webrtc_m130/webrtc/api/localaudiosource_unittest.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 <string>
#include <vector>
#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/base/gunit.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/fakemediaengine.h"
#include "webrtc/media/base/fakevideorenderer.h"
using webrtc::LocalAudioSource;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaSourceInterface;
using webrtc::PeerConnectionFactoryInterface;
TEST(LocalAudioSourceTest, SetValidOptions) {
webrtc::FakeConstraints constraints;
constraints.AddMandatory(
MediaConstraintsInterface::kGoogEchoCancellation, false);
constraints.AddOptional(
MediaConstraintsInterface::kExtendedFilterEchoCancellation, true);
constraints.AddOptional(MediaConstraintsInterface::kDAEchoCancellation, true);
constraints.AddOptional(MediaConstraintsInterface::kAutoGainControl, true);
constraints.AddOptional(
MediaConstraintsInterface::kExperimentalAutoGainControl, true);
constraints.AddMandatory(MediaConstraintsInterface::kNoiseSuppression, false);
constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, true);
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
EXPECT_EQ(rtc::Optional<bool>(false), source->options().echo_cancellation);
EXPECT_EQ(rtc::Optional<bool>(true), source->options().extended_filter_aec);
EXPECT_EQ(rtc::Optional<bool>(true), source->options().delay_agnostic_aec);
EXPECT_EQ(rtc::Optional<bool>(true), source->options().auto_gain_control);
EXPECT_EQ(rtc::Optional<bool>(true), source->options().experimental_agc);
EXPECT_EQ(rtc::Optional<bool>(false), source->options().noise_suppression);
EXPECT_EQ(rtc::Optional<bool>(true), source->options().highpass_filter);
}
TEST(LocalAudioSourceTest, OptionNotSet) {
webrtc::FakeConstraints constraints;
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
EXPECT_EQ(rtc::Optional<bool>(), source->options().highpass_filter);
}
TEST(LocalAudioSourceTest, MandatoryOverridesOptional) {
webrtc::FakeConstraints constraints;
constraints.AddMandatory(
MediaConstraintsInterface::kGoogEchoCancellation, false);
constraints.AddOptional(
MediaConstraintsInterface::kGoogEchoCancellation, true);
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
EXPECT_EQ(rtc::Optional<bool>(false), source->options().echo_cancellation);
}
TEST(LocalAudioSourceTest, InvalidOptional) {
webrtc::FakeConstraints constraints;
constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, false);
constraints.AddOptional("invalidKey", false);
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
EXPECT_EQ(MediaSourceInterface::kLive, source->state());
EXPECT_EQ(rtc::Optional<bool>(false), source->options().highpass_filter);
}
TEST(LocalAudioSourceTest, InvalidMandatory) {
webrtc::FakeConstraints constraints;
constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
constraints.AddMandatory("invalidKey", false);
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
EXPECT_EQ(MediaSourceInterface::kLive, source->state());
EXPECT_EQ(rtc::Optional<bool>(false), source->options().highpass_filter);
}