blob: e3e6fe09142654ef81fff6139f8472e91acb4c21 [file] [log] [blame]
/* Copyright (c) 2018 The Chromium Authors. All rights reserved.
* Use of this source code is governed by a BSD-style license that can be
* found in the LICENSE file.
*/
#include "api/audio/echo_canceller3_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "cras-config/aec_config.h"
#include "cras-config/apm_config.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/task_queue.h"
extern "C" {
#include <errno.h>
#include "webrtc_apm.h"
webrtc_apm webrtc_apm_create(unsigned int num_channels,
unsigned int frame_rate,
dictionary *aec_ini,
dictionary *apm_ini)
{
int err;
webrtc::AudioProcessing *apm;
webrtc::AudioProcessing::ChannelLayout channel_layout;
webrtc::AudioProcessingBuilder apm_builder;
webrtc::EchoCanceller3Config aec3_config;
std::unique_ptr<webrtc::EchoControlFactory> ec3_factory;
switch (num_channels) {
case 1:
channel_layout = webrtc::AudioProcessing::kMono;
break;
case 2:
channel_layout = webrtc::AudioProcessing::kStereo;
break;
default:
return NULL;
}
if (aec_ini) {
aec_config_get(aec_ini, &aec3_config);
ec3_factory.reset(
new webrtc::EchoCanceller3Factory(aec3_config));
} else {
ec3_factory.reset(new webrtc::EchoCanceller3Factory());
}
apm_builder.SetEchoControlFactory(std::move(ec3_factory));
apm = apm_builder.Create();
if (apm_ini)
apm_config_apply(apm_ini, apm);
err = apm->Initialize(frame_rate, frame_rate, frame_rate,
channel_layout, channel_layout, channel_layout);
if (err) {
delete apm;
return NULL;
}
return reinterpret_cast<webrtc_apm>(apm);
}
void webrtc_apm_dump_configs(dictionary *apm_ini,
dictionary *aec_ini)
{
if (apm_ini)
apm_config_dump(apm_ini);
if (aec_ini)
aec_config_dump(aec_ini);
}
int webrtc_apm_process_reverse_stream_f(
webrtc_apm ptr,
int num_channels, int rate,
float *const *data)
{
webrtc::AudioProcessing *apm;
webrtc::StreamConfig config =
webrtc::StreamConfig(rate, num_channels);
apm = reinterpret_cast<webrtc::AudioProcessing *>(ptr);
return apm->ProcessReverseStream(data, config, config, data);
}
int webrtc_apm_process_reverse_stream(webrtc_apm ptr,
int num_channels, int rate,
int16_t *data, int nframes)
{
webrtc::AudioFrame af;
webrtc::AudioProcessing *apm;
apm = reinterpret_cast<webrtc::AudioProcessing *>(ptr);
af.UpdateFrame(0xFFFFFFFF, data, nframes, rate,
webrtc::AudioFrame::kNormalSpeech,
webrtc::AudioFrame::kVadUnknown,
num_channels);
return apm->ProcessReverseStream(&af);
}
int webrtc_apm_process_stream_f(webrtc_apm ptr,
int num_channels,
int rate,
float *const *data)
{
webrtc::AudioProcessing *apm;
webrtc::StreamConfig config =
webrtc::StreamConfig(rate, num_channels);
apm = reinterpret_cast<webrtc::AudioProcessing *>(ptr);
return apm->ProcessStream(data, config, config, data);
}
int webrtc_apm_process_stream(webrtc_apm ptr, int num_channels,
int rate, int16_t *data, int nframes)
{
int ret;
webrtc::AudioFrame af;
webrtc::AudioProcessing *apm;
apm = reinterpret_cast<webrtc::AudioProcessing *>(ptr);
//set stream delay
af.UpdateFrame(0xFFFFFFFF, data, nframes, rate,
webrtc::AudioFrame::kNormalSpeech,
webrtc::AudioFrame::kVadUnknown,
num_channels);
ret = apm->ProcessStream(&af);
if (ret)
return ret;
memcpy(data, af.data(), nframes * num_channels * 2);
return ret;
}
void webrtc_apm_destroy(webrtc_apm ptr)
{
webrtc::AudioProcessing *apm;
apm = reinterpret_cast<webrtc::AudioProcessing *>(ptr);
delete apm;
}
int webrtc_apm_set_stream_delay(webrtc_apm ptr, int delay_ms)
{
webrtc::AudioProcessing *apm;
apm = reinterpret_cast<webrtc::AudioProcessing *>(ptr);
return apm->set_stream_delay_ms(delay_ms);
}
int webrtc_apm_aec_dump(webrtc_apm ptr, void** wq_ptr, int start, FILE *handle)
{
webrtc::AudioProcessing *apm;
rtc::TaskQueue *work_queue;
apm = reinterpret_cast<webrtc::AudioProcessing *>(ptr);
if (start) {
work_queue = new rtc::TaskQueue("aecdump-worker-queue",
rtc::TaskQueue::Priority::LOW);
auto aec_dump = webrtc::AecDumpFactory::Create(handle, -1, work_queue);
if (!aec_dump)
return -ENOMEM;
apm->AttachAecDump(std::move(aec_dump));
*wq_ptr = reinterpret_cast<void *>(work_queue);
} else {
apm->DetachAecDump();
work_queue = reinterpret_cast<rtc::TaskQueue *>(*wq_ptr);
if (work_queue) {
delete work_queue;
work_queue = NULL;
}
}
return 0;
}
} // extern "C"