Julius 4.2
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音声入力および振幅による音区間検出に関する定義 [詳細]
データ構造 | |
struct | DS_FILTER |
down sampling filter [詳細] | |
struct | DS_BUFFER |
down sampling data [詳細] | |
struct | ZEROCROSS |
Work area for zero-cross computation. [詳細] | |
マクロ定義 | |
#define | DEFAULT_WSTEP 1000 |
Default unit size of speech input segment in bytes. | |
#define | SUPPORTED_WAVEFILE_FORMAT "RAW(BE),WAV,AU,SND,NIST,ADPCM and more" |
String describing the list of supported wave file formats. | |
#define | ZMEANSAMPLES 48000 |
Number of samples from beggining of input to be used for computing the zero mean of source channel (for microphone/network input). | |
#define | DS_RBSIZE 512 |
Filter size. | |
#define | DS_BUFSIZE 256 |
Work area buffer size for x[]. | |
#define | DS_BUFSIZE_Y 512 |
Work area buffer size for y[]. | |
#define | ZC_UNDEF 2 |
Undefined mark for zerocross. | |
#define | ZC_POSITIVE 1 |
Positive mark used for zerocross. | |
#define | ZC_NEGATIVE -1 |
Negative mark used for zerocross. | |
列挙型 | |
enum | { INPUT_WAVEFORM, INPUT_VECTOR } |
Speech input type. | |
enum | { SP_RAWFILE, SP_MIC, SP_ADINNET, SP_MFCFILE, SP_NETAUDIO, SP_STDIN, SP_MFCMODULE } |
Speech input source. [詳細] | |
enum | { SP_INPUT_DEFAULT, SP_INPUT_ALSA, SP_INPUT_OSS, SP_INPUT_ESD, SP_INPUT_PULSEAUDIO } |
Input device. | |
関数 | |
boolean | adin_mic_standby (int freq, void *arg) |
Device initialization: check machine capability. | |
boolean | adin_mic_begin (char *pathname) |
Start recording. | |
boolean | adin_mic_end () |
Stop recording. | |
int | adin_mic_read (SP16 *buf, int sampnum) |
Read samples from device. | |
char * | adin_mic_input_name () |
Function to return current input source device name. | |
boolean | adin_alsa_standby (int freq, void *arg) |
Device initialization: check machine capability. | |
boolean | adin_alsa_begin (char *pathname) |
Start recording. | |
boolean | adin_alsa_end () |
Stop recording. | |
int | adin_alsa_read (SP16 *buf, int sampnum) |
Read samples from device. | |
char * | adin_alsa_input_name () |
Function to return current input source device name. | |
boolean | adin_oss_standby (int freq, void *arg) |
Device initialization: check device capability and open for recording. | |
boolean | adin_oss_begin (char *pathname) |
Start recording. | |
boolean | adin_oss_end () |
Stop recording. | |
int | adin_oss_read (SP16 *buf, int sampnum) |
Read samples from device. | |
char * | adin_oss_input_name () |
Function to return current input source device name. | |
boolean | adin_esd_standby (int freq, void *arg) |
Connection initialization: check connectivity and open for recording. | |
boolean | adin_esd_begin (char *pathname) |
Start recording. | |
boolean | adin_esd_end () |
Stop recording. | |
int | adin_esd_read (SP16 *buf, int sampnum) |
Read samples from the daemon. | |
char * | adin_esd_input_name () |
Function to return current input source device name. | |
boolean | adin_pulseaudio_standby (int freq, void *arg) |
Connection initialization: check connectivity and open for recording. | |
boolean | adin_pulseaudio_begin (char *pathname) |
Start recording. | |
boolean | adin_pulseaudio_end () |
Stop recording. | |
int | adin_pulseaudio_read (SP16 *buf, int sampnum) |
Read samples from device. | |
char * | adin_pulseaudio_input_name () |
Function to return current input source device name. | |
boolean | adin_netaudio_standby (int freq, void *arg) |
Connection initialization: check connectivity and open for recording. | |
boolean | adin_netaudio_begin (char *pathname) |
Start recording. | |
boolean | adin_netaudio_end () |
Stop recording. | |
int | adin_netaudio_read (SP16 *buf, int sampnum) |
