Switch to standard integer types in content/renderer/.

content/renderer/media/webrtc_audio_device_impl.cc

 // Copyright 2013 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 "content/renderer/media/webrtc_audio_device_impl.h"
 
 #include "base/bind.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
@@ skipping 72 matching lines @@
   int frames_per_10_ms = (sample_rate / 100);
   int bytes_per_sample = sizeof(render_buffer_[0]);
   const int bytes_per_10_ms =
       audio_bus->channels() * frames_per_10_ms * bytes_per_sample;
   DCHECK_EQ(audio_bus->frames() % frames_per_10_ms, 0);
 
   // Get audio frames in blocks of 10 milliseconds from the registered
   // webrtc::AudioTransport source. Keep reading until our internal buffer
   // is full.
   int accumulated_audio_frames = 0;
-  int16* audio_data = &render_buffer_[0];
+  int16_t* audio_data = &render_buffer_[0];
   while (accumulated_audio_frames < audio_bus->frames()) {
     // Get 10ms and append output to temporary byte buffer.
     int64_t elapsed_time_ms = -1;
     int64_t ntp_time_ms = -1;
     static const int kBitsPerByte = 8;
     audio_transport_callback_->PullRenderData(bytes_per_sample * kBitsPerByte,
                                               sample_rate,
                                               audio_bus->channels(),
                                               frames_per_10_ms,
                                               audio_data,
@@ skipping 405 matching lines @@
   // If there is no capturer or there are more than one open capture devices,
   // return false.
   if (capturers_.size() != 1)
     return false;
 
   return capturers_.back()->GetPairedOutputParameters(
       session_id, output_sample_rate, output_frames_per_buffer);
 }
 
 }  // namespace content