Improved audio buffer handling for iOS

webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

Index: webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
diff --git a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
index 535f16816cb4998230d6394d14b2af9c88007ea6..3ead791986e2739cb60cc0020eec9e0ea38b7dd1 100644
--- a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
+++ b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -13,6 +13,7 @@
 #include <limits.h>
 #include <memory>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
@@ -114,15 +115,16 @@ void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
       .Times(kNumberOfUpdateBufferCalls - 1)
       .WillRepeatedly(Return(kSamplesPer10Ms));
 
-  FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
-                              sample_rate);
+  FineAudioBuffer fine_buffer(
+      &audio_device_buffer, sample_rate, kFrameSizeBytes);
 
   std::unique_ptr<int8_t[]> out_buffer;
   out_buffer.reset(new int8_t[kFrameSizeBytes]);

[kwiberg-webrtc, 2017/05/29 04:09:02]

Replace these two lines with e.g.

  int8_t out_buffer[kFrameSizeBytes];

...

[henrika_webrtc, 2017/05/29 10:33:51]

Done.

   std::unique_ptr<int8_t[]> in_buffer;
   in_buffer.reset(new int8_t[kFrameSizeBytes]);
   for (int i = 0; i < kNumberOfFrames; ++i) {
-    fine_buffer.GetPlayoutData(out_buffer.get());
+    fine_buffer.GetPlayoutData(
+        rtc::ArrayView<int8_t>(out_buffer.get(), kFrameSizeBytes));

[kwiberg-webrtc, 2017/05/29 04:09:02]

...and you'll be able to say just

  fine_buffer.GetPlayoutData(out_buffer);

here.

Oh, and because stack allocations are essentially free, move the allocation into
the loop body, just before first use.

[henrika_webrtc, 2017/05/29 10:33:51]

Done.

[kwiberg-webrtc, 2017/05/29 11:07:01]

You don't appear to have actually done this. :-)

[henrika_webrtc, 2017/05/29 15:01:34]

Actually. I don't really understand your comment. The is no conversion between a
plain array and ArrayView afaik. I tried it and it does not build.

[kwiberg-webrtc, 2017/05/30 07:58:19]

There's supposed to be one. See e.g.
https://cs.chromium.org/chromium/src/third_party/webrtc/base/array_view_unitt...

What's the error message?

[henrika_webrtc, 2017/05/30 08:37:15]

I should have explained better. The issue is that the size or xx_buffer is given
by an input parameter. One test tests when the size is less than 10ms and
another when it is larger that 10ms. I did not come up with any other solution
than using unique_ptr. Hope that is OK. Thanks for all valuable feedback.

[kwiberg-webrtc, 2017/05/30 09:05:16]

OK. The name "kFrameSizeBytes" confused me into thinking it was a constant...

You can get away with less manual twiddling in these situations if you use e.g.
std::vector or rtc::Buffer instead:

  rtc::Buffer out_buffer(kFrameSizeBytes);
  ...
    fine_buffer.GetPlayoutData(out_buffer);

But no need to change it now.

     EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
     UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
     fine_buffer.DeliverRecordedData(in_buffer.get(), kFrameSizeBytes, 0, 0);