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webrtc_m130/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
ossu 10a029e952 Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. 
For backwards compatibility, I've added kept the old interface to
Encode() and EncodeInternal and created default implementations of both
variants of EncodeInternal(), each calling the other. At least one of
the variants must be implemented in a subclass or we'll run out of stack
and explode. Would be nice if we could catch that before runtime. :/

The new interface to EncodeInternal() is protected, since it should
never be called from the outside.

Was unable to mark the old EncodeInternal() as RTC_DEPRECATED, since the
default implementaion of the new variant needs to call it to work around
old implementations. The old Encode() variant is deprecated, at least.

Added a test for backwards compatibility in audio_encoder_unittest.cc.
For the added test I broke out MockEncodeHelper from
audio_encoder_copy_red_unittest.cc and renamed it MockAudioEncoderHelper.

Review URL: https://codereview.webrtc.org/1725143003

Cr-Commit-Position: refs/heads/master@{#11823}
2016-03-01 08:41:39 +00:00

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/*
* 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 "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include <assert.h>
#include <stdlib.h>
#include <memory>
#include <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "webrtc/system_wrappers/include/data_log.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
namespace {
// The absolute difference between the input and output (the first channel) is
// compared vs |tolerance|. The parameter |delay| is used to correct for codec
// delays.
void CompareInputOutput(const std::vector<int16_t>& input,
const std::vector<int16_t>& output,
size_t num_samples,
size_t channels,
int tolerance,
int delay) {
ASSERT_LE(num_samples, input.size());
ASSERT_LE(num_samples * channels, output.size());
for (unsigned int n = 0; n < num_samples - delay; ++n) {
ASSERT_NEAR(input[n], output[channels * n + delay], tolerance)
<< "Exit test on first diff; n = " << n;
DataLog::InsertCell("CodecTest", "input", input[n]);
DataLog::InsertCell("CodecTest", "output", output[channels * n]);
DataLog::NextRow("CodecTest");
}
}
// The absolute difference between the first two channels in |output| is
// compared vs |tolerance|.
void CompareTwoChannels(const std::vector<int16_t>& output,
size_t samples_per_channel,
size_t channels,
int tolerance) {
ASSERT_GE(channels, 2u);
ASSERT_LE(samples_per_channel * channels, output.size());
for (unsigned int n = 0; n < samples_per_channel; ++n)
ASSERT_NEAR(output[channels * n], output[channels * n + 1], tolerance)
<< "Stereo samples differ.";
}
// Calculates mean-squared error between input and output (the first channel).
// The parameter |delay| is used to correct for codec delays.
double MseInputOutput(const std::vector<int16_t>& input,
const std::vector<int16_t>& output,
size_t num_samples,
size_t channels,
int delay) {
assert(delay < static_cast<int>(num_samples));
assert(num_samples <= input.size());
assert(num_samples * channels <= output.size());
if (num_samples == 0)
return 0.0;
double squared_sum = 0.0;
for (unsigned int n = 0; n < num_samples - delay; ++n) {
squared_sum += (input[n] - output[channels * n + delay]) *
(input[n] - output[channels * n + delay]);
}
return squared_sum / (num_samples - delay);
}
} // namespace
class AudioDecoderTest : public ::testing::Test {
protected:
AudioDecoderTest()
: input_audio_(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
32000),
codec_input_rate_hz_(32000), // Legacy default value.
frame_size_(0),
data_length_(0),
channels_(1),
payload_type_(17),
decoder_(NULL) {}
virtual ~AudioDecoderTest() {}
virtual void SetUp() {
if (audio_encoder_)
codec_input_rate_hz_ = audio_encoder_->SampleRateHz();
// Create arrays.
ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
// Logging to view input and output in Matlab.
// Use 'gyp -Denable_data_logging=1' to enable logging.
