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

media/cast/sender/audio_encoder.cc

Index: media/cast/sender/audio_encoder.cc
diff --git a/media/cast/sender/audio_encoder.cc b/media/cast/sender/audio_encoder.cc
index 3c5740c74c4ed4bfadf93c81d9c41e7587aa9260..bacb793fddbb6b9fd1cfc23de5fca3fcc36fcb66 100644
--- a/media/cast/sender/audio_encoder.cc
+++ b/media/cast/sender/audio_encoder.cc
@@ -19,6 +19,8 @@
 #include "base/time/time.h"
 #include "base/trace_event/trace_event.h"
 #include "build/build_config.h"
+#include "media/cast/common/rtp_time.h"
+#include "media/cast/constants.h"
 
 #if !defined(OS_IOS)
 #include "third_party/opus/src/include/opus.h"
@@ -63,8 +65,7 @@ class AudioEncoder::ImplBase
             base::Time::kMicrosecondsPerSecond * samples_per_frame_ /
             sampling_rate)),
         buffer_fill_end_(0),
-        frame_id_(0),
-        frame_rtp_timestamp_(0),
+        frame_id_(kFirstFrameId),
         samples_dropped_from_buffer_(0) {
     // Support for max sampling rate of 48KHz, 2 channels, 100 ms duration.
     const int kMaxSamplesTimesChannelsPerFrame = 48 * 2 * 100;
@@ -107,7 +108,7 @@ class AudioEncoder::ImplBase
         buffer_fill_end_ = 0;
         buffer_fill_duration = base::TimeDelta();
         frame_rtp_timestamp_ +=
-            static_cast<uint32_t>(num_frames_missed * samples_per_frame_);
+            RtpTimeDelta::FromTicks(num_frames_missed * samples_per_frame_);
         DVLOG(1) << "Skipping RTP timestamp ahead to account for "
                  << num_frames_missed * samples_per_frame_
                  << " samples' worth of underrun.";
@@ -146,9 +147,11 @@ class AudioEncoder::ImplBase
       audio_frame->rtp_timestamp = frame_rtp_timestamp_;
       audio_frame->reference_time = frame_capture_time_;
 
-      TRACE_EVENT_ASYNC_BEGIN2("cast.stream", "Audio Encode", audio_frame.get(),
-                               "frame_id", frame_id_,
-                               "rtp_timestamp", frame_rtp_timestamp_);
+      TRACE_EVENT_ASYNC_BEGIN2(
+          "cast.stream",
+          "Audio Encode", audio_frame.get(),
+          "frame_id", frame_id_,
+          "rtp_timestamp", frame_rtp_timestamp_.lower_32_bits());
       if (EncodeFromFilledBuffer(&audio_frame->data)) {
         // Compute deadline utilization as the real-world time elapsed divided
         // by the signal duration.
@@ -174,7 +177,7 @@ class AudioEncoder::ImplBase
       // Reset the internal buffer, frame ID, and timestamps for the next frame.
       buffer_fill_end_ = 0;
       ++frame_id_;
-      frame_rtp_timestamp_ += samples_per_frame_;
+      frame_rtp_timestamp_ += RtpTimeDelta::FromTicks(samples_per_frame_);
       frame_capture_time_ += frame_duration_;
     }
   }
@@ -214,9 +217,8 @@ class AudioEncoder::ImplBase
   // The RTP timestamp for the next frame of encoded audio.  This is defined as
   // the number of audio samples encoded so far, plus the estimated number of
   // samples that were missed due to data underruns.  A receiver uses this value
-  // to detect gaps in the audio signal data being provided.  Per the spec, RTP
-  // timestamp values are allowed to overflow and roll around past zero.
-  uint32_t frame_rtp_timestamp_;
+  // to detect gaps in the audio signal data being provided.
+  RtpTimeTicks frame_rtp_timestamp_;
 
   // The local system time associated with the start of the next frame of
   // encoded audio.  This value is passed on to a receiver as a reference clock