Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 226 insertions, 0 deletions

diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc
new file mode 100644
index 0000000000..226e38366d
--- /dev/null
+++ b/webrtc/voice_engine/utility_unittest.cc
@@ -0,0 +1,226 @@
+/*
+ *  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 "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/format_macros.h"
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/voice_engine/utility.h"
+#include "webrtc/voice_engine/voice_engine_defines.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(AudioFrame* frame, float data, int sample_rate_hz) {
+  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->num_channels_ = 1;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = sample_rate_hz / 100;
+  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(AudioFrame* frame, float data) {
+  SetMonoFrame(frame, data, frame->sample_rate_hz_);
+}
+
+// Sets the signal value to increase by |left| and |right| with every sample in
+// each channel respectively.
+void SetStereoFrame(AudioFrame* frame, float left, float right,
+                    int sample_rate_hz) {
+  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->num_channels_ = 2;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = sample_rate_hz / 100;
+  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(AudioFrame* frame, float left, float right) {
+  SetStereoFrame(frame, left, right, frame->sample_rate_hz_);
+}
+
+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;
+    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);
+  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 kSrcLeft = 30;  // Shouldn't overflow for any used sample rate.
+  const int16_t kSrcRight = 15;
+  const float resampling_factor = (1.0 * src_sample_rate_hz) /
+      dst_sample_rate_hz;
+  const float dst_left = resampling_factor * kSrcLeft;
+  const float dst_right = resampling_factor * kSrcRight;
+  const float dst_mono = (dst_left + dst_right) / 2;
+  if (src_channels == 1)
+    SetMonoFrame(&src_frame_, kSrcLeft, src_sample_rate_hz);
+  else
+    SetStereoFrame(&src_frame_, kSrcLeft, kSrcRight, src_sample_rate_hz);
+
+  if (dst_channels == 1) {
+    SetMonoFrame(&dst_frame_, 0, dst_sample_rate_hz);
+    if (src_channels == 1)
+      SetMonoFrame(&golden_frame_, dst_left, dst_sample_rate_hz);
+    else
+      SetMonoFrame(&golden_frame_, dst_mono, dst_sample_rate_hz);
+  } else {
+    SetStereoFrame(&dst_frame_, 0, 0, dst_sample_rate_hz);
+    if (src_channels == 1)
+      SetStereoFrame(&golden_frame_, dst_left, dst_left, dst_sample_rate_hz);
+    else
+      SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
+  }
+
+  // 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(&src_frame_, 10, 10);
+  SetStereoFrame(&dst_frame_, 0, 0);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(src_frame_, dst_frame_);
+
+  // Mono -> mono.
+  SetMonoFrame(&src_frame_, 20);
+  SetMonoFrame(&dst_frame_, 0);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(src_frame_, dst_frame_);
+}
+
+TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
+  // Stereo -> mono.
+  SetStereoFrame(&dst_frame_, 0, 0);
+  SetMonoFrame(&src_frame_, 10);
+  SetStereoFrame(&golden_frame_, 10, 10);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(dst_frame_, golden_frame_);
+
+  // Mono -> stereo.
+  SetMonoFrame(&dst_frame_, 0);
+  SetStereoFrame(&src_frame_, 10, 20);
+  SetMonoFrame(&golden_frame_, 15);
+  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 = sizeof(kSampleRates) / sizeof(*kSampleRates);
+  const int kChannels[] = {1, 2};
+  const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
+  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 < kChannelsSize; src_channel++) {
+        for (int dst_channel = 0; dst_channel < kChannelsSize; dst_channel++) {
+          RunResampleTest(kChannels[src_channel], kSampleRates[src_rate],
+                          kChannels[dst_channel], kSampleRates[dst_rate]);
+        }
+      }
+    }
+  }
+}
+
+}  // namespace
+}  // namespace voe
+}  // namespace webrtc