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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/audio/remix_resample_unittest.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/remix_resample_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/audio/remix_resample_unittest.cc | 276 |
1 files changed, 276 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/remix_resample_unittest.cc b/third_party/libwebrtc/audio/remix_resample_unittest.cc new file mode 100644 index 0000000000..31dcfac1fe --- /dev/null +++ b/third_party/libwebrtc/audio/remix_resample_unittest.cc @@ -0,0 +1,276 @@ +/* + * 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 "audio/remix_resample.h" + +#include <cmath> + +#include "common_audio/resampler/include/push_resampler.h" +#include "rtc_base/arraysize.h" +#include "rtc_base/checks.h" +#include "test/gtest.h" + +namespace webrtc { +namespace voe { +namespace { + +int GetFrameSize(int sample_rate_hz) { + return sample_rate_hz / 100; +} + +class UtilityTest : public ::testing::Test { + protected: + UtilityTest() { + src_frame_.sample_rate_hz_ = 16000; + src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100; + src_frame_.num_channels_ = 1; + dst_frame_.CopyFrom(src_frame_); + golden_frame_.CopyFrom(src_frame_); + } + + void RunResampleTest(int src_channels, + int src_sample_rate_hz, + int dst_channels, + int dst_sample_rate_hz); + + PushResampler<int16_t> resampler_; + AudioFrame src_frame_; + AudioFrame dst_frame_; + AudioFrame golden_frame_; +}; + +// Sets the signal value to increase by `data` with every sample. Floats are +// used so non-integer values result in rounding error, but not an accumulating +// error. +void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) { + frame->Mute(); + frame->num_channels_ = 1; + frame->sample_rate_hz_ = sample_rate_hz; + frame->samples_per_channel_ = GetFrameSize(sample_rate_hz); + int16_t* frame_data = frame->mutable_data(); + for (size_t i = 0; i < frame->samples_per_channel_; i++) { + frame_data[i] = static_cast<int16_t>(data * i); + } +} + +// Keep the existing sample rate. +void SetMonoFrame(float data, AudioFrame* frame) { + SetMonoFrame(data, frame->sample_rate_hz_, frame); +} + +// Sets the signal value to increase by `left` and `right` with every sample in +// each channel respectively. +void SetStereoFrame(float left, + float right, + int sample_rate_hz, + AudioFrame* frame) { + frame->Mute(); + frame->num_channels_ = 2; + frame->sample_rate_hz_ = sample_rate_hz; + frame->samples_per_channel_ = GetFrameSize(sample_rate_hz); + int16_t* frame_data = frame->mutable_data(); + for (size_t i = 0; i < frame->samples_per_channel_; i++) { + frame_data[i * 2] = static_cast<int16_t>(left * i); + frame_data[i * 2 + 1] = static_cast<int16_t>(right * i); + } +} + +// Keep the existing sample rate. +void SetStereoFrame(float left, float right, AudioFrame* frame) { + SetStereoFrame(left, right, frame->sample_rate_hz_, frame); +} + +// Sets the signal value to increase by `ch1`, `ch2`, `ch3`, `ch4` with every +// sample in each channel respectively. +void SetQuadFrame(float ch1, + float ch2, + float ch3, + float ch4, + int sample_rate_hz, + AudioFrame* frame) { + frame->Mute(); + frame->num_channels_ = 4; + frame->sample_rate_hz_ = sample_rate_hz; + frame->samples_per_channel_ = GetFrameSize(sample_rate_hz); + int16_t* frame_data = frame->mutable_data(); + for (size_t i = 0; i < frame->samples_per_channel_; i++) { + frame_data[i * 4] = static_cast<int16_t>(ch1 * i); + frame_data[i * 4 + 1] = static_cast<int16_t>(ch2 * i); + frame_data[i * 4 + 2] = static_cast<int16_t>(ch3 * i); + frame_data[i * 4 + 3] = static_cast<int16_t>(ch4 * i); + } +} + +void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) { + EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_); + EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_); + EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_); +} + +// Computes the best SNR based on the error between `ref_frame` and +// `test_frame`. It allows for up to a `max_delay` in samples between the +// signals to compensate for the resampling delay. +float ComputeSNR(const AudioFrame& ref_frame, + const AudioFrame& test_frame, + size_t max_delay) { + VerifyParams(ref_frame, test_frame); + float best_snr = 0; + size_t best_delay = 0; + for (size_t delay = 0; delay <= max_delay; delay++) { + float mse = 0; + float variance = 0; + const int16_t* ref_frame_data = ref_frame.data(); + const int16_t* test_frame_data = test_frame.data(); + for (size_t i = 0; + i < ref_frame.samples_per_channel_ * ref_frame.num_channels_ - delay; + i++) { + int error = ref_frame_data[i] - test_frame_data[i + delay]; + mse += error * error; + variance += ref_frame_data[i] * ref_frame_data[i]; + } + float snr = 100; // We assign 100 dB to the zero-error case. + if (mse > 0) + snr = 10 * std::log10(variance / mse); + if (snr > best_snr) { + best_snr = snr; + best_delay = delay; + } + } + printf("SNR=%.1f dB at delay=%zu\n", best_snr, best_delay); + return best_snr; +} + +void VerifyFramesAreEqual(const AudioFrame& ref_frame, + const AudioFrame& test_frame) { + VerifyParams(ref_frame, test_frame); + const int16_t* ref_frame_data = ref_frame.data(); + const int16_t* test_frame_data = test_frame.data(); + for (size_t i = 0; + i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) { + EXPECT_EQ(ref_frame_data[i], test_frame_data[i]); + } +} + +void UtilityTest::RunResampleTest(int src_channels, + int src_sample_rate_hz, + int dst_channels, + int dst_sample_rate_hz) { + PushResampler<int16_t> resampler; // Create a new one with every test. + const int16_t kSrcCh1 = 30; // Shouldn't overflow for any used sample rate. + const int16_t kSrcCh2 = 15; + const int16_t kSrcCh3 = 22; + const int16_t kSrcCh4 = 8; + const float resampling_factor = + (1.0 * src_sample_rate_hz) / dst_sample_rate_hz; + const float dst_ch1 = resampling_factor * kSrcCh1; + const float dst_ch2 = resampling_factor * kSrcCh2; + const float dst_ch3 = resampling_factor * kSrcCh3; + const float dst_ch4 = resampling_factor * kSrcCh4; + const float dst_stereo_to_mono = (dst_ch1 + dst_ch2) / 2; + const float dst_quad_to_mono = (dst_ch1 + dst_ch2 + dst_ch3 + dst_ch4) / 4; + const float dst_quad_to_stereo_ch1 = (dst_ch1 + dst_ch2) / 2; + const float dst_quad_to_stereo_ch2 = (dst_ch3 + dst_ch4) / 2; + if (src_channels == 1) + SetMonoFrame(kSrcCh1, src_sample_rate_hz, &src_frame_); + else if (src_channels == 2) + SetStereoFrame(kSrcCh1, kSrcCh2, src_sample_rate_hz, &src_frame_); + else + SetQuadFrame(kSrcCh1, kSrcCh2, kSrcCh3, kSrcCh4, src_sample_rate_hz, + &src_frame_); + + if (dst_channels == 1) { + SetMonoFrame(0, dst_sample_rate_hz, &dst_frame_); + if (src_channels == 1) + SetMonoFrame(dst_ch1, dst_sample_rate_hz, &golden_frame_); + else if (src_channels == 2) + SetMonoFrame(dst_stereo_to_mono, dst_sample_rate_hz, &golden_frame_); + else + SetMonoFrame(dst_quad_to_mono, dst_sample_rate_hz, &golden_frame_); + } else { + SetStereoFrame(0, 0, dst_sample_rate_hz, &dst_frame_); + if (src_channels == 1) + SetStereoFrame(dst_ch1, dst_ch1, dst_sample_rate_hz, &golden_frame_); + else if (src_channels == 2) + SetStereoFrame(dst_ch1, dst_ch2, dst_sample_rate_hz, &golden_frame_); + else + SetStereoFrame(dst_quad_to_stereo_ch1, dst_quad_to_stereo_ch2, + dst_sample_rate_hz, &golden_frame_); + } + + // The sinc resampler has a known delay, which we compute here. Multiplying by + // two gives us a crude maximum for any resampling, as the old resampler + // typically (but not always) has lower delay. + static const size_t kInputKernelDelaySamples = 16; + const size_t max_delay = static_cast<size_t>( + static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz * + kInputKernelDelaySamples * dst_channels * 2); + printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later. + src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz); + RemixAndResample(src_frame_, &resampler, &dst_frame_); + + if (src_sample_rate_hz == 96000 && dst_sample_rate_hz <= 11025) { + // The sinc resampler gives poor SNR at this extreme conversion, but we + // expect to see this rarely in practice. + EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 14.0f); + } else { + EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 46.0f); + } +} + +TEST_F(UtilityTest, RemixAndResampleCopyFrameSucceeds) { + // Stereo -> stereo. + SetStereoFrame(10, 10, &src_frame_); + SetStereoFrame(0, 0, &dst_frame_); + RemixAndResample(src_frame_, &resampler_, &dst_frame_); + VerifyFramesAreEqual(src_frame_, dst_frame_); + + // Mono -> mono. + SetMonoFrame(20, &src_frame_); + SetMonoFrame(0, &dst_frame_); + RemixAndResample(src_frame_, &resampler_, &dst_frame_); + VerifyFramesAreEqual(src_frame_, dst_frame_); +} + +TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) { + // Stereo -> mono. + SetStereoFrame(0, 0, &dst_frame_); + SetMonoFrame(10, &src_frame_); + SetStereoFrame(10, 10, &golden_frame_); + RemixAndResample(src_frame_, &resampler_, &dst_frame_); + VerifyFramesAreEqual(dst_frame_, golden_frame_); + + // Mono -> stereo. + SetMonoFrame(0, &dst_frame_); + SetStereoFrame(10, 20, &src_frame_); + SetMonoFrame(15, &golden_frame_); + RemixAndResample(src_frame_, &resampler_, &dst_frame_); + VerifyFramesAreEqual(golden_frame_, dst_frame_); +} + +TEST_F(UtilityTest, RemixAndResampleSucceeds) { + const int kSampleRates[] = {8000, 11025, 16000, 22050, + 32000, 44100, 48000, 96000}; + const int kSrcChannels[] = {1, 2, 4}; + const int kDstChannels[] = {1, 2}; + + for (int src_rate : kSampleRates) { + for (int dst_rate : kSampleRates) { + for (size_t src_channels : kSrcChannels) { + for (size_t dst_channels : kDstChannels) { + RunResampleTest(src_channels, src_rate, dst_channels, dst_rate); + } + } + } + } +} + +} // namespace +} // namespace voe +} // namespace webrtc |