PPAPI: Use clang-format on ppapi/shared_impl

ppapi/shared_impl/ppb_audio_config_shared.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "build/build_config.h"
 #include "ppapi/shared_impl/ppb_audio_config_shared.h"
 #include "ppapi/thunk/enter.h"
 #include "ppapi/thunk/ppb_instance_api.h"
 
 namespace ppapi {
 
 // Rounds up requested_size to the nearest multiple of minimum_size.
 static uint32_t CalculateMultipleOfSampleFrameCount(uint32_t minimum_size,
                                                     uint32_t requested_size) {
   const uint32_t multiple = (requested_size + minimum_size - 1) / minimum_size;
   return std::min(minimum_size * multiple,
                   static_cast<uint32_t>(PP_AUDIOMAXSAMPLEFRAMECOUNT));
 }
 
 PPB_AudioConfig_Shared::PPB_AudioConfig_Shared(ResourceObjectType type,
                                                PP_Instance instance)
     : Resource(type, instance),
       sample_rate_(PP_AUDIOSAMPLERATE_NONE),
-      sample_frame_count_(0) {
-}
+      sample_frame_count_(0) {}
 
-PPB_AudioConfig_Shared::~PPB_AudioConfig_Shared() {
-}
+PPB_AudioConfig_Shared::~PPB_AudioConfig_Shared() {}
 
-PP_Resource PPB_AudioConfig_Shared::Create(
-    ResourceObjectType type,
-    PP_Instance instance,
-    PP_AudioSampleRate sample_rate,
-    uint32_t sample_frame_count) {
+PP_Resource PPB_AudioConfig_Shared::Create(ResourceObjectType type,
+                                           PP_Instance instance,
+                                           PP_AudioSampleRate sample_rate,
+                                           uint32_t sample_frame_count) {
   scoped_refptr<PPB_AudioConfig_Shared> object(
       new PPB_AudioConfig_Shared(type, instance));
   if (!object->Init(sample_rate, sample_frame_count))
     return 0;
   return object->GetReference();
 }
 
 // static
 uint32_t PPB_AudioConfig_Shared::RecommendSampleFrameCount_1_0(
     PP_AudioSampleRate sample_rate,
@@ skipping 45 matching lines @@
   // Adobe Flash is affected but not HTML5, WebRTC and WebAudio (they are using
   // real time threads).
   const bool kHighLatencyDevice = true;
 #else
   const bool kHighLatencyDevice = false;
 #endif
 
   // If client is using same sample rate as audio hardware, then recommend a
   // multiple of the audio hardware's sample frame count.
   if (!kHighLatencyDevice && hardware_sample_rate == sample_rate) {
-    return CalculateMultipleOfSampleFrameCount(
-        hardware_sample_frame_count, sample_frame_count);
+    return CalculateMultipleOfSampleFrameCount(hardware_sample_frame_count,
+                                               sample_frame_count);
   }
 
   // If the hardware requires a high latency buffer or we're at a low sample
   // rate w/ a buffer that's larger than 10ms, choose the nearest multiple of
   // the high latency sample frame count.  An example of too low and too large
   // is 16kHz and a sample frame count greater than 160 frames.
   if (kHighLatencyDevice ||
       hardware_sample_frame_count >= kHighLatencySampleFrameCount ||
       (hardware_sample_rate < 44100 &&
        hardware_sample_frame_count > hardware_sample_rate / 100u)) {
     return CalculateMultipleOfSampleFrameCount(
         sample_frame_count,
         std::max(kHighLatencySampleFrameCount, hardware_sample_frame_count));
   }
 
   // All low latency clients should be able to handle a 512 frame buffer with
   // resampling from 44.1kHz and 48kHz to higher sample rates.
   // TODO(dalecurtis): We may need to investigate making the callback thread
   // high priority to handle buffers at the absolute minimum w/o glitching.
   const uint32_t kLowLatencySampleFrameCount = 512;
 
   // Special case for 48kHz -> 44.1kHz and buffer sizes greater than 10ms.  In
   // testing most buffer sizes > 10ms led to glitching, so we choose a size we
   // know won't cause jitter.
   int min_sample_frame_count = kLowLatencySampleFrameCount;
   if (hardware_sample_rate == 44100 && sample_rate == 48000 &&
       hardware_sample_frame_count > hardware_sample_rate / 100u) {
-    min_sample_frame_count = std::max(
-        2 * kLowLatencySampleFrameCount, hardware_sample_frame_count);
+    min_sample_frame_count =
+        std::max(2 * kLowLatencySampleFrameCount, hardware_sample_frame_count);
   }
 
-  return CalculateMultipleOfSampleFrameCount(
-      min_sample_frame_count, sample_frame_count);
+  return CalculateMultipleOfSampleFrameCount(min_sample_frame_count,
+                                             sample_frame_count);
 }
 
 // static
 PP_AudioSampleRate PPB_AudioConfig_Shared::RecommendSampleRate(
     PP_Instance instance) {
   thunk::EnterInstanceNoLock enter(instance);
   if (enter.failed())
     return PP_AUDIOSAMPLERATE_NONE;
   PP_AudioSampleRate hardware_sample_rate = static_cast<PP_AudioSampleRate>(
-    enter.functions()->GetAudioHardwareOutputSampleRate(instance));
+      enter.functions()->GetAudioHardwareOutputSampleRate(instance));
   return hardware_sample_rate;
 }
 
 thunk::PPB_AudioConfig_API* PPB_AudioConfig_Shared::AsPPB_AudioConfig_API() {
   return this;
 }
 
 PP_AudioSampleRate PPB_AudioConfig_Shared::GetSampleRate() {
   return sample_rate_;
 }
@@ skipping 17 matching lines @@
   if (sample_frame_count > PP_AUDIOMAXSAMPLEFRAMECOUNT ||
       sample_frame_count < PP_AUDIOMINSAMPLEFRAMECOUNT)
     return false;
 
   sample_rate_ = sample_rate;
   sample_frame_count_ = sample_frame_count;
   return true;
 }
 
 }  // namespace ppapi