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module-echo-cancel: Move backends to dynamic libaries
Move all backends to dynamic libaries loaded with spa_plugin_loader so new backends not needs changes in pipewire or pipewire dependency to external code Change-Id: I702ce047598d0c318d6dc6ac8248062a5c12f643
This commit is contained in:
Joakim Olsson
Wim Taymans
parent
761199be70
commit
9386c70b3a
11 changed files with 576 additions and 247 deletions
286
spa/plugins/aec/aec-webrtc.cpp
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286
spa/plugins/aec/aec-webrtc.cpp
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/* PipeWire
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*
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* Copyright © 2021 Wim Taymans <wim.taymans@gmail.com>
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* © 2021 Arun Raghavan <arun@asymptotic.io>
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*
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* Permission is hereby granted, free of charge, to any person obtaining a
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* copy of this software and associated documentation files (the "Software"),
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* to deal in the Software without restriction, including without limitation
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* the rights to use, copy, modify, merge, publish, distribute, sublicense,
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* and/or sell copies of the Software, and to permit persons to whom the
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* Software is furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice (including the next
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* paragraph) shall be included in all copies or substantial portions of the
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* Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
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* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
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* DEALINGS IN THE SOFTWARE.
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*/
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#include <memory>
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#include <utility>
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#include <spa/interfaces/audio/aec.h>
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#include <spa/support/log.h>
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#include <spa/utils/string.h>
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#include <spa/utils/names.h>
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#include <spa/support/plugin.h>
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#include <webrtc/modules/audio_processing/include/audio_processing.h>
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#include <webrtc/modules/interface/module_common_types.h>
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#include <webrtc/system_wrappers/include/trace.h>
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struct impl_data {
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struct spa_handle handle;
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struct spa_log *log;
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std::unique_ptr<webrtc::AudioProcessing> apm;
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spa_audio_info_raw info;
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std::unique_ptr<float *[]> play_buffer, rec_buffer, out_buffer;
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};
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static struct spa_log_topic log_topic = SPA_LOG_TOPIC(0, "spa.eac.webrtc");
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#undef SPA_LOG_TOPIC_DEFAULT
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#define SPA_LOG_TOPIC_DEFAULT &log_topic
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static bool webrtc_get_spa_bool(const struct spa_dict *args, const char *key, bool default_value) {
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const char *str_val;
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bool value = default_value;
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str_val = spa_dict_lookup(args, key);
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if (str_val != NULL)
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value =spa_atob(str_val);
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return value;
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}
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static int webrtc_create(struct spa_handle *handle, const struct spa_dict *args, const struct spa_audio_info_raw *info)
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{
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auto impl = reinterpret_cast<struct impl_data*>(handle);
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bool extended_filter = webrtc_get_spa_bool(args, "webrtc.extended_filter", true);
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bool delay_agnostic = webrtc_get_spa_bool(args, "webrtc.delay_agnostic", true);
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bool high_pass_filter = webrtc_get_spa_bool(args, "webrtc.high_pass_filter", true);
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bool noise_suppression = webrtc_get_spa_bool(args, "webrtc.noise_suppression", true);
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bool voice_detection = webrtc_get_spa_bool(args, "webrtc.voice_detection", true);
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// Note: AGC seems to mess up with Agnostic Delay Detection, especially with speech,
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// result in very poor performance, disable by default
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bool gain_control = webrtc_get_spa_bool(args, "webrtc.gain_control", false);
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// Disable experimental flags by default
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bool experimental_agc = webrtc_get_spa_bool(args, "webrtc.experimental_agc", false);
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bool experimental_ns = webrtc_get_spa_bool(args, "webrtc.experimental_ns", false);
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// FIXME: Intelligibility enhancer is not currently supported
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// This filter will modify playback buffer (when calling ProcessReverseStream), but now
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// playback buffer modifications are discarded.
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webrtc::Config config;
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config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(extended_filter));
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config.Set<webrtc::DelayAgnostic>(new webrtc::DelayAgnostic(delay_agnostic));
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config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(experimental_agc));
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config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(experimental_ns));
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webrtc::ProcessingConfig pconfig = {{
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webrtc::StreamConfig(info->rate, info->channels, false), /* input stream */
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webrtc::StreamConfig(info->rate, info->channels, false), /* output stream */
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webrtc::StreamConfig(info->rate, info->channels, false), /* reverse input stream */
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webrtc::StreamConfig(info->rate, info->channels, false), /* reverse output stream */
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}};
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auto apm = std::unique_ptr<webrtc::AudioProcessing>(webrtc::AudioProcessing::Create(config));
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if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
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spa_log_error(impl->log, "Error initialising webrtc audio processing module");
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return -1;
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}
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apm->high_pass_filter()->Enable(high_pass_filter);
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// Always disable drift compensation since it requires drift sampling
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apm->echo_cancellation()->enable_drift_compensation(false);
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apm->echo_cancellation()->Enable(true);
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// TODO: wire up supression levels to args
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apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kHighSuppression);
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apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
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apm->noise_suppression()->Enable(noise_suppression);
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apm->voice_detection()->Enable(voice_detection);
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// TODO: wire up AGC parameters to args
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apm->gain_control()->set_analog_level_limits(0, 255);
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apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
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apm->gain_control()->Enable(gain_control);
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impl->apm = std::move(apm);
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impl->info = *info;
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impl->play_buffer = std::make_unique<float *[]>(info->channels);
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impl->rec_buffer = std::make_unique<float *[]>(info->channels);
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impl->out_buffer = std::make_unique<float *[]>(info->channels);
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return 0;
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}
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static int webrtc_run(struct spa_handle *handle, const float *rec[], const float *play[], float *out[], uint32_t n_samples)
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{
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auto impl = reinterpret_cast<struct impl_data*>(handle);
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webrtc::StreamConfig config =
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webrtc::StreamConfig(impl->info.rate, impl->info.channels, false);
