pulseaudio/src/modules/echo-cancel/webrtc.cc

273 lines
9.2 KiB
C++
Raw Normal View History

/***
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
#define DEFAULT_DRIFT_COMPENSATION FALSE
static const char* const valid_modargs[] = {
"high_pass_filter",
"noise_suppression",
"analog_gain_control",
"digital_gain_control",
"mobile",
"routing_mode",
"comfort_noise",
"drift_compensation",
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;
}
ec->params.drift_compensation = DEFAULT_DRIFT_COMPENSATION;
if (pa_modargs_get_value_boolean(ma, "drift_compensation", &ec->params.drift_compensation) < 0) {
pa_log("Failed to parse drift_compensation value");
goto fail;
}
if (mobile) {
if (ec->params.drift_compensation) {
pa_log("Can't use drift_compensation in mobile mode");
goto fail;
}
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) {
if (ec->params.drift_compensation) {
apm->echo_cancellation()->set_device_sample_rate_hz(source_ss->rate);
apm->echo_cancellation()->enable_drift_compensation(true);
} else {
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_play(pa_echo_canceller *ec, const uint8_t *play) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
webrtc::AudioFrame play_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);
apm->AnalyzeReverseStream(&play_frame);
}
void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
webrtc::AudioFrame out_frame;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
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->set_stream_delay_ms(0);
apm->ProcessStream(&out_frame);
memcpy(out, out_frame._payloadData, ec->params.priv.webrtc.blocksize);
}
void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.priv.webrtc.blocksize / pa_frame_size(ss));
}
void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
pa_webrtc_ec_play(ec, play);
pa_webrtc_ec_record(ec, rec, out);
}
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;
}
}