blob: 44f093ccffa8b6ecdada5e7cae68d659c3598ee0 (
plain)
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
|
/*
* 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 CALL_AUDIO_RECEIVE_STREAM_H_
#define CALL_AUDIO_RECEIVE_STREAM_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/transport.h"
#include "api/optional.h"
#include "api/rtpparameters.h"
#include "api/rtpreceiverinterface.h"
#include "call/rtp_config.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/scoped_ref_ptr.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
class AudioSinkInterface;
// WORK IN PROGRESS
// This class is under development and is not yet intended for for use outside
// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
class AudioReceiveStream {
public:
struct Stats {
uint32_t remote_ssrc = 0;
int64_t bytes_rcvd = 0;
uint32_t packets_rcvd = 0;
uint32_t packets_lost = 0;
float fraction_lost = 0.0f;
std::string codec_name;
rtc::Optional<int> codec_payload_type;
uint32_t ext_seqnum = 0;
uint32_t jitter_ms = 0;
uint32_t jitter_buffer_ms = 0;
uint32_t jitter_buffer_preferred_ms = 0;
uint32_t delay_estimate_ms = 0;
int32_t audio_level = -1;
// Stats below correspond to similarly-named fields in the WebRTC stats
// spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
double total_output_energy = 0.0;
uint64_t total_samples_received = 0;
double total_output_duration = 0.0;
uint64_t concealed_samples = 0;
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
float expand_rate = 0.0f;
float speech_expand_rate = 0.0f;
float secondary_decoded_rate = 0.0f;
float secondary_discarded_rate = 0.0f;
float accelerate_rate = 0.0f;
float preemptive_expand_rate = 0.0f;
int32_t decoding_calls_to_silence_generator = 0;
int32_t decoding_calls_to_neteq = 0;
int32_t decoding_normal = 0;
int32_t decoding_plc = 0;
int32_t decoding_cng = 0;
int32_t decoding_plc_cng = 0;
int32_t decoding_muted_output = 0;
int64_t capture_start_ntp_time_ms = 0;
};
struct Config {
std::string ToString() const;
// Receive-stream specific RTP settings.
struct Rtp {
std::string ToString() const;
// Synchronization source (stream identifier) to be received.
uint32_t remote_ssrc = 0;
// Sender SSRC used for sending RTCP (such as receiver reports).
uint32_t local_ssrc = 0;
// Enable feedback for send side bandwidth estimation.
// See
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
// for details.
bool transport_cc = false;
// See NackConfig for description.
NackConfig nack;
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
Transport* rtcp_send_transport = nullptr;
// Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
// level components.
// TODO(solenberg): Remove when VoiceEngine channels are created outside
// of Call.
int voe_channel_id = -1;
// Identifier for an A/V synchronization group. Empty string to disable.
// TODO(pbos): Synchronize streams in a sync group, not just one video
// stream to one audio stream. Tracked by issue webrtc:4762.
std::string sync_group;
// Decoder specifications for every payload type that we can receive.
std::map<int, SdpAudioFormat> decoder_map;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
};
// Starts stream activity.
// When a stream is active, it can receive, process and deliver packets.
virtual void Start() = 0;
// Stops stream activity.
// When a stream is stopped, it can't receive, process or deliver packets.
virtual void Stop() = 0;
virtual Stats GetStats() const = 0;
// TODO(solenberg): Remove, once AudioMonitor is gone.
virtual int GetOutputLevel() const = 0;
// Sets an audio sink that receives unmixed audio from the receive stream.
// Ownership of the sink is passed to the stream and can be used by the
// caller to do lifetime management (i.e. when the sink's dtor is called).
// Only one sink can be set and passing a null sink clears an existing one.
// NOTE: Audio must still somehow be pulled through AudioTransport for audio
// to stream through this sink. In practice, this happens if mixed audio
// is being pulled+rendered and/or if audio is being pulled for the purposes
// of feeding to the AEC.
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
// Sets playback gain of the stream, applied when mixing, and thus after it
// is potentially forwarded to any attached AudioSinkInterface implementation.
virtual void SetGain(float gain) = 0;
virtual std::vector<RtpSource> GetSources() const = 0;
protected:
virtual ~AudioReceiveStream() {}
};
} // namespace webrtc
#endif // CALL_AUDIO_RECEIVE_STREAM_H_
|