Read samples from the daemon. | |
char * | adin_netaudio_input_name () |
Function to return current input source device name. | |
int | NA_standby (int, char *) |
Initialize NetAudio device. | |
void | NA_start () |
Begin recording. | |
void | NA_stop () |
Pause the recording. | |
int | NA_read (SP16 *buf, int sampnum) |
Read samples from NetAudio port. | |
boolean | adin_file_standby (int freq, void *arg) |
Initialization: if listfile is specified, open it here. | |
boolean | adin_file_begin (char *pathname) |
Begin reading audio data from a file. | |
int | adin_file_read (SP16 *buf, int sampnum) |
Try to read sampnum samples and returns actual sample num recorded. | |
boolean | adin_file_end () |
End recording. | |
boolean | adin_stdin_standby (int freq, void *arg) |
Initialization for speech input via stdin. | |
boolean | adin_stdin_begin (char *pathname) |
Begin reading audio data from stdin. | |
int | adin_stdin_read (SP16 *buf, int sampnum) |
Try to read sampnum samples and returns actual sample num recorded. | |
char * | adin_file_get_current_filename () |
A tiny function to get current input raw speech file name. | |
char * | adin_stdin_input_name () |
A tiny function to get current input raw speech file name. | |
boolean | adin_sndfile_standby (int freq, void *arg) |
Initialization: if listfile is specified, open it here. | |
boolean | adin_sndfile_begin (char *pathname) |
Begin reading audio data from a file. | |
int | adin_sndfile_read (SP16 *buf, int sampnum) |
Try to read sampnum samples and returns actual sample num recorded. | |
boolean | adin_sndfile_end () |
End recording. | |
char * | adin_sndfile_get_current_filename () |
A tiny function to get current input raw speech file name. | |
boolean | adin_tcpip_standby (int freq, void *arg) |
Initialize as adinnet server: prepare to become server. | |
boolean | adin_tcpip_begin (char *pathname) |
Wait for connection from adinnet client and begin audio input stream. | |
boolean | adin_tcpip_end () |
End recording. | |
int | adin_tcpip_read (SP16 *buf, int sampnum) |
Try to read sampnum samples and returns actual sample num recorded. | |
boolean | adin_tcpip_send_pause () |
Tell the adinnet client to pause transfer. | |
boolean | adin_tcpip_send_terminate () |
Tell the adinnet client to terminate transfer. | |
boolean | adin_tcpip_send_resume () |
Tell the adinnet client to resume the paused transfer. | |
char * | adin_tcpip_input_name () |
Function to return current input source device name. | |
void | init_count_zc_e (ZEROCROSS *zc, int length) |
Allocate buffers for zerocross counting. | |
void | reset_count_zc_e (ZEROCROSS *zc, int c_trigger, int c_length, int c_offset) |
Initialize all parameters and buffers for zero-cross counting. | |
void | free_count_zc_e (ZEROCROSS *zc) |
End procedure: free all buffers. | |
int | count_zc_e (ZEROCROSS *zc, SP16 *buf, int step) |
Adding buf[0..step-1] to the cycle buffer and update the count of zero cross. | |
void | zc_copy_buffer (ZEROCROSS *zc, SP16 *newbuf, int *len) |
Flush samples in the current cycle buffer. | |
void | zmean_reset () |
Reset status. | |
void | sub_zmean (SP16 *speech, int samplenum) |
Remove DC offset. | |
DS_BUFFER * | ds48to16_new () |
Setup for down sampling. | |
void | ds48to16_free (DS_BUFFER *ds) |
Free the down sampling buffer. | |
int | ds48to16 (SP16 *dst, SP16 *src, int srclen, int maxdstlen, DS_BUFFER *ds) |
Perform down sampling of input samples to 1/3. |
音声入力および振幅による音区間検出に関する定義
このファイルには, さまざまなソースからの音声入力処理と音声区間の検出 に関連するいくつかの定義が含まれています.
adin.h で定義されています。
#define SUPPORTED_WAVEFILE_FORMAT "RAW(BE),WAV,AU,SND,NIST,ADPCM and more" |
String describing the list of supported wave file formats.
