Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )

This is the second revert. The first attempt in https://codereview.webrtc.org/1423693008/
was missing a subtle curly brace caused by a merge conflict.
I'm going to let this one go through the CQ.

Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.

See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.

Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}

TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1412963007

Cr-Commit-Position: refs/heads/master@{#10532}
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index 1383bbe..fdfaab0 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -636,8 +636,8 @@
         }
 
         if (!raw_output) {
-          // The WAV file needs to be reset every time, because it can't change
-          // its sample rate or number of channels.
+          // The WAV file needs to be reset every time, because it cant change
+          // it's sample rate or number of channels.
           output_wav_file.reset(new WavWriter(out_filename + ".wav",
                                               output_sample_rate,
                                               msg.num_output_channels()));