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Update version to 0.3.81

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This commit is contained in:
Wim Taymans 2023-10-06 12:47:54 +02:00
parent d31dae772d
commit 39c2309d40
3 changed files with 5 additions and 246 deletions
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243
0001-Revert-aec-webrtc-Bump-to-webrtc-audio-processing-1.patch
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@ -1,243 +0,0 @@
From c9914a9f5ead68d67ce7ca56a69d627fbffc3a26 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Thu, 14 Sep 2023 11:27:11 +0200
Subject: [PATCH] Revert "aec-webrtc: Bump to webrtc-audio-processing-1"
This reverts commit c842ef7071d175903df478538bbc1a39440f1b11.
---
.gitignore | 1 -
meson.build | 4 +-
spa/plugins/aec/aec-webrtc.cpp | 136 +++++++++++++++++++----
subprojects/webrtc-audio-processing.wrap | 8 --
4 files changed, 118 insertions(+), 31 deletions(-)
delete mode 100644 subprojects/webrtc-audio-processing.wrap
diff --git a/.gitignore b/.gitignore
index 2ebb460fd..aa606d659 100644
--- a/.gitignore
+++ b/.gitignore
@@ -18,7 +18,6 @@ subprojects/gtest.wrap
subprojects/libyaml.wrap
subprojects/libyaml
subprojects/libcamera
-subprojects/webrtc-audio-processing
# Created by https://www.gitignore.io/api/vim
diff --git a/meson.build b/meson.build
index 756a2ff4c..fce4b9477 100644
--- a/meson.build
+++ b/meson.build
@@ -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-1',
- version : ['>= 1.2' ],
+webrtc_dep = dependency('webrtc-audio-processing',
+ version : ['>= 0.2', '< 1.0'],
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())
diff --git a/spa/plugins/aec/aec-webrtc.cpp b/spa/plugins/aec/aec-webrtc.cpp
index 44719275f..cb19392c4 100644
--- a/spa/plugins/aec/aec-webrtc.cpp
+++ b/spa/plugins/aec/aec-webrtc.cpp
@@ -13,7 +13,9 @@
#include <spa/utils/json.h>
#include <spa/support/plugin.h>
-#include <modules/audio_processing/include/audio_processing.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>
struct impl_data {
struct spa_handle handle;
@@ -39,6 +41,53 @@ 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)
@@ -46,33 +95,69 @@ 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::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::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::ProcessingConfig pconfig = {{
webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */
@@ -81,15 +166,26 @@ 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::AudioProcessingBuilder().Create());
-
- apm->ApplyConfig(config);
-
+ auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(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;
diff --git a/subprojects/webrtc-audio-processing.wrap b/subprojects/webrtc-audio-processing.wrap
deleted file mode 100644
index 1382212d6..000000000
--- a/subprojects/webrtc-audio-processing.wrap
+++ /dev/null
@@ -1,8 +0,0 @@
-[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
--
2.41.0

6
pipewire.spec
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@ -1,6 +1,6 @@
%global majorversion 0
%global minorversion 3
%global microversion 80
%global microversion 81
%global apiversion 0.3
%global spaversion 0.2
@ -80,7 +80,6 @@ Source1: pipewire.sysusers
## upstreamable patches
## fedora patches
Patch0001: 0001-Revert-aec-webrtc-Bump-to-webrtc-audio-processing-1.patch
BuildRequires: gettext
BuildRequires: meson >= 0.59.0
@ -723,6 +722,9 @@ systemctl --no-reload preset --global pipewire.socket >/dev/null 2>&1 || :
%endif
%changelog
* Fri Oct 6 2023 Wim Taymans <wtaymans@redhat.com> - 0.3.81-1
- Update version to 0.3.81
* Thu Sep 14 2023 Wim Taymans <wtaymans@redhat.com> - 0.3.80-1
- Update version to 0.3.80
- Revert webrtc echo-cancel updates until fedora has newer version.

2
sources
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@ -1 +1 @@
SHA512 (pipewire-0.3.80.tar.gz) = b8b1df0034d23d782c94cbafb2f7e4030b26c3370e2f2ce2b1b86d51d3313a23247a186871a81c247e2454ca772f9edc9db8cf08bc3adc0a62fe9191ed238aee
SHA512 (pipewire-0.3.81.tar.gz) = 2cc15b6253989f471c5468bd30add96843efc8263825369c6df94cc7e3b67fe1dbffabc677edc44e063c8c35ff09287615bb4d668999b69c81fc36854af4fd51