diff options
Diffstat (limited to 'lib/libsamplerate/libsamplerate-0.2.2/examples')
5 files changed, 1661 insertions, 0 deletions
diff --git a/lib/libsamplerate/libsamplerate-0.2.2/examples/CMakeLists.txt b/lib/libsamplerate/libsamplerate-0.2.2/examples/CMakeLists.txt new file mode 100755 index 00000000..49d258c8 --- /dev/null +++ b/lib/libsamplerate/libsamplerate-0.2.2/examples/CMakeLists.txt @@ -0,0 +1,31 @@ +find_package(ALSA) +set(HAVE_ALSA ${ALSA_FOUND} PARENT_SCOPE) +# ALSA::ALSA target is exported since CMake >= 3.12, create it for +# old CMake versions +if(ALSA_FOUND) + if(NOT TARGET ALSA::ALSA) + add_library(ALSA::ALSA UNKNOWN IMPORTED) + set_target_properties(ALSA::ALSA PROPERTIES + INTERFACE_INCLUDE_DIRECTORIES "${ALSA_INCLUDE_DIRS}" + IMPORTED_LOCATION "${ALSA_LIBRARIES}") + endif() +endif() + +add_executable(timewarp-file timewarp-file.c) +target_link_libraries(timewarp-file + PRIVATE + samplerate + $<$<BOOL:${SndFile_FOUND}>:${SNDFILE_TARGET}>) + +add_executable(varispeed-play varispeed-play.c audio_out.c audio_out.h) +target_link_libraries(varispeed-play + PRIVATE + samplerate + $<$<BOOL:${SndFile_FOUND}>:${SNDFILE_TARGET}>) +if(WIN32) + target_link_libraries(varispeed-play PRIVATE winmm) +elseif(APPLE) + target_link_libraries(varispeed-play PRIVATE "-framework CoreAudio") +elseif(ALSA_FOUND) + target_link_libraries(varispeed-play PRIVATE ALSA::ALSA) +endif() diff --git a/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.c b/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.c new file mode 100755 index 00000000..e7c28cc5 --- /dev/null +++ b/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.c @@ -0,0 +1,1125 @@ +/* +** Copyright (c) 1999-2016, Erik de Castro Lopo <erikd@mega-nerd.com> +** All rights reserved. +** +** This code is released under 2-clause BSD license. Please see the +** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#ifdef HAVE_UNISTD_H +#include <unistd.h> +#endif + +#ifdef _WIN32 +#ifndef WIN32_LEAN_AN_MEAN +#define WIN32_LEAN_AN_MEAN +#endif +#include <windows.h> +#include <mmsystem.h> +#endif + +#include "audio_out.h" + +#if (HAVE_SNDFILE) + +#include <math.h> + +#include <sndfile.h> + +#define BUFFER_LEN (2048) + +#define MAKE_MAGIC(a,b,c,d,e,f,g,h) \ + ((a) + ((b) << 1) + ((c) << 2) + ((d) << 3) + ((e) << 4) + ((f) << 5) + ((g) << 6) + ((h) << 7)) + +/*------------------------------------------------------------------------------ +** Linux (ALSA and OSS) functions for playing a sound. +*/ + +#if defined (__linux__) + +#if HAVE_ALSA + +#define ALSA_PCM_NEW_HW_PARAMS_API +#define ALSA_PCM_NEW_SW_PARAMS_API + +#include <alsa/asoundlib.h> +#include <sys/time.h> + +#define ALSA_MAGIC MAKE_MAGIC ('L', 'n', 'x', '-', 'A', 'L', 'S', 'A') + +typedef struct AUDIO_OUT +{ int magic ; + snd_pcm_t * dev ; + int channels ; +} ALSA_AUDIO_OUT ; + +static int alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels) ; + +static AUDIO_OUT * +alsa_open (int channels, unsigned samplerate) +{ ALSA_AUDIO_OUT *alsa_out ; + const char * device = "default" ; + snd_pcm_hw_params_t *hw_params ; + snd_pcm_uframes_t buffer_size ; + snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ; + snd_pcm_sw_params_t *sw_params ; + + int err ; + + alsa_period_size = 1024 ; + alsa_buffer_frames = 4 * alsa_period_size ; + + if ((alsa_out = calloc (1, sizeof (ALSA_AUDIO_OUT))) == NULL) + { perror ("alsa_open : malloc ") ; + exit (1) ; + } ; + + alsa_out->magic = ALSA_MAGIC ; + alsa_out->channels = channels ; + + if ((err = snd_pcm_open (&alsa_out->dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) + { fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ; + goto catch_error ; + } ; + + snd_pcm_nonblock (alsa_out->dev, 0) ; + + if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) + { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params_any (alsa_out->dev, hw_params)) < 0) + { fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params_set_access (alsa_out->dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) + { fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params_set_format (alsa_out->dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0) + { fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params_set_rate_near (alsa_out->dev, hw_params, &samplerate, 0)) < 0) + { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params_set_channels (alsa_out->dev, hw_params, channels)) < 0) + { fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_out->dev, hw_params, &alsa_buffer_frames)) < 0) + { fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params_set_period_size_near (alsa_out->dev, hw_params, &alsa_period_size, 0)) < 0) + { fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_hw_params (alsa_out->dev, hw_params)) < 0) + { fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + /* extra check: if we have only one period, this code won't work */ + snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ; + snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ; + if (alsa_period_size == buffer_size) + { fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ; + goto catch_error ; + } ; + + snd_pcm_hw_params_free (hw_params) ; + + if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0) + { fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_sw_params_current (alsa_out->dev, sw_params)) != 0) + { fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ; + goto catch_error ; + } ; + + /* note: set start threshold to delay start until the ring buffer is full */ + snd_pcm_sw_params_current (alsa_out->dev, sw_params) ; + + if ((err = snd_pcm_sw_params_set_start_threshold (alsa_out->dev, sw_params, buffer_size)) < 0) + { fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ; + goto catch_error ; + } ; + + if ((err = snd_pcm_sw_params (alsa_out->dev, sw_params)) != 0) + { fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ; + goto catch_error ; + } ; + + snd_pcm_sw_params_free (sw_params) ; + + snd_pcm_reset (alsa_out->dev) ; + +catch_error : + + if (err < 0 && alsa_out->dev != NULL) + { snd_pcm_close (alsa_out->dev) ; + return NULL ; + } ; + + return (AUDIO_OUT *) alsa_out ; +} /* alsa_open */ + +static void +alsa_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) +{ static float buffer [BUFFER_LEN] ; + ALSA_AUDIO_OUT *alsa_out ; + int read_frames ; + + if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL) + { printf ("alsa_close : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (alsa_out->magic != ALSA_MAGIC) + { printf ("alsa_close : Bad magic number.\n") ; + return ; + } ; + + while ((read_frames = callback (callback_data, buffer, BUFFER_LEN / alsa_out->channels))) + alsa_write_float (alsa_out->dev, buffer, read_frames, alsa_out->channels) ; + + return ; +} /* alsa_play */ + +static int +alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels) +{ static int epipe_count = 0 ; + + int total = 0 ; + int retval ; + + if (epipe_count > 0) + epipe_count -- ; + + while (total < frames) + { retval = snd_pcm_writei (alsa_dev, data + total * channels, frames - total) ; + + if (retval >= 0) + { total += retval ; + if (total == frames) + return total ; + + continue ; + } ; + + switch (retval) + { case -EAGAIN : + puts ("alsa_write_float: EAGAIN") ; + continue ; + break ; + + case -EPIPE : + if (epipe_count > 0) + { printf ("alsa_write_float: EPIPE %d\n", epipe_count) ; + if (epipe_count > 140) + return retval ; + } ; + epipe_count += 100 ; + +#if 0 + if (0) + { snd_pcm_status_t *status ; + + snd_pcm_status_alloca (&status) ; + if ((retval = snd_pcm_status (alsa_dev, status)) < 0) + fprintf (stderr, "alsa_out: xrun. can't determine length\n") ; + else if (snd_pcm_status_get_state (status) == SND_PCM_STATE_XRUN) + { struct timeval now, diff, tstamp ; + + gettimeofday (&now, 0) ; + snd_pcm_status_get_trigger_tstamp (status, &tstamp) ; + timersub (&now, &tstamp, &diff) ; + + fprintf (stderr, "alsa_write_float xrun: of at least %.3f msecs. resetting stream\n", + diff.tv_sec * 1000 + diff.tv_usec / 1000.0) ; + } + else + fprintf (stderr, "alsa_write_float: xrun. can't determine length\n") ; + } ; +#endif + + snd_pcm_prepare (alsa_dev) ; + break ; + + case -EBADFD : + fprintf (stderr, "alsa_write_float: Bad PCM state.n") ; + return 0 ; + break ; + + case -ESTRPIPE : + fprintf (stderr, "alsa_write_float: Suspend event.n") ; + return 0 ; + break ; + + case -EIO : + puts ("alsa_write_float: EIO") ; + return 0 ; + + default : + fprintf (stderr, "alsa_write_float: retval = %d\n", retval) ; + return 0 ; + break ; + } ; /* switch */ + } ; /* while */ + + return total ; +} /* alsa_write_float */ + +static void +alsa_close (AUDIO_OUT *audio_out) +{ ALSA_AUDIO_OUT *alsa_out ; + + if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL) + { printf ("alsa_close : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (alsa_out->magic != ALSA_MAGIC) + { printf ("alsa_close : Bad magic number.\n") ; + return ; + } ; + + memset (alsa_out, 0, sizeof (ALSA_AUDIO_OUT)) ; + + free (alsa_out) ; + + return ; +} /* alsa_close */ + +#endif /* HAVE_ALSA */ + +#include <fcntl.h> +#include <sys/ioctl.h> +#include <sys/soundcard.h> + +#define OSS_MAGIC MAKE_MAGIC ('L', 'i', 'n', 'u', 'x', 'O', 'S', 'S') + +typedef struct +{ int magic ; + int fd ; + int channels ; +} OSS_AUDIO_OUT ; + +static AUDIO_OUT *opensoundsys_open (int channels, int samplerate) ; +static void opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ; +static void opensoundsys_close (AUDIO_OUT *audio_out) ; + + +static AUDIO_OUT * +opensoundsys_open (int channels, int samplerate) +{ OSS_AUDIO_OUT *opensoundsys_out ; + int stereo, fmt, error ; + + if ((opensoundsys_out = calloc (1, sizeof (OSS_AUDIO_OUT))) == NULL) + { perror ("opensoundsys_open : malloc ") ; + exit (1) ; + } ; + + opensoundsys_out->magic = OSS_MAGIC ; + opensoundsys_out->channels = channels ; + + if ((opensoundsys_out->fd = open ("/dev/dsp", O_WRONLY, 0)) == -1) + { perror ("opensoundsys_open : open ") ; + exit (1) ; + } ; + + stereo = 0 ; + if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_STEREO, &stereo) == -1) + { /* Fatal error */ + perror ("opensoundsys_open : stereo ") ; + exit (1) ; + } ; + + if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_RESET, 0)) + { perror ("opensoundsys_open : reset ") ; + exit (1) ; + } ; + + fmt = CPU_IS_BIG_ENDIAN ? AFMT_S16_BE : AFMT_S16_LE ; + if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_SETFMT, &fmt) != 0) + { perror ("opensoundsys_open_dsp_device : set format ") ; + exit (1) ; + } ; + + if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_CHANNELS, &channels)) != 