2011-09-19 13:41:13 +05:30
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/***
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This file is part of PulseAudio.
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Copyright 2011 Collabora Ltd.
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Contributor: Arun Raghavan <arun.raghavan@collabora.co.uk>
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PulseAudio is free software; you can redistribute it and/or modify
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it under the terms of the GNU Lesser General Public License as published
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by the Free Software Foundation; either version 2.1 of the License,
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or (at your option) any later version.
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PulseAudio is distributed in the hope that it will be useful, but
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WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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General Public License for more details.
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You should have received a copy of the GNU Lesser General Public License
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along with PulseAudio; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
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USA.
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***/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <pulse/cdecl.h>
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PA_C_DECL_BEGIN
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#include <pulsecore/core-util.h>
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#include <pulsecore/modargs.h>
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#include <pulse/timeval.h>
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#include "echo-cancel.h"
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PA_C_DECL_END
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#include <audio_processing.h>
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#include <module_common_types.h>
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#define BLOCK_SIZE_US 10000
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#define DEFAULT_HIGH_PASS_FILTER TRUE
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#define DEFAULT_NOISE_SUPPRESSION TRUE
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#define DEFAULT_ANALOG_GAIN_CONTROL FALSE
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#define DEFAULT_DIGITAL_GAIN_CONTROL TRUE
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#define DEFAULT_MOBILE FALSE
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#define DEFAULT_ROUTING_MODE "speakerphone"
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#define DEFAULT_COMFORT_NOISE TRUE
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2011-10-07 16:28:11 +05:30
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#define DEFAULT_DRIFT_COMPENSATION FALSE
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2011-09-19 13:41:13 +05:30
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static const char* const valid_modargs[] = {
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"high_pass_filter",
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"noise_suppression",
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"analog_gain_control",
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"digital_gain_control",
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"mobile",
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"routing_mode",
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"comfort_noise",
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2011-10-07 16:28:11 +05:30
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"drift_compensation",
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2011-09-19 13:41:13 +05:30
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NULL
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};
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static int routing_mode_from_string(const char *rmode) {
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if (pa_streq(rmode, "quiet-earpiece-or-headset"))
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return webrtc::EchoControlMobile::kQuietEarpieceOrHeadset;
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else if (pa_streq(rmode, "earpiece"))
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return webrtc::EchoControlMobile::kEarpiece;
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else if (pa_streq(rmode, "loud-earpiece"))
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return webrtc::EchoControlMobile::kLoudEarpiece;
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else if (pa_streq(rmode, "speakerphone"))
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return webrtc::EchoControlMobile::kSpeakerphone;
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else if (pa_streq(rmode, "loud-speakerphone"))
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return webrtc::EchoControlMobile::kLoudSpeakerphone;
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else
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return -1;
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}
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pa_bool_t pa_webrtc_ec_init(pa_core *c, pa_echo_canceller *ec,
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pa_sample_spec *source_ss, pa_channel_map *source_map,
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pa_sample_spec *sink_ss, pa_channel_map *sink_map,
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uint32_t *blocksize, const char *args)
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{
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webrtc::AudioProcessing *apm = NULL;
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pa_bool_t hpf, ns, agc, dgc, mobile, cn;
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int rm;
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pa_modargs *ma;
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if (!(ma = pa_modargs_new(args, valid_modargs))) {
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pa_log("Failed to parse submodule arguments.");
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goto fail;
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}
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hpf = DEFAULT_HIGH_PASS_FILTER;
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if (pa_modargs_get_value_boolean(ma, "high_pass_filter", &hpf) < 0) {
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pa_log("Failed to parse high_pass_filter value");
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goto fail;
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}
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ns = DEFAULT_NOISE_SUPPRESSION;
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if (pa_modargs_get_value_boolean(ma, "noise_suppression", &ns) < 0) {
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pa_log("Failed to parse noise_suppression value");
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goto fail;
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}
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agc = DEFAULT_ANALOG_GAIN_CONTROL;
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if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &agc) < 0) {
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pa_log("Failed to parse analog_gain_control value");
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goto fail;
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}
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dgc = DEFAULT_DIGITAL_GAIN_CONTROL;
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if (pa_modargs_get_value_boolean(ma, "analog_gain_control", &dgc) < 0) {
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pa_log("Failed to parse digital_gain_control value");
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goto fail;
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}
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if (agc && dgc) {
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pa_log("You must pick only one between analog and digital gain control");
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goto fail;
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}
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mobile = DEFAULT_MOBILE;
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if (pa_modargs_get_value_boolean(ma, "mobile", &mobile) < 0) {
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pa_log("Failed to parse mobile value");
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goto fail;
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}
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2011-10-07 16:28:11 +05:30
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ec->params.drift_compensation = DEFAULT_DRIFT_COMPENSATION;
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if (pa_modargs_get_value_boolean(ma, "drift_compensation", &ec->params.drift_compensation) < 0) {
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pa_log("Failed to parse drift_compensation value");
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goto fail;
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}
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2011-09-19 13:41:13 +05:30
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if (mobile) {
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2011-10-07 16:28:11 +05:30
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if (ec->params.drift_compensation) {
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pa_log("Can't use drift_compensation in mobile mode");
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goto fail;
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}
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2011-09-19 13:41:13 +05:30
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if ((rm = routing_mode_from_string(pa_modargs_get_value(ma, "routing_mode", DEFAULT_ROUTING_MODE))) < 0) {
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pa_log("Failed to parse routing_mode value");
