ccAudio
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The AudioResample class is used to manage linear intropolation buffering for rate conversions. More...
#include <audio2.h>
Public Member Functions | |
AudioResample (Rate mul, Rate div) | |
~AudioResample () | |
size_t | process (Linear from, Linear to, size_t count) |
size_t | estimate (size_t count) |
Protected Attributes | |
unsigned | mfact |
unsigned | dfact |
unsigned | max |
unsigned | gpos |
unsigned | ppos |
Sample | last |
Linear | buffer |
Additional Inherited Members | |
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enum | Rate { rateUnknown , rate6khz = 6000 , rate8khz = 8000 , rate16khz = 16000 , rate32khz = 32000 , rate44khz = 44100 } |
Audio encoding rate, samples per second. More... | |
enum | Mode { modeRead , modeReadAny , modeReadOne , modeWrite , modeCache , modeInfo , modeFeed , modeAppend , modeCreate } |
File processing mode, whether to skip missing files, etc. More... | |
enum | Encoding { unknownEncoding = 0 , g721ADPCM , g722Audio , g722_7bit , g722_6bit , g723_2bit , g723_3bit , g723_5bit , gsmVoice , msgsmVoice , mulawAudio , alawAudio , mp1Audio , mp2Audio , mp3Audio , okiADPCM , voxADPCM , sx73Voice , sx96Voice , cdaStereo , cdaMono , pcm8Stereo , pcm8Mono , pcm16Stereo , pcm16Mono , pcm32Stereo , pcm32Mono , speexVoice , speexAudio , g729Audio , ilbcAudio , speexUltra , speexNarrow = speexVoice , speexWide = speexAudio , g723_4bit = g721ADPCM } |
Audio encoding formats. More... | |
enum | Format { raw , snd , riff , mpeg , wave } |
Audio container file format. More... | |
enum | DeviceMode { PLAY , RECORD , PLAYREC } |
Audio device access mode. More... | |
enum | Error { errSuccess = 0 , errReadLast , errNotOpened , errEndOfFile , errStartOfFile , errRateInvalid , errEncodingInvalid , errReadInterrupt , errWriteInterrupt , errReadFailure , errWriteFailure , errReadIncomplete , errWriteIncomplete , errRequestInvalid , errTOCFailed , errStatFailed , errInvalidTrack , errPlaybackFailed , errNotPlaying , errNoCodec } |
Audio error conditions. More... | |
typedef int16_t | snd16_t |
typedef int32_t | snd32_t |
typedef int16_t | Level |
typedef int16_t | Sample |
typedef int16_t * | Linear |
typedef unsigned long | timeout_t |
typedef unsigned char * | Encoded |
typedef enum Rate | Rate |
typedef enum Mode | Mode |
typedef enum Encoding | Encoding |
typedef enum Format | Format |
typedef enum DeviceMode | DeviceMode |
typedef enum Error | Error |
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static Level | tolevel (float dbm) |
Convert dbm power level to integer value (0-32768). | |
static float | todbm (Level power) |
Convert integer power levels to dbm. | |
static bool | hasDevice (unsigned device=0) |
Test for the presense of a specified (indexed) audio device. | |
static AudioDevice * | getDevice (unsigned device=0, DeviceMode mode=PLAY) |
Get a audio device object that can be used to play or record audio. | |
static const char * | getCodecPath (void) |
Get pathname to where loadable codec modules are stored. | |
static const char * | getMIME (Info &info) |
Get the mime descriptive type for a given Audio encoding description, usually retrieved from a newly opened audio file. | |
static const char * | getName (Encoding encoding) |
Get the short ascii description used for the given audio encoding type. | |
static const char * | getExtension (Encoding encoding) |
Get the preferred file extension name to use for a given audio encoding type. | |
static Encoding | getEncoding (const char *name) |
Get the audio encoding format that is specified by a short ascii name. | |
static Encoding | getStereo (Encoding encoding) |
Get the stereo encoding format associated with the given format. | |
static Encoding | getMono (Encoding encoding) |
Get the mono encoding format associated with the given format. | |
static bool | isLinear (Encoding encoding) |
Test if the audio encoding format is a linear one. | |
static bool | isBuffered (Encoding encoding) |
Test if the audio encoding format must be packetized (that is, has irregular sized frames) and must be processed only through buffered codecs. | |
