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Revert "[ACM] iSAC audio codec removed"

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This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1.

Reason for revert: breaks a downstream project

Original change's description:
> [ACM] iSAC audio codec removed
>
> 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}

Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
This commit is contained in:
Alessio Bazzica
2022-11-16 19:13:25 +00:00
committed by WebRTC LUCI CQ
parent cb2b133bf0
commit fbeb76ab51
164 changed files with 39429 additions and 117 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
modules/audio_coding/test/EncodeDecodeTest.cc
View File

@ -110,7 +110,9 @@ void Receiver::Setup(AudioCodingModule* acm,
EXPECT_EQ(0, acm->InitializeReceiver());
if (channels == 1) {
acm->SetReceiveCodecs({{107, {"L16", 8000, 1}},
acm->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{0, {"PCMU", 8000, 1}},
@ -230,6 +232,7 @@ EncodeDecodeTest::EncodeDecodeTest() = default;
void EncodeDecodeTest::Perform() {
const std::map<int, SdpAudioFormat> send_codecs = {
{103, {"ISAC", 16000, 1}}, {104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}},
{8, {"PCMA", 8000, 1}},

35
modules/audio_coding/test/TestAllCodecs.cc
View File

@ -130,7 +130,9 @@ void TestAllCodecs::Perform() {
acm_a_->InitializeReceiver();
acm_b_->InitializeReceiver();
acm_b_->SetReceiveCodecs({{107, {"L16", 8000, 1}},
acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{111, {"L16", 8000, 2}},
@ -184,6 +186,33 @@ void TestAllCodecs::Perform() {
RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
test_count_++;
OpenOutFile(test_count_);
char codec_isac[] = "ISAC";
RegisterSendCodec('A', codec_isac, 16000, -1, 480, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_isac, 16000, -1, 960, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_isac, 16000, 15000, 480, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_isac, 16000, 32000, 960, kVariableSize);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
#ifdef WEBRTC_CODEC_ISAC
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_isac, 32000, 56000, 960, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_isac, 32000, 37000, 960, kVariableSize);
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_isac, 32000, 32000, 960, kVariableSize);
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
test_count_++;
OpenOutFile(test_count_);
@ -290,11 +319,15 @@ void TestAllCodecs::RegisterSendCodec(char side,
// Store packet-size in samples, used to validate the received packet.
// If G.722, store half the size to compensate for the timestamp bug in the
// RFC for G.722.
// If iSAC runs in adaptive mode, packet size in samples can change on the
// fly, so we exclude this test by setting `packet_size_samples_` to -1.
int clockrate_hz = sampling_freq_hz;
size_t num_channels = 1;
if (absl::EqualsIgnoreCase(codec_name, "G722")) {
packet_size_samples_ = packet_size / 2;
clockrate_hz = sampling_freq_hz / 2;
} else if (absl::EqualsIgnoreCase(codec_name, "ISAC") && (rate == -1)) {
packet_size_samples_ = -1;
} else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) {
packet_size_samples_ = packet_size;
num_channels = 2;

33
modules/audio_coding/test/TestRedFec.cc
View File

@ -22,6 +22,8 @@
#include "api/audio_codecs/g711/audio_encoder_g711.h"
#include "api/audio_codecs/g722/audio_decoder_g722.h"
#include "api/audio_codecs/g722/audio_encoder_g722.h"
#include "api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
@ -36,10 +38,12 @@ namespace webrtc {
TestRedFec::TestRedFec()
: encoder_factory_(CreateAudioEncoderFactory<AudioEncoderG711,
AudioEncoderG722,
AudioEncoderIsacFloat,
AudioEncoderL16,
AudioEncoderOpus>()),
decoder_factory_(CreateAudioDecoderFactory<AudioDecoderG711,
AudioDecoderG722,
AudioDecoderIsacFloat,
AudioDecoderL16,
AudioDecoderOpus>()),
_acmA(AudioCodingModule::Create(
@ -91,6 +95,19 @@ void TestRedFec::Perform() {
Run();
_outFileB.Close();
RegisterSendCodec(_acmA, {"ISAC", 16000, 1}, Vad::kVadVeryAggressive, false);
OpenOutFile(_testCntr);
Run();
_outFileB.Close();
// Switch to a 32 kHz codec; RED should be switched off.
RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, Vad::kVadVeryAggressive, false);
OpenOutFile(_testCntr);
Run();
_outFileB.Close();
RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, absl::nullopt, false);
_channelA2B->SetFECTestWithPacketLoss(true);
// Following tests are under packet losses.
@ -101,6 +118,22 @@ void TestRedFec::Perform() {
Run();
_outFileB.Close();
// Switch to a 16 kHz codec, RED should have been switched off.
RegisterSendCodec(_acmA, {"ISAC", 16000, 1}, Vad::kVadVeryAggressive, false);
OpenOutFile(_testCntr);
Run();
_outFileB.Close();
// Switch to a 32 kHz codec, RED should have been switched off.
RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, Vad::kVadVeryAggressive, false);
OpenOutFile(_testCntr);
Run();
_outFileB.Close();
RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, absl::nullopt, false);
RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false);
// _channelA2B imposes 25% packet loss rate.

