Switch to standard integer types in content/renderer/.

content/renderer/media/media_stream_audio_processor.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/media_stream_audio_processor.h"
 
+#include <stddef.h>
+#include <stdint.h>
+
 #include "base/command_line.h"
 #include "base/metrics/field_trial.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_number_conversions.h"
 #include "base/trace_event/trace_event.h"
+#include "build/build_config.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_converter.h"
 #include "media/base/audio_fifo.h"
 #include "media/base/channel_layout.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
@@ skipping 676 matching lines @@
                                            int volume,
                                            bool key_pressed,
                                            float* const* output_ptrs) {
   DCHECK(audio_processing_);
   DCHECK(capture_thread_checker_.CalledOnValidThread());
 
   TRACE_EVENT0("audio", "MediaStreamAudioProcessor::ProcessData");
 
   base::subtle::Atomic32 render_delay_ms =
       base::subtle::Acquire_Load(&render_delay_ms_);
-  int64 capture_delay_ms = capture_delay.InMilliseconds();
+  int64_t capture_delay_ms = capture_delay.InMilliseconds();
   DCHECK_LT(capture_delay_ms,
             std::numeric_limits<base::subtle::Atomic32>::max());
   int total_delay_ms =  capture_delay_ms + render_delay_ms;
   if (total_delay_ms > 300) {
     LOG(WARNING) << "Large audio delay, capture delay: " << capture_delay_ms
                  << "ms; render delay: " << render_delay_ms << "ms";
   }
 
   webrtc::AudioProcessing* ap = audio_processing_.get();
   ap->set_stream_delay_ms(total_delay_ms);
@@ skipping 25 matching lines @@
   if (echo_information_) {
     echo_information_.get()->UpdateAecDelayStats(ap->echo_cancellation());
   }
 
   // Return 0 if the volume hasn't been changed, and otherwise the new volume.
   return (agc->stream_analog_level() == volume) ?
       0 : agc->stream_analog_level();
 }
 
 }  // namespace content