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-rw-r--r--spa/plugins/aec/aec-webrtc.cpp276
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diff --git a/spa/plugins/aec/aec-webrtc.cpp b/spa/plugins/aec/aec-webrtc.cpp
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+++ b/spa/plugins/aec/aec-webrtc.cpp
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+/* PipeWire
+ *
+ * Copyright © 2021 Wim Taymans <wim.taymans@gmail.com>
+ * © 2021 Arun Raghavan <arun@asymptotic.io>
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a
+ * copy of this software and associated documentation files (the "Software"),
+ * to deal in the Software without restriction, including without limitation
+ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
+ * and/or sell copies of the Software, and to permit persons to whom the
+ * Software is furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice (including the next
+ * paragraph) shall be included in all copies or substantial portions of the
+ * Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
+ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+ * DEALINGS IN THE SOFTWARE.
+ */
+
+#include <memory>
+#include <utility>
+
+#include <spa/interfaces/audio/aec.h>
+#include <spa/support/log.h>
+#include <spa/utils/string.h>
+#include <spa/utils/names.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>
+
+struct impl_data {
+ struct spa_handle handle;
+ struct spa_audio_aec aec;
+
+ struct spa_log *log;
+ std::unique_ptr<webrtc::AudioProcessing> apm;
+ spa_audio_info_raw info;
+ std::unique_ptr<float *[]> play_buffer, rec_buffer, out_buffer;
+};
+
+static struct spa_log_topic log_topic = SPA_LOG_TOPIC(0, "spa.eac.webrtc");
+#undef SPA_LOG_TOPIC_DEFAULT
+#define SPA_LOG_TOPIC_DEFAULT &log_topic
+
+static bool webrtc_get_spa_bool(const struct spa_dict *args, const char *key, bool default_value)
+{
+ if (auto str = spa_dict_lookup(args, key))
+ return spa_atob(str);
+
+ return default_value;
+}
+
+static int webrtc_init(void *object, const struct spa_dict *args, const struct spa_audio_info_raw *info)
+{
+ auto impl = static_cast<struct impl_data*>(object);
+
+ 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 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);
+
+ // 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));
+
+ webrtc::ProcessingConfig pconfig = {{
+ webrtc::StreamConfig(info->rate, info->channels, false), /* input stream */
+ webrtc::StreamConfig(info->rate, info->channels, false), /* output stream */
+ webrtc::StreamConfig(info->rate, info->channels, false), /* reverse input stream */
+ webrtc::StreamConfig(info->rate, info->channels, false), /* reverse output stream */
+ }};
+
+ auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
+ if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
+ spa_log_error(impl->log, "Error initialising webrtc audio processing module");
+ return -1;
+ }
+
+ 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->info = *info;
+ impl->play_buffer = std::make_unique<float *[]>(info->channels);
+ impl->rec_buffer = std::make_unique<float *[]>(info->channels);
+ impl->out_buffer = std::make_unique<float *[]>(info->channels);
+ return 0;
+}
+
+static int webrtc_run(void *object, const float *rec[], const float *play[], float *out[], uint32_t n_samples)
+{
+ auto impl = static_cast<struct impl_data*>(object);
+ webrtc::StreamConfig config =
+ webrtc::StreamConfig(impl->info.rate, impl->info.channels, false);
+ unsigned int num_blocks = n_samples * 1000 / impl->info.rate / 10;
+
+ if (n_samples * 1000 / impl->info.rate % 10 != 0) {
+ spa_log_error(impl->log, "Buffers must be multiples of 10ms in length (currently %u samples)", n_samples);
+ return -1;
+ }
+
+ for (size_t i = 0; i < num_blocks; i ++) {