14
modules/audio_coding/test/TestVADDTX.cc
View File

@ -18,6 +18,8 @@
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include "api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
@ -66,10 +68,12 @@ void MonitoringAudioPacketizationCallback::GetStatistics(uint32_t* counter) {
}
TestVadDtx::TestVadDtx()
: encoder_factory_(
CreateAudioEncoderFactory<AudioEncoderIlbc, AudioEncoderOpus>()),
decoder_factory_(
CreateAudioDecoderFactory<AudioDecoderIlbc, AudioDecoderOpus>()),
: encoder_factory_(CreateAudioEncoderFactory<AudioEncoderIlbc,
AudioEncoderIsacFloat,
AudioEncoderOpus>()),
decoder_factory_(CreateAudioDecoderFactory<AudioDecoderIlbc,
AudioDecoderIsacFloat,
AudioDecoderOpus>()),
acm_send_(AudioCodingModule::Create(
AudioCodingModule::Config(decoder_factory_))),
acm_receive_(AudioCodingModule::Create(
@ -178,6 +182,8 @@ void TestVadDtx::Run(absl::string_view in_filename,
TestWebRtcVadDtx::TestWebRtcVadDtx() : output_file_num_(0) {}
void TestWebRtcVadDtx::Perform() {
RunTestCases({"ISAC", 16000, 1});
RunTestCases({"ISAC", 32000, 1});
RunTestCases({"ILBC", 8000, 1});
RunTestCases({"opus", 48000, 2});
}

20
modules/audio_coding/test/Tester.cc
View File

@ -21,6 +21,7 @@
#include "modules/audio_coding/test/TestStereo.h"
#include "modules/audio_coding/test/TestVADDTX.h"
#include "modules/audio_coding/test/TwoWayCommunication.h"
#include "modules/audio_coding/test/iSACTest.h"
#include "modules/audio_coding/test/opus_test.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@ -41,6 +42,25 @@ TEST(AudioCodingModuleTest, TestRedFec) {
webrtc::TestRedFec().Perform();
}
#if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TestIsac) {
#else
TEST(AudioCodingModuleTest, TestIsac) {
#endif
webrtc::ISACTest().Perform();
}
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC)
#if defined(WEBRTC_ANDROID)
TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
#else
TEST(AudioCodingModuleTest, TwoWayCommunication) {
#endif
webrtc::TwoWayCommunication().Perform();
}
#endif
// Disabled on ios as flaky, see https://crbug.com/webrtc/7057
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
TEST(AudioCodingModuleTest, DISABLED_TestStereo) {