DataLog::CreateLog();
DataLog::AddTable("CodecTest");
DataLog::AddColumn("CodecTest", "input", 1);
DataLog::AddColumn("CodecTest", "output", 1);
}
virtual void TearDown() {
delete decoder_;
decoder_ = NULL;
// Close log.
DataLog::ReturnLog();
}
virtual void InitEncoder() { }
// TODO(henrik.lundin) Change return type to size_t once most/all overriding
// implementations are gone.
virtual int EncodeFrame(const int16_t* input,
size_t input_len_samples,
rtc::Buffer* output) {
AudioEncoder::EncodedInfo encoded_info;
const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100;
RTC_CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
input_len_samples);
std::unique_ptr<int16_t[]> interleaved_input(
new int16_t[channels_ * samples_per_10ms]);
for (size_t i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
EXPECT_EQ(0u, encoded_info.encoded_bytes);
// Duplicate the mono input signal to however many channels the test
// wants.
test::InputAudioFile::DuplicateInterleaved(input + i * samples_per_10ms,
samples_per_10ms, channels_,
interleaved_input.get());
encoded_info = audio_encoder_->Encode(
0, rtc::ArrayView<const int16_t>(interleaved_input.get(),
audio_encoder_->NumChannels() *
audio_encoder_->SampleRateHz() /
100),
output);
}
EXPECT_EQ(payload_type_, encoded_info.payload_type);
return static_cast<int>(encoded_info.encoded_bytes);
}
// Encodes and decodes audio. The absolute difference between the input and
// output is compared vs |tolerance|, and the mean-squared error is compared
// with |mse|. The encoded stream should contain |expected_bytes|. For stereo
// audio, the absolute difference between the two channels is compared vs
// |channel_diff_tolerance|.
void EncodeDecodeTest(size_t expected_bytes, int tolerance, double mse,
int delay = 0, int channel_diff_tolerance = 0) {
ASSERT_GE(tolerance, 0) << "Test must define a tolerance >= 0";
ASSERT_GE(channel_diff_tolerance, 0) <<
"Test must define a channel_diff_tolerance >= 0";
size_t processed_samples = 0u;
rtc::Buffer encoded;
size_t encoded_bytes = 0u;
InitEncoder();
std::vector<int16_t> input;
std::vector<int16_t> decoded;
while (processed_samples + frame_size_ <= data_length_) {
// Extend input vector with |frame_size_|.
input.resize(input.size() + frame_size_, 0);
// Read from input file.
ASSERT_GE(input.size() - processed_samples, frame_size_);
ASSERT_TRUE(input_audio_.Read(
frame_size_, codec_input_rate_hz_, &input[processed_samples]));
size_t enc_len = EncodeFrame(
&input[processed_samples], frame_size_, &encoded);
// Make sure that frame_size_ * channels_ samples are allocated and free.
decoded.resize((processed_samples + frame_size_) * channels_, 0);
AudioDecoder::SpeechType speech_type;
size_t dec_len = decoder_->Decode(
&encoded.data()[encoded_bytes], enc_len, codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
&decoded[processed_samples * channels_], &speech_type);
EXPECT_EQ(frame_size_ * channels_, dec_len);
encoded_bytes += enc_len;
processed_samples += frame_size_;
}
// For some codecs it doesn't make sense to check expected number of bytes,
// since the number can vary for different platforms. Opus and iSAC are
// such codecs. In this case expected_bytes is set to 0.
if (expected_bytes) {
EXPECT_EQ(expected_bytes, encoded_bytes);
}
CompareInputOutput(
input, decoded, processed_samples, channels_, tolerance, delay);
if (channels_ == 2)
CompareTwoChannels(
decoded, processed_samples, channels_, channel_diff_tolerance);
EXPECT_LE(
MseInputOutput(input, decoded, processed_samples, channels_, delay),
mse);
}
// Encodes a payload and decodes it twice with decoder re-init before each
// decode. Verifies that the decoded result is the same.