0) + { perror ("opensoundsys_open : channels ") ; + exit (1) ; + } ; + + if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SPEED, &samplerate)) != 0) + { perror ("opensoundsys_open : sample rate ") ; + exit (1) ; + } ; + + if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SYNC, 0)) != 0) + { perror ("opensoundsys_open : sync ") ; + exit (1) ; + } ; + + return (AUDIO_OUT*) opensoundsys_out ; +} /* opensoundsys_open */ + +static void +opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) +{ OSS_AUDIO_OUT *opensoundsys_out ; + static float float_buffer [BUFFER_LEN] ; + static short buffer [BUFFER_LEN] ; + int k, read_frames ; + + if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL) + { printf ("opensoundsys_play : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (opensoundsys_out->magic != OSS_MAGIC) + { printf ("opensoundsys_play : Bad magic number.\n") ; + return ; + } ; + + while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / opensoundsys_out->channels))) + { for (k = 0 ; k < read_frames * opensoundsys_out->channels ; k++) + buffer [k] = lrint (32767.0 * float_buffer [k]) ; + if (write (opensoundsys_out->fd, buffer, read_frames * opensoundsys_out->channels * sizeof (short))) {} + } ; + + return ; +} /* opensoundsys_play */ + +static void +opensoundsys_close (AUDIO_OUT *audio_out) +{ OSS_AUDIO_OUT *opensoundsys_out ; + + if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL) + { printf ("opensoundsys_close : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (opensoundsys_out->magic != OSS_MAGIC) + { printf ("opensoundsys_close : Bad magic number.\n") ; + return ; + } ; + + memset (opensoundsys_out, 0, sizeof (OSS_AUDIO_OUT)) ; + + free (opensoundsys_out) ; + + return ; +} /* opensoundsys_close */ + +#endif /* __linux__ */ + +/*------------------------------------------------------------------------------ +** Mac OS X functions for playing a sound. +*/ + +#if (defined (__MACH__) && defined (__APPLE__)) /* MacOSX */ + +#include <AvailabilityMacros.h> +#include <CoreAudio/AudioHardware.h> +#ifndef MAC_OS_VERSION_12_0 +#define kAudioObjectPropertyElementMain kAudioObjectPropertyElementMaster +#endif + +#define MACOSX_MAGIC MAKE_MAGIC ('M', 'a', 'c', ' ', 'O', 'S', ' ', 'X') + +typedef struct +{ int magic ; + AudioStreamBasicDescription format ; + + UInt32 buf_size ; + AudioDeviceID device ; + + int channels ; + int samplerate ; + int buffer_size ; + int done_playing ; + + get_audio_callback_t callback ; + + void *callback_data ; + + AudioDeviceIOProcID ioprocid; + +} MACOSX_AUDIO_OUT ; + +static AUDIO_OUT *macosx_open (int channels, int samplerate) ; +static void macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ; +static void macosx_close (AUDIO_OUT *audio_out) ; + +static OSStatus +macosx_audio_out_callback (AudioDeviceID device, const AudioTimeStamp* current_time, + const AudioBufferList* data_in, const AudioTimeStamp* time_in, + AudioBufferList* data_out, const AudioTimeStamp* time_out, void* client_data) ; + + +static AUDIO_OUT * +macosx_open (int channels, int samplerate) +{ MACOSX_AUDIO_OUT *macosx_out ; + OSStatus err ; + UInt32 count ; + AudioObjectPropertyAddress propertyAddress ; + + if ((macosx_out = calloc (1, sizeof (MACOSX_AUDIO_OUT))) == NULL) + { perror ("macosx_open : malloc ") ; + exit (1) ; + } ; + + macosx_out->magic = MACOSX_MAGIC ; + macosx_out->channels = channels ; + macosx_out->samplerate = samplerate ; + + macosx_out->device = kAudioDeviceUnknown ; + + /* get the default output device for the HAL */ + propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; + propertyAddress.mScope = kAudioDevicePropertyScopeOutput; + propertyAddress.mElement = kAudioObjectPropertyElementMain; + + count = sizeof (AudioDeviceID) ; + if ((err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, + &count, &(macosx_out->device))) != noErr) + { printf ("AudioObjectGetPropertyData (kAudioHardwarePropertyDefaultOutputDevice) failed.\n") ; + free (macosx_out) ; + return NULL ; + } ; + + /* get the buffersize that the default device uses for IO */ + count = sizeof (UInt32) ; + propertyAddress.mSelector = kAudioDevicePropertyBufferSize ; + if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL, + &count, &(macosx_out->buffer_size))) != noErr) + { printf ("AudioObjectGetPropertyData (kAudioDevicePropertyBufferSize) (AudioDeviceGetProperty) failed.\n") ; + free (macosx_out) ; + return NULL ; + } ; + + /* get a description of the data format used by the default device */ + count = sizeof (AudioStreamBasicDescription) ; + propertyAddress.mSelector = kAudioDevicePropertyStreamFormat ; + if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL, + &count, &(macosx_out->format))) != noErr) + { printf ("AudioObjectGetPropertyData (kAudioDevicePropertyStreamFormat) failed.\n") ; + free (macosx_out) ; + return NULL ; + } ; + + macosx_out->format.mSampleRate = samplerate ; + macosx_out->format.mChannelsPerFrame = channels ; + propertyAddress.mSelector = kAudioDevicePropertyStreamFormat ; + count = sizeof (AudioStreamBasicDescription) ; + if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL, + &count, &(macosx_out->format))) != noErr) + { printf ("AudioObjectGetPropertyData (kAudioDevicePropertyStreamFormat) failed.