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
|
/*
* Copyright 2019 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 PC_AUDIO_RTP_RECEIVER_H_
#define PC_AUDIO_RTP_RECEIVER_H_
#include <stdint.h>
#include <optional>
#include <string>
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/rtp/rtp_source.h"
#include "media/base/media_channel.h"
#include "pc/audio_track.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/media_stream_track_proxy.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_receiver.h"
#include "rtc_base/system/no_unique_address.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class AudioRtpReceiver : public ObserverInterface,
public AudioSourceInterface::AudioObserver,
public RtpReceiverInternal {
public:
// The constructor supports optionally passing the voice channel to the
// instance at construction time without having to call `SetMediaChannel()`
// on the worker thread straight after construction.
// However, when using that, the assumption is that right after construction,
// a call to either `SetupUnsignaledMediaChannel` or `SetupMediaChannel`
// will be made, which will internally start the source on the worker thread.
AudioRtpReceiver(Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids,
bool is_unified_plan,
VoiceMediaReceiveChannelInterface* voice_channel = nullptr);
// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
AudioRtpReceiver(
Thread* worker_thread,
const std::string& receiver_id,
const std::vector<scoped_refptr<MediaStreamInterface>>& streams,
bool is_unified_plan,
VoiceMediaReceiveChannelInterface* media_channel = nullptr);
virtual ~AudioRtpReceiver();
// ObserverInterface implementation
void OnChanged() override;
// AudioSourceInterface::AudioObserver implementation
void OnSetVolume(double volume) override;
scoped_refptr<AudioTrackInterface> audio_track() const { return track_; }
// RtpReceiverInterface implementation
scoped_refptr<MediaStreamTrackInterface> track() const override {
return track_;
}
scoped_refptr<DtlsTransportInterface> dtls_transport() const override;
std::vector<std::string> stream_ids() const override;
std::vector<scoped_refptr<MediaStreamInterface>> streams() const override;
webrtc::MediaType media_type() const override {
return webrtc::MediaType::AUDIO;
}
std::string id() const override { return id_; }
RtpParameters GetParameters() const override;
void SetFrameDecryptor(
scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;
scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const override;
// RtpReceiverInternal implementation.
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
std::optional<uint32_t> ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(
scoped_refptr<DtlsTransportInterface> dtls_transport) override;
void SetStreams(
const std::vector<scoped_refptr<MediaStreamInterface>>& streams) override;
void SetObserver(RtpReceiverObserverInterface* observer) override;
void SetJitterBufferMinimumDelay(
std::optional<double> delay_seconds) override;
void SetMediaChannel(MediaReceiveChannelInterface* media_channel) override;
std::vector<RtpSource> GetSources() const override;
int AttachmentId() const override { return attachment_id_; }
void SetFrameTransformer(
scoped_refptr<FrameTransformerInterface> frame_transformer) override;
private:
void RestartMediaChannel(std::optional<uint32_t> ssrc)
RTC_RUN_ON(&signaling_thread_checker_);
void RestartMediaChannel_w(std::optional<uint32_t> ssrc,
bool track_enabled,
MediaSourceInterface::SourceState state)
RTC_RUN_ON(worker_thread_);
void Reconfigure(bool track_enabled) RTC_RUN_ON(worker_thread_);
void SetOutputVolume_w(double volume) RTC_RUN_ON(worker_thread_);
RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
Thread* const worker_thread_;
const std::string id_;
const scoped_refptr<RemoteAudioSource> source_;
const scoped_refptr<AudioTrackProxyWithInternal<AudioTrack>> track_;
VoiceMediaReceiveChannelInterface* media_channel_
RTC_GUARDED_BY(worker_thread_) = nullptr;
std::optional<uint32_t> signaled_ssrc_ RTC_GUARDED_BY(worker_thread_);
std::vector<scoped_refptr<MediaStreamInterface>> streams_
RTC_GUARDED_BY(&signaling_thread_checker_);
bool cached_track_enabled_ RTC_GUARDED_BY(&signaling_thread_checker_);
double cached_volume_ RTC_GUARDED_BY(worker_thread_) = 1.0;
RtpReceiverObserverInterface* observer_
RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;
bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =
false;
const int attachment_id_;
scoped_refptr<FrameDecryptorInterface> frame_decryptor_
RTC_GUARDED_BY(worker_thread_);
scoped_refptr<DtlsTransportInterface> dtls_transport_
RTC_GUARDED_BY(&signaling_thread_checker_);
// Stores and updates the playout delay. Handles caching cases if
// `SetJitterBufferMinimumDelay` is called before start.
JitterBufferDelay delay_ RTC_GUARDED_BY(worker_thread_);
scoped_refptr<FrameTransformerInterface> frame_transformer_
RTC_GUARDED_BY(worker_thread_);
const scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_;
};
} // namespace webrtc
#endif // PC_AUDIO_RTP_RECEIVER_H_
|
