| /* |
| * 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. |
| */ |
| |
| // Note: the class cannot be used for reading and writing at the same time. |
| #ifndef WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_ |
| #define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_ |
| |
| #include <stdio.h> |
| |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/media_file/media_file_defines.h" |
| |
| namespace webrtc { |
| class InStream; |
| class OutStream; |
| |
| class ModuleFileUtility |
| { |
| public: |
| |
| ModuleFileUtility(); |
| ~ModuleFileUtility(); |
| |
| // Prepare for playing audio from stream. |
| // startPointMs and stopPointMs, unless zero, specify what part of the file |
| // should be read. From startPointMs ms to stopPointMs ms. |
| int32_t InitWavReading(InStream& stream, |
| const uint32_t startPointMs = 0, |
| const uint32_t stopPointMs = 0); |
| |
| // Put 10-60ms of audio data from stream into the audioBuffer depending on |
| // codec frame size. dataLengthInBytes indicates the size of audioBuffer. |
| // The return value is the number of bytes written to audioBuffer. |
| // Note: This API only play mono audio but can be used on file containing |
| // audio with more channels (in which case the audio will be converted to |
| // mono). |
| int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer, |
| const size_t dataLengthInBytes); |
| |
| // Put 10-60ms, depending on codec frame size, of audio data from file into |
| // audioBufferLeft and audioBufferRight. The buffers contain the left and |
| // right channel of played out stereo audio. |
| // dataLengthInBytes indicates the size of both audioBufferLeft and |
| // audioBufferRight. |
| // The return value is the number of bytes read for each buffer. |
| // Note: This API can only be successfully called for WAV files with stereo |
| // audio. |
| int32_t ReadWavDataAsStereo(InStream& wav, |
| int8_t* audioBufferLeft, |
| int8_t* audioBufferRight, |
| const size_t bufferLength); |
| |
| // Prepare for recording audio to stream. |
| // codecInst specifies the encoding of the audio data. |
| // Note: codecInst.channels should be set to 2 for stereo (and 1 for |
| // mono). Stereo is only supported for WAV files. |
| int32_t InitWavWriting(OutStream& stream, const CodecInst& codecInst); |
| |
| // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, |
| // to file. The audio frame size is determined by the codecInst.pacsize |
| // parameter of the last sucessfull StartRecordingAudioFile(..) call. |
| // The return value is the number of bytes written to audioBuffer. |
| int32_t WriteWavData(OutStream& stream, |
| const int8_t* audioBuffer, |
| const size_t bufferLength); |
| |
| // Finalizes the WAV header so that it is correct if nothing more will be |
| // written to stream. |
| // Note: this API must be called before closing stream to ensure that the |
| // WAVE header is updated with the file size. Don't call this API |
| // if more samples are to be written to stream. |
| int32_t UpdateWavHeader(OutStream& stream); |
| |
| // Prepare for playing audio from stream. |
| // startPointMs and stopPointMs, unless zero, specify what part of the file |
| // should be read. From startPointMs ms to stopPointMs ms. |
| // freqInHz is the PCM sampling frequency. |
| // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz) |
| int32_t InitPCMReading(InStream& stream, |
| const uint32_t startPointMs = 0, |
| const uint32_t stopPointMs = 0, |
| const uint32_t freqInHz = 16000); |
| |
| // Put 10-60ms of audio data from stream into the audioBuffer depending on |
| // codec frame size. dataLengthInBytes indicates the size of audioBuffer. |
| // The return value is the number of bytes written to audioBuffer. |
| int32_t ReadPCMData(InStream& stream, int8_t* audioBuffer, |
| const size_t dataLengthInBytes); |
| |
| // Prepare for recording audio to stream. |
| // freqInHz is the PCM sampling frequency. |
| // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz) |
| int32_t InitPCMWriting(OutStream& stream, const uint32_t freqInHz = 16000); |
| |
| // Write one 10ms audio frame, i.e. the bufferLength first bytes of |
| // audioBuffer, to file. The audio frame size is determined by the freqInHz |
| // parameter of the last sucessfull InitPCMWriting(..) call. |
| // The return value is the number of bytes written to audioBuffer. |
| int32_t WritePCMData(OutStream& stream, |
| const int8_t* audioBuffer, |
| size_t bufferLength); |
| |
| // Prepare for playing audio from stream. |
| // startPointMs and stopPointMs, unless zero, specify what part of the file |
| // should be read. From startPointMs ms to stopPointMs ms. |
| int32_t InitCompressedReading(InStream& stream, |
| const uint32_t startPointMs = 0, |
| const uint32_t stopPointMs = 0); |
| |
| // Put 10-60ms of audio data from stream into the audioBuffer depending on |
| // codec frame size. dataLengthInBytes indicates the size of audioBuffer. |
| // The return value is the number of bytes written to audioBuffer. |
| int32_t ReadCompressedData(InStream& stream, |
| int8_t* audioBuffer, |
| const size_t dataLengthInBytes); |
| |
| // Prepare for recording audio to stream. |
| // codecInst specifies the encoding of the audio data. |
| int32_t InitCompressedWriting(OutStream& stream, |
