Reformat existing code. There should be no functional effects.

This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)

BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1172163004

Cr-Commit-Position: refs/heads/master@{#9420}

1 files changed, 18 insertions, 18 deletions

diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index 821b44caa4..56f1d070ed 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -55,16 +55,16 @@ class FakeAdmTest : public testing::Test,
 
   // Callbacks inherited from webrtc::AudioTransport.
   // ADM is pushing data.
-  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                          const uint32_t nSamples,
-                                          const uint8_t nBytesPerSample,
-                                          const uint8_t nChannels,
-                                          const uint32_t samplesPerSec,
-                                          const uint32_t totalDelayMS,
-                                          const int32_t clockDrift,
-                                          const uint32_t currentMicLevel,
-                                          const bool keyPressed,
-                                          uint32_t& newMicLevel) {
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  const uint32_t nSamples,
+                                  const uint8_t nBytesPerSample,
+                                  const uint8_t nChannels,
+                                  const uint32_t samplesPerSec,
+                                  const uint32_t totalDelayMS,
+                                  const int32_t clockDrift,
+                                  const uint32_t currentMicLevel,
+                                  const bool keyPressed,
+                                  uint32_t& newMicLevel) override {
     rec_buffer_bytes_ = nSamples * nBytesPerSample;
     if ((rec_buffer_bytes_ == 0) ||
         (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
@@ -79,14 +79,14 @@ class FakeAdmTest : public testing::Test,
   }
 
   // ADM is pulling data.
-  virtual int32_t NeedMorePlayData(const uint32_t nSamples,
-                                   const uint8_t nBytesPerSample,
-                                   const uint8_t nChannels,
-                                   const uint32_t samplesPerSec,
-                                   void* audioSamples,
-                                   uint32_t& nSamplesOut,
-                                   int64_t* elapsed_time_ms,
-                                   int64_t* ntp_time_ms) {
+  int32_t NeedMorePlayData(const uint32_t nSamples,
+                           const uint8_t nBytesPerSample,
+                           const uint8_t nChannels,
+                           const uint32_t samplesPerSec,
+                           void* audioSamples,
+                           uint32_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
     ++pull_iterations_;
     const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
     const uint32_t bytes_out = RecordedDataReceived() ?