Switch to standard integer types in chrome/browser/, part 2 of 4.

chrome/browser/media/webrtc_logging_handler_host.h

 // 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.
 
 #ifndef CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_
 #define CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_
 
-#include "base/basictypes.h"
+#include <stddef.h>
+#include <stdint.h>
+
+#include "base/macros.h"
 #include "base/memory/shared_memory.h"
+#include "build/build_config.h"
 #include "chrome/browser/media/rtp_dump_type.h"
 #include "chrome/browser/media/webrtc_rtp_dump_handler.h"
 #include "chrome/common/media/webrtc_logging_message_data.h"
 #include "chrome/common/partial_circular_buffer.h"
 #include "content/public/browser/browser_message_filter.h"
 #include "content/public/browser/render_process_host.h"
 #include "net/base/network_interfaces.h"
 
 namespace net {
 class URLRequestContextGetter;
@@ skipping 31 matching lines @@
 
   // Switches the buffer to read-only mode, where access to the internal
   // buffer is allowed from different threads than were used to contribute
   // to the log.  Calls to Log() won't be allowed after calling
   // SetComplete() and the call to SetComplete() must be done on the same
   // thread as constructed the buffer and calls Log().
   void SetComplete();
 
  private:
   base::ThreadChecker thread_checker_;
-  uint8 buffer_[kWebRtcLogSize];
+  uint8_t buffer_[kWebRtcLogSize];
   PartialCircularBuffer circular_;
   bool read_only_;
 };
 
 // WebRtcLoggingHandlerHost handles operations regarding the WebRTC logging:
 // - Opens a shared memory buffer that the handler in the render process
 //   writes to.
 // - Writes basic machine info to the log.
 // - Informs the handler in the render process when to stop logging.
 // - Closes the shared memory (and thereby discarding it) or triggers uploading
@@ skipping 66 matching lines @@
                         stop_callback);
 
   // Stops dumping the RTP headers for the specified direction. Must be called
   // on the IO thread. |type| specifies which direction(s) of RTP packet dumping
   // should be stopped. |callback| will be called when stopping the dump is
   // done.
   void StopRtpDump(RtpDumpType type, const GenericDoneCallback& callback);
 
   // Called when an RTP packet is sent or received. Must be called on the UI
   // thread.
-  void OnRtpPacket(scoped_ptr<uint8[]> packet_header,
+  void OnRtpPacket(scoped_ptr<uint8_t[]> packet_header,
                    size_t header_length,
                    size_t packet_length,
                    bool incoming);
 
   // Starts an audio debug recording. The recording lasts the given |delay|,
   // unless |delay| is zero, in which case recording will continue until
   // StopAudioDebugRecordings() is explicitly invoked.
   // |callback| is invoked once recording stops. If |delay| is zero
   // |callback| is invoked once recording starts.
   // If a recording was already in progress, |error_callback| is invoked instead
@@ skipping 79 matching lines @@
   // sure the log directory exists.
   void CreateRtpDumpHandlerAndStart(RtpDumpType type,
                                     const GenericDoneCallback& callback,
                                     const base::FilePath& dump_dir);
 
   // A helper for starting RTP dump assuming the RTP dump handler has been
   // created.
   void DoStartRtpDump(RtpDumpType type, const GenericDoneCallback& callback);
 
   // Adds the packet to the dump on IO thread.
-  void DumpRtpPacketOnIOThread(scoped_ptr<uint8[]> packet_header,
+  void DumpRtpPacketOnIOThread(scoped_ptr<uint8_t[]> packet_header,
                                size_t header_length,
                                size_t packet_length,
                                bool incoming);
 
   bool ReleaseRtpDumps(WebRtcLogPaths* log_paths);
 
   void FireGenericDoneCallback(
       const WebRtcLoggingHandlerHost::GenericDoneCallback& callback,
       bool success,
       const std::string& error_message);
@@ skipping 59 matching lines @@
   // Must be accessed on the UI thread.
   bool is_audio_debug_recordings_in_progress_;
 
   // This counter allows saving each debug recording in separate files.
   uint64_t current_audio_debug_recordings_id_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLoggingHandlerHost);
 };
 
 #endif  // CHROME_BROWSER_MEDIA_WEBRTC_LOGGING_HANDLER_HOST_H_