| const CodecInst& codecInst); |
| |
| // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, |
| // to file. The audio frame size is determined by the codecInst.pacsize |
| // parameter of the last sucessfull InitCompressedWriting(..) call. |
| // The return value is the number of bytes written to stream. |
| // Note: bufferLength must be exactly one frame. |
| int32_t WriteCompressedData(OutStream& stream, |
| const int8_t* audioBuffer, |
| const size_t bufferLength); |
| |
| // Prepare for playing audio from stream. |
| // codecInst specifies the encoding of the audio data. |
| int32_t InitPreEncodedReading(InStream& stream, |
| const CodecInst& codecInst); |
| |
| // Put 10-60ms of audio data from stream into the audioBuffer depending on |
| // codec frame size. dataLengthInBytes indicates the size of audioBuffer. |
| // The return value is the number of bytes written to audioBuffer. |
| int32_t ReadPreEncodedData(InStream& stream, |
| int8_t* audioBuffer, |
| const size_t dataLengthInBytes); |
| |
| // Prepare for recording audio to stream. |
| // codecInst specifies the encoding of the audio data. |
| int32_t InitPreEncodedWriting(OutStream& stream, |
| const CodecInst& codecInst); |
| |
| // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, |
| // to stream. The audio frame size is determined by the codecInst.pacsize |
| // parameter of the last sucessfull InitPreEncodedWriting(..) call. |
| // The return value is the number of bytes written to stream. |
| // Note: bufferLength must be exactly one frame. |
| int32_t WritePreEncodedData(OutStream& stream, |
| const int8_t* inData, |
| const size_t dataLengthInBytes); |
| |
| // Set durationMs to the size of the file (in ms) specified by fileName. |
| // freqInHz specifies the sampling frequency of the file. |
| int32_t FileDurationMs(const char* fileName, |
| const FileFormats fileFormat, |
| const uint32_t freqInHz = 16000); |
| |
| // Return the number of ms that have been played so far. |
| uint32_t PlayoutPositionMs(); |
| |
| // Update codecInst according to the current audio codec being used for |
| // reading or writing. |
| int32_t codec_info(CodecInst& codecInst); |
| |
| private: |
| // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio. |
| static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2; |
| |
| |
| int32_t InitWavCodec(uint32_t samplesPerSec, |
| size_t channels, |
| uint32_t bitsPerSample, |
| uint32_t formatTag); |
| |
| // Parse the WAV header in stream. |
| int32_t ReadWavHeader(InStream& stream); |
| |
| // Update the WAV header. freqInHz, bytesPerSample, channels, format, |
| // lengthInBytes specify characterists of the audio data. |
| // freqInHz is the sampling frequency. bytesPerSample is the sample size in |
| // bytes. channels is the number of channels, e.g. 1 is mono and 2 is |
| // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc). |
| // lengthInBytes is the number of bytes the audio samples are using up. |
| int32_t WriteWavHeader(OutStream& stream, |
| uint32_t freqInHz, |
| size_t bytesPerSample, |
| size_t channels, |
| uint32_t format, |
| size_t lengthInBytes); |
| |
| // Put dataLengthInBytes of audio data from stream into the audioBuffer. |
| // The return value is the number of bytes written to audioBuffer. |
| int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer, |
| size_t dataLengthInBytes); |
| |
| // Update the current audio codec being used for reading or writing |
| // according to codecInst. |
| int32_t set_codec_info(const CodecInst& codecInst); |
| |
| struct WAVE_FMTINFO_header |
| { |
| int16_t formatTag; |
| int16_t nChannels; |
| int32_t nSamplesPerSec; |
| int32_t nAvgBytesPerSec; |
| int16_t nBlockAlign; |
| int16_t nBitsPerSample; |
| }; |
| // Identifiers for preencoded files. |
| enum MediaFileUtility_CodecType |
| { |
| kCodecNoCodec = 0, |
| kCodecIsac, |
| kCodecIsacSwb, |
| kCodecIsacLc, |
| kCodecL16_8Khz, |
| kCodecL16_16kHz, |
| kCodecL16_32Khz, |
| kCodecPcmu, |
| kCodecPcma, |
| kCodecIlbc20Ms, |
| kCodecIlbc30Ms, |
| kCodecG722, |
| kCodecG722_1_32Kbps, |
| kCodecG722_1_24Kbps, |
| kCodecG722_1_16Kbps, |
| kCodecG722_1c_48, |
| kCodecG722_1c_32, |
| kCodecG722_1c_24, |
| kCodecAmr, |
| kCodecAmrWb, |
| kCodecG729, |
| kCodecG729_1, |
| kCodecG726_40, |
| kCodecG726_32, |
| kCodecG726_24, |
| kCodecG726_16 |
| }; |
| |
| // TODO (hellner): why store multiple formats. Just store either codec_info_ |
| // or _wavFormatObj and supply conversion functions. |
| WAVE_FMTINFO_header _wavFormatObj; |
| size_t _dataSize; // Chunk size if reading a WAV file |
| // Number of bytes to read. I.e. frame size in bytes. May be multiple |
| // chunks if reading WAV. |
| size_t _readSizeBytes; |
| |
| uint32_t _stopPointInMs; |
| uint32_t _startPointInMs; |
| uint32_t _playoutPositionMs; |
| size_t _bytesWritten; |
| |
| CodecInst codec_info_; |
| MediaFileUtility_CodecType _codecId; |
| |
| // The amount of bytes, on average, used for one audio sample. |
| size_t _bytesPerSample; |
| size_t _readPos; |
| |
| // Only reading or writing can be enabled, not both. |
| bool _reading; |
| bool _writing; |
| |
| // Scratch buffer used for turning stereo audio to mono. |
| uint8_t _tempData[WAV_MAX_BUFFER_SIZE]; |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_ |