It depends on HAVE_LIBSNDFILE.
anonymous enum |
boolean adin_mic_standby | ( | int | sfreq, |
void * | dummy | ||
) |
Device initialization: check machine capability.
sfreq | [in] required sampling frequency. |
arg | [in] a dummy data |
Device initialization: check machine capability.
sfreq | [in] required sampling frequency. |
dummy | [in] a dummy data |
Device initialization: check machine capability.
sfreq | [in] required sampling frequency. |
arg | [in] a dummy data |
adin_mic_darwin_coreaudio.c の 288 行で定義されています。
参照元 adin_select().
boolean adin_mic_begin | ( | char * | arg | ) |
Start recording.
pathname | [in] path name to open or NULL for default |
adin_mic_darwin_coreaudio.c の 578 行で定義されています。
参照元 adin_select().
boolean adin_mic_end | ( | ) |
Stop recording.
adin_mic_darwin_coreaudio.c の 579 行で定義されています。
参照元 adin_select().
int adin_mic_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Read samples from device.
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least one sample can be obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least one sample can be obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least some samples are obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_mic_freebsd.c の 255 行で定義されています。
参照元 adin_select().
char* adin_mic_input_name | ( | ) |
Function to return current input source device name.
adin_mic_darwin_coreaudio.c の 675 行で定義されています。
参照元 adin_select().
boolean adin_alsa_standby | ( | int | sfreq, |
void * | dummy | ||
) |
Device initialization: check machine capability.
sfreq | [in] required sampling frequency. |
dummy | [in] a dummy data |
adin_mic_linux_alsa.c の 164 行で定義されています。
参照元 adin_mic_standby(), と adin_select().
boolean adin_alsa_begin | ( | char * | pathname | ) |
Start recording.
pathname | [in] device name to open or NULL for default |
adin_mic_linux_alsa.c の 430 行で定義されています。
参照元 adin_mic_begin(), と adin_select().
boolean adin_alsa_end | ( | ) |
Stop recording.
adin_mic_linux_alsa.c の 499 行で定義されています。
参照元 adin_mic_end(), と adin_select().
int adin_alsa_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Read samples from device.
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least one sample can be obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_mic_linux_alsa.c の 523 行で定義されています。
参照元 adin_mic_read(), と adin_select().
char* adin_alsa_input_name | ( | ) |
Function to return current input source device name.
adin_mic_linux_alsa.c の 613 行で定義されています。
参照元 adin_mic_input_name(), と adin_select().
boolean adin_oss_standby | ( | int | sfreq, |
void * | dummy | ||
) |
Device initialization: check device capability and open for recording.
sfreq | [in] required sampling frequency. |
dummy | [in] a dummy data |
adin_mic_linux_oss.c の 118 行で定義されています。
参照元 adin_mic_standby(), と adin_select().
boolean adin_oss_begin | ( | char * | pathname | ) |
Start recording.
pathname | [in] path name to open or NULL for default |
adin_mic_linux_oss.c の 358 行で定義されています。
参照元 adin_mic_begin(), と adin_select().
boolean adin_oss_end | ( | ) |
Stop recording.
adin_mic_linux_oss.c の 394 行で定義されています。
参照元 adin_mic_end(), と adin_select().
int adin_oss_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Read samples from device.
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block at most MAXPOLLINTERVAL msec, until at least one sample can be obtained. If no data has been obtained after waiting for MAXPOLLINTERVAL msec, returns 0.