void ReInitTest() {
InitEncoder();
std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
ASSERT_TRUE(
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
rtc::Buffer encoded;
size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);
size_t dec_len;
AudioDecoder::SpeechType speech_type1, speech_type2;
decoder_->Reset();
std::unique_ptr<int16_t[]> output1(new int16_t[frame_size_ * channels_]);
dec_len = decoder_->Decode(encoded.data(), enc_len, codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
output1.get(), &speech_type1);
ASSERT_LE(dec_len, frame_size_ * channels_);
EXPECT_EQ(frame_size_ * channels_, dec_len);
// Re-init decoder and decode again.
decoder_->Reset();
std::unique_ptr<int16_t[]> output2(new int16_t[frame_size_ * channels_]);
dec_len = decoder_->Decode(encoded.data(), enc_len, codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
output2.get(), &speech_type2);
ASSERT_LE(dec_len, frame_size_ * channels_);
EXPECT_EQ(frame_size_ * channels_, dec_len);
for (unsigned int n = 0; n < frame_size_; ++n) {
ASSERT_EQ(output1[n], output2[n]) << "Exit test on first diff; n = " << n;
}
EXPECT_EQ(speech_type1, speech_type2);
}
// Call DecodePlc and verify that the correct number of samples is produced.
void DecodePlcTest() {
InitEncoder();
std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
ASSERT_TRUE(
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
rtc::Buffer encoded;
size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);
AudioDecoder::SpeechType speech_type;
decoder_->Reset();
std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
size_t dec_len = decoder_->Decode(encoded.data(), enc_len,
codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
output.get(), &speech_type);
EXPECT_EQ(frame_size_ * channels_, dec_len);
// Call DecodePlc and verify that we get one frame of data.
// (Overwrite the output from the above Decode call, but that does not
// matter.)
dec_len = decoder_->DecodePlc(1, output.get());
EXPECT_EQ(frame_size_ * channels_, dec_len);
}
test::ResampleInputAudioFile input_audio_;
int codec_input_rate_hz_;
size_t frame_size_;
size_t data_length_;
size_t channels_;
const int payload_type_;
AudioDecoder* decoder_;
std::unique_ptr<AudioEncoder> audio_encoder_;
};
class AudioDecoderPcmUTest : public AudioDecoderTest {
protected:
AudioDecoderPcmUTest() : AudioDecoderTest() {
frame_size_ = 160;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderPcmU(1);
AudioEncoderPcmU::Config config;
config.frame_size_ms = static_cast<int>(frame_size_ / 8);
config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderPcmU(config));
}
};
class AudioDecoderPcmATest : public AudioDecoderTest {
protected:
AudioDecoderPcmATest() : AudioDecoderTest() {
frame_size_ = 160;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderPcmA(1);
AudioEncoderPcmA::Config config;
config.frame_size_ms = static_cast<int>(frame_size_ / 8);
config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderPcmA(config));
}
};
class AudioDecoderPcm16BTest : public AudioDecoderTest {
protected:
AudioDecoderPcm16BTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 16000;
frame_size_ = 20 * codec_input_rate_hz_ / 1000;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderPcm16B(1);
assert(decoder_);
AudioEncoderPcm16B::Config config;
config.sample_rate_hz = codec_input_rate_hz_;
config.frame_size_ms =
static_cast<int>(frame_size_ / (config.sample_rate_hz / 1000));
config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderPcm16B(config));
}
};
class AudioDecoderIlbcTest : public AudioDecoderTest {
protected:
AudioDecoderIlbcTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 8000;
frame_size_ = 240;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderIlbc;
assert(decoder_);
AudioEncoderIlbc::Config config;
config.frame_size_ms = 30;
config.payload_type = payload_type_;
audio_encoder_.reset(new AudioEncoderIlbc(config));
}
// Overload the default test since iLBC's function WebRtcIlbcfix_NetEqPlc does
// not return any data. It simply resets a few states and returns 0.
void DecodePlcTest() {
InitEncoder();
std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
ASSERT_TRUE(
input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
rtc::Buffer encoded;
size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);
AudioDecoder::SpeechType speech_type;
decoder_->Reset();
std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
size_t dec_len = decoder_->Decode(encoded.data(), enc_len,
codec_input_rate_hz_,
frame_size_ * channels_ * sizeof(int16_t),
output.get(), &speech_type);
EXPECT_EQ(frame_size_, dec_len);
// Simply call DecodePlc and verify that we get 0 as return value.
