Convert usage of ARRAY_SIZE to arraysize.

webrtc/sound/alsasoundsystem.cc

 /*
  *  Copyright 2004 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.
  */
 
 #include "webrtc/sound/alsasoundsystem.h"
 
 #include <algorithm>
-#include "webrtc/sound/sounddevicelocator.h"
-#include "webrtc/sound/soundinputstreaminterface.h"
-#include "webrtc/sound/soundoutputstreaminterface.h"
+
+#include "webrtc/base/arraysize.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/worker.h"
+#include "webrtc/sound/sounddevicelocator.h"
+#include "webrtc/sound/soundinputstreaminterface.h"
+#include "webrtc/sound/soundoutputstreaminterface.h"
 
 namespace rtc {
 
 // Lookup table from the rtc format enum in soundsysteminterface.h to
 // ALSA's enums.
 static const snd_pcm_format_t kCricketFormatToAlsaFormatTable[] = {
   // The order here must match the order in soundsysteminterface.h
   SND_PCM_FORMAT_S16_LE,
 };
 
@@ skipping 566 matching lines @@
 
 bool AlsaSoundSystem::GetDefaultDevice(SoundDeviceLocator **device) {
   if (!IsInitialized()) {
     return false;
   }
   *device = new AlsaDeviceLocator("Default device", "default");
   return true;
 }
 
 inline size_t AlsaSoundSystem::FrameSize(const OpenParams &params) {
-  ASSERT(static_cast<int>(params.format) <
-         ARRAY_SIZE(kCricketFormatToSampleSizeTable));
   return kCricketFormatToSampleSizeTable[params.format] * params.channels;
 }
 
 template <typename StreamInterface>
 StreamInterface *AlsaSoundSystem::OpenDevice(
     const SoundDeviceLocator *device,
     const OpenParams &params,
     snd_pcm_stream_t type,
     StreamInterface *(AlsaSoundSystem::*start_fn)(
         snd_pcm_t *handle,
@@ skipping 34 matching lines @@
     // kLowLatency is 0, so we treat it the same as a request for zero latency.
     // Compute what the user asked for.
     latency = rtc::kNumMicrosecsPerSec *
         params.latency /
         params.freq /
         FrameSize(params);
     // And this is what we'll actually use.
     latency = std::max(latency, kMinimumLatencyUsecs);
   }
 
-  ASSERT(static_cast<int>(params.format) <
-         ARRAY_SIZE(kCricketFormatToAlsaFormatTable));
+  ASSERT(params.format < arraysize(kCricketFormatToAlsaFormatTable));
 
   err = symbol_table_.snd_pcm_set_params()(
       handle,
       kCricketFormatToAlsaFormatTable[params.format],
       // SoundSystemInterface only supports interleaved audio.
       SND_PCM_ACCESS_RW_INTERLEAVED,
       params.channels,
       params.freq,
       1,  // Allow ALSA to resample.
       latency);
@@ skipping 57 matching lines @@
   }
   return new AlsaInputStream(
       this, handle, frame_size, wait_timeout_ms, flags, freq);
 }
 
 inline const char *AlsaSoundSystem::GetError(int err) {
   return symbol_table_.snd_strerror()(err);
 }
 
 }  // namespace rtc