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| 1 /* | |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ | |
| 12 #define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ | |
| 13 | |
| 14 #include "webrtc/common_types.h" | |
| 15 #include "webrtc/modules/include/module.h" | |
| 16 #include "webrtc/modules/include/module_common_types.h" | |
| 17 #include "webrtc/modules/media_file/include/media_file_defines.h" | |
| 18 #include "webrtc/typedefs.h" | |
| 19 | |
| 20 namespace webrtc { | |
| 21 class MediaFile : public Module | |
| 22 { | |
| 23 public: | |
| 24 // Factory method. Constructor disabled. id is the identifier for the | |
| 25 // MediaFile instance. | |
| 26 static MediaFile* CreateMediaFile(const int32_t id); | |
| 27 static void DestroyMediaFile(MediaFile* module); | |
| 28 | |
| 29 // Put 10-60ms of audio data from file into the audioBuffer depending on | |
| 30 // codec frame size. dataLengthInBytes is both an input and output | |
| 31 // parameter. As input parameter it indicates the size of audioBuffer. | |
| 32 // As output parameter it indicates the number of bytes written to | |
| 33 // audioBuffer. | |
| 34 // Note: This API only play mono audio but can be used on file containing | |
| 35 // audio with more channels (in which case the audio will be converted to | |
| 36 // mono). | |
| 37 virtual int32_t PlayoutAudioData( | |
| 38 int8_t* audioBuffer, | |
| 39 size_t& dataLengthInBytes) = 0; | |
| 40 | |
| 41 // Put 10-60ms, depending on codec frame size, of audio data from file into | |
| 42 // audioBufferLeft and audioBufferRight. The buffers contain the left and | |
| 43 // right channel of played out stereo audio. | |
| 44 // dataLengthInBytes is both an input and output parameter. As input | |
| 45 // parameter it indicates the size of both audioBufferLeft and | |
| 46 // audioBufferRight. As output parameter it indicates the number of bytes | |
| 47 // written to both audio buffers. | |
| 48 // Note: This API can only be successfully called for WAV files with stereo | |
| 49 // audio. | |
| 50 virtual int32_t PlayoutStereoData( | |
| 51 int8_t* audioBufferLeft, | |
| 52 int8_t* audioBufferRight, | |
| 53 size_t& dataLengthInBytes) = 0; | |
| 54 | |
| 55 // Open the file specified by fileName (relative path is allowed) for | |
| 56 // reading. FileCallback::PlayNotification(..) will be called after | |
| 57 // notificationTimeMs of the file has been played if notificationTimeMs is | |
| 58 // greater than zero. If loop is true the file will be played until | |
| 59 // StopPlaying() is called. When end of file is reached the file is read | |
| 60 // from the start. format specifies the type of file fileName refers to. | |
| 61 // codecInst specifies the encoding of the audio data. Note that | |
| 62 // file formats that contain this information (like WAV files) don't need to | |
| 63 // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero, | |
| 64 // specify what part of the file should be read. From startPointMs ms to | |
| 65 // stopPointMs ms. | |
| 66 // Note: codecInst.channels should be set to 2 for stereo (and 1 for | |
| 67 // mono). Stereo audio is only supported for WAV files. | |
| 68 virtual int32_t StartPlayingAudioFile( | |
| 69 const char* fileName, | |
| 70 const uint32_t notificationTimeMs = 0, | |
| 71 const bool loop = false, | |
| 72 const FileFormats format = kFileFormatPcm16kHzFile, | |
| 73 const CodecInst* codecInst = NULL, | |
| 74 const uint32_t startPointMs = 0, | |
| 75 const uint32_t stopPointMs = 0) = 0; | |
| 76 | |
| 77 // Prepare for playing audio from stream. | |
| 78 // FileCallback::PlayNotification(..) will be called after | |
| 79 // notificationTimeMs of the file has been played if notificationTimeMs is | |
| 80 // greater than zero. format specifies the type of file fileName refers to. | |
| 81 // codecInst specifies the encoding of the audio data. Note that | |
| 82 // file formats that contain this information (like WAV files) don't need to | |
| 83 // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero, | |
| 84 // specify what part of the file should be read. From startPointMs ms to | |
| 85 // stopPointMs ms. | |
| 86 // Note: codecInst.channels should be set to 2 for stereo (and 1 for | |
| 87 // mono). Stereo audio is only supported for WAV files. | |
| 88 virtual int32_t StartPlayingAudioStream( | |
| 89 InStream& stream, | |