EXPECT_EQ(0U, decoder_->DecodePlc(1, output.get()));
}
};
class AudioDecoderIsacFloatTest : public AudioDecoderTest {
protected:
AudioDecoderIsacFloatTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 16000;
frame_size_ = 480;
data_length_ = 10 * frame_size_;
AudioEncoderIsac::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsac(config));
decoder_ = new AudioDecoderIsac();
}
};
class AudioDecoderIsacSwbTest : public AudioDecoderTest {
protected:
AudioDecoderIsacSwbTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 32000;
frame_size_ = 960;
data_length_ = 10 * frame_size_;
AudioEncoderIsac::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsac(config));
decoder_ = new AudioDecoderIsac();
}
};
class AudioDecoderIsacFixTest : public AudioDecoderTest {
protected:
AudioDecoderIsacFixTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 16000;
frame_size_ = 480;
data_length_ = 10 * frame_size_;
AudioEncoderIsacFix::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
audio_encoder_.reset(new AudioEncoderIsacFix(config));
decoder_ = new AudioDecoderIsacFix();
}
};
class AudioDecoderG722Test : public AudioDecoderTest {
protected:
AudioDecoderG722Test() : AudioDecoderTest() {
codec_input_rate_hz_ = 16000;
frame_size_ = 160;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderG722;
assert(decoder_);
AudioEncoderG722::Config config;
config.frame_size_ms = 10;
config.payload_type = payload_type_;
config.num_channels = 1;
audio_encoder_.reset(new AudioEncoderG722(config));
}
};
class AudioDecoderG722StereoTest : public AudioDecoderTest {
protected:
AudioDecoderG722StereoTest() : AudioDecoderTest() {
channels_ = 2;
codec_input_rate_hz_ = 16000;
frame_size_ = 160;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderG722Stereo;
assert(decoder_);
AudioEncoderG722::Config config;
config.frame_size_ms = 10;
config.payload_type = payload_type_;
config.num_channels = 2;
audio_encoder_.reset(new AudioEncoderG722(config));
}
};
class AudioDecoderOpusTest : public AudioDecoderTest {
protected:
AudioDecoderOpusTest() : AudioDecoderTest() {
codec_input_rate_hz_ = 48000;
frame_size_ = 480;
data_length_ = 10 * frame_size_;
decoder_ = new AudioDecoderOpus(1);
AudioEncoderOpus::Config config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.payload_type = payload_type_;
config.application = AudioEncoderOpus::kVoip;
audio_encoder_.reset(new AudioEncoderOpus(config));
}
};
class AudioDecoderOpusStereoTest : public AudioDecoderOpusTest {
protected:
AudioDecoderOpusStereoTest() : AudioDecoderOpusTest() {
channels_ = 2;
delete decoder_;
decoder_ = new AudioDecoderOpus(2);
AudioEncoderOpus::Config config;
config.frame_size_ms = static_cast<int>(frame_size_) / 48;
config.num_channels = 2;
config.payload_type = payload_type_;
config.application = AudioEncoderOpus::kAudio;
audio_encoder_.reset(new AudioEncoderOpus(config));
}
};
TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
int tolerance = 251;
double mse = 1734.0;
EncodeDecodeTest(data_length_, tolerance, mse);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
namespace {
int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
audio_encoder->SetTargetBitrate(rate);
return audio_encoder->GetTargetBitrate();
}
void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
int fixed_rate) {
EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, 32000));
EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate - 1));
EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate));
EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate + 1));
}
} // namespace
TEST_F(AudioDecoderPcmUTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
}
TEST_F(AudioDecoderPcmATest, EncodeDecode) {
int tolerance = 308;
double mse = 1931.0;
EncodeDecodeTest(data_length_, tolerance, mse);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderPcmATest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
}
TEST_F(AudioDecoderPcm16BTest, EncodeDecode) {
int tolerance = 0;
double mse = 0.0;
EncodeDecodeTest(2 * data_length_, tolerance, mse);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderPcm16BTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(),
codec_input_rate_hz_ * 16);
}
TEST_F(AudioDecoderIlbcTest, EncodeDecode) {
int tolerance = 6808;
double mse = 2.13e6;
int delay = 80; // Delay from input to output.
