Support 4 channel mic in Windows Core Audio

webrtc/voice_engine/utility_unittest.cc

Index: webrtc/voice_engine/utility_unittest.cc
diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc
index ecd0baaeb30cc100a770fdc14ce65ecde4a33ca3..317a16dc0366d5d8dce189f5e43f3461536887a4 100644
--- a/webrtc/voice_engine/utility_unittest.cc
+++ b/webrtc/voice_engine/utility_unittest.cc
@@ -74,6 +74,20 @@ void SetStereoFrame(AudioFrame* frame, float left, float right,
   }
 }
 
+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_);
@@ -128,30 +142,45 @@ void UtilityTest::RunResampleTest(int src_channels,
                                   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
@@ -207,14 +236,19 @@ TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
 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]);
         }
       }
     }