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
|
/*
* 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_SEND_STREAM_H_
#define AUDIO_AUDIO_SEND_STREAM_H_
#include <memory>
#include <vector>
#include "audio/time_interval.h"
#include "audio/transport_feedback_packet_loss_tracker.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/bitrate_allocator.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
class RtcEventLog;
class RtcpBandwidthObserver;
class RtcpRttStats;
class RtpTransportControllerSendInterface;
namespace voe {
class ChannelProxy;
} // namespace voe
namespace internal {
class AudioState;
class AudioSendStream final : public webrtc::AudioSendStream,
public webrtc::BitrateAllocatorObserver,
public webrtc::PacketFeedbackObserver {
public:
AudioSendStream(const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
ProcessThread* module_process_thread,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const rtc::Optional<RtpState>& suspended_rtp_state,
TimeInterval* overall_call_lifetime);
// For unit tests, which need to supply a mock channel proxy.
AudioSendStream(const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
rtc::TaskQueue* worker_queue,
RtpTransportControllerSendInterface* transport,
BitrateAllocator* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const rtc::Optional<RtpState>& suspended_rtp_state,
TimeInterval* overall_call_lifetime,
std::unique_ptr<voe::ChannelProxy> channel_proxy);
~AudioSendStream() override;
// webrtc::AudioSendStream implementation.
const webrtc::AudioSendStream::Config& GetConfig() const override;
void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
void Start() override;
void Stop() override;
void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
int duration_ms) override;
void SetMuted(bool muted) override;
webrtc::AudioSendStream::Stats GetStats() const override;
webrtc::AudioSendStream::Stats GetStats(
bool has_remote_tracks) 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,
int64_t bwe_period_ms) override;
// From PacketFeedbackObserver.
void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
void OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) override;
void SetTransportOverhead(int transport_overhead_per_packet);
RtpState GetRtpState() const;
const voe::ChannelProxy& GetChannelProxy() const;
private:
class TimedTransport;
internal::AudioState* audio_state();
const internal::AudioState* audio_state() const;
void StoreEncoderProperties(int sample_rate_hz, size_t num_channels);
// These are all static to make it less likely that (the old) config_ is
// accessed unintentionally.
static void ConfigureStream(AudioSendStream* stream,
const Config& new_config,
bool first_time);
static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config);
static bool ReconfigureSendCodec(AudioSendStream* stream,
const Config& new_config);
static void ReconfigureANA(AudioSendStream* stream, const Config& new_config);
static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config);
static void ReconfigureBitrateObserver(AudioSendStream* stream,
const Config& new_config);
void ConfigureBitrateObserver(int min_bitrate_bps,
int max_bitrate_bps,
double bitrate_priority,
bool has_packet_feedback);
void RemoveBitrateObserver();
void RegisterCngPayloadType(int payload_type, int clockrate_hz);
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker pacer_thread_checker_;
rtc::RaceChecker audio_capture_race_checker_;
rtc::TaskQueue* worker_queue_;
webrtc::AudioSendStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
RtcEventLog* const event_log_;
int encoder_sample_rate_hz_ = 0;
size_t encoder_num_channels_ = 0;
bool sending_ = false;
BitrateAllocator* const bitrate_allocator_;
RtpTransportControllerSendInterface* const transport_;
rtc::CriticalSection packet_loss_tracker_cs_;
TransportFeedbackPacketLossTracker packet_loss_tracker_
RTC_GUARDED_BY(&packet_loss_tracker_cs_);
RtpRtcp* rtp_rtcp_module_;
rtc::Optional<RtpState> const suspended_rtp_state_;
std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
TimeInterval active_lifetime_;
TimeInterval* overall_call_lifetime_ = nullptr;
// RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
// reserved for padding and MUST NOT be used as a local identifier.
// So it should be safe to use 0 here to indicate "not configured".
struct ExtensionIds {
int audio_level = 0;
int transport_sequence_number = 0;
int mid = 0;
};
static ExtensionIds FindExtensionIds(
const std::vector<RtpExtension>& extensions);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_SEND_STREAM_H_
|
