fe0fd41bb9
Also did some slight refactoring of the code that turns constraints into rtc::Optionals. Used a template method to avoid code duplication, and used the same pattern for "CopyConstraintsIntoAudioOptions" as was being used for "CopyConstraintsIntoRtcConfiguration". BUG=webrtc:6752 Review-Url: https://codereview.webrtc.org/2628523003 Cr-Commit-Position: refs/heads/master@{#16063}
57 lines
1.7 KiB
C++
57 lines
1.7 KiB
C++
/*
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* Copyright 2013 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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#include "webrtc/api/localaudiosource.h"
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#include <vector>
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#include "webrtc/api/mediaconstraintsinterface.h"
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#include "webrtc/media/base/mediaengine.h"
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using webrtc::MediaConstraintsInterface;
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using webrtc::MediaSourceInterface;
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namespace webrtc {
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rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
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const PeerConnectionFactoryInterface::Options& options,
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const MediaConstraintsInterface* constraints) {
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rtc::scoped_refptr<LocalAudioSource> source(
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new rtc::RefCountedObject<LocalAudioSource>());
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source->Initialize(options, constraints);
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return source;
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}
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rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
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const PeerConnectionFactoryInterface::Options& options,
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const cricket::AudioOptions* audio_options) {
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rtc::scoped_refptr<LocalAudioSource> source(
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new rtc::RefCountedObject<LocalAudioSource>());
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source->Initialize(options, audio_options);
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return source;
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}
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void LocalAudioSource::Initialize(
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const PeerConnectionFactoryInterface::Options& options,
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const MediaConstraintsInterface* constraints) {
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CopyConstraintsIntoAudioOptions(constraints, &options_);
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}
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void LocalAudioSource::Initialize(
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const PeerConnectionFactoryInterface::Options& options,
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const cricket::AudioOptions* audio_options) {
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if (!audio_options)
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return;
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options_ = *audio_options;
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}
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} // namespace webrtc
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