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https://gitlab.freedesktop.org/pipewire/pipewire.git
synced 2025-10-29 05:40:27 -04:00
aec: support both webrtc versions
Version 1 does not seem to be packaged in many distros and so they would need to revert the patch or disable AEC. Enabling both allows for things to move forwards gracefully.
This commit is contained in:
Wim Taymans
parent
a4f3b78dff
commit
1f1c308c97
2 changed files with 143 additions and 4 deletions
14
meson.build
14
meson.build
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@ -377,9 +377,17 @@ cdata.set('HAVE_GSTREAMER_DEVICE_PROVIDER', get_option('gstreamer-device-provide
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webrtc_dep = dependency('webrtc-audio-processing-1',
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version : ['>= 1.2' ],
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required : get_option('echo-cancel-webrtc'))
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summary({'WebRTC Echo Canceling': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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cdata.set('HAVE_WEBRTC', webrtc_dep.found())
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required : false)
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cdata.set('HAVE_WEBRTC1', webrtc_dep.found())
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if webrtc_dep.found()
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summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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else
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webrtc_dep = dependency('webrtc-audio-processing',
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version : ['>= 0.2', '< 1.0'],
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required : get_option('echo-cancel-webrtc'))
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cdata.set('HAVE_WEBRTC', webrtc_dep.found())
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summary({'WebRTC Echo Canceling < 1.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
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endif
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# On FreeBSD and MidnightBSD, epoll-shim library is required for eventfd() and timerfd()
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epoll_shim_dep = (host_machine.system() == 'freebsd' or host_machine.system() == 'midnightbsd'
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@ -3,6 +3,8 @@
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/* SPDX-FileCopyrightText: Copyright © 2021 Arun Raghavan <arun@asymptotic.io> */
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/* SPDX-License-Identifier: MIT */
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#include "config.h"
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#include <memory>
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#include <utility>
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@ -13,7 +15,13 @@
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#include <spa/utils/json.h>
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#include <spa/support/plugin.h>
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#ifdef HAVE_WEBRTC
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#include <webrtc/modules/audio_processing/include/audio_processing.h>
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#include <webrtc/modules/interface/module_common_types.h>
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#include <webrtc/system_wrappers/include/trace.h>
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#else
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#include <modules/audio_processing/include/audio_processing.h>
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#endif
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struct impl_data {
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struct spa_handle handle;
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@ -39,6 +47,54 @@ static bool webrtc_get_spa_bool(const struct spa_dict *args, const char *key, bo
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return default_value;
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}
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#ifdef HAVE_WEBRTC
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/* [ f0 f1 f2 ] */
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static int parse_point(struct spa_json *it, float (&f)[3])
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{
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struct spa_json arr;
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int i, res;
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if (spa_json_enter_array(it, &arr) <= 0)
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return -EINVAL;
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for (i = 0; i < 3; i++) {
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if ((res = spa_json_get_float(&arr, &f[i])) <= 0)
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return -EINVAL;
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}
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return 0;
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}
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/* [ point1 point2 ... ] */
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static int parse_mic_geometry(struct impl_data *impl, const char *mic_geometry,
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std::vector<webrtc::Point>& geometry)
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{
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int res;
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size_t i;
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struct spa_json it[2];
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spa_json_init(&it[0], mic_geometry, strlen(mic_geometry));
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if (spa_json_enter_array(&it[0], &it[1]) <= 0) {
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spa_log_error(impl->log, "Error: webrtc.mic-geometry expects an array");
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return -EINVAL;
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}
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for (i = 0; i < geometry.size(); i++) {
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float f[3];
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if ((res = parse_point(&it[1], f)) < 0) {
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spa_log_error(impl->log, "Error: can't parse webrtc.mic-geometry points: %d", res);
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return res;
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}
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spa_log_info(impl->log, "mic %zd position: (%g %g %g)", i, f[0], f[1], f[2]);
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geometry[i].c[0] = f[0];
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geometry[i].c[1] = f[1];
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geometry[i].c[2] = f[2];
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}
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return 0;
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}
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#endif
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static int webrtc_init2(void *object, const struct spa_dict *args,
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struct spa_audio_info_raw *rec_info, struct spa_audio_info_raw *out_info,
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struct spa_audio_info_raw *play_info)
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@ -48,9 +104,18 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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bool high_pass_filter = webrtc_get_spa_bool(args, "webrtc.high_pass_filter", true);
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bool noise_suppression = webrtc_get_spa_bool(args, "webrtc.noise_suppression", true);
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bool transient_suppression = webrtc_get_spa_bool(args, "webrtc.transient_suppression", true);
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bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
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#ifdef HAVE_WEBRTC
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bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true);
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bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true);
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// Disable experimental flags by default
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bool experimental_agc = webrtc_get_spa_bool(args, "webrtc.experimental_agc", false);
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bool experimental_ns = webrtc_get_spa_bool(args, "webrtc.experimental_ns", false);
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bool beamforming = webrtc_get_spa_bool(args, "webrtc.beamforming", false);
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#else
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bool transient_suppression = webrtc_get_spa_bool(args, "webrtc.transient_suppression", true);
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#endif
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// Note: AGC seems to mess up with Agnostic Delay Detection, especially with speech,
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// result in very poor performance, disable by default
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bool gain_control = webrtc_get_spa_bool(args, "webrtc.gain_control", false);
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@ -59,6 +124,51 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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// This filter will modify playback buffer (when calling ProcessReverseStream), but now
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// playback buffer modifications are discarded.