EncodeDecodeTest(500, tolerance, mse, delay);
ReInitTest();
EXPECT_TRUE(decoder_->HasDecodePlc());
DecodePlcTest();
}
TEST_F(AudioDecoderIlbcTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 13333);
}
TEST_F(AudioDecoderIsacFloatTest, EncodeDecode) {
int tolerance = 3399;
double mse = 434951.0;
int delay = 48; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderIsacFloatTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 32000);
}
TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) {
int tolerance = 19757;
double mse = 8.18e6;
int delay = 160; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderIsacSwbTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 32000);
}
TEST_F(AudioDecoderIsacFixTest, EncodeDecode) {
int tolerance = 11034;
double mse = 3.46e6;
int delay = 54; // Delay from input to output.
#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM)
static const int kEncodedBytes = 685;
#elif defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM64)
static const int kEncodedBytes = 673;
#else
static const int kEncodedBytes = 671;
#endif
EncodeDecodeTest(kEncodedBytes, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderIsacFixTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 32000);
}
TEST_F(AudioDecoderG722Test, EncodeDecode) {
int tolerance = 6176;
double mse = 238630.0;
int delay = 22; // Delay from input to output.
EncodeDecodeTest(data_length_ / 2, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderG722Test, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
}
TEST_F(AudioDecoderG722StereoTest, EncodeDecode) {
int tolerance = 6176;
int channel_diff_tolerance = 0;
double mse = 238630.0;
int delay = 22; // Delay from input to output.
EncodeDecodeTest(data_length_, tolerance, mse, delay, channel_diff_tolerance);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderG722StereoTest, SetTargetBitrate) {
TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 128000);
}
TEST_F(AudioDecoderOpusTest, EncodeDecode) {
int tolerance = 6176;
double mse = 238630.0;
int delay = 22; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
namespace {
void TestOpusSetTargetBitrates(AudioEncoder* audio_encoder) {
EXPECT_EQ(500, SetAndGetTargetBitrate(audio_encoder, 499));
EXPECT_EQ(500, SetAndGetTargetBitrate(audio_encoder, 500));
EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder, 32000));
EXPECT_EQ(512000, SetAndGetTargetBitrate(audio_encoder, 512000));
EXPECT_EQ(512000, SetAndGetTargetBitrate(audio_encoder, 513000));
}
} // namespace
TEST_F(AudioDecoderOpusTest, SetTargetBitrate) {
TestOpusSetTargetBitrates(audio_encoder_.get());
}
TEST_F(AudioDecoderOpusStereoTest, EncodeDecode) {
int tolerance = 6176;
int channel_diff_tolerance = 0;
double mse = 238630.0;
int delay = 22; // Delay from input to output.
