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/** @brief Encapsulates sample waves used for playback. |
/** @brief Encapsulates sample waves used for playback. |
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* |
* |
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* In case you created a new sample with File::AddSample(), you should |
* In case you created a new sample with File::AddSample(), you should |
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* first call Resize() with the desired sample size. This will create |
* first update all attributes with the desired meta informations |
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* (amount of channels, bit depth, sample rate, etc.), then call |
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* Resize() with the desired sample size. The latter will create |
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* the mandatory RIFF chunk which will hold the sample wave data. |
* the mandatory RIFF chunk which will hold the sample wave data. |
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*/ |
*/ |
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class Sample : public Resource { |
class Sample : public Resource { |
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public: |
public: |
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uint16_t FormatTag; ///< Format ID of the waveform data (should be WAVE_FORMAT_PCM for DLS1 compliant files). |
uint16_t FormatTag; ///< Format ID of the waveform data (should be WAVE_FORMAT_PCM for DLS1 compliant files, this is also the default value if Sample was created with Instrument::AddSample()). |
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uint16_t Channels; ///< Number of channels represented in the waveform data, e.g. 1 for mono, 2 for stereo (). |
uint16_t Channels; ///< Number of channels represented in the waveform data, e.g. 1 for mono, 2 for stereo (defaults to 1=mono if Sample was created with Instrument::AddSample() previously). |
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uint32_t SamplesPerSecond; ///< Sampling rate at which each channel should be played. |
uint32_t SamplesPerSecond; ///< Sampling rate at which each channel should be played (defaults to 44100 if Sample was created with Instrument::AddSample() previously). |
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uint32_t AverageBytesPerSecond; ///< The average number of bytes per second at which the waveform data should be transferred (Playback software can estimate the buffer size using this value). |
uint32_t AverageBytesPerSecond; ///< The average number of bytes per second at which the waveform data should be transferred (Playback software can estimate the buffer size using this value). |
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uint16_t BlockAlign; ///< The block alignment (in bytes) of the waveform data. Playback software needs to process a multiple of <i>BlockAlign</i> bytes of data at a time, so the value of <i>BlockAlign</i> can be used for buffer alignment. |
uint16_t BlockAlign; ///< The block alignment (in bytes) of the waveform data. Playback software needs to process a multiple of <i>BlockAlign</i> bytes of data at a time, so the value of <i>BlockAlign</i> can be used for buffer alignment. |
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uint16_t BitDepth; ///< Size of each sample per channel (only if known sample data format is used, 0 otherwise). |
uint16_t BitDepth; ///< Size of each sample per channel (only if known sample data format is used, 0 otherwise). |
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unsigned long SamplesTotal; ///< Reflects total number of sample points (only if known sample data format is used, 0 otherwise), do not bother to change this value, it will not be saved. |
unsigned long SamplesTotal; ///< Reflects total number of sample points (only if known sample data format is used, 0 otherwise), do not bother to change this value, it will not be saved. |
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uint FrameSize; ///< Reflects the size (in bytes) of one single sample point (only if known sample data format is used, 0 otherwise). |
uint FrameSize; ///< Reflects the size (in bytes) of one single sample point (only if known sample data format is used, 0 otherwise). <b>Caution:</b> with the current version of libgig you have to upate this field by yourself whenever you change one of the following fields: Channels, BitDepth ! Ignoring this might lead to undesired behavior when i.e. calling Resize(), SetPos(), Write() or Read(). |
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void* LoadSampleData(); |
void* LoadSampleData(); |
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void ReleaseSampleData(); |
void ReleaseSampleData(); |