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

265 lines
9.4 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/TestVADDTX.h"
#include <string>
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/audio_codecs/audio_decoder_factory_template.h"
#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/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/test/PCMFile.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
MonitoringAudioPacketizationCallback::MonitoringAudioPacketizationCallback(
AudioPacketizationCallback* next)
: next_(next) {
ResetStatistics();
}
int32_t MonitoringAudioPacketizationCallback::SendData(
AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) {
counter_[static_cast<int>(frame_type)]++;
return next_->SendData(frame_type, payload_type, timestamp, payload_data,
payload_len_bytes, absolute_capture_timestamp_ms);
}
void MonitoringAudioPacketizationCallback::PrintStatistics() {
printf("\n");
printf("kEmptyFrame %u\n",
counter_[static_cast<int>(AudioFrameType::kEmptyFrame)]);
printf("kAudioFrameSpeech %u\n",
counter_[static_cast<int>(AudioFrameType::kAudioFrameSpeech)]);
printf("kAudioFrameCN %u\n",
counter_[static_cast<int>(AudioFrameType::kAudioFrameCN)]);
printf("\n\n");
}
void MonitoringAudioPacketizationCallback::ResetStatistics() {
memset(counter_, 0, sizeof(counter_));
}
void MonitoringAudioPacketizationCallback::GetStatistics(uint32_t* counter) {
memcpy(counter, counter_, sizeof(counter_));
}
TestVadDtx::TestVadDtx()
: encoder_factory_(
CreateAudioEncoderFactory<AudioEncoderIlbc, AudioEncoderOpus>()),
decoder_factory_(
CreateAudioDecoderFactory<AudioDecoderIlbc, AudioDecoderOpus>()),
acm_send_(AudioCodingModule::Create(
AudioCodingModule::Config(decoder_factory_))),
acm_receive_(AudioCodingModule::Create(
AudioCodingModule::Config(decoder_factory_))),
channel_(std::make_unique<Channel>()),
packetization_callback_(
std::make_unique<MonitoringAudioPacketizationCallback>(
channel_.get())) {
EXPECT_EQ(
0, acm_send_->RegisterTransportCallback(packetization_callback_.get()));
channel_->RegisterReceiverACM(acm_receive_.get());
}
bool TestVadDtx::RegisterCodec(const SdpAudioFormat& codec_format,
absl::optional<Vad::Aggressiveness> vad_mode) {
constexpr int payload_type = 17, cn_payload_type = 117;
bool added_comfort_noise = false;
auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
absl::nullopt);
if (vad_mode.has_value() &&
!absl::EqualsIgnoreCase(codec_format.name, "opus")) {
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));
added_comfort_noise = true;
}
channel_->SetIsStereo(encoder->NumChannels() > 1);
acm_send_->SetEncoder(std::move(encoder));
std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
acm_receive_->SetReceiveCodecs(receive_codecs);
return added_comfort_noise;
}
// Encoding a file and see if the numbers that various packets occur follow
// the expectation.
void TestVadDtx::Run(absl::string_view in_filename,
int frequency,
int channels,
absl::string_view out_filename,
bool append,
const int* expects) {
packetization_callback_->ResetStatistics();
PCMFile in_file;
in_file.Open(in_filename, frequency, "rb");
in_file.ReadStereo(channels > 1);
// Set test length to 1000 ms (100 blocks of 10 ms each).
in_file.SetNum10MsBlocksToRead(100);
// Fast-forward both files 500 ms (50 blocks). The first second of the file is
// silence, but we want to keep half of that to test silence periods.
