webrtc_m130/webrtc/api/localaudiosource.cc

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/*
* Copyright 2013 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.
*/
#include "webrtc/api/localaudiosource.h"
#include <vector>
#include "webrtc/api/mediaconstraintsinterface.h"
Move talk/media to webrtc/media I removed the 'libjingle' target in talk/libjingle.gyp and replaced all users of it with base/base.gyp:rtc_base. It seems the jsoncpp and expat dependencies were not used by it's previous references. The files in talk/media/testdata were uploaded to Google Storage and added .sha1 files in resources/media instead of simply moving them. The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL in order to not break Git history. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc. * Unused GYP reference to libjingle_tests_additional_deps was removed. * Removed duplicated GYP entries of webrtc/base/testutils.cc webrtc/base/testutils.h The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media, so they were moved to the media.gyp. I also checked that none of EXPAT_RELATIVE_PATH, FEATURE_ENABLE_VOICEMAIL, GTEST_RELATIVE_PATH, JSONCPP_RELATIVE_PATH, LOGGING=1, SRTP_RELATIVE_PATH, FEATURE_ENABLE_SSL, FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN, HAVE_SCTP, HAVE_SRTP, are used by the talk/media code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/ BUG=webrtc:5420 NOPRESUBMIT=True TBR=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1587193006 Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
#include "webrtc/media/base/mediaengine.h"
using webrtc::MediaConstraintsInterface;
using webrtc::MediaSourceInterface;
namespace webrtc {
rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints) {
rtc::scoped_refptr<LocalAudioSource> source(
new rtc::RefCountedObject<LocalAudioSource>());
source->Initialize(options, constraints);
return source;
}
rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
const PeerConnectionFactoryInterface::Options& options,
const cricket::AudioOptions* audio_options) {
rtc::scoped_refptr<LocalAudioSource> source(
new rtc::RefCountedObject<LocalAudioSource>());
source->Initialize(options, audio_options);
return source;
}
void LocalAudioSource::Initialize(
const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints) {
CopyConstraintsIntoAudioOptions(constraints, &options_);
}
void LocalAudioSource::Initialize(
const PeerConnectionFactoryInterface::Options& options,
const cricket::AudioOptions* audio_options) {
if (!audio_options)
return;
options_ = *audio_options;
}
} // namespace webrtc