| /* 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 "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" |