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
222
223
224
225
226
227
228
229
230
231
232
233
234
|
/***
This file is part of PulseAudio.
Copyright 2011 Collabora Ltd.
Contributor: Arun Raghavan <arun.raghavan@collabora.co.uk>
PulseAudio is free software; you can redistribute it and/or modify
it under the terms of the GNU Lesser General Public License as published
by the Free Software Foundation; either version 2.1 of the License,
or (at your option) any later version.
PulseAudio is distributed in the hope that it will be useful, but
WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
General Public License for more details.
You should have received a copy of the GNU Lesser General Public License
along with PulseAudio; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
USA.
***/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <pulse/cdecl.h>
PA_C_DECL_BEGIN
#include <pulsecore/core-util.h>
#include <pulsecore/modargs.h>
#include <pulse/timeval.h>
#include "echo-cancel.h"
PA_C_DECL_END
#include <audio_processing.h>
#include <module_common_types.h>
#define BLOCK_SIZE_US 10000
#define DEFAULT_HIGH_PASS_FILTER TRUE
#define DEFAULT_NOISE_SUPPRESSION TRUE
#define DEFAULT_ANALOG_GAIN_CONTROL FALSE
#define DEFAULT_DIGITAL_GAIN_CONTROL TRUE
#define DEFAULT_MOBILE FALSE
#define DEFAULT_ROUTING_MODE "speakerphone"
#define DEFAULT_COMFORT_NOISE TRUE
static const char* const valid_modargs[] = {
"high_pass_filter",
"noise_suppression",
"analog_gain_control",
"digital_gain_control",
"mobile",
"routing_mode",
"comfort_noise",
NULL
};
static int routing_mode_from_string(const char *rmode) {
if (pa_streq(rmode, "quiet-earpiece-or-headset"))
return webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
else if (pa_streq(rmode, "earpiece"))
return webrtc::EchoControlMobile::kEarpiece;
else if (pa_streq(rmode, "loud-earpiece"))
return webrtc::EchoControlMobile::kLoudEarpiece;
else if (pa_streq(rmode, "speakerphone"))
return webrtc::EchoControlMobile::kSpeakerphone;
else if (pa_streq(rmode, "loud-speakerphone"))
return webrtc::EchoControlMobile::kLoudSpeakerphone;
else
return -1;
}
pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
pa_sample_spec *source_ss, pa_channel_map *source_map,
pa_sample_spec *sink_ss, pa_channel_map *sink_map,
uint32_t *blocksize, const char *args)
{
webrtc::AudioProcessing *apm = NULL;
pa_bool_t hpf, ns, agc, dgc, mobile, cn;
int rm;
pa_modargs *ma;
if (!(ma = pa_modargs_new(args, valid_modargs))) {
pa_log("Failed to parse submodule arguments.");
goto fail;
}
hpf = DEFAULT_HIGH_PASS_FILTER;
if (pa_modargs_get_value_boolean(ma, "high_pass_filter", &hpf) < 0) {
pa_log("Failed to parse high_pass_filter value");
goto fail;
}
ns = DEFAULT_NOISE_SUPPRESSION;
if (pa_modargs_get_value_boolean(ma, "noise_suppression", &ns) < 0) {
pa_log("Failed to parse noise_suppression value");
goto fail;
}
agc = DEFAULT_ANALOG_GAIN_CONTROL;
if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &agc) < 0) {
pa_log("Failed to parse analog_gain_control value");
goto fail;
}
dgc = DEFAULT_DIGITAL_GAIN_CONTROL;
if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &dgc) < 0) {
pa_log("Failed to parse digital_gain_control value");
goto fail;
}
if (agc && dgc) {
pa_log("You must pick only one between analog and digital gain control");
goto fail;
}
mobile = DEFAULT_MOBILE;
if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
pa_log("Failed to parse mobile value");
goto fail;
}
if (mobile) {
if ((rm = routing_mode_from_string(pa_modargs_get_value(ma, "routing_mode", DEFAULT_ROUTING_MODE))) < 0) {
pa_log("Failed to parse routing_mode value");
goto fail;
}
cn = DEFAULT_COMFORT_NOISE;
if (pa_modargs_get_value_boolean(ma, "comfort_noise", &cn) < 0) {
pa_log("Failed to parse cn value");
goto fail;
}
} else {
if (pa_modargs_get_value(ma, "comfort_noise", NULL) || pa_modargs_get_value(ma, "routing_mode", NULL)) {
pa_log("The routing_mode and comfort_noise options are only valid with mobile=true");
goto fail;
}
}
apm = webrtc::AudioProcessing::Create(0);
source_ss->format = PA_SAMPLE_S16NE;
*sink_ss = *source_ss;
/* FIXME: the implementation actually allows a different number of
* source/sink channels. Do we want to support that? */
*sink_map = *source_map;
apm->set_sample_rate_hz(source_ss->rate);
apm->set_num_channels(source_ss->channels, source_ss->channels);
apm->set_num_reverse_channels(sink_ss->channels);
if (hpf)
apm->high_pass_filter()->Enable(true);
if (!mobile) {
apm->echo_cancellation()->enable_drift_compensation(false);
apm->echo_cancellation()->Enable(true);
} else {
apm->echo_control_mobile()->set_routing_mode(static_cast<webrtc::EchoControlMobile::RoutingMode>(rm));
apm->echo_control_mobile()->enable_comfort_noise(cn);
apm->echo_control_mobile()->Enable(true);
}
if (ns) {
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
apm->noise_suppression()->Enable(true);
}
if (agc || dgc) {
if (mobile && rm <= webrtc::EchoControlMobile::kEarpiece)
/* Maybe this should be a knob, but we've got a lot of knobs already */
apm->gain_control()->set_mode(webrtc::GainControl::kFixedDigital);
else if (dgc)
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
else {
/* FIXME: Hook up for analog AGC */
pa_log("Analog gain control isn't implemented yet -- using ditital gain control.");
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
}
}
apm->voice_detection()->Enable(true);
ec->params.priv.webrtc.apm = apm;
ec->params.priv.webrtc.sample_spec = *source_ss;
ec->params.priv.webrtc.blocksize = *blocksize = (uint64_t)pa_bytes_per_second(source_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
pa_modargs_free(ma);
return TRUE;
fail:
if (ma)
pa_modargs_free(ma);
if (apm)
webrtc::AudioProcessing::Destroy(apm);
return FALSE;
}
void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
webrtc::AudioFrame play_frame, out_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
play_frame._audioChannel = ss->channels;
play_frame._frequencyInHz = ss->rate;
play_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
memcpy(play_frame._payloadData, play, ec->params.priv.webrtc.blocksize);
out_frame._audioChannel = ss->channels;
out_frame._frequencyInHz = ss->rate;
out_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
memcpy(out_frame._payloadData, rec, ec->params.priv.webrtc.blocksize);
apm->AnalyzeReverseStream(&play_frame);
apm->set_stream_delay_ms(0);
apm->ProcessStream(&out_frame);
memcpy(out, out_frame._payloadData, ec->params.priv.webrtc.blocksize);
}
void pa_webrtc_ec_done(pa_echo_canceller *ec) {
if (ec->params.priv.webrtc.apm) {
webrtc::AudioProcessing::Destroy((webrtc::AudioProcessing*)ec->params.priv.webrtc.apm);
ec->params.priv.webrtc.apm = NULL;
}
}
|
