Source code
Revision control
Copy as Markdown
Other Tools
/*
* Copyright (c) 2013 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 <memory>
#include <string>
#include "modules/audio_coding/codecs/opus/opus_inst.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace {
// Equivalent to SDP params
// {{"channel_mapping", "0,1,2,3"}, {"coupled_streams", "2"}}.
constexpr unsigned char kQuadChannelMapping[] = {0, 1, 2, 3};
constexpr int kQuadTotalStreams = 2;
constexpr int kQuadCoupledStreams = 2;
constexpr unsigned char kStereoChannelMapping[] = {0, 1};
constexpr int kStereoTotalStreams = 1;
constexpr int kStereoCoupledStreams = 1;
constexpr unsigned char kMonoChannelMapping[] = {0};
constexpr int kMonoTotalStreams = 1;
constexpr int kMonoCoupledStreams = 0;
void CreateSingleOrMultiStreamEncoder(WebRtcOpusEncInst** opus_encoder,
int channels,
int application,
bool use_multistream,
int encoder_sample_rate_hz) {
EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
if (use_multistream) {
EXPECT_EQ(encoder_sample_rate_hz, 48000);
if (channels == 1) {
EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
opus_encoder, channels, application, kMonoTotalStreams,
kMonoCoupledStreams, kMonoChannelMapping));
} else if (channels == 2) {
EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
opus_encoder, channels, application, kStereoTotalStreams,
kStereoCoupledStreams, kStereoChannelMapping));
} else if (channels == 4) {
EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
opus_encoder, channels, application, kQuadTotalStreams,
kQuadCoupledStreams, kQuadChannelMapping));
} else {
EXPECT_TRUE(false) << channels;
}
} else {
EXPECT_EQ(0, WebRtcOpus_EncoderCreate(opus_encoder, channels, application,
encoder_sample_rate_hz));
}
}
void CreateSingleOrMultiStreamDecoder(WebRtcOpusDecInst** opus_decoder,
int channels,
bool use_multistream,
int decoder_sample_rate_hz) {
EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
if (use_multistream) {
EXPECT_EQ(decoder_sample_rate_hz, 48000);
if (channels == 1) {
EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
opus_decoder, channels, kMonoTotalStreams,
kMonoCoupledStreams, kMonoChannelMapping));
} else if (channels == 2) {
EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
opus_decoder, channels, kStereoTotalStreams,
kStereoCoupledStreams, kStereoChannelMapping));
} else if (channels == 4) {
EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
opus_decoder, channels, kQuadTotalStreams,
kQuadCoupledStreams, kQuadChannelMapping));
} else {
EXPECT_TRUE(false) << channels;
}
} else {
EXPECT_EQ(0, WebRtcOpus_DecoderCreate(opus_decoder, channels,
decoder_sample_rate_hz));
}
}
int SamplesPerChannel(int sample_rate_hz, int duration_ms) {
const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz, 1000);
return samples_per_ms * duration_ms;
}
using test::AudioLoop;
using ::testing::Combine;
using ::testing::TestWithParam;
using ::testing::Values;
// Maximum number of bytes in output bitstream.
const size_t kMaxBytes = 2000;
class OpusTest
: public TestWithParam<::testing::tuple<size_t, int, bool, int, int>> {
protected:
OpusTest() = default;
void TestDtxEffect(bool dtx, int block_length_ms);
void TestCbrEffect(bool dtx, int block_length_ms);
// Prepare `speech_data_` for encoding, read from a hard-coded file.
// After preparation, `speech_data_.GetNextBlock()` returns a pointer to a
// block of `block_length_ms` milliseconds. The data is looped every
// `loop_length_ms` milliseconds.
void PrepareSpeechData(int block_length_ms, int loop_length_ms);
int EncodeDecode(WebRtcOpusEncInst* encoder,
rtc::ArrayView<const int16_t> input_audio,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type);
void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
opus_int32 expect,
int32_t set);
void CheckAudioBounded(const int16_t* audio,
size_t samples,
size_t channels,
uint16_t bound) const;
WebRtcOpusEncInst* opus_encoder_ = nullptr;
WebRtcOpusDecInst* opus_decoder_ = nullptr;
AudioLoop speech_data_;
uint8_t bitstream_[kMaxBytes];
size_t encoded_bytes_ = 0;
const size_t channels_{std::get<0>(GetParam())};
const int application_{std::get<1>(GetParam())};
const bool use_multistream_{std::get<2>(GetParam())};
const int encoder_sample_rate_hz_{std::get<3>(GetParam())};
const int decoder_sample_rate_hz_{std::get<4>(GetParam())};
};
} // namespace
// Singlestream: Try all combinations.