+ for (size_t j = 0; j < impl->info.channels; j++) {
+ impl->play_buffer[j] = const_cast<float *>(play[j]) + config.num_frames() * i;
+ impl->rec_buffer[j] = const_cast<float *>(rec[j]) + config.num_frames() * i;
+ impl->out_buffer[j] = out[j] + config.num_frames() * i;
+ }
+ /* FIXME: ProcessReverseStream may change the playback buffer, in which
+ * case we should use that, if we ever expose the intelligibility
+ * enhancer */
+ if (impl->apm->ProcessReverseStream(impl->play_buffer.get(), config, config, impl->play_buffer.get()) !=
+ webrtc::AudioProcessing::kNoError) {
+ spa_log_error(impl->log, "Processing reverse stream failed");
+ }
+
+ // Extra delay introduced by multiple frames
+ impl->apm->set_stream_delay_ms((num_blocks - 1) * 10);
+
+ if (impl->apm->ProcessStream(impl->rec_buffer.get(), config, config, impl->out_buffer.get()) !=
+ webrtc::AudioProcessing::kNoError) {
+ spa_log_error(impl->log, "Processing stream failed");
+ }
+ }
+
+ return 0;
+}
+
+static const struct spa_audio_aec_methods impl_aec = {
+ SPA_VERSION_AUDIO_AEC_METHODS,
+ .add_listener = NULL,
+ .init = webrtc_init,
+ .run = webrtc_run,
+};
+
+static int impl_get_interface(struct spa_handle *handle, const char *type, void **interface)
+{
+ auto impl = reinterpret_cast<struct impl_data*>(handle);
+
+ spa_return_val_if_fail(handle != NULL, -EINVAL);
+ spa_return_val_if_fail(interface != NULL, -EINVAL);
+
+ if (spa_streq(type, SPA_TYPE_INTERFACE_AUDIO_AEC))
+ *interface = &impl->aec;
+ else
+ return -ENOENT;
+
+ return 0;
+}
+
+static int impl_clear(struct spa_handle *handle)
+{
+ spa_return_val_if_fail(handle != NULL, -EINVAL);
+ auto impl = reinterpret_cast<struct impl_data*>(handle);
+ impl->~impl_data();
+ return 0;
+}
+
+static size_t
+impl_get_size(const struct spa_handle_factory *factory,
+ const struct spa_dict *params)
+{
+ return sizeof(struct impl_data);
+}
+
+static int
+impl_init(const struct spa_handle_factory *factory,
+ struct spa_handle *handle,
+ const struct spa_dict *info,
+ const struct spa_support *support,
+ uint32_t n_support)
+{
+ spa_return_val_if_fail(factory != NULL, -EINVAL);
+ spa_return_val_if_fail(handle != NULL, -EINVAL);
+
+ auto impl = new (handle) impl_data();
+
+ impl->handle.get_interface = impl_get_interface;
+ impl->handle.clear = impl_clear;
+
+ impl->aec.iface = SPA_INTERFACE_INIT(
+ SPA_TYPE_INTERFACE_AUDIO_AEC,
+ SPA_VERSION_AUDIO_AEC,
+ &impl_aec, impl);
+ impl->aec.name = "webrtc",
+ impl->aec.info = NULL;
+ impl->aec.latency = "480/48000",
+
+ impl->log = static_cast<struct spa_log *>(spa_support_find(support, n_support, SPA_TYPE_INTERFACE_Log));
+ spa_log_topic_init(impl->log, &log_topic);
+
+ return 0;
+}
+
+static const struct spa_interface_info impl_interfaces[] = {
+ {SPA_TYPE_INTERFACE_AUDIO_AEC,},
+};
+
+static int
+impl_enum_interface_info(const struct spa_handle_factory *factory,
+ const struct spa_interface_info **info,
+ uint32_t *index)
+{
+ spa_return_val_if_fail(factory != NULL, -EINVAL);
+ spa_return_val_if_fail(info != NULL, -EINVAL);
+ spa_return_val_if_fail(index != NULL, -EINVAL);
+
+ switch (*index) {
+ case 0:
+ *info = &impl_interfaces[*index];
+ break;
+ default:
+ return 0;
+ }
+ (*index)++;
+ return 1;
+}
+
+static const struct spa_handle_factory spa_aec_webrtc_factory = {
+ SPA_VERSION_HANDLE_FACTORY,
+ SPA_NAME_AEC,
+ NULL,
+ impl_get_size,
+ impl_init,
+ impl_enum_interface_info,
+};
+
+SPA_EXPORT
+int spa_handle_factory_enum(const struct spa_handle_factory **factory, uint32_t *index)
+{
+ spa_return_val_if_fail(factory != NULL, -EINVAL);
+ spa_return_val_if_fail(index != NULL, -EINVAL);
+
+ switch (*index) {
+ case 0:
+ *factory = &spa_aec_webrtc_factory;
+ break;
+ default:
+ return 0;
+ }
+ (*index)++;
+ return 1;
+}