet_router,
NetEqFactory* neteq_factory,
const webrtc::AudioReceiveStreamInterface::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.
AudioReceiveStreamImpl(
Clock* clock,
PacketRouter* packet_router,
const webrtc::AudioReceiveStreamInterface::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log,
std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
AudioReceiveStreamImpl() = delete;
AudioReceiveStreamImpl(const AudioReceiveStreamImpl&) = delete;
AudioReceiveStreamImpl& operator=(const AudioReceiveStreamImpl&) = delete;
// Destruction happens on the worker thread. Prior to destruction the caller
// must ensure that a registration with the transport has been cleared. See
// `RegisterWithTransport` for details.
// TODO(tommi): As a further improvement to this, performing the full
// destruction on the network thread could be made the default.
~AudioReceiveStreamImpl() override;
// Called on the network thread to register/unregister with the network
// transport.
void RegisterWithTransport(
RtpStreamReceiverControllerInterface* receiver_controller);
// If registration has previously been done (via `RegisterWithTransport`) then
// `UnregisterFromTransport` must be called prior to destruction, on the
// network thread.
void UnregisterFromTransport();
// webrtc::AudioReceiveStreamInterface implementation.
void Start() override;
void Stop() override;
bool IsRunning() const override;
void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
override;
void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
void SetNackHistory(int history_ms) override;
void SetNonSenderRttMeasurement(bool enabled) override;
void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
frame_decryptor) override;
void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
RtpHeaderExtensionMap GetRtpExtensionMap() const override;
webrtc::AudioReceiveStreamInterface::Stats GetStats(
bool get_and_clear_legacy_stats) 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;
// AudioMixer::Source
AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
AudioFrame* audio_frame) override;
int Ssrc() const override;
int PreferredSampleRate() const override;
// Syncable
uint32_t 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;
bool SetMinimumPlayoutDelay(int delay_ms) override;
void AssociateSendStream(internal::AudioSendStream* send_stream);
void DeliverRtcp(const uint8_t* packet, size_t length);
void SetSyncGroup(absl::string_view sync_group);
void SetLocalSsrc(uint32_t local_ssrc);
uint32_t local_ssrc() const;
uint32_t remote_ssrc() const override {
// The remote_ssrc member variable of config_ will never change and can be
// considered const.
return config_.rtp.remote_ssrc;
}
// Returns a reference to the currently set sync group of the stream.
// Must be called on the packet delivery thread.
const std::string& sync_group() const;
const AudioSendStream* GetAssociatedSendStreamForTesting() const;
// TODO(tommi): Remove this method.
void ReconfigureForTesting(
const webrtc::AudioReceiveStreamInterface::Config& config);
private:
internal::AudioState* audio_state() const;
RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
// TODO(bugs.webrtc.org/11993): This checker conceptually represents
// operations that belong to the network thread. The Call class is currently
// moving towards handling network packets on the network thread and while
// that work is ongoing, this checker may in practice represent the worker
// thread, but still serves as a mechanism of grouping together concepts
// that belong to the network thread. Once the packets are fully delivered
// on the network thread, this comment will be deleted.
RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;
webrtc::AudioReceiveStreamInterface::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
SourceTracker source_tracker_;
const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
AudioSendStream* associated_send_stream_
RTC_GUARDED_BY(packet_sequence_checker_) = nullptr;
bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_
RTC_GUARDED_BY(packet_sequence_checker_);
};
} // namespace webrtc
#endif // AUDIO_AUDIO_RECEIVE_STREAM_H_
|