INSTANTIATE_TEST_SUITE_P(Singlestream,
OpusTest,
testing::Combine(testing::Values(1, 2),
testing::Values(0, 1),
testing::Values(false),
testing::Values(16000, 48000),
testing::Values(16000, 48000)));
// Multistream: Some representative cases (only 48 kHz for now).
INSTANTIATE_TEST_SUITE_P(
Multistream,
OpusTest,
testing::Values(std::make_tuple(1, 0, true, 48000, 48000),
std::make_tuple(2, 1, true, 48000, 48000),
std::make_tuple(4, 0, true, 48000, 48000),
std::make_tuple(4, 1, true, 48000, 48000)));
void OpusTest::PrepareSpeechData(int block_length_ms, int loop_length_ms) {
std::map<int, std::string> channel_to_basename = {
{1, "audio_coding/testfile32kHz"},
{2, "audio_coding/teststereo32kHz"},
{4, "audio_coding/speech_4_channels_48k_one_second"}};
std::map<int, std::string> channel_to_suffix = {
{1, "pcm"}, {2, "pcm"}, {4, "wav"}};
const std::string file_name = webrtc::test::ResourcePath(
channel_to_basename[channels_], channel_to_suffix[channels_]);
if (loop_length_ms < block_length_ms) {
loop_length_ms = block_length_ms;
}
const int sample_rate_khz =
rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000);
EXPECT_TRUE(speech_data_.Init(file_name,
loop_length_ms * sample_rate_khz * channels_,
block_length_ms * sample_rate_khz * channels_));
}
void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
opus_int32 expect,
int32_t set) {
opus_int32 bandwidth;
EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
EXPECT_EQ(0, WebRtcOpus_GetMaxPlaybackRate(opus_encoder_, &bandwidth));
EXPECT_EQ(expect, bandwidth);
}
void OpusTest::CheckAudioBounded(const int16_t* audio,
size_t samples,
size_t channels,
uint16_t bound) const {
for (size_t i = 0; i < samples; ++i) {
for (size_t c = 0; c < channels; ++c) {
ASSERT_GE(audio[i * channels + c], -bound);
ASSERT_LE(audio[i * channels + c], bound);
}
}
}
int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
rtc::ArrayView<const int16_t> input_audio,
WebRtcOpusDecInst* decoder,
int16_t* output_audio,
int16_t* audio_type) {
const int input_samples_per_channel =
rtc::CheckedDivExact(input_audio.size(), channels_);
int encoded_bytes_int =
WebRtcOpus_Encode(encoder, input_audio.data(), input_samples_per_channel,
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
if (encoded_bytes_ != 0) {
int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
int act_len = WebRtcOpus_Decode(decoder, bitstream_, encoded_bytes_,
output_audio, audio_type);
EXPECT_EQ(est_len, act_len);
return act_len;
} else {
int total_dtx_len = 0;
const int output_samples_per_channel = input_samples_per_channel *
decoder_sample_rate_hz_ /
encoder_sample_rate_hz_;
while (total_dtx_len < output_samples_per_channel) {
int est_len = WebRtcOpus_DurationEst(decoder, NULL, 0);
int act_len = WebRtcOpus_Decode(decoder, NULL, 0,
&output_audio[total_dtx_len * channels_],
audio_type);
EXPECT_EQ(est_len, act_len);
total_dtx_len += act_len;
}
return total_dtx_len;
}
}
// Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
// they should not. This test is signal dependent.
void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
PrepareSpeechData(block_length_ms, 2000);
const size_t input_samples =
rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000) * block_length_ms;
const size_t output_samples =
rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Set input audio as silence.
std::vector<int16_t> silence(input_samples * channels_, 0);
// Setting DTX.
EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_)
: WebRtcOpus_DisableDtx(opus_encoder_));
int16_t audio_type;
int16_t* output_data_decode = new int16_t[output_samples * channels_];
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(output_samples,
static_cast<size_t>(EncodeDecode(
opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
output_data_decode, &audio_type)));
// If not DTX, it should never enter DTX mode. If DTX, we do not care since
// whether it enters DTX depends on the signal type.
if (!dtx) {
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
}
// We input some silent segments. In DTX mode, the encoder will stop sending.
// However, DTX may happen after a while.
for (int i = 0; i < 30; ++i) {
EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, opus_decoder_,
output_data_decode, &audio_type)));
if (!dtx) {
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
} else if (encoded_bytes_ == 1) {
EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
EXPECT_EQ(2, audio_type); // Comfort noise.
break;
}
}
// When Opus is in DTX, it wakes up in a regular basis. It sends two packets,
// one with an arbitrary size and the other of 1-byte, then stops sending for
// a certain number of frames.
// `max_dtx_frames` is the maximum number of frames Opus can stay in DTX.
// TODO(kwiberg): Why does this number depend on the encoding sample rate?
const int max_dtx_frames =
(encoder_sample_rate_hz_ == 16000 ? 800 : 400) / block_length_ms + 1;
// We run `kRunTimeMs` milliseconds of pure silence.
const int kRunTimeMs = 4500;
// We check that, after a `kCheckTimeMs` milliseconds (given that the CNG in
// Opus needs time to adapt), the absolute values of DTX decoded signal are
// bounded by `kOutputValueBound`.
const int kCheckTimeMs = 4000;
#if defined(OPUS_FIXED_POINT)
// Fixed-point Opus generates a random (comfort) noise, which has a less
// predictable value bound than its floating-point Opus. This value depends on
// input signal, and the time window for checking the output values (between
// `kCheckTimeMs` and `kRunTimeMs`).
const uint16_t kOutputValueBound = 30;
#else
const uint16_t kOutputValueBound = 2;
#endif
int time = 0;
while (time < kRunTimeMs) {
// DTX mode is maintained for maximum `max_dtx_frames` frames.
int i = 0;
for (; i < max_dtx_frames; ++i) {
time += block_length_ms;
EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, opus_decoder_,
output_data_decode, &audio_type)));
if (dtx) {
if (encoded_bytes_ > 1)
break;
EXPECT_EQ(0U, encoded_bytes_) // Send 0 byte.
<< "Opus should have entered DTX mode.";
EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
EXPECT_EQ(2, audio_type); // Comfort noise.
if (time >= kCheckTimeMs) {
CheckAudioBounded(output_data_decode, output_samples, channels_,
kOutputValueBound);
}
} else {
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
}
if (dtx) {
// With DTX, Opus must stop transmission for some time.
EXPECT_GT(i, 1);
}
// We expect a normal payload.
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
// Enters DTX again immediately.
time += block_length_ms;
EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, opus_decoder_,
output_data_decode, &audio_type)));
if (dtx) {
EXPECT_EQ(1U, encoded_bytes_); // Send 1 byte.
EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
EXPECT_EQ(2, audio_type); // Comfort noise.
if (time >= kCheckTimeMs) {
CheckAudioBounded(output_data_decode, output_samples, channels_,
kOutputValueBound);
}
} else {
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
}
silence[0] = 10000;
if (dtx) {
// Verify that encoder/decoder can jump out from DTX mode.
EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
opus_encoder_, silence, opus_decoder_,
output_data_decode, &audio_type)));
EXPECT_GT(encoded_bytes_, 1U);
EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
EXPECT_EQ(0, audio_type); // Speech.
}
// Free memory.
delete[] output_data_decode;
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
// Test if CBR does what we expect.
void OpusTest::TestCbrEffect(bool cbr, int block_length_ms) {
PrepareSpeechData(block_length_ms, 2000);
const size_t output_samples =
rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
int32_t max_pkt_size_diff = 0;
int32_t prev_pkt_size = 0;
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Setting CBR.