273
modules/audio_coding/test/iSACTest.cc Normal file
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@ -0,0 +1,273 @@
/*
* Copyright (c) 2012 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.
*/
#include "modules/audio_coding/test/iSACTest.h"
#include <stdio.h>
#include <string.h>
#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
using ::testing::AnyOf;
using ::testing::Eq;
using ::testing::StrCaseEq;
namespace {
constexpr int kISAC16kPayloadType = 103;
constexpr int kISAC32kPayloadType = 104;
const SdpAudioFormat kISAC16kFormat = {"ISAC", 16000, 1};
const SdpAudioFormat kISAC32kFormat = {"ISAC", 32000, 1};
AudioEncoderIsacFloat::Config TweakConfig(
AudioEncoderIsacFloat::Config config,
const ACMTestISACConfig& test_config) {
if (test_config.currentRateBitPerSec > 0) {
config.bit_rate = test_config.currentRateBitPerSec;
}
if (test_config.currentFrameSizeMsec != 0) {
config.frame_size_ms = test_config.currentFrameSizeMsec;
}
EXPECT_THAT(config.IsOk(), Eq(true));
return config;
}
void SetISACConfigDefault(ACMTestISACConfig& isacConfig) {
isacConfig.currentRateBitPerSec = 0;
isacConfig.currentFrameSizeMsec = 0;
isacConfig.encodingMode = -1;
isacConfig.initRateBitPerSec = 0;
isacConfig.initFrameSizeInMsec = 0;
isacConfig.enforceFrameSize = false;
}
} // namespace
ISACTest::ISACTest()
: _acmA(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
_acmB(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {}
ISACTest::~ISACTest() {}
void ISACTest::Setup() {
// Register both iSAC-wb & iSAC-swb in both sides as receiver codecs.
std::map<int, SdpAudioFormat> receive_codecs = {
{kISAC16kPayloadType, kISAC16kFormat},
{kISAC32kPayloadType, kISAC32kFormat}};
_acmA->SetReceiveCodecs(receive_codecs);
_acmB->SetReceiveCodecs(receive_codecs);
//--- Set A-to-B channel
_channel_A2B.reset(new Channel);
EXPECT_EQ(0, _acmA->RegisterTransportCallback(_channel_A2B.get()));
_channel_A2B->RegisterReceiverACM(_acmB.get());
//--- Set B-to-A channel
_channel_B2A.reset(new Channel);
EXPECT_EQ(0, _acmB->RegisterTransportCallback(_channel_B2A.get()));
_channel_B2A->RegisterReceiverACM(_acmA.get());
file_name_swb_ =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
_acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
kISAC16kPayloadType));
_acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
kISAC32kPayloadType));
_inFileA.Open(file_name_swb_, 32000, "rb");
// Set test length to 500 ms (50 blocks of 10 ms each).
_inFileA.SetNum10MsBlocksToRead(50);
// Fast-forward 1 second (100 blocks) since the files start with silence.
_inFileA.FastForward(100);
std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm";
std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm";
_outFileA.Open(fileNameA, 32000, "wb");
_outFileB.Open(fileNameB, 32000, "wb");
while (!_inFileA.EndOfFile()) {
Run10ms();
}
_inFileA.Close();
_outFileA.Close();
_outFileB.Close();
}
void ISACTest::Perform() {
Setup();
int16_t testNr = 0;
ACMTestISACConfig wbISACConfig;
ACMTestISACConfig swbISACConfig;
SetISACConfigDefault(wbISACConfig);
SetISACConfigDefault(swbISACConfig);
wbISACConfig.currentRateBitPerSec = -1;
swbISACConfig.currentRateBitPerSec = -1;
testNr++;
EncodeDecode(testNr, wbISACConfig, swbISACConfig);
SetISACConfigDefault(wbISACConfig);
SetISACConfigDefault(swbISACConfig);
testNr++;
EncodeDecode(testNr, wbISACConfig, swbISACConfig);
testNr++;
SwitchingSamplingRate(testNr, 4);
}
void ISACTest::Run10ms() {
AudioFrame audioFrame;
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
bool muted;
EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame, &muted));
ASSERT_FALSE(muted);
_outFileA.Write10MsData(audioFrame);
EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame, &muted));
ASSERT_FALSE(muted);
_outFileB.Write10MsData(audioFrame);
}
void ISACTest::EncodeDecode(int testNr,
ACMTestISACConfig& wbISACConfig,
ACMTestISACConfig& swbISACConfig) {
// Files in Side A and B
_inFileA.Open(file_name_swb_, 32000, "rb", true);
_inFileB.Open(file_name_swb_, 32000, "rb", true);
std::string file_name_out;
rtc::StringBuilder file_stream_a;
rtc::StringBuilder file_stream_b;
file_stream_a << webrtc::test::OutputPath();
file_stream_b << webrtc::test::OutputPath();
file_stream_a << "out_iSACTest_A_" << testNr << ".pcm";
file_stream_b << "out_iSACTest_B_" << testNr << ".pcm";
file_name_out = file_stream_a.str();
_outFileA.Open(file_name_out, 32000, "wb");
file_name_out = file_stream_b.str();
_outFileB.Open(file_name_out, 32000, "wb");
// Side A is sending super-wideband, and side B is sending wideband.
_acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
swbISACConfig),
kISAC32kPayloadType));
_acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
wbISACConfig),
kISAC16kPayloadType));
_channel_A2B->ResetStats();
_channel_B2A->ResetStats();
while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
Run10ms();
}
_channel_A2B->ResetStats();
_channel_B2A->ResetStats();
_outFileA.Close();
_outFileB.Close();
_inFileA.Close();
_inFileB.Close();
}
void ISACTest::SwitchingSamplingRate(int testNr, int maxSampRateChange) {
// Files in Side A
_inFileA.Open(file_name_swb_, 32000, "rb");
_inFileB.Open(file_name_swb_, 32000, "rb");
std::string file_name_out;
rtc::StringBuilder file_stream_a;
rtc::StringBuilder file_stream_b;
file_stream_a << webrtc::test::OutputPath();
file_stream_b << webrtc::test::OutputPath();
file_stream_a << "out_iSACTest_A_" << testNr << ".pcm";
file_stream_b << "out_iSACTest_B_" << testNr << ".pcm";
file_name_out = file_stream_a.str();
_outFileA.Open(file_name_out, 32000, "wb");
file_name_out = file_stream_b.str();
_outFileB.Open(file_name_out, 32000, "wb");
// Start with side A sending super-wideband and side B seding wideband.
// Toggle sending wideband/super-wideband in this test.
_acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
kISAC32kPayloadType));
_acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
kISAC16kPayloadType));
int numSendCodecChanged = 0;
while (numSendCodecChanged < (maxSampRateChange << 1)) {
Run10ms();
if (_inFileA.EndOfFile()) {
if (_inFileA.SamplingFrequency() == 16000) {
// Switch side A to send super-wideband.
_inFileA.Close();
_inFileA.Open(file_name_swb_, 32000, "rb");
_acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
kISAC32kPayloadType));
} else {
// Switch side A to send wideband.
_inFileA.Close();
_inFileA.Open(file_name_swb_, 32000, "rb");
_acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
kISAC16kPayloadType));
}
numSendCodecChanged++;
}
if (_inFileB.EndOfFile()) {
if (_inFileB.SamplingFrequency() == 16000) {
// Switch side B to send super-wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
_acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
kISAC32kPayloadType));
} else {
// Switch side B to send wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
_acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
kISAC16kPayloadType));
}
numSendCodecChanged++;
}
}
_outFileA.Close();
_outFileB.Close();
_inFileA.Close();
_inFileB.Close();
}
} // namespace webrtc

