blob: 246c485afe62846baf0d4085ad77b878281698df [file] [log] [blame]
/*
* 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