Move remaining traces of VoiceEngine
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}diff --git a/audio/remix_resample_unittest.cc b/audio/remix_resample_unittest.cc
new file mode 100644
index 0000000..753584b
--- /dev/null
+++ b/audio/remix_resample_unittest.cc
@@ -0,0 +1,275 @@
+/*
+ * 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 <math.h>
+
+#include "audio/remix_resample.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/format_macros.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace voe {
+namespace {
+
+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_ = rtc::CheckedDivExact(sample_rate_hz, 100);
+ 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_ = rtc::CheckedDivExact(sample_rate_hz, 100);
+ 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_ = rtc::CheckedDivExact(sample_rate_hz, 100);
+ 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 * log10(variance / mse);
+ if (snr > best_snr) {
+ best_snr = snr;
+ best_delay = delay;
+ }
+ }
+ printf("SNR=%.1f dB at delay=%" PRIuS "\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 == 8000) {
+ // 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, 16000, 32000, 44100, 48000, 96000};
+ const int kSampleRatesSize = arraysize(kSampleRates);
+ const int kSrcChannels[] = {1, 2, 4};
+ const int kSrcChannelsSize = arraysize(kSrcChannels);
+ const int kDstChannels[] = {1, 2};
+ const int kDstChannelsSize = arraysize(kDstChannels);
+
+ for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
+ for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
+ for (int src_channel = 0; src_channel < kSrcChannelsSize;
+ src_channel++) {
+ for (int dst_channel = 0; dst_channel < kDstChannelsSize;
+ dst_channel++) {
+ RunResampleTest(kSrcChannels[src_channel], kSampleRates[src_rate],
+ kDstChannels[dst_channel], kSampleRates[dst_rate]);
+ }
+ }
+ }
+ }
+}
+
+} // namespace
+} // namespace voe
+} // namespace webrtc