68
modules/audio_coding/test/iSACTest.h Normal file
View File

@ -0,0 +1,68 @@
/*
* Copyright (c) 2012 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.
*/
#ifndef MODULES_AUDIO_CODING_TEST_ISACTEST_H_
#define MODULES_AUDIO_CODING_TEST_ISACTEST_H_
#include <string.h>
#include <memory>
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
namespace webrtc {
struct ACMTestISACConfig {
int32_t currentRateBitPerSec;
int16_t currentFrameSizeMsec;
int16_t encodingMode;
uint32_t initRateBitPerSec;
int16_t initFrameSizeInMsec;
bool enforceFrameSize;
};
class ISACTest {
public:
ISACTest();
~ISACTest();
void Perform();
private:
void Setup();
void Run10ms();
void EncodeDecode(int testNr,
ACMTestISACConfig& wbISACConfig,
ACMTestISACConfig& swbISACConfig);
void SwitchingSamplingRate(int testNr, int maxSampRateChange);
std::unique_ptr<AudioCodingModule> _acmA;
std::unique_ptr<AudioCodingModule> _acmB;
std::unique_ptr<Channel> _channel_A2B;
std::unique_ptr<Channel> _channel_B2A;
PCMFile _inFileA;
PCMFile _inFileB;
PCMFile _outFileA;
PCMFile _outFileB;
std::string file_name_swb_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_ISACTEST_H_