blob: cf1f91365fc5147cd8045792e0ba4ce7540a31bf [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/TestStereo.h"
#include <string>
#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
// Class for simulating packet handling
TestPackStereo::TestPackStereo()
: receiver_acm_(NULL),
seq_no_(0),
timestamp_diff_(0),
last_in_timestamp_(0),
total_bytes_(0),
payload_size_(0),
lost_packet_(false) {}
TestPackStereo::~TestPackStereo() {}
void TestPackStereo::RegisterReceiverACM(acm2::AcmReceiver* acm_receiver) {
receiver_acm_ = acm_receiver;
return;
}
int32_t TestPackStereo::SendData(const AudioFrameType frame_type,
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
const size_t payload_size,
int64_t absolute_capture_timestamp_ms) {
RTPHeader rtp_header;
int32_t status = 0;
rtp_header.markerBit = false;
rtp_header.ssrc = 0;
rtp_header.sequenceNumber = seq_no_++;
rtp_header.payloadType = payload_type;
rtp_header.timestamp = timestamp;
if (frame_type == AudioFrameType::kEmptyFrame) {
// Skip this frame
return 0;
}
if (lost_packet_ == false) {
status = receiver_acm_->InsertPacket(
rtp_header, rtc::ArrayView<const uint8_t>(payload_data, payload_size),
/*receive_time=*/Timestamp::MinusInfinity());
if (frame_type != AudioFrameType::kAudioFrameCN) {
payload_size_ = static_cast<int>(payload_size);
} else {
payload_size_ = -1;
}
timestamp_diff_ = timestamp - last_in_timestamp_;
last_in_timestamp_ = timestamp;
total_bytes_ += payload_size;
}
return status;
}
uint16_t TestPackStereo::payload_size() {
return static_cast<uint16_t>(payload_size_);
}
uint32_t TestPackStereo::timestamp_diff() {
return timestamp_diff_;
}
void TestPackStereo::reset_payload_size() {
payload_size_ = 0;
}
void TestPackStereo::set_codec_mode(enum StereoMonoMode mode) {
codec_mode_ = mode;
}
void TestPackStereo::set_lost_packet(bool lost) {
lost_packet_ = lost;
}
TestStereo::TestStereo()
: acm_a_(AudioCodingModule::Create()),
acm_b_(std::make_unique<acm2::AcmReceiver>(
acm2::AcmReceiver::Config(CreateBuiltinAudioDecoderFactory()))),
channel_a2b_(NULL),
test_cntr_(0),
pack_size_samp_(0),
pack_size_bytes_(0),
counter_(0) {}
TestStereo::~TestStereo() {
if (channel_a2b_ != NULL) {
delete channel_a2b_;
channel_a2b_ = NULL;
}
}
void TestStereo::Perform() {
uint16_t frequency_hz;
int audio_channels;
int codec_channels;
// Open both mono and stereo test files in 32 kHz.
const std::string file_name_stereo =
webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
const std::string file_name_mono =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
frequency_hz = 32000;
in_file_stereo_ = new PCMFile();
in_file_mono_ = new PCMFile();
in_file_stereo_->Open(file_name_stereo, frequency_hz, "rb");
in_file_stereo_->ReadStereo(true);
in_file_mono_->Open(file_name_mono, frequency_hz, "rb");
in_file_mono_->ReadStereo(false);
// Create and initialize two ACMs, one for each side of a one-to-one call.
ASSERT_TRUE((acm_a_.get() != NULL) && (acm_b_.get() != NULL));
acm_b_->FlushBuffers();
acm_b_->SetCodecs({{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{111, {"L16", 8000, 2}},
{112, {"L16", 16000, 2}},
{113, {"L16", 32000, 2}},
{0, {"PCMU", 8000, 1}},
{110, {"PCMU", 8000, 2}},
{8, {"PCMA", 8000, 1}},
{118, {"PCMA", 8000, 2}},
{102, {"ILBC", 8000, 1}},
{9, {"G722", 8000, 1}},
{119, {"G722", 8000, 2}},
{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
{13, {"CN", 8000, 1}},
{98, {"CN", 16000, 1}},
{99, {"CN", 32000, 1}}});
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
EXPECT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_));
channel_a2b_->RegisterReceiverACM(acm_b_.get());
char codec_pcma_temp[] = "PCMA";
RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2);
//
// Test Stereo-To-Stereo for all codecs.
//
audio_channels = 2;
codec_channels = 2;
// TODO(bugs.webrtc.org/345525069): Either fix/enable or remove G722.
