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 |
new file mode 100644 |
index 0000000000000000000000000000000000000000..b064e3021b54a40b85919043e7353f051e49ec66 |
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+++ b/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_ |