Support 4 channel mic in Windows Core Audio

webrtc/voice_engine/utility_unittest.cc

 /*
  *  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.
  */
 
@@ skipping 56 matching lines @@
   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);
   }
 }
 
+void SetQuadFrame(AudioFrame* frame, float ch1, float ch2,
+                  float ch3, float ch4, int sample_rate_hz) {

[aleloi, 2017/02/24 10:39:32]

Please change param order.

[jens.nielsen, 2017/02/24 14:55:49]

Done.

+  memset(frame->data_, 0, sizeof(frame->data_));
+  frame->num_channels_ = 4;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = sample_rate_hz / 100;

[aleloi, 2017/02/24 10:39:32]

Please use rtc::CheckedDivExact<int>(sample_rate_hz, 100) from
webrtc/base/checks.h

[jens.nielsen, 2017/02/24 14:55:49]

Done.

+  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);
+  }
+}
+
 // 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_);
 }
@@ skipping 34 matching lines @@
        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 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_left = resampling_factor * kSrcLeft;
-  const float dst_right = resampling_factor * kSrcRight;
-  const float dst_mono = (dst_left + dst_right) / 2;
+  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(&src_frame_, kSrcLeft, src_sample_rate_hz);
+    SetMonoFrame(&src_frame_, kSrcCh1, src_sample_rate_hz);
+  else if (src_channels == 2)
+    SetStereoFrame(&src_frame_, kSrcCh1, kSrcCh2, src_sample_rate_hz);
   else
-    SetStereoFrame(&src_frame_, kSrcLeft, kSrcRight, src_sample_rate_hz);
+    SetQuadFrame(&src_frame_, kSrcCh1, kSrcCh2, kSrcCh3, kSrcCh4,
+                 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);
+      SetMonoFrame(&golden_frame_, dst_ch1, dst_sample_rate_hz);
+    else if (src_channels == 2)
+      SetMonoFrame(&golden_frame_, dst_stereo_to_mono, dst_sample_rate_hz);
     else
-      SetMonoFrame(&golden_frame_, dst_mono, dst_sample_rate_hz);
+      SetMonoFrame(&golden_frame_, dst_quad_to_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);
+      SetStereoFrame(&golden_frame_, dst_ch1, dst_ch1, dst_sample_rate_hz);
+    else if (src_channels == 2)
+      SetStereoFrame(&golden_frame_, dst_ch1, dst_ch2, dst_sample_rate_hz);
     else
-      SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
+      SetStereoFrame(&golden_frame_, dst_quad_to_stereo_ch1,
+                     dst_quad_to_stereo_ch2, 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.
@@ skipping 35 matching lines @@
   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);
+  const int kSrcChannels[] = {1, 2, 4};
+  const int kDstChannels[] = {1, 2};
+
   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]);
+      for (int src_channel = 0;
+           src_channel < sizeof(kSrcChannels) / sizeof(*kSrcChannels);

[hlundin-webrtc, 2017/02/24 09:46:36]

Use arraysize() from webrtc/base/arraysize.h.

[jens.nielsen, 2017/02/24 14:55:49]

Done.

+           src_channel++) {
+        for (int dst_channel = 0;
+             dst_channel < sizeof(kDstChannels) / sizeof(*kDstChannels);
+             dst_channel++) {
+          RunResampleTest(kSrcChannels[src_channel], kSampleRates[src_rate],
+                          kDstChannels[dst_channel], kSampleRates[dst_rate]);
         }
       }
     }
   }
 }
 
 }  // namespace
 }  // namespace voe
 }  // namespace webrtc