Index: webrtc/modules/utility/include/file_player.h |
diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h |
deleted file mode 100644 |
index b064e3021b54a40b85919043e7353f051e49ec66..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/utility/include/file_player.h |
+++ /dev/null |
@@ -1,86 +0,0 @@ |
-/* |
- * 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_ |