pipewire: update to 0.3.81

srcpkgs/pipewire/patches/support-old-webrtc.patch

@@ -1,226 +0,0 @@
-# Allows building with webrtc above and below v1, will be in v0.3.81
-# Upstream: https://gitlab.freedesktop.org/pipewire/pipewire/-/commit/1f1c308c9766312e684f0b53fc2d1422c7414d31
-diff --git a/meson.build b/meson.build
-index 756a2ff4c5253baf6e6dc4cda74a9abb818a9465..ff09002cb8565bbdcd020763e27abdd2e909994f 100644
---- 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'
-diff --git a/spa/plugins/aec/aec-webrtc.cpp b/spa/plugins/aec/aec-webrtc.cpp
-index 44719275fc4bece08907e6589c7c93e993ebe222..1fb34dd8c3e5b3cd4f5af33bd1bca44fc444c24a 100644
---- 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;

srcpkgs/pipewire/template

@@ -1,6 +1,6 @@
 # Template file for 'pipewire'
 pkgname=pipewire
-version=0.3.80
+version=0.3.81
 revision=1
 build_style=meson
 configure_args="
@@ -35,7 +35,7 @@ license="MIT"
 homepage="https://pipewire.org/"
 changelog="https://gitlab.freedesktop.org/pipewire/pipewire/-/raw/master/NEWS"
 distfiles="https://gitlab.freedesktop.org/pipewire/pipewire/-/archive/${version}/pipewire-${version}.tar.gz"
-checksum=0bbaf3d46649ac2217f0bbf68dda7e62f1f2cf6b7f3025e364cf7205d1499ef9
+checksum=241bd06703e41e30d26e096301337547ef5be1201919996c10045758673e691d
 make_dirs="/var/lib/pipewire 0755 _pipewire _pipewire"
 system_accounts="_pipewire"