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#ifdef HAVE_WEBRTC
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webrtc::Config config;
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config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(extended_filter));
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config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(delay_agnostic));
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config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(experimental_agc));
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config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(experimental_ns));
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if (beamforming) {
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std::vector<webrtc::Point> geometry(rec_info->channels);
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const char *mic_geometry, *target_direction;
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/* The beamformer gives a single mono channel */
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out_info->channels = 1;
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out_info->position[0] = SPA_AUDIO_CHANNEL_MONO;
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if ((mic_geometry = spa_dict_lookup(args, "webrtc.mic-geometry")) == NULL) {
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spa_log_error(impl->log, "Error: webrtc.beamforming requires webrtc.mic-geometry");
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return -EINVAL;
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}
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if ((res = parse_mic_geometry(impl, mic_geometry, geometry)) < 0)
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return res;
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if ((target_direction = spa_dict_lookup(args, "webrtc.target-direction")) != NULL) {
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webrtc::SphericalPointf direction(0.0f, 0.0f, 0.0f);
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struct spa_json it;
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float f[3];
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spa_json_init(&it, target_direction, strlen(target_direction));
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if (parse_point(&it, f) < 0) {
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spa_log_error(impl->log, "Error: can't parse target-direction %s",
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target_direction);
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return -EINVAL;
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}
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direction.s[0] = f[0];
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direction.s[1] = f[1];
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direction.s[2] = f[2];
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config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry, direction));
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} else {
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config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
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}
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}
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#else
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webrtc::AudioProcessing::Config config;
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config.echo_canceller.enabled = true;
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// FIXME: Example code enables both gain controllers, but that seems sus
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@ -73,6 +183,7 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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// FIXME: expose pre/postamp gain
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config.transient_suppression.enabled = transient_suppression;
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config.voice_detection.enabled = voice_detection;
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#endif
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webrtc::ProcessingConfig pconfig = {{
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webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */
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@ -81,15 +192,35 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
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webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse output stream */
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}};
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#ifdef HAVE_WEBRTC
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auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
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#else
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auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessingBuilder().Create());
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apm->ApplyConfig(config);
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#endif
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if ((res = apm->Initialize(pconfig)) != webrtc::AudioProcessing::kNoError) {
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spa_log_error(impl->log, "Error initialising webrtc audio processing module: %d", res);
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return -EINVAL;
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}
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#ifdef HAVE_WEBRTC
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apm->high_pass_filter()->Enable(high_pass_filter);
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// Always disable drift compensation since PipeWire will already do
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// drift compensation on all sinks and sources linked to this echo-canceler
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apm->echo_cancellation()->enable_drift_compensation(false);
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apm->echo_cancellation()->Enable(true);
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// TODO: wire up supression levels to args
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apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kHighSuppression);
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apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
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apm->noise_suppression()->Enable(noise_suppression);
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apm->voice_detection()->Enable(voice_detection);
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// TODO: wire up AGC parameters to args
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apm->gain_control()->set_analog_level_limits(0, 255);
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apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
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apm->gain_control()->Enable(gain_control);
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#endif
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impl->apm = std::move(apm);
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impl->rec_info = *rec_info;
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impl->out_info = *out_info;
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