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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/voip/voip_engine_factory.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/voip/voip_engine_factory.h | 68 |
1 files changed, 68 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/voip/voip_engine_factory.h b/third_party/libwebrtc/api/voip/voip_engine_factory.h new file mode 100644 index 0000000000..62fe8011a6 --- /dev/null +++ b/third_party/libwebrtc/api/voip/voip_engine_factory.h @@ -0,0 +1,68 @@ +/* + * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_VOIP_VOIP_ENGINE_FACTORY_H_ +#define API_VOIP_VOIP_ENGINE_FACTORY_H_ + +#include <memory> + +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/scoped_refptr.h" +#include "api/task_queue/task_queue_factory.h" +#include "api/voip/voip_engine.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" + +namespace webrtc { + +// VoipEngineConfig is a struct that defines parameters to instantiate a +// VoipEngine instance through CreateVoipEngine factory method. Each member is +// marked with comments as either mandatory or optional and default +// implementations that applications can use. +struct VoipEngineConfig { + // Mandatory (e.g. api/audio_codec/builtin_audio_encoder_factory). + // AudioEncoderFactory provides a set of audio codecs for VoipEngine to encode + // the audio input sample. Application can choose to limit the set to reduce + // application footprint. + rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; + + // Mandatory (e.g. api/audio_codec/builtin_audio_decoder_factory). + // AudioDecoderFactory provides a set of audio codecs for VoipEngine to decode + // the received RTP packets from remote media endpoint. Application can choose + // to limit the set to reduce application footprint. + rtc::scoped_refptr<AudioDecoderFactory> decoder_factory; + + // Mandatory (e.g. api/task_queue/default_task_queue_factory). + // TaskQeueuFactory provided for VoipEngine to work asynchronously on its + // encoding flow. + std::unique_ptr<TaskQueueFactory> task_queue_factory; + + // Mandatory (e.g. modules/audio_device/include). + // AudioDeviceModule that periocally provides audio input samples from + // recording device (e.g. microphone) and requests audio output samples to + // play through its output device (e.g. speaker). + rtc::scoped_refptr<AudioDeviceModule> audio_device_module; + + // Optional (e.g. modules/audio_processing/include). + // AudioProcessing provides audio procesing functionalities (e.g. acoustic + // echo cancellation, noise suppression, gain control, etc) on audio input + // samples for VoipEngine. When optionally not set, VoipEngine will not have + // such functionalities to perform on audio input samples received from + // AudioDeviceModule. + rtc::scoped_refptr<AudioProcessing> audio_processing; +}; + +// Creates a VoipEngine instance with provided VoipEngineConfig. +std::unique_ptr<VoipEngine> CreateVoipEngine(VoipEngineConfig config); + +} // namespace webrtc + +#endif // API_VOIP_VOIP_ENGINE_FACTORY_H_ |