EncodeDecodeTest(0, tolerance, mse, delay, channel_diff_tolerance);
ReInitTest();
EXPECT_FALSE(decoder_->HasDecodePlc());
}
TEST_F(AudioDecoderOpusStereoTest, SetTargetBitrate) {
TestOpusSetTargetBitrates(audio_encoder_.get());
}
namespace {
#ifdef WEBRTC_CODEC_ILBC
const bool has_ilbc = true;
#else
const bool has_ilbc = false;
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
const bool has_isac = true;
#else
const bool has_isac = false;
#endif
#ifdef WEBRTC_CODEC_ISAC
const bool has_isac_swb = true;
#else
const bool has_isac_swb = false;
#endif
#ifdef WEBRTC_CODEC_G722
const bool has_g722 = true;
#else
const bool has_g722 = false;
#endif
#ifdef WEBRTC_CODEC_OPUS
const bool has_opus = true;
#else
const bool has_opus = false;
#endif
} // namespace
TEST(AudioDecoder, CodecSampleRateHz) {
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMu));
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMa));
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMu_2ch));
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCMa_2ch));
EXPECT_EQ(has_ilbc ? 8000 : -1,
CodecSampleRateHz(NetEqDecoder::kDecoderILBC));
EXPECT_EQ(has_isac ? 16000 : -1,
CodecSampleRateHz(NetEqDecoder::kDecoderISAC));
EXPECT_EQ(has_isac_swb ? 32000 : -1,
CodecSampleRateHz(NetEqDecoder::kDecoderISACswb));
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16B));
EXPECT_EQ(16000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bwb));
EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb32kHz));
EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb48kHz));
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16B_2ch));
EXPECT_EQ(16000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bwb_2ch));
EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch));
EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch));
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderPCM16B_5ch));
EXPECT_EQ(has_g722 ? 16000 : -1,
CodecSampleRateHz(NetEqDecoder::kDecoderG722));
EXPECT_EQ(has_g722 ? 16000 : -1,
CodecSampleRateHz(NetEqDecoder::kDecoderG722_2ch));
EXPECT_EQ(-1, CodecSampleRateHz(NetEqDecoder::kDecoderRED));
EXPECT_EQ(-1, CodecSampleRateHz(NetEqDecoder::kDecoderAVT));
EXPECT_EQ(8000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGnb));
EXPECT_EQ(16000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGwb));
EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGswb32kHz));
EXPECT_EQ(has_opus ? 48000 : -1,
CodecSampleRateHz(NetEqDecoder::kDecoderOpus));
EXPECT_EQ(has_opus ? 48000 : -1,
CodecSampleRateHz(NetEqDecoder::kDecoderOpus_2ch));
EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderOpus));
EXPECT_EQ(48000, CodecSampleRateHz(NetEqDecoder::kDecoderOpus_2ch));
// TODO(tlegrand): Change 32000 to 48000 below once ACM has 48 kHz support.
EXPECT_EQ(32000, CodecSampleRateHz(NetEqDecoder::kDecoderCNGswb48kHz));
EXPECT_EQ(-1, CodecSampleRateHz(NetEqDecoder::kDecoderArbitrary));
}
TEST(AudioDecoder, CodecSupported) {
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMu));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMa));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMu_2ch));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCMa_2ch));
EXPECT_EQ(has_ilbc, CodecSupported(NetEqDecoder::kDecoderILBC));
EXPECT_EQ(has_isac, CodecSupported(NetEqDecoder::kDecoderISAC));
EXPECT_EQ(has_isac_swb, CodecSupported(NetEqDecoder::kDecoderISACswb));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16B));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bwb));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb32kHz));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb48kHz));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16B_2ch));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bwb_2ch));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderPCM16B_5ch));
EXPECT_EQ(has_g722, CodecSupported(NetEqDecoder::kDecoderG722));
EXPECT_EQ(has_g722, CodecSupported(NetEqDecoder::kDecoderG722_2ch));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderRED));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderAVT));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGnb));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGwb));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb32kHz));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderCNGswb48kHz));
EXPECT_TRUE(CodecSupported(NetEqDecoder::kDecoderArbitrary));
EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus));
EXPECT_EQ(has_opus, CodecSupported(NetEqDecoder::kDecoderOpus_2ch));
}
} // namespace webrtc
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