mirror of
https://github.com/mollyim/webrtc.git
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b46c4bf27b
Note: this CL has to leave behind one part of iSAC, which is its VAD currently used by AGC1 in APM. The target visibility has been restricted and the VAD will be removed together with AGC1 when the time comes. Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319 Bug: webrtc:14450 Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38652}
102 lines
2.9 KiB
C++
102 lines
2.9 KiB
C++
/*
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* Copyright (c) 2012 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 <stdio.h>
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#include <string>
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#include <vector>
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_coding/test/EncodeDecodeTest.h"
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#include "modules/audio_coding/test/PacketLossTest.h"
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#include "modules/audio_coding/test/TestAllCodecs.h"
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#include "modules/audio_coding/test/TestRedFec.h"
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#include "modules/audio_coding/test/TestStereo.h"
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#include "modules/audio_coding/test/TestVADDTX.h"
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#include "modules/audio_coding/test/TwoWayCommunication.h"
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#include "modules/audio_coding/test/opus_test.h"
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#include "test/gtest.h"
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#include "test/testsupport/file_utils.h"
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TEST(AudioCodingModuleTest, TestAllCodecs) {
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webrtc::TestAllCodecs().Perform();
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}
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#if defined(WEBRTC_ANDROID)
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TEST(AudioCodingModuleTest, DISABLED_TestEncodeDecode) {
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#else
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TEST(AudioCodingModuleTest, TestEncodeDecode) {
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#endif
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webrtc::EncodeDecodeTest().Perform();
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}
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TEST(AudioCodingModuleTest, TestRedFec) {
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webrtc::TestRedFec().Perform();
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}
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// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
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#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
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TEST(AudioCodingModuleTest, DISABLED_TestStereo) {
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#else
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TEST(AudioCodingModuleTest, TestStereo) {
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#endif
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webrtc::TestStereo().Perform();
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}
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TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
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webrtc::TestWebRtcVadDtx().Perform();
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}
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TEST(AudioCodingModuleTest, TestOpusDtx) {
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webrtc::TestOpusDtx().Perform();
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}
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// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
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#if defined(WEBRTC_IOS)
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TEST(AudioCodingModuleTest, DISABLED_TestOpus) {
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#else
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TEST(AudioCodingModuleTest, TestOpus) {
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#endif
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webrtc::OpusTest().Perform();
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}
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TEST(AudioCodingModuleTest, TestPacketLoss) {
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webrtc::PacketLossTest(1, 10, 10, 1).Perform();
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}
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TEST(AudioCodingModuleTest, TestPacketLossBurst) {
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webrtc::PacketLossTest(1, 10, 10, 2).Perform();
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}
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// Disabled on ios as flake, see https://crbug.com/webrtc/7057
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#if defined(WEBRTC_IOS)
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TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereo) {
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#else
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TEST(AudioCodingModuleTest, TestPacketLossStereo) {
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#endif
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webrtc::PacketLossTest(2, 10, 10, 1).Perform();
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}
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// Disabled on ios as flake, see https://crbug.com/webrtc/7057
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#if defined(WEBRTC_IOS)
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TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereoBurst) {
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#else
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TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
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#endif
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webrtc::PacketLossTest(2, 10, 10, 2).Perform();
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}
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// The full API test is too long to run automatically on bots, but can be used
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// for offline testing. User interaction is needed.
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#ifdef ACM_TEST_FULL_API
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TEST(AudioCodingModuleTest, TestAPI) {
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webrtc::APITest().Perform();
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}
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#endif
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