Adding "rtp_timebase" variable in lieu of "frequency" for FrameReceiverConfig structure.

chrome/browser/extensions/api/cast_streaming/cast_streaming_apitest.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 <algorithm>
 #include <cmath>
 #include <vector>
 
 #include "base/callback_helpers.h"
 #include "base/command_line.h"
@@ skipping 158 matching lines @@
     // Assume the audio signal is a single sine wave (it can have some
     // low-amplitude noise).  Count zero crossings, and extrapolate the
     // frequency of the sine wave in |audio_frame|.
     int crossings = 0;
     for (int ch = 0; ch < audio_frame->channels(); ++ch) {
       crossings += media::cast::CountZeroCrossings(audio_frame->channel(ch),
                                                    audio_frame->frames());
     }
     crossings /= audio_frame->channels();  // Take the average.
     const float seconds_per_frame =
-        audio_frame->frames() / static_cast<float>(audio_config().frequency);
+        audio_frame->frames() / static_cast<float>(audio_config().rtp_timebase);
     const float frequency = crossings / seconds_per_frame / 2.0f;
     VLOG(1) << "Current audio tone frequency: " << frequency;
 
     const int kTargetWindowHz = 20;
     for (std::vector<int>::iterator it = expected_tones_.begin();
          it != expected_tones_.end(); ++it) {
       if (abs(static_cast<int>(frequency) - *it) < kTargetWindowHz) {
         LOG(INFO) << "Heard tone at frequency " << *it << " Hz.";
         expected_tones_.erase(it);
         MaybeRunDoneCallback();
@@ skipping 195 matching lines @@
 
   delete receiver;
   cast_environment->Shutdown();
 }
 
 IN_PROC_BROWSER_TEST_F(CastStreamingApiTestWithPixelOutput, RtpStreamError) {
   ASSERT_TRUE(RunExtensionSubtest("cast_streaming", "rtp_stream_error.html"));
 }
 
 }  // namespace extensions