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aec-webrtc: Bump to webrtc-audio-processing-1

Upstream updated drops beamforming, adds a new gain controller and
includes a bunch of updates to the AEC engine (internally AEC3).
This commit is contained in:
Arun Raghavan 2023-09-04 11:27:52 -04:00 committed by Wim Taymans
parent be943ca9db
commit c842ef7071
4 changed files with 31 additions and 118 deletions

1
.gitignore vendored
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@ -18,6 +18,7 @@ subprojects/gtest.wrap
subprojects/libyaml.wrap
subprojects/libyaml
subprojects/libcamera
subprojects/webrtc-audio-processing
# Created by https://www.gitignore.io/api/vim

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@ -375,8 +375,8 @@ summary({'gstreamer-device-provider': gst_dp_found}, bool_yn: true, section: 'Ba
cdata.set('HAVE_GSTREAMER_DEVICE_PROVIDER', get_option('gstreamer-device-provider').allowed())
webrtc_dep = dependency('webrtc-audio-processing',
version : ['>= 0.2', '< 1.0'],
webrtc_dep = dependency('webrtc-audio-processing-1',
version : ['>= 1.2' ],
required : get_option('echo-cancel-webrtc'))
summary({'WebRTC Echo Canceling': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
cdata.set('HAVE_WEBRTC', webrtc_dep.found())