When stereo input, only left channel will be used.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_mic_linux_oss.c の 418 行で定義されています。
参照元 adin_mic_read(), と adin_select().
char* adin_oss_input_name | ( | ) |
Function to return current input source device name.
adin_mic_linux_oss.c の 485 行で定義されています。
参照元 adin_mic_input_name(), と adin_select().
boolean adin_esd_standby | ( | int | sfreq, |
void * | dummy | ||
) |
Connection initialization: check connectivity and open for recording.
sfreq | [in] required sampling frequency |
dummy | [in] a dummy data |
adin_esd.c の 54 行で定義されています。
参照元 adin_mic_standby(), と adin_select().
boolean adin_esd_begin | ( | char * | pathname | ) |
Start recording.
pathname is dummy.
pathname | [in] path name to open or NULL for default |
adin_esd.c の 86 行で定義されています。
参照元 adin_mic_begin(), と adin_select().
boolean adin_esd_end | ( | ) |
Stop recording.
adin_esd.c の 97 行で定義されています。
参照元 adin_mic_end(), と adin_select().
int adin_esd_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Read samples from the daemon.
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least one sample can be obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_esd.c の 115 行で定義されています。
参照元 adin_mic_read(), と adin_select().
char* adin_esd_input_name | ( | ) |
Function to return current input source device name.
adin_esd.c の 146 行で定義されています。
参照元 adin_mic_input_name(), と adin_select().
boolean adin_pulseaudio_standby | ( | int | sfreq, |
void * | dummy | ||
) |
Connection initialization: check connectivity and open for recording.
sfreq | [in] required sampling frequency |
dummy | [in] a dummy data |
adin_pulseaudio.c の 54 行で定義されています。
参照元 adin_mic_standby(), と adin_select().
boolean adin_pulseaudio_begin | ( | char * | arg | ) |
Start recording.
pathname is dummy.
arg | [in] argument |
adin_pulseaudio.c の 74 行で定義されています。
参照元 adin_mic_begin(), と adin_select().
boolean adin_pulseaudio_end | ( | ) |
Stop recording.
adin_pulseaudio.c の 101 行で定義されています。
参照元 adin_mic_end(), と adin_select().
int adin_pulseaudio_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Read samples from device.
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least one sample was obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_pulseaudio.c の 128 行で定義されています。
参照元 adin_mic_read(), と adin_select().
char* adin_pulseaudio_input_name | ( | ) |
Function to return current input source device name.
adin_pulseaudio.c の 160 行で定義されています。
参照元 adin_mic_input_name(), と adin_select().
boolean adin_netaudio_standby | ( | int | sfreq, |
void * | arg | ||
) |
Connection initialization: check connectivity and open for recording.
sfreq | [in] required sampling frequency |
arg | [in] server device name string to connect |
adin_netaudio.c の 60 行で定義されています。
参照元 adin_select().
boolean adin_netaudio_begin | ( | char * | pathname | ) |
Start recording.
pathname | [in] path name to open or NULL for default |
adin_netaudio.c の 75 行で定義されています。
参照元 adin_select().
boolean adin_netaudio_end | ( | ) |
Stop recording.
adin_netaudio.c の 87 行で定義されています。
参照元 adin_select().
int adin_netaudio_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Read samples from the daemon.
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least one sample can be obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_netaudio.c の 106 行で定義されています。
参照元 adin_select().
char* adin_netaudio_input_name | ( | ) |
Function to return current input source device name.
adin_netaudio.c の 125 行で定義されています。
参照元 adin_select().
int NA_standby | ( | int | sfreq, |
char * | server_devname | ||
) |
Initialize NetAudio device.
sfreq | [in] sampling frequency |
server_devname | [in] server host name |
int NA_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Read samples from NetAudio port.
