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}

webrtc/modules/audio_coding/acm2/call_statistics.h

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2013 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_CALL_STATISTICS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+//
+// This class is for book keeping of calls to ACM. It is not useful to log API
+// calls which are supposed to be called every 10ms, e.g. PlayoutData10Ms(),
+// however, it is useful to know the number of such calls in a given time
+// interval. The current implementation covers calls to PlayoutData10Ms() with
+// detailed accounting of the decoded speech type.
+//
+// Thread Safety
+// =============
+// Please note that this class in not thread safe. The class must be protected
+// if different APIs are called from different threads.
+//
+
+namespace webrtc {
+
+namespace acm2 {
+
+class CallStatistics {
+ public:
+  CallStatistics() {}
+  ~CallStatistics() {}
+
+  // Call this method to indicate that NetEq engaged in decoding. |speech_type|
+  // is the audio-type according to NetEq.
+  void DecodedByNetEq(AudioFrame::SpeechType speech_type);
+
+  // Call this method to indicate that a decoding call resulted in generating
+  // silence, i.e. call to NetEq is bypassed and the output audio is zero.
+  void DecodedBySilenceGenerator();
+
+  // Get statistics for decoding. The statistics include the number of calls to
+  // NetEq and silence generator, as well as the type of speech pulled of off
+  // NetEq, c.f. declaration of AudioDecodingCallStats for detailed description.
+  const AudioDecodingCallStats& GetDecodingStatistics() const;
+
+ private:
+  // Reset the decoding statistics.
+  void ResetDecodingStatistics();
+
+  AudioDecodingCallStats decoding_stat_;
+};
+
+}  // namespace acm2
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_