Replace uses of raw uint32's with a type-checked RtpTimeTicks data type.

media/cast/net/cast_transport_config.h

 // Copyright 2014 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 MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
 #define MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_
 
 #include <string>
 
 #include "base/basictypes.h"
 #include "base/callback.h"
 #include "base/stl_util.h"
+#include "media/cast/common/rtp_time.h"
 #include "media/cast/net/cast_transport_defines.h"
 
 namespace media {
 namespace cast {
 
 enum Codec {
   CODEC_UNKNOWN,
   CODEC_AUDIO_OPUS,
   CODEC_AUDIO_PCM16,
   CODEC_AUDIO_AAC,
@@ skipping 70 matching lines @@
   // The label associated with the frame upon which this frame depends.  If
   // this frame does not require any other frame in order to become decodable
   // (e.g., key frames), |referenced_frame_id| must equal |frame_id|.
   uint32 referenced_frame_id;
 
   // The stream timestamp, on the timeline of the signal data.  For example, RTP
   // timestamps for audio are usually defined as the total number of audio
   // samples encoded in all prior frames.  A playback system uses this value to
   // detect gaps in the stream, and otherwise stretch the signal to match
   // playout targets.
-  uint32 rtp_timestamp;
+  RtpTimeTicks rtp_timestamp;
 
   // The common reference clock timestamp for this frame.  This value originates
   // from a sender and is used to provide lip synchronization between streams in
   // a receiver.  Thus, in the sender context, this is set to the time at which
   // the frame was captured/recorded.  In the receiver context, this is set to
   // the target playout time.  Over a sequence of frames, this time value is
   // expected to drift with respect to the elapsed time implied by the RTP
   // timestamps; and it may not necessarily increment with precise regularity.
   base::TimeTicks reference_time;
 
@@ skipping 24 matching lines @@
   virtual ~PacketSender() {}
 };
 
 struct RtcpSenderInfo {
   RtcpSenderInfo();
   ~RtcpSenderInfo();
   // First three members are used for lipsync.
   // First two members are used for rtt.
   uint32 ntp_seconds;
   uint32 ntp_fraction;
-  uint32 rtp_timestamp;
+  RtpTimeTicks rtp_timestamp;
   uint32 send_packet_count;
   size_t send_octet_count;
 };
 
 struct RtcpReportBlock {
   RtcpReportBlock();
   ~RtcpReportBlock();
   uint32 remote_ssrc;  // SSRC of sender of this report.
   uint32 media_ssrc;   // SSRC of the RTP packet sender.
   uint8 fraction_lost;
@@ skipping 16 matching lines @@
          lhs.ntp_fraction == rhs.ntp_fraction &&
          lhs.rtp_timestamp == rhs.rtp_timestamp &&
          lhs.send_packet_count == rhs.send_packet_count &&
          lhs.send_octet_count == rhs.send_octet_count;
 }
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_