in_file.FastForward(50);
PCMFile out_file;
if (append) {
out_file.Open(out_filename, kOutputFreqHz, "ab");
} else {
out_file.Open(out_filename, kOutputFreqHz, "wb");
}
uint16_t frame_size_samples = in_file.PayloadLength10Ms();
AudioFrame audio_frame;
while (!in_file.EndOfFile()) {
in_file.Read10MsData(audio_frame);
audio_frame.timestamp_ = time_stamp_;
time_stamp_ += frame_size_samples;
EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
bool muted;
acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted);
ASSERT_FALSE(muted);
out_file.Write10MsData(audio_frame);
}
in_file.Close();
out_file.Close();
#ifdef PRINT_STAT
packetization_callback_->PrintStatistics();
#endif
uint32_t stats[3];
packetization_callback_->GetStatistics(stats);
packetization_callback_->ResetStatistics();
for (const auto& st : stats) {
int i = &st - stats; // Calculate the current position in stats.
switch (expects[i]) {
case 0: {
EXPECT_EQ(0u, st) << "stats[" << i << "] error.";
break;
}
case 1: {
EXPECT_GT(st, 0u) << "stats[" << i << "] error.";
break;
}
}
}
}
// Following is the implementation of TestWebRtcVadDtx.
TestWebRtcVadDtx::TestWebRtcVadDtx() : output_file_num_(0) {}
void TestWebRtcVadDtx::Perform() {
RunTestCases({"ILBC", 8000, 1});
RunTestCases({"opus", 48000, 2});
}
// Test various configurations on VAD/DTX.
void TestWebRtcVadDtx::RunTestCases(const SdpAudioFormat& codec_format) {
Test(/*new_outfile=*/true,
/*expect_dtx_enabled=*/RegisterCodec(codec_format, absl::nullopt));
Test(/*new_outfile=*/false,
/*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadAggressive));
Test(/*new_outfile=*/false,
/*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadLowBitrate));
Test(/*new_outfile=*/false, /*expect_dtx_enabled=*/RegisterCodec(
codec_format, Vad::kVadVeryAggressive));
Test(/*new_outfile=*/false,
/*expect_dtx_enabled=*/RegisterCodec(codec_format, Vad::kVadNormal));
}
// Set the expectation and run the test.
void TestWebRtcVadDtx::Test(bool new_outfile, bool expect_dtx_enabled) {
int expects[] = {-1, 1, expect_dtx_enabled, 0, 0};
if (new_outfile) {
output_file_num_++;
}
rtc::StringBuilder out_filename;
out_filename << webrtc::test::OutputPath() << "testWebRtcVadDtx_outFile_"
<< output_file_num_ << ".pcm";
Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
out_filename.str(), !new_outfile, expects);
}
// Following is the implementation of TestOpusDtx.
void TestOpusDtx::Perform() {
int expects[] = {0, 1, 0, 0, 0};
// Register Opus as send codec
std::string out_filename =
webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm";
RegisterCodec({"opus", 48000, 2}, absl::nullopt);
acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) {
(*encoder_ptr)->SetDtx(false);
});
Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
out_filename, false, expects);
acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) {
(*encoder_ptr)->SetDtx(true);
});
expects[static_cast<int>(AudioFrameType::kEmptyFrame)] = 1;
expects[static_cast<int>(AudioFrameType::kAudioFrameCN)] = 1;
Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
out_filename, true, expects);
// Register stereo Opus as send codec
out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
RegisterCodec({"opus", 48000, 2, {{"stereo", "1"}}}, absl::nullopt);
acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) {
(*encoder_ptr)->SetDtx(false);
});
expects[static_cast<int>(AudioFrameType::kEmptyFrame)] = 0;
expects[static_cast<int>(AudioFrameType::kAudioFrameCN)] = 0;
Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 32000,
2, out_filename, false, expects);
acm_send_->ModifyEncoder([](std::unique_ptr<AudioEncoder>* encoder_ptr) {
(*encoder_ptr)->SetDtx(true);
// The default bitrate will not generate frames recognized as CN on desktop
// since the frames will be encoded as CELT. Set a low target bitrate to get
// consistent behaviour across platforms.
(*encoder_ptr)->OnReceivedTargetAudioBitrate(24000);
});
expects[static_cast<int>(AudioFrameType::kEmptyFrame)] = 1;
expects[static_cast<int>(AudioFrameType::kAudioFrameCN)] = 1;
Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 32000,
2, out_filename, true, expects);
}
} // namespace webrtc