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@ -13,9 +13,7 @@
#include <spa/utils/json.h>
#include <spa/support/plugin.h>
#include <webrtc/modules/audio_processing/include/audio_processing.h>
#include <webrtc/modules/interface/module_common_types.h>
#include <webrtc/system_wrappers/include/trace.h>
#include <modules/audio_processing/include/audio_processing.h>
struct impl_data {
struct spa_handle handle;
@ -41,53 +39,6 @@ static bool webrtc_get_spa_bool(const struct spa_dict *args, const char *key, bo
return default_value;
}
/* [ f0 f1 f2 ] */
static int parse_point(struct spa_json *it, float (&f)[3])
{
struct spa_json arr;
int i, res;
if (spa_json_enter_array(it, &arr) <= 0)
return -EINVAL;
for (i = 0; i < 3; i++) {
if ((res = spa_json_get_float(&arr, &f[i])) <= 0)
return -EINVAL;
}
return 0;
}
/* [ point1 point2 ... ] */
static int parse_mic_geometry(struct impl_data *impl, const char *mic_geometry,
std::vector<webrtc::Point>& geometry)
{
int res;
size_t i;
struct spa_json it[2];
spa_json_init(&it[0], mic_geometry, strlen(mic_geometry));
if (spa_json_enter_array(&it[0], &it[1]) <= 0) {
spa_log_error(impl->log, "Error: webrtc.mic-geometry expects an array");
return -EINVAL;
}
for (i = 0; i < geometry.size(); i++) {
float f[3];
if ((res = parse_point(&it[1], f)) < 0) {
spa_log_error(impl->log, "Error: can't parse webrtc.mic-geometry points: %d", res);
return res;
}
spa_log_info(impl->log, "mic %zd position: (%g %g %g)", i, f[0], f[1], f[2]);
geometry[i].c[0] = f[0];
geometry[i].c[1] = f[1];
geometry[i].c[2] = f[2];
}
return 0;
}
static int webrtc_init2(void *object, const struct spa_dict *args,
struct spa_audio_info_raw *rec_info, struct spa_audio_info_raw *out_info,
struct spa_audio_info_raw *play_info)
@ -95,69 +46,33 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
auto impl = static_cast<struct impl_data*>(object);
int res;
bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true);
bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true);
bool high_pass_filter = webrtc_get_spa_bool(args, "webrtc.high_pass_filter", true);
bool noise_suppression = webrtc_get_spa_bool(args, "webrtc.noise_suppression", true);
bool transient_suppression = webrtc_get_spa_bool(args, "webrtc.transient_suppression", true);
bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
// Note: AGC seems to mess up with Agnostic Delay Detection, especially with speech,
// result in very poor performance, disable by default
bool gain_control = webrtc_get_spa_bool(args, "webrtc.gain_control", false);
// Disable experimental flags by default
bool experimental_agc = webrtc_get_spa_bool(args, "webrtc.experimental_agc", false);
bool experimental_ns = webrtc_get_spa_bool(args, "webrtc.experimental_ns", false);
bool beamforming = webrtc_get_spa_bool(args, "webrtc.beamforming", false);
// FIXME: Intelligibility enhancer is not currently supported
// This filter will modify playback buffer (when calling ProcessReverseStream), but now
// playback buffer modifications are discarded.
webrtc::Config config;
config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(extended_filter));
config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(delay_agnostic));
config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(experimental_agc));
config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(experimental_ns));
if (beamforming) {
std::vector<webrtc::Point> geometry(rec_info->channels);
const char *mic_geometry, *target_direction;
/* The beamformer gives a single mono channel */
out_info->channels = 1;
out_info->position[0] = SPA_AUDIO_CHANNEL_MONO;
if ((mic_geometry = spa_dict_lookup(args, "webrtc.mic-geometry")) == NULL) {
spa_log_error(impl->log, "Error: webrtc.beamforming requires webrtc.mic-geometry");
return -EINVAL;
}
if ((res = parse_mic_geometry(impl, mic_geometry, geometry)) < 0)
return res;
if ((target_direction = spa_dict_lookup(args, "webrtc.target-direction")) != NULL) {
webrtc::SphericalPointf direction(0.0f, 0.0f, 0.0f);
struct spa_json it;
float f[3];
spa_json_init(&it, target_direction, strlen(target_direction));
if (parse_point(&it, f) < 0) {
spa_log_error(impl->log, "Error: can't parse target-direction %s",
target_direction);
return -EINVAL;
}
direction.s[0] = f[0];
direction.s[1] = f[1];
direction.s[2] = f[2];
config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry, direction));
} else {
config.Set<webrtc::Beamforming>(new webrtc::Beamforming(true, geometry));
}
}
webrtc::AudioProcessing::Config config;
config.echo_canceller.enabled = true;
// FIXME: Example code enables both gain controllers, but that seems sus
config.gain_controller1.enabled = gain_control;
config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital;
config.gain_controller1.analog_level_minimum = 0;
config.gain_controller1.analog_level_maximum = 255;
config.gain_controller2.enabled = gain_control;
config.high_pass_filter.enabled = high_pass_filter;
config.noise_suppression.enabled = noise_suppression;
config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kHigh;
// FIXME: expose pre/postamp gain
config.transient_suppression.enabled = transient_suppression;
config.voice_detection.enabled = voice_detection;
webrtc::ProcessingConfig pconfig = {{
webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */
@ -166,26 +81,15 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse output stream */
}};
auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessingBuilder().Create());
apm->ApplyConfig(config);
if ((res = apm->Initialize(pconfig)) != webrtc::AudioProcessing::kNoError) {
spa_log_error(impl->log, "Error initialising webrtc audio processing module: %d", res);
return -EINVAL;
}
apm->high_pass_filter()->Enable(high_pass_filter);
// Always disable drift compensation since PipeWire will already do
// drift compensation on all sinks and sources linked to this echo-canceler
apm->echo_cancellation()->enable_drift_compensation(false);
apm->echo_cancellation()->Enable(true);
// TODO: wire up supression levels to args
apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kHighSuppression);
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
apm->noise_suppression()->Enable(noise_suppression);
apm->voice_detection()->Enable(voice_detection);
// TODO: wire up AGC parameters to args
apm->gain_control()->set_analog_level_limits(0, 255);
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
apm->gain_control()->Enable(gain_control);
impl->apm = std::move(apm);
impl->rec_info = *rec_info;
impl->out_info = *out_info;

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@ -0,0 +1,8 @@
[wrap-git]
directory = webrtc-audio-processing
url = https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git
push-url = git@gitlab.freedesktop.org:pulseaudio/webrtc-audio-processing.git
revision = v1.3
[provide]
dependency_names = webrtc-audio-coding-1, webrtc-audio-processing-1