\n") ; + free (macosx_out) ; + return NULL ; + } ; + + /* we want linear pcm */ + if (macosx_out->format.mFormatID != kAudioFormatLinearPCM) + { free (macosx_out) ; + return NULL ; + } ; + + macosx_out->done_playing = 0 ; + + /* Fire off the device. */ + if ((err = AudioDeviceCreateIOProcID (macosx_out->device, macosx_audio_out_callback, + (void *) macosx_out, &macosx_out->ioprocid)) != noErr) + { printf ("AudioDeviceAddIOProc failed.\n") ; + free (macosx_out) ; + return NULL ; + } ; + + return (AUDIO_OUT *) macosx_out ; +} /* macosx_open */ + +static void +macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) +{ MACOSX_AUDIO_OUT *macosx_out ; + OSStatus err ; + + if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL) + { printf ("macosx_play : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (macosx_out->magic != MACOSX_MAGIC) + { printf ("macosx_play : Bad magic number.\n") ; + return ; + } ; + + /* Set the callback function and callback data. */ + macosx_out->callback = callback ; + macosx_out->callback_data = callback_data ; + + err = AudioDeviceStart (macosx_out->device, macosx_audio_out_callback) ; + if (err != noErr) + printf ("AudioDeviceStart failed.\n") ; + + while (macosx_out->done_playing == SF_FALSE) + usleep (10 * 1000) ; /* 10 000 milliseconds. */ + + return ; +} /* macosx_play */ + +static void +macosx_close (AUDIO_OUT *audio_out) +{ MACOSX_AUDIO_OUT *macosx_out ; + OSStatus err ; + + if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL) + { printf ("macosx_close : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (macosx_out->magic != MACOSX_MAGIC) + { printf ("macosx_close : Bad magic number.\n") ; + return ; + } ; + + + if ((err = AudioDeviceStop (macosx_out->device, macosx_audio_out_callback)) != noErr) + { printf ("AudioDeviceStop failed.\n") ; + return ; + } ; + + err = AudioDeviceDestroyIOProcID(macosx_out->device, + macosx_out->ioprocid); + if (err != noErr) + { printf ("AudioDeviceRemoveIOProc failed.\n") ; + return ; + } ; + +} /* macosx_close */ + +static OSStatus +macosx_audio_out_callback (AudioDeviceID device, const AudioTimeStamp* current_time, + const AudioBufferList* data_in, const AudioTimeStamp* time_in, + AudioBufferList* data_out, const AudioTimeStamp* time_out, void* client_data) +{ MACOSX_AUDIO_OUT *macosx_out ; + int size, frame_count, read_count ; + float *buffer ; + + /* unused params: */ + (void) device; + (void) current_time; + (void) data_in; + (void) time_in; + (void) time_out; + + if ((macosx_out = (MACOSX_AUDIO_OUT*) client_data) == NULL) + { printf ("macosx_play : AUDIO_OUT is NULL.\n") ; + return 42 ; + } ; + + if (macosx_out->magic != MACOSX_MAGIC) + { printf ("macosx_play : Bad magic number.\n") ; + return 42 ; + } ; + + size = data_out->mBuffers [0].mDataByteSize ; + frame_count = size / sizeof (float) / macosx_out->channels ; + + buffer = (float*) data_out->mBuffers [0].mData ; + + read_count = macosx_out->callback (macosx_out->callback_data, buffer, frame_count) ; + + if (read_count < frame_count) + { memset (&(buffer [read_count]), 0, (frame_count - read_count) * sizeof (float)) ; + macosx_out->done_playing = 1 ; + } ; + + return noErr ; +} /* macosx_audio_out_callback */ + +#endif /* MacOSX */ + + +/*------------------------------------------------------------------------------ +** Win32 functions for playing a sound. +** +** This API sucks. Its needlessly complicated and is *WAY* too loose with +** passing pointers arounf in integers and and using char* pointers to +** point to data instead of short*. It plain sucks! +*/ + +#if (defined (_WIN32) || defined (WIN32)) + +#define WIN32_BUFFER_LEN (1<<15) +#define WIN32_MAGIC MAKE_MAGIC ('W', 'i', 'n', '3', '2', 's', 'u', 'x') + +typedef struct +{ int magic ; + + HWAVEOUT hwave ; + WAVEHDR whdr [2] ; + + HANDLE Event ; + + short short_buffer [WIN32_BUFFER_LEN / sizeof (short)] ; + float float_buffer [WIN32_BUFFER_LEN / sizeof (short) / 2] ; + + int bufferlen, current ; + + int channels ; + + get_audio_callback_t callback ; + + void *callback_data ; +} WIN32_AUDIO_OUT ; + +static AUDIO_OUT *win32_open (int channels, int samplerate) ; +static void win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ; +static void win32_close (AUDIO_OUT *audio_out) ; + +static DWORD CALLBACK + win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD_PTR param1, DWORD_PTR param2) ; + +static AUDIO_OUT* +win32_open (int channels, int samplerate) +{ WIN32_AUDIO_OUT *win32_out ; + + WAVEFORMATEX wf ; + int error ; + + if ((win32_out = calloc (1, sizeof (WIN32_AUDIO_OUT))) == NULL) + { perror ("win32_open : malloc ") ; + exit (1) ; + } ; + + win32_out->magic = WIN32_MAGIC ; + win32_out->channels = channels ; + + win32_out->current = 0 ; + + win32_out->Event = CreateEvent (0, FALSE, FALSE, 0) ; + + wf.nChannels = channels ; + wf.nSamplesPerSec = samplerate ; + wf.nBlockAlign = (WORD) (channels * sizeof (short)) ; + + wf.wFormatTag = WAVE_FORMAT_PCM ; + wf.cbSize = 0 ; + wf.wBitsPerSample = 16 ; + wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec ; + + error = waveOutOpen (&(win32_out->hwave), WAVE_MAPPER, &wf, (DWORD_PTR) win32_audio_out_callback, + (DWORD_PTR) win32_out, CALLBACK_FUNCTION) ; + if (error) + { puts ("waveOutOpen failed.") ; + free (win32_out) ; + return NULL ; + } ; + + waveOutPause (win32_out->hwave) ; + + return (AUDIO_OUT *) win32_out ; +} /* win32_open */ + +static void +win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) +{ WIN32_AUDIO_OUT *win32_out ; + int error ; + + if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL) + { printf ("win32_play : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (win32_out->magic != WIN32_MAGIC) + { printf ("win32_play : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ; + return ; + } ; + + /* Set the callback function and callback data. */ + win32_out->callback = callback ; + win32_out->callback_data = callback_data ; + + win32_out->whdr [0].lpData = (char*) win32_out->short_buffer ; + win32_out->whdr [1].lpData = ((char*) win32_out->short_buffer) + sizeof (win32_out->short_buffer) / 2 ; + + win32_out->whdr [0].