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84f75699c6
This change makes AudioCodingModule a pure sender and AcmReceiver a pure receiver. The Config struct is in practice no longer used by AudioCodingModule, so a new definition is included in AcmReceiver. The old definition remains in AudioCodingModule while downstream clients are being updated. Bug: webrtc:14867 Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533 Reviewed-by: Tomas Lundqvist <tomasl@google.com> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39244}
101 lines
2.9 KiB
C++
101 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/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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