ec0f45be11
This reverts commit 056f9738bf7a3d16da45398239656e165c4e0851. Reason for revert: breaks downstream Original change's description: > Remove CodecInst pt.1 > > Update audio_coding tests to not use CodecInst. > > Bug: webrtc:7626 > Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2 > Reviewed-on: https://webrtc-review.googlesource.com/c/112594 > Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25879} TBR=solenberg@webrtc.org,kwiberg@webrtc.org Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7626 Reviewed-on: https://webrtc-review.googlesource.com/c/112906 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25881}
126 lines
3.6 KiB
C++
126 lines
3.6 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/iSACTest.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/fileutils.h"
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// This parameter is used to describe how to run the tests. It is normally
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// set to 0, and all tests are run in quite mode.
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#define ACM_TEST_MODE 0
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TEST(AudioCodingModuleTest, TestAllCodecs) {
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webrtc::TestAllCodecs(ACM_TEST_MODE).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(ACM_TEST_MODE).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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#if defined(WEBRTC_ANDROID)
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TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
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#else
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TEST(AudioCodingModuleTest, TestIsac) {
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#endif
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webrtc::ISACTest(ACM_TEST_MODE).Perform();
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}
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#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
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defined(WEBRTC_CODEC_ILBC)
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#if defined(WEBRTC_ANDROID)
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TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
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#else
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TEST(AudioCodingModuleTest, TwoWayCommunication) {
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#endif
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webrtc::TwoWayCommunication().Perform();
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}
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#endif
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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(ACM_TEST_MODE).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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