git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/utility/interface/file_player.h b/modules/utility/interface/file_player.h
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+/*
+ * Copyright (c) 2011 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_INTERFACE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+
+#include "common_types.h"
+#include "engine_configurations.h"
+#include "module_common_types.h"
+#include "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 video file formats (e.g. AVI) if the flag
+ // WEBRTC_MODULE_UTILITY_VIDEO is not defined.
+ static FilePlayer* CreateFilePlayer(const WebRtc_UWord32 instanceID,
+ const FileFormats fileFormat);
+
+ static void DestroyFilePlayer(FilePlayer* player);
+
+ virtual WebRtc_Word32 Get10msAudioFromFile(
+ WebRtc_Word16* decodedDataBuffer,
+ WebRtc_UWord32& decodedDataLengthInSamples,
+ const WebRtc_UWord32 frequencyInHz) = 0;
+
+ // Register callback for receiving file playing notifications.
+ virtual WebRtc_Word32 RegisterModuleFileCallback(
+ FileCallback* callback) = 0;
+
+ // API for playing audio from fileName to channel.
+ // Note: codecInst is used for pre-encoded files.
+ virtual WebRtc_Word32 StartPlayingFile(
+ const WebRtc_Word8* fileName,
+ bool loop,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
+
+ // Note: codecInst is used for pre-encoded files.
+ virtual WebRtc_Word32 StartPlayingFile(
+ InStream& sourceStream,
+ WebRtc_UWord32 startPosition,
+ float volumeScaling,
+ WebRtc_UWord32 notification,
+ WebRtc_UWord32 stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
+
+ virtual WebRtc_Word32 StopPlayingFile() = 0;
+
+ virtual bool IsPlayingFile() const = 0;
+
+ virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs) = 0;
+
+ // Set audioCodec to the currently used audio codec.
+ virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const = 0;
+
+ virtual WebRtc_Word32 Frequency() const = 0;
+
+ // Note: scaleFactor is in the range [0.0 - 2.0]
+ virtual WebRtc_Word32 SetAudioScaling(float scaleFactor) = 0;
+
+ // Return the time in ms until next video frame should be pulled (by
+ // calling GetVideoFromFile(..)).
+ // Note: this API reads one video frame from file. This means that it should
+ // be called exactly once per GetVideoFromFile(..) API call.
+ virtual WebRtc_Word32 TimeUntilNextVideoFrame() { return -1;}
+
+ virtual WebRtc_Word32 StartPlayingVideoFile(
+ const WebRtc_Word8* /*fileName*/,
+ bool /*loop*/,
+ bool /*videoOnly*/) { return -1;}
+
+ virtual WebRtc_Word32 video_codec_info(VideoCodec& /*videoCodec*/) const
+ {return -1;}
+
+ virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/)
+ { return -1;}
+
+ // Same as GetVideoFromFile(). videoFrame will have the resolution specified
+ // by the width outWidth and height outHeight in pixels.
+ virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/,
+ const WebRtc_UWord32 /*outWidth*/,
+ const WebRtc_UWord32 /*outHeight*/)
+ {return -1;}
+protected:
+ virtual ~FilePlayer() {}
+
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_