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
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
|
/*
* 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.
*/
#include "webrtc/audio/audio_send_stream.h"
#include <string>
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/call/congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
std::string AudioSendStream::Config::Rtp::ToString() const {
std::stringstream ss;
ss << "{ssrc: " << ssrc;
ss << ", extensions: [";
for (size_t i = 0; i < extensions.size(); ++i) {
ss << extensions[i].ToString();
if (i != extensions.size() - 1) {
ss << ", ";
}
}
ss << ']';
ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
std::string AudioSendStream::Config::ToString() const {
std::stringstream ss;
ss << "{rtp: " << rtp.ToString();
ss << ", voe_channel_id: " << voe_channel_id;
// TODO(solenberg): Encoder config.
ss << ", cng_payload_type: " << cng_payload_type;
ss << ", red_payload_type: " << red_payload_type;
ss << '}';
return ss.str();
}
namespace internal {
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
CongestionController* congestion_controller)
: config_(config), audio_state_(audio_state) {
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
RTC_DCHECK(congestion_controller);
VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
channel_proxy_->SetCongestionControlObjects(
congestion_controller->pacer(),
congestion_controller->GetTransportFeedbackObserver(),
congestion_controller->packet_router());
channel_proxy_->SetRTCPStatus(true);
channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
for (const auto& extension : config.rtp.extensions) {
if (extension.name == RtpExtension::kAbsSendTime) {
channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
} else if (extension.name == RtpExtension::kAudioLevel) {
channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
} else if (extension.name == RtpExtension::kTransportSequenceNumber) {
channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
} else {
RTC_NOTREACHED() << "Registering unsupported RTP extension.";
}
}
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
}
void AudioSendStream::Start() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioSendStream::Stop() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
void AudioSendStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
}
bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.CalledOnValidThread());
return false;
}
bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
uint32_t duration_ms) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
}
webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
webrtc::AudioSendStream::Stats stats;
stats.local_ssrc = config_.rtp.ssrc;
ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
ScopedVoEInterface<VoECodec> codec(voice_engine());
ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
stats.bytes_sent = call_stats.bytesSent;
stats.packets_sent = call_stats.packetsSent;
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
// returns 0 to indicate an error value.
if (call_stats.rttMs > 0) {
stats.rtt_ms = call_stats.rttMs;
}
// TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
// implementation.
stats.aec_quality_min = -1;
webrtc::CodecInst codec_inst = {0};
if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
RTC_DCHECK_NE(codec_inst.pltype, -1);
stats.codec_name = codec_inst.plname;
// Get data from the last remote RTCP report.
for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
// Lookup report for send ssrc only.
if (block.source_SSRC == stats.local_ssrc) {
stats.packets_lost = block.cumulative_num_packets_lost;
stats.fraction_lost = Q8ToFloat(block.fraction_lost);
stats.ext_seqnum = block.extended_highest_sequence_number;
// Convert samples to milliseconds.
if (codec_inst.plfreq / 1000 > 0) {
stats.jitter_ms =
block.interarrival_jitter / (codec_inst.plfreq / 1000);
}
break;
}
}
}
// Local speech level.
{
unsigned int level = 0;
int error = volume->GetSpeechInputLevelFullRange(level);
RTC_DCHECK_EQ(0, error);
stats.audio_level = static_cast<int32_t>(level);
}
bool echo_metrics_on = false;
int error = processing->GetEcMetricsStatus(echo_metrics_on);
RTC_DCHECK_EQ(0, error);
if (echo_metrics_on) {
// These can also be negative, but in practice -1 is only used to signal
// insufficient data, since the resolution is limited to multiples of 4 ms.
int median = -1;
int std = -1;
float dummy = 0.0f;
error = processing->GetEcDelayMetrics(median, std, dummy);
RTC_DCHECK_EQ(0, error);
stats.echo_delay_median_ms = median;
stats.echo_delay_std_ms = std;
// These can take on valid negative values, so use the lowest possible level
// as default rather than -1.
int erl = -100;
int erle = -100;
int dummy1 = 0;
int dummy2 = 0;
error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
RTC_DCHECK_EQ(0, error);
stats.echo_return_loss = erl;
stats.echo_return_loss_enhancement = erle;
}
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
stats.typing_noise_detected = audio_state->typing_noise_detected();
return stats;
}
const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_;
}
VoiceEngine* AudioSendStream::voice_engine() const {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(audio_state_.get());
VoiceEngine* voice_engine = audio_state->voice_engine();
RTC_DCHECK(voice_engine);
return voice_engine;
}
} // namespace internal
} // namespace webrtc
|
