gentoo auto-resync : 15:09:2023 - 10:49:51

1 files changed, 233 insertions, 0 deletions

diff --git a/media-video/pipewire/files/0.3.80/0001-aes-support-both-webrtc-versions.patch b/media-video/pipewire/files/0.3.80/0001-aes-support-both-webrtc-versions.patch
new file mode 100644
index 000000000000..f2afdde4f4be
--- /dev/null
+++ b/media-video/pipewire/files/0.3.80/0001-aes-support-both-webrtc-versions.patch
@@ -0,0 +1,233 @@
+https://bugs.gentoo.org/913693
+https://gitlab.freedesktop.org/pipewire/pipewire/-/commit/1f1c308c9766312e684f0b53fc2d1422c7414d31
+
+From 1f1c308c9766312e684f0b53fc2d1422c7414d31 Mon Sep 17 00:00:00 2001
+From: Wim Taymans <wtaymans@redhat.com>
+Date: Thu, 14 Sep 2023 15:35:40 +0200
+Subject: [PATCH] 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.
+--- a/meson.build
++++ b/meson.build
+@@ -377,9 +377,17 @@ cdata.set('HAVE_GSTREAMER_DEVICE_PROVIDER', get_option('gstreamer-device-provide
+ 
+ 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())
++  required : false)
++cdata.set('HAVE_WEBRTC1', webrtc_dep.found())
++if webrtc_dep.found()
++  summary({'WebRTC Echo Canceling >= 1.2': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
++else
++  webrtc_dep = dependency('webrtc-audio-processing',
++    version : ['>= 0.2', '< 1.0'],
++    required : get_option('echo-cancel-webrtc'))
++  cdata.set('HAVE_WEBRTC', webrtc_dep.found())
++  summary({'WebRTC Echo Canceling < 1.0': webrtc_dep.found()}, bool_yn: true, section: 'Misc dependencies')
++endif
+ 
+ # On FreeBSD and MidnightBSD, epoll-shim library is required for eventfd() and timerfd()
+ epoll_shim_dep = (host_machine.system() == 'freebsd' or host_machine.system() == 'midnightbsd'
+--- a/spa/plugins/aec/aec-webrtc.cpp
++++ b/spa/plugins/aec/aec-webrtc.cpp
+@@ -3,6 +3,8 @@
+ /* SPDX-FileCopyrightText: Copyright © 2021 Arun Raghavan <arun@asymptotic.io> */
+ /* SPDX-License-Identifier: MIT */
+ 
++#include "config.h"
++
+ #include <memory>
+ #include <utility>
+ 
+@@ -13,7 +15,13 @@
+ #include <spa/utils/json.h>
+ #include <spa/support/plugin.h>
+ 
++#ifdef HAVE_WEBRTC
++#include <webrtc/modules/audio_processing/include/audio_processing.h>
++#include <webrtc/modules/interface/module_common_types.h>
++#include <webrtc/system_wrappers/include/trace.h>
++#else
+ #include <modules/audio_processing/include/audio_processing.h>
++#endif
+ 
+ struct impl_data {
+ 	struct spa_handle handle;
+@@ -39,6 +47,54 @@ static bool webrtc_get_spa_bool(const struct spa_dict *args, const char *key, bo
+ 	return default_value;
+ }
+ 
++#ifdef HAVE_WEBRTC
++/* [ 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;
++}
++#endif
++
+ 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)
+@@ -48,9 +104,18 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
+ 
+ 	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);
+-
++#ifdef HAVE_WEBRTC
++	bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true);
++	bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true);
++	// 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);
++#else
++	bool transient_suppression = webrtc_get_spa_bool(args, "webrtc.transient_suppression", true);
++#endif
+ 	// 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);
+@@ -59,6 +124,51 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
+ 	// This filter will modify playback buffer (when calling ProcessReverseStream), but now
+ 	// playback buffer modifications are discarded.
+ 
++#ifdef HAVE_WEBRTC
++	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));
++		}
++	}
++#else
+ 	webrtc::AudioProcessing::Config config;
+ 	config.echo_canceller.enabled = true;
+ 	// FIXME: Example code enables both gain controllers, but that seems sus
+@@ -73,6 +183,7 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
+ 	// FIXME: expose pre/postamp gain
+ 	config.transient_suppression.enabled = transient_suppression;
+ 	config.voice_detection.enabled = voice_detection;
++#endif
+ 
+ 	webrtc::ProcessingConfig pconfig = {{
+ 		webrtc::StreamConfig(rec_info->rate, rec_info->channels, false), /* input stream */
+@@ -81,15 +192,35 @@ static int webrtc_init2(void *object, const struct spa_dict *args,
+ 		webrtc::StreamConfig(play_info->rate, play_info->channels, false), /* reverse output stream */
+ 	}};
+ 
++#ifdef HAVE_WEBRTC
++	auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
++#else
+ 	auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessingBuilder().Create());
+ 
+ 	apm->ApplyConfig(config);
++#endif
+ 
+ 	if ((res = apm->Initialize(pconfig)) != webrtc::AudioProcessing::kNoError) {
+ 		spa_log_error(impl->log, "Error initialising webrtc audio processing module: %d", res);
+ 		return -EINVAL;
+ 	}
+ 
++#ifdef HAVE_WEBRTC
++	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);
++#endif
+ 	impl->apm = std::move(apm);
+ 	impl->rec_info = *rec_info;
+ 	impl->out_info = *out_info;
+-- 
+GitLab