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https://github.com/mollyim/webrtc.git
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254ad1b914
Starting from Android N, mobile app may not be able to access microphone while in background where it fails the call. In order to mitigate the issue, delay the ADM initialization as late as possible. Bug: webrtc:12120 Change-Id: I0fbf0300299b6c53413dfaaf88f748edc0a06bc1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191100 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32598}
68 lines
2.9 KiB
C++
68 lines
2.9 KiB
C++
/*
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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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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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