diff options
Diffstat (limited to 'media-video/pipewire/files/0.3.80/0001-aes-support-both-webrtc-versions.patch')
| -rw-r--r-- | media-video/pipewire/files/0.3.80/0001-aes-support-both-webrtc-versions.patch | 233 |
1 files changed, 0 insertions, 233 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 deleted file mode 100644 index f2afdde4f4be..000000000000 --- a/media-video/pipewire/files/0.3.80/0001-aes-support-both-webrtc-versions.patch +++ /dev/null @@ -1,233 +0,0 @@ -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 |