audio_coding: remove "main" directory

This is the last piece of the old directory layout of the modules.

Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1481493004

Cr-Commit-Position: refs/heads/master@{#10803}
diff --git a/webrtc/modules/audio_coding/acm2/acm_resampler.h b/webrtc/modules/audio_coding/acm2/acm_resampler.h
new file mode 100644
index 0000000..700fefa
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/acm_resampler.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace acm2 {
+
+class ACMResampler {
+ public:
+  ACMResampler();
+  ~ACMResampler();
+
+  int Resample10Msec(const int16_t* in_audio,
+                     int in_freq_hz,
+                     int out_freq_hz,
+                     int num_audio_channels,
+                     size_t out_capacity_samples,
+                     int16_t* out_audio);
+
+ private:
+  PushResampler<int16_t> resampler_;
+};
+
+}  // namespace acm2
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_