Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )

Reason for revert:
Breaks downstream code, so revert again. Yay.

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}

TBR=perkj@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}

webrtc/modules/utility/include/file_player.h

@@ -0,0 +1,86 @@
+/*
+ *  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_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+class FileCallback;
+
+class FilePlayer
+{
+public:
+    // The largest decoded frame size in samples (60ms with 32kHz sample rate).
+    enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
+    enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
+
+    // Note: will return NULL for unsupported formats.
+    static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
+                                        const FileFormats fileFormat);
+
+    static void DestroyFilePlayer(FilePlayer* player);
+
+    // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
+    // will be set to the number of samples read (not the number of samples per
+    // channel).
+    virtual int Get10msAudioFromFile(
+        int16_t* outBuffer,
+        size_t& lengthInSamples,
+        int frequencyInHz) = 0;
+
+    // Register callback for receiving file playing notifications.
+    virtual int32_t RegisterModuleFileCallback(
+        FileCallback* callback) = 0;
+
+    // API for playing audio from fileName to channel.
+    // Note: codecInst is used for pre-encoded files.
+    virtual int32_t StartPlayingFile(
+        const char* fileName,
+        bool loop,
+        uint32_t startPosition,
+        float volumeScaling,
+        uint32_t notification,
+        uint32_t stopPosition = 0,
+        const CodecInst* codecInst = NULL) = 0;
+
+    // Note: codecInst is used for pre-encoded files.
+    virtual int32_t StartPlayingFile(
+        InStream& sourceStream,
+        uint32_t startPosition,
+        float volumeScaling,
+        uint32_t notification,
+        uint32_t stopPosition = 0,
+        const CodecInst* codecInst = NULL) = 0;
+
+    virtual int32_t StopPlayingFile() = 0;
+
+    virtual bool IsPlayingFile() const = 0;
+
+    virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
+
+    // Set audioCodec to the currently used audio codec.
+    virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
+
+    virtual int32_t Frequency() const = 0;
+
+    // Note: scaleFactor is in the range [0.0 - 2.0]
+    virtual int32_t SetAudioScaling(float scaleFactor) = 0;
+
+protected:
+    virtual ~FilePlayer() {}
+
+};
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_