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unsigned int num_blocks = n_samples * 1000 / impl->info.rate / 10;
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if (n_samples * 1000 / impl->info.rate % 10 != 0) {
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spa_log_error(impl->log, "Buffers must be multiples of 10ms in length (currently %u samples)", n_samples);
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return -1;
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}
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for (size_t i = 0; i < num_blocks; i ++) {
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for (size_t j = 0; j < impl->info.channels; j++) {
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impl->play_buffer[j] = const_cast<float *>(play[j]) + config.num_frames() * i;
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impl->rec_buffer[j] = const_cast<float *>(rec[j]) + config.num_frames() * i;
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impl->out_buffer[j] = out[j] + config.num_frames() * i;
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}
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/* FIXME: ProcessReverseStream may change the playback buffer, in which
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* case we should use that, if we ever expose the intelligibility
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* enhancer */
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if (impl->apm->ProcessReverseStream(impl->play_buffer.get(), config, config, impl->play_buffer.get()) !=
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webrtc::AudioProcessing::kNoError) {
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spa_log_error(impl->log, "Processing reverse stream failed");
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}
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// Extra delay introduced by multiple frames
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impl->apm->set_stream_delay_ms((num_blocks - 1) * 10);
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if (impl->apm->ProcessStream(impl->rec_buffer.get(), config, config, impl->out_buffer.get()) !=
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webrtc::AudioProcessing::kNoError) {
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spa_log_error(impl->log, "Processing stream failed");
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}
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}
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return 0;
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}
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struct spa_dict *webrtc_get_properties(SPA_UNUSED struct spa_handle *handle)
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{
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/* Not supported */
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return NULL;
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}
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int webrtc_set_properties(SPA_UNUSED struct spa_handle *handle, SPA_UNUSED const struct spa_dict *args)
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{
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/* Not supported */
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return -1;
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}
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static struct echo_cancel_info echo_cancel_webrtc_impl = {
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.name = "webrtc",
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.info = SPA_DICT_INIT(NULL, 0),
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.latency = "480/48000",
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.create = webrtc_create,
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.run = webrtc_run,
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.get_properties = webrtc_get_properties,
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.set_properties = webrtc_set_properties,
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};
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static int impl_get_interface(struct spa_handle *handle, const char *type, void **interface)
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{
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spa_return_val_if_fail(handle != NULL, -EINVAL);
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spa_return_val_if_fail(interface != NULL, -EINVAL);
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if (spa_streq(type, SPA_TYPE_INTERFACE_AEC))
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*interface = &echo_cancel_webrtc_impl;
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else
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return -ENOENT;
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return 0;
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}
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static int impl_clear(struct spa_handle *handle)
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{
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spa_return_val_if_fail(handle != NULL, -EINVAL);
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auto impl = reinterpret_cast<struct impl_data*>(handle);
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impl->~impl_data();
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return 0;
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}
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static size_t
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impl_get_size(const struct spa_handle_factory *factory,
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const struct spa_dict *params)
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{
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return sizeof(struct impl_data);
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}
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static int
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impl_init(const struct spa_handle_factory *factory,
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struct spa_handle *handle,
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const struct spa_dict *info,
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const struct spa_support *support,
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uint32_t n_support)
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{
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spa_return_val_if_fail(factory != NULL, -EINVAL);
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spa_return_val_if_fail(handle != NULL, -EINVAL);
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echo_cancel_webrtc_impl.iface = SPA_INTERFACE_INIT(
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SPA_TYPE_INTERFACE_AEC,
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SPA_VERSION_AUDIO_AEC,
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NULL,
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NULL);
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auto impl = new (handle) impl_data();
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impl->handle.get_interface = impl_get_interface;
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impl->handle.clear = impl_clear;
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impl->log = (struct spa_log*)spa_support_find(support, n_support, SPA_TYPE_INTERFACE_Log);
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spa_log_topic_init(impl->log, &log_topic);
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return 0;
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}
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static const struct spa_interface_info impl_interfaces[] = {
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{SPA_TYPE_INTERFACE_AEC,},
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};
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static int
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impl_enum_interface_info(const struct spa_handle_factory *factory,
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const struct spa_interface_info **info,
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uint32_t *index)
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{
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spa_return_val_if_fail(factory != NULL, -EINVAL);
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spa_return_val_if_fail(info != NULL, -EINVAL);
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spa_return_val_if_fail(index != NULL, -EINVAL);
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switch (*index) {
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case 0:
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*info = &impl_interfaces[*index];
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break;
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default:
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return 0;
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}
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(*index)++;
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return 1;
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}
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const struct spa_handle_factory spa_aec_exaudio_factory = {
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SPA_VERSION_HANDLE_FACTORY,
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SPA_NAME_AEC,
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NULL,
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impl_get_size,
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impl_init,
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impl_enum_interface_info,
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};
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SPA_EXPORT
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int spa_handle_factory_enum(const struct spa_handle_factory **factory, uint32_t *index)
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{
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spa_return_val_if_fail(factory != NULL, -EINVAL);
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spa_return_val_if_fail(index != NULL, -EINVAL);
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switch (*index) {
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case 0:
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*factory = &spa_aec_exaudio_factory;
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break;
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default:
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return 0;
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}
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(*index)++;
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return 1;
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}
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