EXPECT_EQ(0, cbr ? WebRtcOpus_EnableCbr(opus_encoder_)
: WebRtcOpus_DisableCbr(opus_encoder_));
int16_t audio_type;
std::vector<int16_t> audio_out(output_samples * channels_);
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(output_samples,
static_cast<size_t>(
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
opus_decoder_, audio_out.data(), &audio_type)));
if (prev_pkt_size > 0) {
int32_t diff = std::abs((int32_t)encoded_bytes_ - prev_pkt_size);
max_pkt_size_diff = std::max(max_pkt_size_diff, diff);
}
prev_pkt_size = rtc::checked_cast<int32_t>(encoded_bytes_);
}
if (cbr) {
EXPECT_EQ(max_pkt_size_diff, 0);
} else {
EXPECT_GT(max_pkt_size_diff, 0);
}
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
// Test failing Create.
TEST(OpusTest, OpusCreateFail) {
WebRtcOpusEncInst* opus_encoder;
WebRtcOpusDecInst* opus_decoder;
// Test to see that an invalid pointer is caught.
EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(NULL, 1, 0, 48000));
// Invalid channel number.
EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 257, 0, 48000));
// Invalid applciation mode.
EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 2, 48000));
// Invalid sample rate.
EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 0, 12345));
EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1, 48000));
// Invalid channel number.
EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257, 48000));
// Invalid sample rate.
EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 1, 12345));
}
// Test failing Free.
TEST(OpusTest, OpusFreeFail) {
// Test to see that an invalid pointer is caught.
EXPECT_EQ(-1, WebRtcOpus_EncoderFree(NULL));
EXPECT_EQ(-1, WebRtcOpus_DecoderFree(NULL));
}
// Test normal Create and Free.
TEST_P(OpusTest, OpusCreateFree) {
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
EXPECT_TRUE(opus_encoder_ != NULL);
EXPECT_TRUE(opus_decoder_ != NULL);
// Free encoder and decoder memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
#define ENCODER_CTL(inst, vargs) \
inst->encoder \
? opus_encoder_ctl(inst->encoder, vargs) \
: opus_multistream_encoder_ctl(inst->multistream_encoder, vargs)
TEST_P(OpusTest, OpusEncodeDecode) {
PrepareSpeechData(20, 20);
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Check number of channels for decoder.
EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
// Check application mode.
opus_int32 app;
ENCODER_CTL(opus_encoder_, OPUS_GET_APPLICATION(&app));
EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
app);
// Encode & decode.
int16_t audio_type;
const int decode_samples_per_channel =
SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
int16_t* output_data_decode =
new int16_t[decode_samples_per_channel * channels_];
EXPECT_EQ(decode_samples_per_channel,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
opus_decoder_, output_data_decode, &audio_type));
// Free memory.
delete[] output_data_decode;
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
TEST_P(OpusTest, OpusSetBitRate) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
// Create encoder memory, try with different bitrates.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 30000));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 300000));
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 600000));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
TEST_P(OpusTest, OpusSetComplexity) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 9));
// Create encoder memory, try with different complexities.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 0));
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 10));
EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 11));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
TEST_P(OpusTest, OpusSetBandwidth) {
if (channels_ > 2) {
// TODO(webrtc:10217): investigate why multi-stream Opus reports
// narrowband when it's configured with FULLBAND.
return;
}
PrepareSpeechData(20, 20);
int16_t audio_type;
const int decode_samples_per_channel =
SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
std::unique_ptr<int16_t[]> output_data_decode(
new int16_t[decode_samples_per_channel * channels_]());
// Test without creating encoder memory.