| 90 const uint32_t notificationTimeMs = 0, | |
| 91 const FileFormats format = kFileFormatPcm16kHzFile, | |
| 92 const CodecInst* codecInst = NULL, | |
| 93 const uint32_t startPointMs = 0, | |
| 94 const uint32_t stopPointMs = 0) = 0; | |
| 95 | |
| 96 // Stop playing from file or stream. | |
| 97 virtual int32_t StopPlaying() = 0; | |
| 98 | |
| 99 // Return true if playing. | |
| 100 virtual bool IsPlaying() = 0; | |
| 101 | |
| 102 | |
| 103 // Set durationMs to the number of ms that has been played from file. | |
| 104 virtual int32_t PlayoutPositionMs( | |
| 105 uint32_t& durationMs) const = 0; | |
| 106 | |
| 107 // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, | |
| 108 // to file. The audio frame size is determined by the codecInst.pacsize | |
| 109 // parameter of the last sucessfull StartRecordingAudioFile(..) call. | |
| 110 // Note: bufferLength must be exactly one frame. | |
| 111 virtual int32_t IncomingAudioData( | |
| 112 const int8_t* audioBuffer, | |
| 113 const size_t bufferLength) = 0; | |
| 114 | |
| 115 // Open/creates file specified by fileName for writing (relative path is | |
| 116 // allowed). FileCallback::RecordNotification(..) will be called after | |
| 117 // notificationTimeMs of audio data has been recorded if | |
| 118 // notificationTimeMs is greater than zero. | |
| 119 // format specifies the type of file that should be created/opened. | |
| 120 // codecInst specifies the encoding of the audio data. maxSizeBytes | |
| 121 // specifies the number of bytes allowed to be written to file if it is | |
| 122 // greater than zero. | |
| 123 // Note: codecInst.channels should be set to 2 for stereo (and 1 for | |
| 124 // mono). Stereo is only supported for WAV files. | |
| 125 virtual int32_t StartRecordingAudioFile( | |
| 126 const char* fileName, | |
| 127 const FileFormats format, | |
| 128 const CodecInst& codecInst, | |
| 129 const uint32_t notificationTimeMs = 0, | |
| 130 const uint32_t maxSizeBytes = 0) = 0; | |
| 131 | |
| 132 // Prepare for recording audio to stream. | |
| 133 // FileCallback::RecordNotification(..) will be called after | |
| 134 // notificationTimeMs of audio data has been recorded if | |
| 135 // notificationTimeMs is greater than zero. | |
| 136 // format specifies the type of file that stream should correspond to. | |
| 137 // codecInst specifies the encoding of the audio data. | |
| 138 // Note: codecInst.channels should be set to 2 for stereo (and 1 for | |
| 139 // mono). Stereo is only supported for WAV files. | |
| 140 virtual int32_t StartRecordingAudioStream( | |
| 141 OutStream& stream, | |
| 142 const FileFormats format, | |
| 143 const CodecInst& codecInst, | |
| 144 const uint32_t notificationTimeMs = 0) = 0; | |
| 145 | |
| 146 // Stop recording to file or stream. | |
| 147 virtual int32_t StopRecording() = 0; | |
| 148 | |
| 149 // Return true if recording. | |
| 150 virtual bool IsRecording() = 0; | |
| 151 | |
| 152 // Set durationMs to the number of ms that has been recorded to file. | |
| 153 virtual int32_t RecordDurationMs(uint32_t& durationMs) = 0; | |
| 154 | |
| 155 // Return true if recording or playing is stereo. | |
| 156 virtual bool IsStereo() = 0; | |
| 157 | |
| 158 // Register callback to receive media file related notifications. Disables | |
| 159 // callbacks if callback is NULL. | |
| 160 virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0; | |
| 161 | |
| 162 // Set durationMs to the size of the file (in ms) specified by fileName. | |
| 163 // format specifies the type of file fileName refers to. freqInHz specifies | |
| 164 // the sampling frequency of the file. | |
| 165 virtual int32_t FileDurationMs( | |
| 166 const char* fileName, | |
| 167 uint32_t& durationMs, | |
| 168 const FileFormats format, | |
| 169 const uint32_t freqInHz = 16000) = 0; | |
| 170 | |
| 171 // Update codecInst according to the current audio codec being used for | |
| 172 // reading or writing. | |
| 173 virtual int32_t codec_info(CodecInst& codecInst) const = 0; | |
| 174 | |
| 175 protected: | |
| 176 MediaFile() {} | |
| 177 virtual ~MediaFile() {} | |
| 178 }; | |
| 179 } // namespace webrtc | |
| 180 #endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ | |
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Chromium Code Reviews
Issue 1435093002:
Remove include dirs from modules/{media_file,pacing} (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master