#if defined(__has_feature) && !__has_feature(undefined_behavior_sanitizer)
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
channel_a2b_->set_codec_mode(kStereo);
test_cntr_++;
OpenOutFile(test_cntr_);
char codec_g722[] = "G722";
RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_g722, 16000, 64000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_g722, 16000, 64000, 480, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_g722, 16000, 64000, 640, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_g722, 16000, 64000, 800, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_g722, 16000, 64000, 960, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
channel_a2b_->set_codec_mode(kStereo);
test_cntr_++;
OpenOutFile(test_cntr_);
char codec_l16[] = "L16";
RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_l16, 8000, 128000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_l16, 8000, 128000, 240, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_l16, 8000, 128000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_l16, 16000, 256000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_l16, 16000, 256000, 480, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_l16, 16000, 256000, 640, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_l16, 32000, 512000, 640, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#ifdef PCMA_AND_PCMU
channel_a2b_->set_codec_mode(kStereo);
audio_channels = 2;
codec_channels = 2;
test_cntr_++;
OpenOutFile(test_cntr_);
char codec_pcma[] = "PCMA";
RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
test_cntr_++;
OpenOutFile(test_cntr_);
char codec_pcmu[] = "PCMU";
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
#ifdef WEBRTC_CODEC_OPUS
channel_a2b_->set_codec_mode(kStereo);
audio_channels = 2;
codec_channels = 2;
test_cntr_++;
OpenOutFile(test_cntr_);
char codec_opus[] = "opus";
// Run Opus with 10 ms frame size.
RegisterSendCodec('A', codec_opus, 48000, 64000, 480, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
// Run Opus with 20 ms frame size.
RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 2, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
// Run Opus with 40 ms frame size.
RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
// Run Opus with 60 ms frame size.
RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 6, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
// Run Opus with 20 ms frame size and different bitrates.
RegisterSendCodec('A', codec_opus, 48000, 40000, 960, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_opus, 48000, 510000, 960, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
//
// Test Mono-To-Stereo for all codecs.
//
audio_channels = 1;
codec_channels = 2;
// TODO(bugs.webrtc.org/345525069): Either fix/enable or remove G722.
#if defined(__has_feature) && !__has_feature(undefined_behavior_sanitizer)
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#ifdef PCMA_AND_PCMU
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
#ifdef WEBRTC_CODEC_OPUS
// Keep encode and decode in stereo.
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_opus, 48000, 64000, 960, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
// Encode in mono, decode in stereo mode.
RegisterSendCodec('A', codec_opus, 48000, 64000, 960, 1);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
//
// Test Stereo-To-Mono for all codecs.
//
audio_channels = 2;
codec_channels = 1;
channel_a2b_->set_codec_mode(kMono);
// TODO(bugs.webrtc.org/345525069): Either fix/enable or remove G722.
#if defined(__has_feature) && !__has_feature(undefined_behavior_sanitizer)
// Run stereo audio and mono codec.
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#ifdef PCMA_AND_PCMU
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
#ifdef WEBRTC_CODEC_OPUS
test_cntr_++;
OpenOutFile(test_cntr_);
// Encode and decode in mono.
RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels);
acm_b_->SetCodecs({{120, {"OPUS", 48000, 2}}});
Run(channel_a2b_, audio_channels, codec_channels);
// Encode in stereo, decode in mono.
RegisterSendCodec('A', codec_opus, 48000, 32000, 960, 2);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
// Test switching between decoding mono and stereo for Opus.
// Decode in mono.
test_cntr_++;
OpenOutFile(test_cntr_);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
// Decode in stereo.
test_cntr_++;
OpenOutFile(test_cntr_);
acm_b_->SetCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
Run(channel_a2b_, audio_channels, 2);
out_file_.Close();
// Decode in mono.
test_cntr_++;
OpenOutFile(test_cntr_);
acm_b_->SetCodecs({{120, {"OPUS", 48000, 2}}});
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#endif
// Delete the file pointers.
delete in_file_stereo_;
delete in_file_mono_;
}
// Register Codec to use in the test
//
// Input: side - which ACM to use, 'A' or 'B'
// codec_name - name to use when register the codec
// sampling_freq_hz - sampling frequency in Herz
// rate - bitrate in bytes
// pack_size - packet size in samples
// channels - number of channels; 1 for mono, 2 for stereo
void TestStereo::RegisterSendCodec(char side,
char* codec_name,
int32_t sampling_freq_hz,
int rate,
int pack_size,
int channels) {
// Store packet size in samples, used to validate the received packet
pack_size_samp_ = pack_size;
// Store the expected packet size in bytes, used to validate the received
// packet. Add 0.875 to always round up to a whole byte.
pack_size_bytes_ = (uint16_t)(static_cast<float>(pack_size * rate) /
static_cast<float>(sampling_freq_hz * 8) +
0.875);
// Set pointer to the ACM where to register the codec
AudioCodingModule* my_acm = NULL;
switch (side) {
case 'A': {
my_acm = acm_a_.get();
break;
}
case 'B': {
// We no longer use this case. Refactor code to avoid the switch.