EXPECT_EQ(-1,
WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
EXPECT_EQ(-1, WebRtcOpus_GetBandwidth(opus_encoder_));
// Create encoder memory, try with different bandwidths.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
EXPECT_EQ(-1, WebRtcOpus_SetBandwidth(opus_encoder_,
OPUS_BANDWIDTH_NARROWBAND - 1));
EXPECT_EQ(0,
WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
output_data_decode.get(), &audio_type);
EXPECT_EQ(OPUS_BANDWIDTH_NARROWBAND, WebRtcOpus_GetBandwidth(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND));
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
output_data_decode.get(), &audio_type);
EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
: OPUS_BANDWIDTH_FULLBAND,
WebRtcOpus_GetBandwidth(opus_encoder_));
EXPECT_EQ(
-1, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND + 1));
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
output_data_decode.get(), &audio_type);
EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
: OPUS_BANDWIDTH_FULLBAND,
WebRtcOpus_GetBandwidth(opus_encoder_));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
TEST_P(OpusTest, OpusForceChannels) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
ASSERT_NE(nullptr, opus_encoder_);
if (channels_ >= 2) {
EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 3));
EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
} else {
EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
}
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
// Encode and decode one frame, initialize the decoder and
// decode once more.
TEST_P(OpusTest, OpusDecodeInit) {
PrepareSpeechData(20, 20);
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
// Encode & decode.
int16_t audio_type;
const int decode_samples_per_channel =
SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
int16_t* output_data_decode =
new int16_t[decode_samples_per_channel * channels_];
EXPECT_EQ(decode_samples_per_channel,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
opus_decoder_, output_data_decode, &audio_type));
WebRtcOpus_DecoderInit(opus_decoder_);
EXPECT_EQ(decode_samples_per_channel,
WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
output_data_decode, &audio_type));
// Free memory.
delete[] output_data_decode;
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
TEST_P(OpusTest, OpusEnableDisableFec) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_EnableFec(opus_encoder_));
EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_encoder_));
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
TEST_P(OpusTest, OpusEnableDisableDtx) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_EnableDtx(opus_encoder_));
EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_encoder_));
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
opus_int32 dtx;
// DTX is off by default.
ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Test to enable DTX.
EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
EXPECT_EQ(1, dtx);
// Test to disable DTX.
EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_));
ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
EXPECT_EQ(0, dtx);
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
TEST_P(OpusTest, OpusDtxOff) {
TestDtxEffect(false, 10);
TestDtxEffect(false, 20);
TestDtxEffect(false, 40);
}
TEST_P(OpusTest, OpusDtxOn) {
if (channels_ > 2 || application_ != 0) {
// DTX does not work with OPUS_APPLICATION_AUDIO at low complexity settings.
// TODO(webrtc:10218): adapt the test to the sizes and order of multi-stream
// DTX packets.
return;
}
TestDtxEffect(true, 10);
TestDtxEffect(true, 20);
TestDtxEffect(true, 40);
}
TEST_P(OpusTest, OpusCbrOff) {
TestCbrEffect(false, 10);
TestCbrEffect(false, 20);
TestCbrEffect(false, 40);
}
TEST_P(OpusTest, OpusCbrOn) {
TestCbrEffect(true, 10);
TestCbrEffect(true, 20);
TestCbrEffect(true, 40);
}
TEST_P(OpusTest, OpusSetPacketLossRate) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, -1));
EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 101));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
TEST_P(OpusTest, OpusSetMaxPlaybackRate) {
// Test without creating encoder memory.
EXPECT_EQ(-1, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, 20000));
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 48000);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 24001);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 24000);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 16001);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 16000);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 12001);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 12000);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 8001);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 8000);
SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 4000);
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
}
// Test PLC.
TEST_P(OpusTest, OpusDecodePlc) {
PrepareSpeechData(20, 20);
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
// Set bitrate.
EXPECT_EQ(
0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Check number of channels for decoder.
EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
// Encode & decode.
int16_t audio_type;
const int decode_samples_per_channel =
SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
int16_t* output_data_decode =
new int16_t[decode_samples_per_channel * channels_];
EXPECT_EQ(decode_samples_per_channel,
EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
opus_decoder_, output_data_decode, &audio_type));
// Call decoder PLC.
constexpr int kPlcDurationMs = 10;
const int plc_samples = decoder_sample_rate_hz_ * kPlcDurationMs / 1000;
int16_t* plc_buffer = new int16_t[plc_samples * channels_];
EXPECT_EQ(plc_samples,
WebRtcOpus_Decode(opus_decoder_, NULL, 0, plc_buffer, &audio_type));
// Free memory.
delete[] plc_buffer;
delete[] output_data_decode;
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
// Duration estimation.
