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Diffstat (limited to 'webrtc/modules/utility/interface/file_player.h')
-rw-r--r-- | webrtc/modules/utility/interface/file_player.h | 111 |
1 files changed, 0 insertions, 111 deletions
diff --git a/webrtc/modules/utility/interface/file_player.h b/webrtc/modules/utility/interface/file_player.h deleted file mode 100644 index 44f03e475a..0000000000 --- a/webrtc/modules/utility/interface/file_player.h +++ /dev/null @@ -1,111 +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_INTERFACE_FILE_PLAYER_H_ -#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_ - -#include "webrtc/common_types.h" -#include "webrtc/engine_configurations.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/typedefs.h" -#include "webrtc/video_frame.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; - - // 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 int32_t TimeUntilNextVideoFrame() { return -1;} - - virtual int32_t StartPlayingVideoFile( - const char* /*fileName*/, - bool /*loop*/, - bool /*videoOnly*/) { return -1;} - - virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const - {return -1;} - - virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; } - - // Same as GetVideoFromFile(). videoFrame will have the resolution specified - // by the width outWidth and height outHeight in pixels. - virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/, - const uint32_t /*outWidth*/, - const uint32_t /*outHeight*/) { - return -1; - } - -protected: - virtual ~FilePlayer() {} - -}; -} // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_ |