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 (c) 2012 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 "modules/audio_mixer/audio_mixer_impl.h"
#include <stdint.h>
#include <algorithm>
#include <iterator>
#include <type_traits>
#include <utility>
#include "modules/audio_mixer/audio_frame_manipulator.h"
#include "modules/audio_mixer/default_output_rate_calculator.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
struct AudioMixerImpl::SourceStatus {
explicit SourceStatus(Source* audio_source) : audio_source(audio_source) {}
Source* audio_source = nullptr;
// A frame that will be passed to audio_source->GetAudioFrameWithInfo.
AudioFrame audio_frame;
};
namespace {
std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>>::const_iterator
FindSourceInList(
AudioMixerImpl::Source const* audio_source,
std::vector<std::unique_ptr<AudioMixerImpl::SourceStatus>> const*
audio_source_list) {
return std::find_if(
audio_source_list->begin(), audio_source_list->end(),
[audio_source](const std::unique_ptr<AudioMixerImpl::SourceStatus>& p) {
return p->audio_source == audio_source;
});
}
} // namespace
struct AudioMixerImpl::HelperContainers {
void resize(size_t size) {
audio_to_mix.resize(size);
preferred_rates.resize(size);
}
std::vector<AudioFrame*> audio_to_mix;
std::vector<int> preferred_rates;
};
AudioMixerImpl::AudioMixerImpl(
std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter)
: output_rate_calculator_(std::move(output_rate_calculator)),
audio_source_list_(),
helper_containers_(std::make_unique<HelperContainers>()),
frame_combiner_(use_limiter) {}
AudioMixerImpl::~AudioMixerImpl() {}
rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create() {
return Create(std::unique_ptr<DefaultOutputRateCalculator>(
new DefaultOutputRateCalculator()),
/*use_limiter=*/true);
}
rtc::scoped_refptr<AudioMixerImpl> AudioMixerImpl::Create(
std::unique_ptr<OutputRateCalculator> output_rate_calculator,
bool use_limiter) {
return rtc::make_ref_counted<AudioMixerImpl>(
std::move(output_rate_calculator), use_limiter);
}
void AudioMixerImpl::Mix(size_t number_of_channels,
AudioFrame* audio_frame_for_mixing) {
TRACE_EVENT0("webrtc", "AudioMixerImpl::Mix");
RTC_DCHECK(number_of_channels >= 1);
MutexLock lock(&mutex_);
size_t number_of_streams = audio_source_list_.size();
std::transform(audio_source_list_.begin(), audio_source_list_.end(),
helper_containers_->preferred_rates.begin(),
[&](std::unique_ptr<SourceStatus>& a) {
return a->audio_source->PreferredSampleRate();
});
int output_frequency = output_rate_calculator_->CalculateOutputRateFromRange(
rtc::ArrayView<const int>(helper_containers_->preferred_rates.data(),
number_of_streams));
frame_combiner_.Combine(GetAudioFromSources(output_frequency),
number_of_channels, output_frequency,
number_of_streams, audio_frame_for_mixing);
}
bool AudioMixerImpl::AddSource(Source* audio_source) {
RTC_DCHECK(audio_source);
MutexLock lock(&mutex_);
RTC_DCHECK(FindSourceInList(audio_source, &audio_source_list_) ==
audio_source_list_.end())
<< "Source already added to mixer";
audio_source_list_.emplace_back(new SourceStatus(audio_source));
helper_containers_->resize(audio_source_list_.size());
UpdateSourceCountStats();
return true;
}
void AudioMixerImpl::RemoveSource(Source* audio_source) {
RTC_DCHECK(audio_source);
MutexLock lock(&mutex_);
const auto iter = FindSourceInList(audio_source, &audio_source_list_);
RTC_DCHECK(iter != audio_source_list_.end()) << "Source not present in mixer";
audio_source_list_.erase(iter);
}
rtc::ArrayView<AudioFrame* const> AudioMixerImpl::GetAudioFromSources(
int output_frequency) {
int audio_to_mix_count = 0;
for (auto& source_and_status : audio_source_list_) {
const auto audio_frame_info =
source_and_status->audio_source->GetAudioFrameWithInfo(
output_frequency, &source_and_status->audio_frame);
switch (audio_frame_info) {
case Source::AudioFrameInfo::kError:
RTC_LOG_F(LS_WARNING)
<< "failed to GetAudioFrameWithInfo() from source";
break;
case Source::AudioFrameInfo::kMuted:
break;
case Source::AudioFrameInfo::kNormal:
helper_containers_->audio_to_mix[audio_to_mix_count++] =
&source_and_status->audio_frame;
}
}
return rtc::ArrayView<AudioFrame* const>(
helper_containers_->audio_to_mix.data(), audio_to_mix_count);
}
void AudioMixerImpl::UpdateSourceCountStats() {
size_t current_source_count = audio_source_list_.size();
// Log to the histogram whenever the maximum number of sources increases.
if (current_source_count > max_source_count_ever_) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AudioMixer.NewHighestSourceCount",
current_source_count, 1, 20, 20);
max_source_count_ever_ = current_source_count;
}
}
} // namespace webrtc
|
