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) 2020 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 "audio/voip/audio_egress.h"
#include <utility>
#include <vector>
#include "rtc_base/logging.h"
namespace webrtc {
AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp,
Clock* clock,
TaskQueueFactory* task_queue_factory)
: rtp_rtcp_(rtp_rtcp),
rtp_sender_audio_(clock, rtp_rtcp_->RtpSender()),
audio_coding_(AudioCodingModule::Create(AudioCodingModule::Config())),
encoder_queue_(task_queue_factory->CreateTaskQueue(
"AudioEncoder",
TaskQueueFactory::Priority::NORMAL)) {
audio_coding_->RegisterTransportCallback(this);
}
AudioEgress::~AudioEgress() {
audio_coding_->RegisterTransportCallback(nullptr);
}
bool AudioEgress::IsSending() const {
return rtp_rtcp_->SendingMedia();
}
void AudioEgress::SetEncoder(int payload_type,
const SdpAudioFormat& encoder_format,
std::unique_ptr<AudioEncoder> encoder) {
RTC_DCHECK_GE(payload_type, 0);
RTC_DCHECK_LE(payload_type, 127);
SetEncoderFormat(encoder_format);
// The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
// as well as some other things, so we collect this info and send it along.
rtp_rtcp_->RegisterSendPayloadFrequency(payload_type,
encoder->RtpTimestampRateHz());
rtp_sender_audio_.RegisterAudioPayload("audio", payload_type,
encoder->RtpTimestampRateHz(),
encoder->NumChannels(), 0);
audio_coding_->SetEncoder(std::move(encoder));
}
void AudioEgress::StartSend() {
rtp_rtcp_->SetSendingMediaStatus(true);
}
void AudioEgress::StopSend() {
rtp_rtcp_->SetSendingMediaStatus(false);
}
void AudioEgress::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
RTC_DCHECK_LE(audio_frame->num_channels_, 8);
encoder_queue_.PostTask(
[this, audio_frame = std::move(audio_frame)]() mutable {
RTC_DCHECK_RUN_ON(&encoder_queue_);
if (!rtp_rtcp_->SendingMedia()) {
return;
}
AudioFrameOperations::Mute(audio_frame.get(),
encoder_context_.previously_muted_,
encoder_context_.mute_);
encoder_context_.previously_muted_ = encoder_context_.mute_;
audio_frame->timestamp_ = encoder_context_.frame_rtp_timestamp_;
// This call will trigger AudioPacketizationCallback::SendData if
// encoding is done and payload is ready for packetization and
// transmission. Otherwise, it will return without invoking the
// callback.
if (audio_coding_->Add10MsData(*audio_frame) < 0) {
RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
return;
}
encoder_context_.frame_rtp_timestamp_ +=
rtc::dchecked_cast<uint32_t>(audio_frame->samples_per_channel_);
});
}
int32_t AudioEgress::SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_size) {
RTC_DCHECK_RUN_ON(&encoder_queue_);
rtc::ArrayView<const uint8_t> payload(payload_data, payload_size);
// Currently we don't get a capture time from downstream modules (ADM,
// AudioTransportImpl).
// TODO(natim@webrtc.org): Integrate once it's ready.
constexpr uint32_t kUndefinedCaptureTime = -1;
// Push data from ACM to RTP/RTCP-module to deliver audio frame for
// packetization.
if (!rtp_rtcp_->OnSendingRtpFrame(timestamp, kUndefinedCaptureTime,
payload_type,
/*force_sender_report=*/false)) {
return -1;
}
const uint32_t rtp_timestamp = timestamp + rtp_rtcp_->StartTimestamp();
// This call will trigger Transport::SendPacket() from the RTP/RTCP module.
if (!rtp_sender_audio_.SendAudio(frame_type, payload_type, rtp_timestamp,
payload.data(), payload.size())) {
RTC_DLOG(LS_ERROR)
<< "AudioEgress::SendData() failed to send data to RTP/RTCP module";
return -1;
}
return 0;
}
void AudioEgress::RegisterTelephoneEventType(int rtp_payload_type,
int sample_rate_hz) {
RTC_DCHECK_GE(rtp_payload_type, 0);
RTC_DCHECK_LE(rtp_payload_type, 127);
rtp_rtcp_->RegisterSendPayloadFrequency(rtp_payload_type, sample_rate_hz);
rtp_sender_audio_.RegisterAudioPayload("telephone-event", rtp_payload_type,
sample_rate_hz, 0, 0);
}
bool AudioEgress::SendTelephoneEvent(int dtmf_event, int duration_ms) {
RTC_DCHECK_GE(dtmf_event, 0);
RTC_DCHECK_LE(dtmf_event, 255);
RTC_DCHECK_GE(duration_ms, 0);
RTC_DCHECK_LE(duration_ms, 65535);
if (!IsSending()) {
return false;
}
constexpr int kTelephoneEventAttenuationdB = 10;
if (rtp_sender_audio_.SendTelephoneEvent(dtmf_event, duration_ms,
kTelephoneEventAttenuationdB) != 0) {
RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
return false;
}
return true;
}
void AudioEgress::SetMute(bool mute) {
encoder_queue_.PostTask([this, mute] {
RTC_DCHECK_RUN_ON(&encoder_queue_);
encoder_context_.mute_ = mute;
});
}
} // namespace webrtc
|