static bool | isMono (Encoding encoding) |
Test if the audio encoding format is a mono format. | |
static bool | isStereo (Encoding encoding) |
Test if the audio encoding format is a stereo format. | |
static Rate | getRate (Encoding encoding) |
Return default sample rate associated with the specified audio encoding format. | |
static Rate | getRate (Encoding e, Rate request) |
Return optional rate setting effect. | |
static timeout_t | getFraming (Encoding encoding, timeout_t timeout=0) |
Return frame timing for an audio encoding format. | |
static timeout_t | getFraming (Info &info, timeout_t timeout=0) |
Return frame time for an audio source description. | |
static bool | isEndian (Encoding encoding) |
Test if the endian byte order of the encoding format is different from the machine's native byte order. | |
static bool | isEndian (Info &info) |
Test if the endian byte order of the audio source description is different from the machine's native byte order. | |
static bool | swapEndian (Encoding encoding, void *buffer, unsigned number) |
Optionally swap endian of audio data if the encoding format endian byte order is different from the machine's native endian. | |
static void | swapEncoded (Info &info, Encoded data, size_t bytes) |
Optionally swap endian of encoded audio data based on the audio encoding type, and relationship to native byte order. | |
static bool | swapEndian (Info &info, void *buffer, unsigned number) |
Optionally swap endian of audio data if the audio source description byte order is different from the machine's native endian byte order. | |
static Level | getImpulse (Encoding encoding, void *buffer, unsigned number) |
Get the energey impulse level of a frame of audio data. | |
static Level | getImpulse (Info &info, void *buffer, unsigned number=0) |
Get the energey impulse level of a frame of audio data. | |
static Level | getPeak (Encoding encoding, void *buffer, unsigned number) |
Get the peak (highest energy) level found in a frame of audio data. | |
static Level | getPeak (Info &info, void *buffer, unsigned number=0) |
Get the peak (highest energy) level found in a frame of audio data. | |
static void | toTimestamp (timeout_t duration, char *address, size_t size) |
Provide ascii timestamp representation of a timeout value. | |
static timeout_t | toTimeout (const char *timestamp) |
Convert ascii timestamp representation to a timeout number. | |
static int | getFrame (Encoding encoding, int samples=0) |
Returns the number of bytes in a sample frame for the given encoding type, rounded up to the nearest integer. | |
static int | getCount (Encoding encoding) |
Returns the number of samples in all channels for a frame in the given encoding. | |
static unsigned long | toSamples (Encoding encoding, size_t bytes) |
Compute byte counts of audio data into number of samples based on the audio encoding format used. | |
static unsigned long | toSamples (Info &info, size_t bytes) |
Compute byte counts of audio data into number of samples based on the audio source description used. | |
static size_t | toBytes (Info &info, unsigned long number) |
Compute the number of bytes a given number of samples in a given audio encoding will occupy. | |
static size_t | toBytes (Encoding encoding, unsigned long number) |
Compute the number of bytes a given number of samples in a given audio encoding will occupy. | |
static void | fill (unsigned char *address, int number, Encoding encoding) |
Fill an audio buffer with "empty" (silent) audio data, based on the audio encoding format. | |
static bool | loadPlugin (const char *path) |
Load a dso plugin (codec plugin), used internally... | |
static size_t | maxFramesize (Info &info) |
Maximum framesize for a given coding that may be needed to store a result. | |
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static const unsigned | ndata |
The AudioResample class is used to manage linear intropolation buffering for rate conversions.
linear intropolation and rate conversion.
ost::AudioResample::~AudioResample | ( | ) |
size_t ost::AudioResample::estimate | ( | size_t | count | ) |
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