TEST_P(OpusTest, OpusDurationEstimation) {
PrepareSpeechData(20, 20);
// Create.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
// 10 ms. We use only first 10 ms of a 20 ms block.
auto speech_block = speech_data_.GetNextBlock();
int encoded_bytes_int = WebRtcOpus_Encode(
opus_encoder_, speech_block.data(),
rtc::CheckedDivExact(speech_block.size(), 2 * channels_), kMaxBytes,
bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/10),
WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
static_cast<size_t>(encoded_bytes_int)));
// 20 ms
speech_block = speech_data_.GetNextBlock();
encoded_bytes_int =
WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
rtc::CheckedDivExact(speech_block.size(), channels_),
kMaxBytes, bitstream_);
EXPECT_GE(encoded_bytes_int, 0);
EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20),
WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
static_cast<size_t>(encoded_bytes_int)));
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
TEST_P(OpusTest, OpusDecodeRepacketized) {
if (channels_ > 2) {
// As per the Opus documentation
// https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__repacketizer.html#details,
// multiple streams are not supported.
return;
}
constexpr size_t kPackets = 6;
PrepareSpeechData(20, 20 * kPackets);
// Create encoder memory.
CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
use_multistream_, encoder_sample_rate_hz_);
ASSERT_NE(nullptr, opus_encoder_);
CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
decoder_sample_rate_hz_);
ASSERT_NE(nullptr, opus_decoder_);
// Set bitrate.
EXPECT_EQ(
0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
// Check number of channels for decoder.
EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
// Encode & decode.
int16_t audio_type;
const int decode_samples_per_channel =
SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
std::unique_ptr<int16_t[]> output_data_decode(
new int16_t[kPackets * decode_samples_per_channel * channels_]);
OpusRepacketizer* rp = opus_repacketizer_create();
size_t num_packets = 0;
constexpr size_t kMaxCycles = 100;
for (size_t idx = 0; idx < kMaxCycles; ++idx) {
auto speech_block = speech_data_.GetNextBlock();
encoded_bytes_ =
WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
rtc::CheckedDivExact(speech_block.size(), channels_),
kMaxBytes, bitstream_);
if (opus_repacketizer_cat(rp, bitstream_,
rtc::checked_cast<opus_int32>(encoded_bytes_)) ==
OPUS_OK) {
++num_packets;
if (num_packets == kPackets) {
break;
}
} else {
// Opus repacketizer cannot guarantee a success. We try again if it fails.
opus_repacketizer_init(rp);
num_packets = 0;
}
}
EXPECT_EQ(kPackets, num_packets);
encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);
EXPECT_EQ(decode_samples_per_channel * kPackets,
static_cast<size_t>(WebRtcOpus_DurationEst(
opus_decoder_, bitstream_, encoded_bytes_)));
EXPECT_EQ(decode_samples_per_channel * kPackets,
static_cast<size_t>(
WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
output_data_decode.get(), &audio_type)));
// Free memory.
opus_repacketizer_destroy(rp);
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
TEST(OpusVadTest, CeltUnknownStatus) {
const uint8_t celt[] = {0x80};
EXPECT_EQ(WebRtcOpus_PacketHasVoiceActivity(celt, 1), -1);
}
TEST(OpusVadTest, Mono20msVadSet) {
uint8_t silk20msMonoVad[] = {0x78, 0x80};
EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoVad, 2));
}
TEST(OpusVadTest, Mono20MsVadUnset) {
uint8_t silk20msMonoSilence[] = {0x78, 0x00};
EXPECT_FALSE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoSilence, 2));
}
TEST(OpusVadTest, Stereo20MsVadOnSideChannel) {
uint8_t silk20msStereoVadSideChannel[] = {0x78 | 0x04, 0x20};
EXPECT_TRUE(
WebRtcOpus_PacketHasVoiceActivity(silk20msStereoVadSideChannel, 2));
}
TEST(OpusVadTest, TwoOpusMonoFramesVadOnSecond) {
uint8_t twoMonoFrames[] = {0x78 | 0x1, 0x00, 0x80};
EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(twoMonoFrames, 3));
}
} // namespace webrtc