72 lines
3.1 KiB
C
72 lines
3.1 KiB
C
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/*
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* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_VOIP_VOIP_ENGINE_FACTORY_H_
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#define API_VOIP_VOIP_ENGINE_FACTORY_H_
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#include <memory>
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/audio_codecs/audio_encoder_factory.h"
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#include "api/scoped_refptr.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "api/voip/voip_engine.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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// VoipEngineConfig is a struct that defines parameters to instantiate a
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// VoipEngine instance through CreateVoipEngine factory method. Each member is
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// marked with comments as either mandatory or optional and default
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// implementations that applications can use.
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struct VoipEngineConfig {
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// Mandatory (e.g. api/audio_codec/builtin_audio_encoder_factory).
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// AudioEncoderFactory provides a set of audio codecs for VoipEngine to encode
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// the audio input sample. Application can choose to limit the set to reduce
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// application footprint.
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rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
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// Mandatory (e.g. api/audio_codec/builtin_audio_decoder_factory).
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// AudioDecoderFactory provides a set of audio codecs for VoipEngine to decode
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// the received RTP packets from remote media endpoint. Application can choose
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// to limit the set to reduce application footprint.
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rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
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// Mandatory (e.g. api/task_queue/default_task_queue_factory).
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// TaskQeueuFactory provided for VoipEngine to work asynchronously on its
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// encoding flow.
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std::unique_ptr<TaskQueueFactory> task_queue_factory;
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// Mandatory (e.g. modules/audio_device/include).
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// AudioDeviceModule that periocally provides audio input samples from
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// recording device (e.g. microphone) and requests audio output samples to
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// play through its output device (e.g. speaker).
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rtc::scoped_refptr<AudioDeviceModule> audio_device_module;
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// Optional (e.g. modules/audio_processing/include).
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// AudioProcessing provides audio procesing functionalities (e.g. acoustic
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// echo cancellation, noise suppression, gain control, etc) on audio input
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// samples for VoipEngine. When optionally not set, VoipEngine will not have
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// such functionalities to perform on audio input samples received from
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// AudioDeviceModule.
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rtc::scoped_refptr<AudioProcessing> audio_processing;
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};
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// Creates a VoipEngine instance with provided VoipEngineConfig.
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// This could return nullptr if AudioDeviceModule (ADM) initialization fails
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// during construction of VoipEngine which would render VoipEngine
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// nonfunctional.
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std::unique_ptr<VoipEngine> CreateVoipEngine(VoipEngineConfig config);
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} // namespace webrtc
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#endif // API_VOIP_VOIP_ENGINE_FACTORY_H_
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