a96e2d77cb
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}
118 lines
5.2 KiB
C++
118 lines
5.2 KiB
C++
/*
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* libjingle
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* Copyright 2013 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "talk/app/webrtc/localaudiosource.h"
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#include <string>
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#include <vector>
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#include "talk/app/webrtc/test/fakeconstraints.h"
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#include "webrtc/base/gunit.h"
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#include "webrtc/media/base/fakemediaengine.h"
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#include "webrtc/media/base/fakevideorenderer.h"
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using webrtc::LocalAudioSource;
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using webrtc::MediaConstraintsInterface;
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using webrtc::MediaSourceInterface;
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using webrtc::PeerConnectionFactoryInterface;
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TEST(LocalAudioSourceTest, SetValidOptions) {
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webrtc::FakeConstraints constraints;
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constraints.AddMandatory(
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MediaConstraintsInterface::kGoogEchoCancellation, false);
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constraints.AddOptional(
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MediaConstraintsInterface::kExtendedFilterEchoCancellation, true);
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constraints.AddOptional(MediaConstraintsInterface::kDAEchoCancellation, true);
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constraints.AddOptional(MediaConstraintsInterface::kAutoGainControl, true);
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constraints.AddOptional(
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MediaConstraintsInterface::kExperimentalAutoGainControl, true);
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constraints.AddMandatory(MediaConstraintsInterface::kNoiseSuppression, false);
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constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, true);
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constraints.AddOptional(MediaConstraintsInterface::kAecDump, true);
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rtc::scoped_refptr<LocalAudioSource> source =
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LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
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&constraints);
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EXPECT_EQ(rtc::Optional<bool>(false), source->options().echo_cancellation);
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EXPECT_EQ(rtc::Optional<bool>(true), source->options().extended_filter_aec);
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EXPECT_EQ(rtc::Optional<bool>(true), source->options().delay_agnostic_aec);
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EXPECT_EQ(rtc::Optional<bool>(true), source->options().auto_gain_control);
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EXPECT_EQ(rtc::Optional<bool>(true), source->options().experimental_agc);
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EXPECT_EQ(rtc::Optional<bool>(false), source->options().noise_suppression);
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EXPECT_EQ(rtc::Optional<bool>(true), source->options().highpass_filter);
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EXPECT_EQ(rtc::Optional<bool>(true), source->options().aec_dump);
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}
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TEST(LocalAudioSourceTest, OptionNotSet) {
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webrtc::FakeConstraints constraints;
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rtc::scoped_refptr<LocalAudioSource> source =
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LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
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&constraints);
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EXPECT_EQ(rtc::Optional<bool>(), source->options().highpass_filter);
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}
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TEST(LocalAudioSourceTest, MandatoryOverridesOptional) {
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webrtc::FakeConstraints constraints;
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constraints.AddMandatory(
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MediaConstraintsInterface::kGoogEchoCancellation, false);
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constraints.AddOptional(
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MediaConstraintsInterface::kGoogEchoCancellation, true);
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rtc::scoped_refptr<LocalAudioSource> source =
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LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
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&constraints);
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EXPECT_EQ(rtc::Optional<bool>(false), source->options().echo_cancellation);
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}
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TEST(LocalAudioSourceTest, InvalidOptional) {
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webrtc::FakeConstraints constraints;
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constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, false);
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constraints.AddOptional("invalidKey", false);
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rtc::scoped_refptr<LocalAudioSource> source =
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LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
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&constraints);
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EXPECT_EQ(MediaSourceInterface::kLive, source->state());
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EXPECT_EQ(rtc::Optional<bool>(false), source->options().highpass_filter);
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}
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TEST(LocalAudioSourceTest, InvalidMandatory) {
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webrtc::FakeConstraints constraints;
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constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
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constraints.AddMandatory("invalidKey", false);
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rtc::scoped_refptr<LocalAudioSource> source =
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LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
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&constraints);
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EXPECT_EQ(MediaSourceInterface::kLive, source->state());
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EXPECT_EQ(rtc::Optional<bool>(false), source->options().highpass_filter);
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}
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