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ; + win32_out->whdr [1].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ; + + win32_out->bufferlen = sizeof (win32_out->short_buffer) / 2 / sizeof (short) ; + + /* Prepare the WAVEHDRs */ + if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)))) + { printf ("waveOutPrepareHeader [0] failed : %08X\n", error) ; + waveOutClose (win32_out->hwave) ; + return ; + } ; + + if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR)))) + { printf ("waveOutPrepareHeader [1] failed : %08X\n", error) ; + waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ; + waveOutClose (win32_out->hwave) ; + return ; + } ; + + waveOutRestart (win32_out->hwave) ; + + /* Fake 2 calls to the callback function to queue up enough audio. */ + win32_audio_out_callback (0, MM_WOM_DONE, (DWORD_PTR) win32_out, 0, 0) ; + win32_audio_out_callback (0, MM_WOM_DONE, (DWORD_PTR) win32_out, 0, 0) ; + + /* Wait for playback to finish. The callback notifies us when all + ** wave data has been played. + */ + WaitForSingleObject (win32_out->Event, INFINITE) ; + + waveOutPause (win32_out->hwave) ; + waveOutReset (win32_out->hwave) ; + + waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ; + waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR)) ; + + waveOutClose (win32_out->hwave) ; + win32_out->hwave = 0 ; + + return ; +} /* win32_play */ + +static void +win32_close (AUDIO_OUT *audio_out) +{ WIN32_AUDIO_OUT *win32_out ; + + if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL) + { printf ("win32_close : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (win32_out->magic != WIN32_MAGIC) + { printf ("win32_close : Bad magic number.\n") ; + return ; + } ; + + memset (win32_out, 0, sizeof (WIN32_AUDIO_OUT)) ; + + free (win32_out) ; +} /* win32_close */ + +static DWORD CALLBACK +win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD_PTR param1, DWORD_PTR param2) +{ + UNREFERENCED_PARAMETER (hwave) ; + UNREFERENCED_PARAMETER (param1) ; + UNREFERENCED_PARAMETER (param2) ; + WIN32_AUDIO_OUT *win32_out ; + int read_count, frame_count, k ; + short *sptr ; + + /* + ** I consider this technique of passing a pointer via an integer as + ** fundamentally broken but thats the way microsoft has defined the + ** interface. + */ + if ((win32_out = (WIN32_AUDIO_OUT*) data) == NULL) + { printf ("win32_audio_out_callback : AUDIO_OUT is NULL.\n") ; + return 1 ; + } ; + + if (win32_out->magic != WIN32_MAGIC) + { printf ("win32_audio_out_callback : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ; + return 1 ; + } ; + + if (msg != MM_WOM_DONE) + return 0 ; + + /* Do the actual audio. */ + frame_count = win32_out->bufferlen / win32_out->channels ; + + read_count = win32_out->callback (win32_out->callback_data, win32_out->float_buffer, frame_count) ; + + sptr = (short*) win32_out->whdr [win32_out->current].lpData ; + + for (k = 0 ; k < read_count ; k++) + sptr [k] = (short) lrint (32767.0 * win32_out->float_buffer [k]) ; + + if (read_count > 0) + { /* Fix buffer length is only a partial block. */ + if (read_count * (int) sizeof (short) < win32_out->bufferlen) + win32_out->whdr [win32_out->current].dwBufferLength = read_count * sizeof (short) ; + + /* Queue the WAVEHDR */ + waveOutWrite (win32_out->hwave, (LPWAVEHDR) &(win32_out->whdr [win32_out->current]), sizeof (WAVEHDR)) ; + } + else + { /* Stop playback */ + waveOutPause (win32_out->hwave) ; + + SetEvent (win32_out->Event) ; + } ; + + win32_out->current = (win32_out->current + 1) % 2 ; + + return 0 ; +} /* win32_audio_out_callback */ + +#endif /* Win32 */ + +/*------------------------------------------------------------------------------ +** Solaris. +*/ + +#if (defined (sun) && defined (unix)) /* ie Solaris */ + +#include <fcntl.h> +#include <sys/ioctl.h> +#include <sys/audioio.h> + +#define SOLARIS_MAGIC MAKE_MAGIC ('S', 'o', 'l', 'a', 'r', 'i', 's', ' ') + +typedef struct +{ int magic ; + int fd ; + int channels ; + int samplerate ; +} SOLARIS_AUDIO_OUT ; + +static AUDIO_OUT *solaris_open (int channels, int samplerate) ; +static void solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ; +static void solaris_close (AUDIO_OUT *audio_out) ; + +static AUDIO_OUT * +solaris_open (int channels, int samplerate) +{ SOLARIS_AUDIO_OUT *solaris_out ; + audio_info_t audio_info ; + int error ; + + if ((solaris_out = calloc (1, sizeof (SOLARIS_AUDIO_OUT))) == NULL) + { perror ("solaris_open : malloc ") ; + exit (1) ; + } ; + + solaris_out->magic = SOLARIS_MAGIC ; + solaris_out->channels = channels ; + solaris_out->samplerate = channels ; + + /* open the audio device - write only, non-blocking */ + if ((solaris_out->fd = open ("/dev/audio", O_WRONLY | O_NONBLOCK)) < 0) + { perror ("open (/dev/audio) failed") ; + exit (1) ; + } ; + + /* Retrive standard values. */ + AUDIO_INITINFO (&audio_info) ; + + audio_info.play.sample_rate = samplerate ; + audio_info.play.channels = channels ; + audio_info.play.precision = 16 ; + audio_info.play.encoding = AUDIO_ENCODING_LINEAR ; + audio_info.play.gain = AUDIO_MAX_GAIN ; + audio_info.play.balance = AUDIO_MID_BALANCE ; + + if ((error = ioctl (solaris_out->fd, AUDIO_SETINFO, &audio_info))) + { perror ("ioctl (AUDIO_SETINFO) failed") ; + exit (1) ; + } ; + + return (AUDIO_OUT*) solaris_out ; +} /* solaris_open */ + +static void +solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) +{ SOLARIS_AUDIO_OUT *solaris_out ; + static float float_buffer [BUFFER_LEN] ; + static short buffer [BUFFER_LEN] ; + int k, read_frames ; + + if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL) + { printf ("solaris_play : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (solaris_out->magic != SOLARIS_MAGIC) + { printf ("solaris_play : Bad magic number.\n") ; + return ; + } ; + + while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / solaris_out->channels))) + { for (k = 0 ; k < read_frames * solaris_out->channels ; k++) + buffer [k] = psf_lrint (32767.0 * float_buffer [k]) ; + write (solaris_out->fd, buffer, read_frames * solaris_out->channels * sizeof (short)) ; + } ; + + return ; +} /* solaris_play */ + +static void +solaris_close (AUDIO_OUT *audio_out) +{ SOLARIS_AUDIO_OUT *solaris_out ; + + if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL) + { printf ("solaris_close : AUDIO_OUT is NULL.\n") ; + return ; + } ; + + if (solaris_out->magic != SOLARIS_MAGIC) + { printf ("solaris_close : Bad magic number.\n") ; + return ; + } ; + + memset (solaris_out, 0, sizeof (SOLARIS_AUDIO_OUT)) ; + + free (solaris_out) ; + + return ; +} /* solaris_close */ + +#endif /* Solaris */ + +/*============================================================================== +** Main function. +*/ + +AUDIO_OUT * +audio_open (int channels, int samplerate) +{ +#if defined (__linux__) + #if HAVE_ALSA + if (access ("/proc/asound/cards", R_OK) == 0) + return alsa_open (channels, samplerate) ; + #endif + return opensoundsys_open (channels, samplerate) ; +#elif (defined (__MACH__) && defined (__APPLE__)) + return macosx_open (channels, samplerate) ; +#elif (defined (sun) && defined (unix)) + return solaris_open (channels, samplerate) ; +#elif (defined (_WIN32) || defined (WIN32)) + return win32_open (channels, samplerate) ; +#else + #warning "*** Playing sound not yet supported on this platform." + #warning "*** Please feel free to submit a patch." + printf ("Error : Playing sound not yet supported on this platform.\n") ; + return NULL ; +#endif + + + return NULL ; +} /* audio_open */ + +void +audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) +{ + + if (callback == NULL) + { printf ("Error : bad callback pointer.\n") ; + return ; + } ; + + if (audio_out == NULL) + { printf ("Error : bad audio_out pointer.\n") ; + return ; + } ; + + if (callback_data == NULL) + { printf ("Error : bad callback_data pointer.\n") ; + return ; + } ; + +#if defined (__linux__) + #if HAVE_ALSA + if (audio_out->magic == ALSA_MAGIC) + alsa_play (callback, audio_out, callback_data) ; + #endif + opensoundsys_play (callback, audio_out, callback_data) ; +#elif (defined (__MACH__) && defined (__APPLE__)) + macosx_play (callback, audio_out, callback_data) ; +#elif (defined (sun) && defined (unix)) + solaris_play (callback, audio_out, callback_data) ; +#elif (defined (_WIN32) || defined (WIN32)) + win32_play (callback, audio_out, callback_data) ; +#else + #warning "*** Playing sound not yet supported on this platform." + #warning "*** Please feel free to submit a patch." + printf ("Error : Playing sound not yet supported on this platform.\n") ; + return ; +#endif + + return ; +} /* audio_play */ + +void +audio_close (AUDIO_OUT *audio_out) +{ +#if defined (__linux__) + #if HAVE_ALSA + if (audio_out->magic == ALSA_MAGIC) + alsa_close (audio_out) ; + #endif + opensoundsys_close (audio_out) ; +#elif (defined (__MACH__) && defined (__APPLE__)) + macosx_close (audio_out) ; +#elif (defined (sun) && defined (unix)) + solaris_close (audio_out) ; +#elif (defined (_WIN32) || defined (WIN32)) + win32_close (audio_out) ; +#else + #warning "*** Playing sound not yet supported on this platform." + #warning "*** Please feel free to submit a patch." + printf ("Error : Playing sound not yet supported on this platform.