Try to read sampnum samples and returns actual number of recorded samples currently available. This function will block until at least some samples are obtained.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
参照元 adin_netaudio_read().
boolean adin_file_standby | ( | int | freq, |
void * | arg | ||
) |
Initialization: if listfile is specified, open it here.
freq | [in] required sampling frequency. |
arg | [in] file name of listfile, or NULL if not use |
adin_file.c の 326 行で定義されています。
参照元 adin_select().
boolean adin_file_begin | ( | char * | filename | ) |
Begin reading audio data from a file.
If listfile was specified in adin_file_standby(), the next filename will be read from the listfile. Otherwise, the filename will be obtained from stdin. Then the file will be opened here.
filename | [in] file name to open or NULL for prompt |
adin_file.c の 358 行で定義されています。
参照元 adin_select().
int adin_file_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Try to read sampnum samples and returns actual sample num recorded.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_file.c の 410 行で定義されています。
参照元 adin_select().
boolean adin_file_end | ( | ) |
End recording.
adin_file.c の 473 行で定義されています。
参照元 adin_select().
boolean adin_stdin_standby | ( | int | freq, |
void * | arg | ||
) |
Initialization for speech input via stdin.
freq | [in] required sampling frequency. |
arg | dummy, ignored |
adin_file.c の 489 行で定義されています。
参照元 adin_select().
boolean adin_stdin_begin | ( | char * | pathname | ) |
Begin reading audio data from stdin.
pathname | [in] dummy |
adin_file.c の 504 行で定義されています。
参照元 adin_select().
int adin_stdin_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Try to read sampnum samples and returns actual sample num recorded.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_file.c の 529 行で定義されています。
参照元 adin_select().
char* adin_file_get_current_filename | ( | ) |
A tiny function to get current input raw speech file name.
adin_file.c の 585 行で定義されています。
参照元 adin_select().
char* adin_stdin_input_name | ( | ) |
A tiny function to get current input raw speech file name.
adin_file.c の 597 行で定義されています。
参照元 adin_select().
boolean adin_sndfile_standby | ( | int | freq, |
void * | arg | ||
) |
Initialization: if listfile is specified, open it here.
Else, just store the required frequency.
freq | [in] required sampling frequency |
arg | [in] file name of listfile, or NULL if not use |
adin_sndfile.c の 202 行で定義されています。
参照元 adin_select().
boolean adin_sndfile_begin | ( | char * | filename | ) |
Begin reading audio data from a file.
If listfile was specified in adin_sndfile_standby(), the next filename will be read from the listfile. Otherwise, the filename will be obtained from stdin. Then the file will be opened here.
filename | [in] file name to open or NULL for prompt |
adin_sndfile.c の 287 行で定義されています。
参照元 adin_select().
int adin_sndfile_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Try to read sampnum samples and returns actual sample num recorded.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_sndfile.c の 341 行で定義されています。
参照元 adin_select().
boolean adin_sndfile_end | ( | ) |
End recording.
adin_sndfile.c の 362 行で定義されています。
参照元 adin_select().
char* adin_sndfile_get_current_filename | ( | ) |
A tiny function to get current input raw speech file name.
adin_sndfile.c の 381 行で定義されています。
参照元 adin_select().
boolean adin_tcpip_standby | ( | int | freq, |
void * | port_str | ||
) |
Initialize as adinnet server: prepare to become server.
freq | [in] required sampling frequency |
port_str | [in] port number in string |
adin_tcpip.c の 78 行で定義されています。
参照元 adin_select().
boolean adin_tcpip_begin | ( | char * | pathname | ) |
Wait for connection from adinnet client and begin audio input stream.
pathname | [in] path name to open or NULL for default |
adin_tcpip.c の 102 行で定義されています。
参照元 adin_select().
boolean adin_tcpip_end | ( | ) |
End recording.
If last input segment was segmented by end of connection, close socket and wait for next connection. Otherwise, it means that the last input segment was segmented by either end-of-segment signal from client side or speech detection function in server side, so just wait for a next input in the current socket.
adin_tcpip.c の 151 行で定義されています。
参照元 adin_select().
int adin_tcpip_read | ( | SP16 * | buf, |
int | sampnum | ||
) |
Try to read sampnum samples and returns actual sample num recorded.