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goto fail;
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}
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cn = DEFAULT_COMFORT_NOISE;
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if (pa_modargs_get_value_boolean(ma, "comfort_noise", &cn) < 0) {
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pa_log("Failed to parse cn value");
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goto fail;
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}
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} else {
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if (pa_modargs_get_value(ma, "comfort_noise", NULL) || pa_modargs_get_value(ma, "routing_mode", NULL)) {
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pa_log("The routing_mode and comfort_noise options are only valid with mobile=true");
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goto fail;
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}
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}
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apm = webrtc::AudioProcessing::Create(0);
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source_ss->format = PA_SAMPLE_S16NE;
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*sink_ss = *source_ss;
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/* FIXME: the implementation actually allows a different number of
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* source/sink channels. Do we want to support that? */
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*sink_map = *source_map;
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apm->set_sample_rate_hz(source_ss->rate);
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apm->set_num_channels(source_ss->channels, source_ss->channels);
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apm->set_num_reverse_channels(sink_ss->channels);
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if (hpf)
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apm->high_pass_filter()->Enable(true);
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if (!mobile) {
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2011-10-07 16:28:11 +05:30
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if (ec->params.drift_compensation) {
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apm->echo_cancellation()->set_device_sample_rate_hz(source_ss->rate);
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apm->echo_cancellation()->enable_drift_compensation(true);
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} else {
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apm->echo_cancellation()->enable_drift_compensation(false);
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}
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2011-09-19 13:41:13 +05:30
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apm->echo_cancellation()->Enable(true);
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} else {
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apm->echo_control_mobile()->set_routing_mode(static_cast<webrtc::EchoControlMobile::RoutingMode>(rm));
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apm->echo_control_mobile()->enable_comfort_noise(cn);
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apm->echo_control_mobile()->Enable(true);
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}
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if (ns) {
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apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
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apm->noise_suppression()->Enable(true);
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}
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if (agc || dgc) {
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if (mobile && rm <= webrtc::EchoControlMobile::kEarpiece)
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/* Maybe this should be a knob, but we've got a lot of knobs already */
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apm->gain_control()->set_mode(webrtc::GainControl::kFixedDigital);
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else if (dgc)
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apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
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else {
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/* FIXME: Hook up for analog AGC */
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pa_log("Analog gain control isn't implemented yet -- using ditital gain control.");
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apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
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}
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}
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apm->voice_detection()->Enable(true);
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ec->params.priv.webrtc.apm = apm;
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ec->params.priv.webrtc.sample_spec = *source_ss;
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ec->params.priv.webrtc.blocksize = *blocksize = (uint64_t)pa_bytes_per_second(source_ss) * BLOCK_SIZE_US / PA_USEC_PER_SEC;
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pa_modargs_free(ma);
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return TRUE;
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fail:
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if (ma)
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pa_modargs_free(ma);
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if (apm)
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webrtc::AudioProcessing::Destroy(apm);
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return FALSE;
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}
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2011-10-07 16:28:11 +05:30
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void pa_webrtc_ec_play(pa_echo_canceller *ec, const uint8_t *play) {
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2011-09-19 13:41:13 +05:30
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webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
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2011-10-07 16:28:11 +05:30
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webrtc::AudioFrame play_frame;
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2011-09-19 13:41:13 +05:30
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const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
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play_frame._audioChannel = ss->channels;
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play_frame._frequencyInHz = ss->rate;
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play_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
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memcpy(play_frame._payloadData, play, ec->params.priv.webrtc.blocksize);
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2011-10-07 16:28:11 +05:30
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apm->AnalyzeReverseStream(&play_frame);
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}
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void pa_webrtc_ec_record(pa_echo_canceller *ec, const uint8_t *rec, uint8_t *out) {
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webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
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webrtc::AudioFrame out_frame;
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const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
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2011-09-19 13:41:13 +05:30
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out_frame._audioChannel = ss->channels;
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out_frame._frequencyInHz = ss->rate;
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out_frame._payloadDataLengthInSamples = ec->params.priv.webrtc.blocksize / pa_frame_size(ss);
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memcpy(out_frame._payloadData, rec, ec->params.priv.webrtc.blocksize);
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apm->set_stream_delay_ms(0);
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apm->ProcessStream(&out_frame);
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memcpy(out, out_frame._payloadData, ec->params.priv.webrtc.blocksize);
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}
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2011-10-07 16:28:11 +05:30
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void pa_webrtc_ec_set_drift(pa_echo_canceller *ec, float drift) {
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webrtc::AudioProcessing *apm = (webrtc::AudioProcessing*)ec->params.priv.webrtc.apm;
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const pa_sample_spec *ss = &ec->params.priv.webrtc.sample_spec;
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apm->echo_cancellation()->set_stream_drift_samples(drift * ec->params.priv.webrtc.blocksize / pa_frame_size(ss));
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}
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void pa_webrtc_ec_run(pa_echo_canceller *ec, const uint8_t *rec, const uint8_t *play, uint8_t *out) {
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pa_webrtc_ec_play(ec, play);
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pa_webrtc_ec_record(ec, rec, out);
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}
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2011-09-19 13:41:13 +05:30
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void pa_webrtc_ec_done(pa_echo_canceller *ec) {
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if (ec->params.priv.webrtc.apm) {
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webrtc::AudioProcessing::Destroy((webrtc::AudioProcessing*)ec->params.priv.webrtc.apm);
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ec->params.priv.webrtc.apm = NULL;
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}
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}
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