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
|
// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "content/renderer/media/webaudio_capturer_source.h"
#include "base/logging.h"
#include "base/time/time.h"
#include "content/renderer/media/webrtc_audio_capturer.h"
#include "content/renderer/media/webrtc_local_audio_track.h"
using media::AudioBus;
using media::AudioFifo;
using media::AudioParameters;
using media::ChannelLayout;
using media::CHANNEL_LAYOUT_MONO;
using media::CHANNEL_LAYOUT_STEREO;
static const int kMaxNumberOfBuffersInFifo = 5;
namespace content {
WebAudioCapturerSource::WebAudioCapturerSource()
: track_(NULL),
capturer_(NULL),
audio_format_changed_(false) {
}
WebAudioCapturerSource::~WebAudioCapturerSource() {
}
void WebAudioCapturerSource::setFormat(
size_t number_of_channels, float sample_rate) {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
<< sample_rate << ")";
if (number_of_channels > 2) {
// TODO(xians): Handle more than just the mono and stereo cases.
LOG(WARNING) << "WebAudioCapturerSource::setFormat() : unhandled format.";
return;
}
ChannelLayout channel_layout =
number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
base::AutoLock auto_lock(lock_);
// Set the format used by this WebAudioCapturerSource. We are using 10ms data
// as buffer size since that is the native buffer size of WebRtc packet
// running on.
params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
channel_layout, number_of_channels, 0, sample_rate, 16,
sample_rate / 100);
audio_format_changed_ = true;
wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
capture_bus_ = AudioBus::Create(params_);
fifo_.reset(new AudioFifo(
params_.channels(),
kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
}
void WebAudioCapturerSource::Start(
WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(track);
base::AutoLock auto_lock(lock_);
track_ = track;
capturer_ = capturer;
}
void WebAudioCapturerSource::Stop() {
DCHECK(thread_checker_.CalledOnValidThread());
base::AutoLock auto_lock(lock_);
track_ = NULL;
capturer_ = NULL;
}
void WebAudioCapturerSource::consumeAudio(
const blink::WebVector<const float*>& audio_data,
size_t number_of_frames) {
base::AutoLock auto_lock(lock_);
if (!track_)
return;
// Update the downstream client if the audio format has been changed.
if (audio_format_changed_) {
track_->OnSetFormat(params_);
audio_format_changed_ = false;
}
wrapper_bus_->set_frames(number_of_frames);
// Make sure WebKit is honoring what it told us up front
// about the channels.
DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
for (size_t i = 0; i < audio_data.size(); ++i)
wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
// Handle mismatch between WebAudio buffer-size and WebRTC.
int available = fifo_->max_frames() - fifo_->frames();
if (available < static_cast<int>(number_of_frames)) {
NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
return;
}
fifo_->Push(wrapper_bus_.get());
int capture_frames = params_.frames_per_buffer();
base::TimeDelta delay;
int volume = 0;
bool key_pressed = false;
if (capturer_) {
capturer_->GetAudioProcessingParams(&delay, &volume, &key_pressed);
}
while (fifo_->frames() >= capture_frames) {
fifo_->Consume(capture_bus_.get(), 0, capture_frames);
track_->Capture(capture_bus_.get(), delay.InMilliseconds(),
volume, key_pressed);
}
}
} // namespace content
|