This function will not block.
If data of zero length has been received, it is treated as a end-of-segment marker from the client. If data below zero length has been received, it means the client has finished the overall input stream transmission and want to disconnect.
buf | [out] samples obtained in this function |
sampnum | [in] wanted number of samples to be read |
adin_tcpip.c の 182 行で定義されています。
参照元 adin_select().
boolean adin_tcpip_send_pause | ( | ) |
Tell the adinnet client to pause transfer.
adin_tcpip.c の 226 行で定義されています。
参照元 adin_select().
boolean adin_tcpip_send_terminate | ( | ) |
Tell the adinnet client to terminate transfer.
adin_tcpip.c の 292 行で定義されています。
参照元 adin_select().
boolean adin_tcpip_send_resume | ( | ) |
Tell the adinnet client to resume the paused transfer.
adin_tcpip.c の 244 行で定義されています。
参照元 adin_select().
char* adin_tcpip_input_name | ( | ) |
Function to return current input source device name.
adin_tcpip.c の 312 行で定義されています。
参照元 adin_select().
void init_count_zc_e | ( | ZEROCROSS * | zc, |
int | length | ||
) |
Allocate buffers for zerocross counting.
zc | [i/o] zerocross work area |
length | [in] Cycle buffer size = Number of samples to hold |
参照元 adin_setup_param(), と reset_count_zc_e().
void reset_count_zc_e | ( | ZEROCROSS * | zc, |
int | c_trigger, | ||
int | c_length, | ||
int | c_offset | ||
) |
Initialize all parameters and buffers for zero-cross counting.
zc | [i/o] zerocross work area |
c_trigger | [in] Tgigger level threshold |
c_length | [in] Cycle buffer size = Number of samples to hold |
c_offset | [in] Static DC offset of input data |
参照元 adin_cut().
void free_count_zc_e | ( | ZEROCROSS * | zc | ) |
End procedure: free all buffers.
zc | [i/o] zerocross work area |
参照元 adin_free_param(), と reset_count_zc_e().
Adding buf[0..step-1] to the cycle buffer and update the count of zero cross.
Also swap them with the oldest ones in the cycle buffer. Also get the maximum level in the cycle buffer.
zc | [i/o] zerocross work area |
buf | [I/O] new samples, will be swapped by old samples when returned. |
step | [in] length of above. |
参照元 adin_cut().
Flush samples in the current cycle buffer.
zc | [i/o] zerocross work area |
newbuf | [out] the samples in teh cycle buffer will be written here. |
len | [out] length of above. |
参照元 adin_cut().
void sub_zmean | ( | SP16 * | speech, |
int | samplenum | ||
) |
Remove DC offset.
The DC offset is estimated by the first samples after zmean_reset() was called. If the first input segment is longer than ZMEANSAMPLES, the whole input is used to estimate the zero mean. Otherwise, the zero mean will continue to be updated until the read length exceed ZMEANSAMPLES.
speech | [I/O] input speech data, will be subtracted by DC offset. |
samplenum | [in] length of above. |
参照元 adin_cut().
DS_BUFFER* ds48to16_new | ( | ) |
Setup for down sampling.
ds48to16.c の 269 行で定義されています。
参照元 adin_setup_all().
void ds48to16_free | ( | DS_BUFFER * | ds | ) |
Free the down sampling buffer.
ds | [i/o] down sampling buffer to free |
ds48to16.c の 307 行で定義されています。
参照元 adin_free_param().
Perform down sampling of input samples to 1/3.
dst | [out] store the resulting samples |
src | [in] input samples |
srclen | [in] number of input samples |
maxdstlen | [in] maximum length of dst |
ds | [i/o] down sampling buffer |
ds48to16.c の 332 行で定義されています。
参照元 adin_cut().