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module-echo-cancel: Wire up the webrtc echo canceller

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Mostly uses the existing infrastructure, but the webrtc canceller has a
fixed blocksize, so we:

  1. Use the canceller blocksize if configured
  2. Accumulate output data in a ringbuffer
  3. Push out the data in the required chunk size
This commit is contained in:
Arun Raghavan 2021-05-28 11:28:03 -04:00
parent 1349d5334d
commit d95870d8d3
9 changed files with 247 additions and 69 deletions
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85
src/modules/module-echo-cancel/aec-webrtc.cc
View file

@ -1,6 +1,7 @@
/* 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"),
@ -24,35 +25,105 @@
#include "echo-cancel.h"
#include <pipewire/pipewire.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 {
uint32_t channels;
webrtc::AudioProcessing *apm = NULL;
spa_audio_info_raw info;
};
static void *webrtc_create(struct spa_dict *info, uint32_t channels)
static void *webrtc_create(const struct pw_properties *args, const spa_audio_info_raw *info)
{
struct impl *impl;
webrtc::AudioProcessing *apm;
webrtc::ProcessingConfig pconfig;
webrtc::Config config;
apm = webrtc::AudioProcessing::Create(config);
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 */
}};
if (apm->Initialize(pconfig) != webrtc::AudioProcessing::kNoError) {
pw_log_error("Error initialising webrtc audio processing module");
goto error;
}
// TODO: wire up args to control these
apm->high_pass_filter()->Enable(true);
apm->echo_cancellation()->enable_drift_compensation(false);
apm->echo_cancellation()->Enable(true);
apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
apm->noise_suppression()->Enable(true);
apm->gain_control()->set_analog_level_limits(0, 255);
// FIXME: can we hook up AGC?
apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);
apm->gain_control()->Enable(true);
apm->voice_detection()->Enable(true);
impl = (struct impl *)calloc(1, sizeof(struct impl));
impl->channels = channels;
impl->info = *info;
impl->apm = apm;
return impl;
error:
if (apm)
delete apm;
return NULL;
}
static void webrtc_destroy(void *ec)
{
free(ec);
struct impl *impl = (struct impl*)ec;
delete impl->apm;
free(impl);
}
static int webrtc_run(void *ec, const float *rec[], const float *play[], float *out[], uint32_t n_samples)
{
struct impl *impl = (struct impl*)ec;
uint32_t i;
for (i = 0; i < impl->channels; i++)
memcpy(out[i], rec[i], n_samples * sizeof(float));
webrtc::StreamConfig config =
webrtc::StreamConfig(impl->info.rate, impl->info.channels, false);
if (n_samples * 1000 / impl->info.rate != 10) {
pw_log_error("Buffers must be 10ms in length (currently %u samples)", n_samples);
return -1;
}
/* 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(play, config, config, (float**)play) !=
webrtc::AudioProcessing::kNoError) {
pw_log_error("Processing reverse stream failed");
}
impl->apm->set_stream_delay_ms(0);
if (impl->apm->ProcessStream(rec, config, config, out) !=
webrtc::AudioProcessing::kNoError) {
pw_log_error("Processing stream failed");
}
return 0;
}
static const struct echo_cancel_info echo_cancel_webrtc_impl = {
.name = "webrtc",
.info = SPA_DICT_INIT(NULL, 0),
.latency = "480/48000",
.create = webrtc_create,
.destroy = webrtc_destroy,