Alessio Bazzica b46c4bf27b [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}
2022-11-16 16:42:55 +00:00

201 lines
7.0 KiB
C++

/*
* 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/TestRedFec.h"
#include <memory>
#include <utility>
#include "absl/strings/match.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_decoder_factory_template.h"
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/g711/audio_decoder_g711.h"
#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/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
TestRedFec::TestRedFec()
: encoder_factory_(CreateAudioEncoderFactory<AudioEncoderG711,
AudioEncoderG722,
AudioEncoderL16,
AudioEncoderOpus>()),
decoder_factory_(CreateAudioDecoderFactory<AudioDecoderG711,
AudioDecoderG722,
AudioDecoderL16,
AudioDecoderOpus>()),
_acmA(AudioCodingModule::Create(
AudioCodingModule::Config(decoder_factory_))),
_acmB(AudioCodingModule::Create(
AudioCodingModule::Config(decoder_factory_))),
_channelA2B(NULL),
_testCntr(0) {}
TestRedFec::~TestRedFec() {
if (_channelA2B != NULL) {
delete _channelA2B;
_channelA2B = NULL;
}
}
void TestRedFec::Perform() {
const std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
_inFileA.Open(file_name, 32000, "rb");
ASSERT_EQ(0, _acmA->InitializeReceiver());
ASSERT_EQ(0, _acmB->InitializeReceiver());
// Create and connect the channel
_channelA2B = new Channel;
_acmA->RegisterTransportCallback(_channelA2B);
_channelA2B->RegisterReceiverACM(_acmB.get());
RegisterSendCodec(_acmA, {"L16", 8000, 1}, Vad::kVadAggressive, true);
OpenOutFile(_testCntr);
Run();
_outFileB.Close();
// Switch to another 8 kHz codec; RED should remain switched on.
RegisterSendCodec(_acmA, {"PCMU", 8000, 1}, Vad::kVadAggressive, true);
OpenOutFile(_testCntr);
Run();
_outFileB.Close();
// Switch to a 16 kHz codec; RED should be switched off.
RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
OpenOutFile(_testCntr);
RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
Run();
RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
Run();
_outFileB.Close();
_channelA2B->SetFECTestWithPacketLoss(true);
// Following tests are under packet losses.
// Switch to a 16 kHz codec; RED should be switched off.
RegisterSendCodec(_acmA, {"G722", 8000, 1}, Vad::kVadAggressive, false);
OpenOutFile(_testCntr);
Run();
_outFileB.Close();
RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false);
// _channelA2B imposes 25% packet loss rate.
EXPECT_EQ(0, _acmA->SetPacketLossRate(25));
_acmA->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
EXPECT_EQ(true, (*enc)->SetFec(true));
});
OpenOutFile(_testCntr);
Run();
// Switch to L16 with RED.
RegisterSendCodec(_acmA, {"L16", 8000, 1}, absl::nullopt, true);
Run();
// Switch to Opus again.
RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false);
_acmA->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
EXPECT_EQ(true, (*enc)->SetFec(false));
});
Run();
_acmA->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* enc) {
EXPECT_EQ(true, (*enc)->SetFec(true));
});
_outFileB.Close();
}
void TestRedFec::RegisterSendCodec(
const std::unique_ptr<AudioCodingModule>& acm,
const SdpAudioFormat& codec_format,
absl::optional<Vad::Aggressiveness> vad_mode,
bool use_red) {
constexpr int payload_type = 17, cn_payload_type = 27, red_payload_type = 37;
const auto& other_acm = &acm == &_acmA ? _acmB : _acmA;
auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
absl::nullopt);
EXPECT_NE(encoder, nullptr);
std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
if (!absl::EqualsIgnoreCase(codec_format.name, "opus")) {
if (vad_mode.has_value()) {
AudioEncoderCngConfig config;
config.speech_encoder = std::move(encoder);
config.num_channels = 1;
config.payload_type = cn_payload_type;
config.vad_mode = vad_mode.value();
encoder = CreateComfortNoiseEncoder(std::move(config));
receive_codecs.emplace(std::make_pair(
cn_payload_type, SdpAudioFormat("CN", codec_format.clockrate_hz, 1)));
}
if (use_red) {
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type;
config.speech_encoder = std::move(encoder);
encoder = std::make_unique<AudioEncoderCopyRed>(std::move(config),
field_trials_);
receive_codecs.emplace(
std::make_pair(red_payload_type,
SdpAudioFormat("red", codec_format.clockrate_hz, 1)));
}
}
acm->SetEncoder(std::move(encoder));
other_acm->SetReceiveCodecs(receive_codecs);
}
void TestRedFec::Run() {
AudioFrame audioFrame;
int32_t outFreqHzB = _outFileB.SamplingFrequency();
// Set test length to 500 ms (50 blocks of 10 ms each).
_inFileA.SetNum10MsBlocksToRead(50);
// Fast-forward 1 second (100 blocks) since the file starts with silence.
_inFileA.FastForward(100);
while (!_inFileA.EndOfFile()) {
EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
bool muted;
EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame, &muted));
ASSERT_FALSE(muted);
_outFileB.Write10MsData(audioFrame.data(), audioFrame.samples_per_channel_);
}
_inFileA.Rewind();
}
void TestRedFec::OpenOutFile(int16_t test_number) {
std::string file_name;
rtc::StringBuilder file_stream;
file_stream << webrtc::test::OutputPath();
file_stream << "TestRedFec_outFile_";
file_stream << test_number << ".pcm";
file_name = file_stream.str();
_outFileB.Open(file_name, 16000, "wb");
}
} // namespace webrtc