webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

45 lines
1.4 KiB
C++

/*
* Copyright (c) 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.
*/
// TODO(bjornv): Make this a comprehensive test.
#include "modules/audio_processing/aec/echo_cancellation.h"
#include <stdlib.h>
#include <time.h>
#include "modules/audio_processing/aec/aec_core.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"
namespace webrtc {
TEST(EchoCancellationTest, CreateAndFreeHasExpectedBehavior) {
void* handle = WebRtcAec_Create();
ASSERT_TRUE(handle);
WebRtcAec_Free(nullptr);
WebRtcAec_Free(handle);
}
TEST(EchoCancellationTest, ApplyAecCoreHandle) {
void* handle = WebRtcAec_Create();
ASSERT_TRUE(handle);
EXPECT_TRUE(WebRtcAec_aec_core(NULL) == NULL);
AecCore* aec_core = WebRtcAec_aec_core(handle);
EXPECT_TRUE(aec_core != NULL);
// A simple test to verify that we can set and get a value from the lower
// level |aec_core| handle.
int delay = 111;
WebRtcAec_SetSystemDelay(aec_core, delay);
EXPECT_EQ(delay, WebRtcAec_system_delay(aec_core));
WebRtcAec_Free(handle);
}
} // namespace webrtc