ASSERT_TRUE(false);
// my_acm = acm_b_.get();
break;
}
default:
break;
}
ASSERT_TRUE(my_acm != NULL);
auto encoder_factory = CreateBuiltinAudioEncoderFactory();
const int clockrate_hz = absl::EqualsIgnoreCase(codec_name, "g722")
? sampling_freq_hz / 2
: sampling_freq_hz;
const std::string ptime = rtc::ToString(rtc::CheckedDivExact(
pack_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
CodecParameterMap params = {{"ptime", ptime}};
RTC_CHECK(channels == 1 || channels == 2);
if (absl::EqualsIgnoreCase(codec_name, "opus")) {
if (channels == 2) {
params["stereo"] = "1";
}
channels = 2;
params["maxaveragebitrate"] = rtc::ToString(rate);
}
constexpr int payload_type = 17;
auto encoder = encoder_factory->MakeAudioEncoder(
payload_type, SdpAudioFormat(codec_name, clockrate_hz, channels, params),
absl::nullopt);
EXPECT_NE(nullptr, encoder);
my_acm->SetEncoder(std::move(encoder));
send_codec_name_ = codec_name;
}
void TestStereo::Run(TestPackStereo* channel,
int in_channels,
int out_channels,
int percent_loss) {
AudioFrame audio_frame;
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
uint16_t rec_size;
uint32_t time_stamp_diff;
channel->reset_payload_size();
int error_count = 0;
int variable_bytes = 0;
int variable_packets = 0;
// Set test length to 500 ms (50 blocks of 10 ms each).
in_file_mono_->SetNum10MsBlocksToRead(50);
in_file_stereo_->SetNum10MsBlocksToRead(50);
// Fast-forward 1 second (100 blocks) since the files start with silence.
in_file_stereo_->FastForward(100);
in_file_mono_->FastForward(100);
while (true) {
// Simulate packet loss by setting `packet_loss_` to "true" in
// `percent_loss` percent of the loops.
if (percent_loss > 0) {
if (counter_ == floor((100 / percent_loss) + 0.5)) {
counter_ = 0;
channel->set_lost_packet(true);
} else {
channel->set_lost_packet(false);
}
counter_++;
}
// Add 10 msec to ACM
if (in_channels == 1) {
if (in_file_mono_->EndOfFile()) {
break;
}
in_file_mono_->Read10MsData(audio_frame);
} else {
if (in_file_stereo_->EndOfFile()) {
break;
}
in_file_stereo_->Read10MsData(audio_frame);
}
EXPECT_GE(acm_a_->Add10MsData(audio_frame), 0);
// Verify that the received packet size matches the settings.
rec_size = channel->payload_size();
if ((0 < rec_size) & (rec_size < 65535)) {
if (strcmp(send_codec_name_, "opus") == 0) {
// Opus is a variable rate codec, hence calculate the average packet
// size, and later make sure the average is in the right range.
variable_bytes += rec_size;
variable_packets++;
} else {
// For fixed rate codecs, check that packet size is correct.
if ((rec_size != pack_size_bytes_ * out_channels) &&
(pack_size_bytes_ < 65535)) {
error_count++;
}
}
// Verify that the timestamp is updated with expected length
time_stamp_diff = channel->timestamp_diff();
if ((counter_ > 10) && (time_stamp_diff != pack_size_samp_)) {
error_count++;
}
}
// Run receive side of ACM
bool muted;
EXPECT_EQ(0, acm_b_->GetAudio(out_freq_hz_b, &audio_frame, &muted));
ASSERT_FALSE(muted);
// Write output speech to file
out_file_.Write10MsData(
audio_frame.data(),
audio_frame.samples_per_channel_ * audio_frame.num_channels_);
}
EXPECT_EQ(0, error_count);
// Check that packet size is in the right range for variable rate codecs,
// such as Opus.
if (variable_packets > 0) {
variable_bytes /= variable_packets;
EXPECT_NEAR(variable_bytes, pack_size_bytes_, 18);
}
if (in_file_mono_->EndOfFile()) {
in_file_mono_->Rewind();
}
if (in_file_stereo_->EndOfFile()) {
in_file_stereo_->Rewind();
}
// Reset in case we ended with a lost packet
channel->set_lost_packet(false);
}
void TestStereo::OpenOutFile(int16_t test_number) {
std::string file_name;
rtc::StringBuilder file_stream;
file_stream << webrtc::test::OutputPath() << "teststereo_out_" << test_number
<< ".pcm";
file_name = file_stream.str();
out_file_.Open(file_name, 32000, "wb");
}
} // namespace webrtc