\n") ; + return ; +#endif + + return ; +} /* audio_close */ + +#else /* (HAVE_SNDFILE == 0) */ + +/* Do not have libsndfile installed so just return. */ + +AUDIO_OUT * +audio_open (int channels, int samplerate) +{ + (void) channels ; + (void) samplerate ; + + return NULL ; +} /* audio_open */ + +void +audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) +{ + (void) callback ; + (void) audio_out ; + (void) callback_data ; + + return ; +} /* audio_play */ + +void +audio_close (AUDIO_OUT *audio_out) +{ + audio_out = audio_out ; + + return ; +} /* audio_close */ + +#endif /* HAVE_SNDFILE */ + diff --git a/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.h b/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.h new file mode 100755 index 00000000..993da707 --- /dev/null +++ b/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.h @@ -0,0 +1,25 @@ +/* +** Copyright (c) 1999-2016, Erik de Castro Lopo <erikd@mega-nerd.com> +** All rights reserved. +** +** This code is released under 2-clause BSD license. Please see the +** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING +*/ + +typedef struct AUDIO_OUT AUDIO_OUT ; + +typedef int (*get_audio_callback_t) (void *callback_data, float *samples, int frames) ; + +/* A general audio output function (Linux/ALSA, Linux/OSS, Win32, MacOSX, +** Solaris) which retrieves data using the callback function in the above +** struct. +** +** audio_open - opens the device and returns an anonymous pointer to its +** own private data. +*/ + +AUDIO_OUT *audio_open (int channels, int samplerate) ; + +void audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ; + +void audio_close (AUDIO_OUT *audio_data) ; diff --git a/lib/libsamplerate/libsamplerate-0.2.2/examples/timewarp-file.c b/lib/libsamplerate/libsamplerate-0.2.2/examples/timewarp-file.c new file mode 100755 index 00000000..4f9f3fa7 --- /dev/null +++ b/lib/libsamplerate/libsamplerate-0.2.2/examples/timewarp-file.c @@ -0,0 +1,234 @@ +/* +** Copyright (c) 2005-2016, Erik de Castro Lopo <erikd@mega-nerd.com> +** All rights reserved. +** +** This code is released under 2-clause BSD license. Please see the +** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdio.h> +#include <stdlib.h> +#ifdef HAVE_UNISTD_H +#include <unistd.h> +#endif +#include <string.h> +#include <math.h> + +#if (HAVE_SNDFILE) + +#include <samplerate.h> +#include <sndfile.h> + +#define ARRAY_LEN(x) ((int) (sizeof (x) / sizeof ((x) [0]))) + +#define DEFAULT_CONVERTER SRC_SINC_MEDIUM_QUALITY + +#define BUFFER_LEN 1024 +#define INPUT_STEP_SIZE 8 + +typedef struct +{ sf_count_t index ; + double ratio ; +} TIMEWARP_FACTOR ; + +static void usage_exit (const char *progname) ; +static sf_count_t timewarp_convert (SNDFILE *infile, SNDFILE *outfile, int converter, int channels) ; + +int +main (int argc, char *argv []) +{ SNDFILE *infile, *outfile ; + SF_INFO sfinfo ; + sf_count_t count ; + + if (argc != 3) + usage_exit (argv [0]) ; + + putchar ('\n') ; + printf ("Input File : %s\n", argv [argc - 2]) ; + if ((infile = sf_open (argv [argc - 2], SFM_READ, &sfinfo)) == NULL) + { printf ("Error : Not able to open input file '%s'\n", argv [argc - 2]) ; + exit (1) ; + } ; + + if (INPUT_STEP_SIZE * sfinfo.channels > BUFFER_LEN) + { printf ("\n\nError : INPUT_STEP_SIZE * sfinfo.channels > BUFFER_LEN\n\n") ; + exit (1) ; + } ; + + + /* Delete the output file length to zero if already exists. */ + remove (argv [argc - 1]) ; + + if ((outfile = sf_open (argv [argc - 1], SFM_WRITE, &sfinfo)) == NULL) + { printf ("Error : Not able to open output file '%s'\n", argv [argc - 1]) ; + sf_close (infile) ; + exit (1) ; + } ; + + sf_command (outfile, SFC_SET_CLIPPING, NULL, SF_TRUE) ; + + printf ("Output file : %s\n", argv [argc - 1]) ; + printf ("Converter : %s\n", src_get_name (DEFAULT_CONVERTER)) ; + + count = timewarp_convert (infile, outfile, DEFAULT_CONVERTER, sfinfo.channels) ; + + printf ("Output Frames : %ld\n\n", (long) count) ; + + sf_close (infile) ; + sf_close (outfile) ; + + return 0 ; +} /* main */ + +/*============================================================================== +*/ + +static TIMEWARP_FACTOR warp [] = +{ { 0 , 1.00000001 }, + { 20000 , 1.01000000 }, + { 20200 , 1.00000001 }, + { 40000 , 1.20000000 }, + { 40300 , 1.00000001 }, + { 60000 , 1.10000000 }, + { 60400 , 1.00000001 }, + { 80000 , 1.50000000 }, + { 81000 , 1.00000001 }, +} ; + +static sf_count_t +timewarp_convert (SNDFILE *infile, SNDFILE *outfile, int converter, int channels) +{ static float input [BUFFER_LEN] ; + static float output [BUFFER_LEN] ; + + SRC_STATE *src_state ; + SRC_DATA src_data ; + int error, warp_index = 0 ; + sf_count_t input_count = 0, output_count = 0 ; + + sf_seek (infile, 0, SEEK_SET) ; + sf_seek (outfile, 0, SEEK_SET) ; + + /* Initialize the sample rate converter. */ + if ((src_state = src_new (converter, channels, &error)) == NULL) + { printf ("\n\nError : src_new() failed : %s.\n\n", src_strerror (error)) ; + exit (1) ; + } ; + + src_data.end_of_input = 0 ; /* Set this later. */ + + /* Start with zero to force load in while loop. */ + src_data.input_frames = 0 ; + src_data.data_in = input ; + + if (warp [0].index > 0) + src_data.src_ratio = 1.0 ; + else + { src_data.src_ratio = warp [0].ratio ; + warp_index ++ ; + } ; + + src_data.data_out = output ; + src_data.output_frames = BUFFER_LEN /channels ; + + while (1) + { + if (warp_index < ARRAY_LEN (warp) - 1 && input_count >= warp [warp_index].index) + { src_data.src_ratio = warp [warp_index].ratio ; + warp_index ++ ; + } ; + + /* If the input buffer is empty, refill it. */ + if (src_data.input_frames == 0) + { src_data.input_frames = (long) sf_readf_float (infile, input, INPUT_STEP_SIZE) ; + input_count += src_data.input_frames ; + src_data.data_in = input ; + + /* The last read will not be a full buffer, so snd_of_input. */ + if (src_data.input_frames < INPUT_STEP_SIZE) + src_data.end_of_input = SF_TRUE ; + } ; + + /* Process current block. */ + if ((error = src_process (src_state, &src_data))) + { printf ("\nError : %s\n", src_strerror (error)) ; + exit (1) ; + } ; + + /* Terminate if done. */ + if (src_data.end_of_input && src_data.output_frames_gen == 0) + break ; + + /* Write output. */ + sf_writef_float (outfile, output, src_data.output_frames_gen) ; + output_count += src_data.output_frames_gen ; + + src_data.data_in += src_data.input_frames_used * channels ; + src_data.input_frames -= src_data.input_frames_used ; + } ; + + src_delete (src_state) ; + + return output_count ; +} /* timewarp_convert */ + +/*------------------------------------------------------------------------------ +*/ + +static void +usage_exit (const char *progname) +{ const char *cptr ; + + if ((cptr = strrchr (progname, '/')) != NULL) + progname = cptr + 1 ; + + if ((cptr = strrchr (progname, '\\')) != NULL) + progname = cptr + 1 ; + + printf ("\n" + " A program demonstrating the time warping capabilities of libsamplerate." + " It uses libsndfile for file I/O and Secret Rabbit Code (aka libsamplerate)" + " for performing the warping.\n" + " It works on any file format supported by libsndfile with any \n" + " number of channels (limited only by host memory).\n" + "\n" + " The warping is dependant on a table hard code into the source code.\n" + "\n" + " libsamplerate version : %s\n" + "\n" + " Usage : \n" + " %s <input file> <output file>\n" + "\n", src_get_version (), progname) ; + + puts ("") ; + + exit (1) ; +} /* usage_exit */ + +/*============================================================================== +*/ + +#else /* (HAVE_SNFILE == 0) */ + +/* Alternative main function when libsndfile is not available. */ + +int +main (void) +{ puts ( + "\n" + "****************************************************************\n" + " This example program was compiled without libsndfile \n" + " (https://github.com/libsndfile/libsndfile/).\n" + " It is therefore completely broken and non-functional.\n" + "****************************************************************\n" + "\n" + ) ; + + return 0 ; +} /* main */ + +#endif + diff --git a/lib/libsamplerate/libsamplerate-0.2.2/examples/varispeed-play.c b/lib/libsamplerate/libsamplerate-0.2.2/examples/varispeed-play.c new file mode 100755 index 00000000..72a5a9c3 --- /dev/null +++ b/lib/libsamplerate/libsamplerate-0.2.2/examples/varispeed-play.c @@ -0,0 +1,246 @@ +/* +** Copyright (c) 2002-2016, Erik de Castro Lopo <erikd@mega-nerd.com> +** All rights reserved. +** +** This code is released under 2-clause BSD license. Please see the +** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING +*/ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <stdio.h> +#include <stdlib.h> +#ifdef HAVE_UNISTD_H +#include <unistd.h> +#endif +#include <string.h> +#include <math.h> + +#if (HAVE_SNDFILE) + +#include <samplerate.h> +#include <sndfile.h> + +#include "audio_out.h" + +#define ARRAY_LEN(x) ((int) (sizeof (x) / sizeof ((x) [0]))) + +#define BUFFER_LEN 4096 +#define VARISPEED_BLOCK_LEN 64 + +#define MIN(a,b) ((a) < (b) ? (a) : (b)) + +#define SRC_MAGIC ((int) ('S' << 16) + ('R' << 8) + ('C')) +#define SNDFILE_MAGIC ((int) ('s' << 24) + ('n' << 20) + ('d' << 16) + ('f' << 12) + ('i' << 8) + ('l' << 4) + 'e') + +#ifndef M_PI +#define M_PI 3.14159265358979323846264338 +#endif + + +typedef struct +{ int magic ; + SNDFILE *sndfile ; + SF_INFO sfinfo ; + + float buffer [BUFFER_LEN] ; +} SNDFILE_CB_DATA ; + +typedef struct +{ int magic ; + + SNDFILE_CB_DATA sf ; + + int freq_point ; + + SRC_STATE *src_state ; + +} SRC_CB_DATA ; + +static int varispeed_get_data (SRC_CB_DATA *data, float *samples, int frames) ; +static void varispeed_play (const char *filename, int converter) ; + +static long src_input_callback (void *cb_data, float **data) ; + +int +main (int argc, char *argv []) +{ const char *cptr, *progname, *filename ; + int k, converter ; + + converter = SRC_SINC_FASTEST ; + + progname = argv [0] ; + + if ((cptr = strrchr (progname, '/')) != NULL) + progname = cptr + 1 ; + + if ((cptr = strrchr (progname, '\\')) != NULL) + progname = cptr + 1 ; + + printf ("\n" + " %s\n" + "\n" + " This is a demo program which plays the given file at a slowly \n" + " varying speed. Lots of fun with drum loops and full mixes.\n" + "\n" + " It uses Secret Rabbit Code (aka libsamplerate) to perform the \n" + " vari-speeding and libsndfile for file I/O.\n" + "\n", progname) ; + + if (argc == 2) + filename = argv [1] ; + else if (argc == 4 && strcmp (argv [1], "-c") == 0) + { filename = argv [3] ; + converter = atoi (argv [2]) ; + } + else + { printf (" Usage :\n\n %s [-c <number>] <input file>\n\n", progname) ; + puts ( + " The optional -c argument allows the converter type to be chosen from\n" + " the following list :" + "\n" + ) ; + + for (k = 0 ; (cptr = src_get_name (k)) != NULL ; k++) + printf (" %d : %s\n", k, cptr) ; + + puts ("") ; + exit (1) ; + } ; + + varispeed_play (filename, converter) ; + + return 0 ; +} /* main */ + +/*============================================================================== +*/ + +static void +varispeed_play (const char *filename, int converter) +{ SRC_CB_DATA data ; + AUDIO_OUT *audio_out ; + int error ; + + memset (&data, 0, sizeof (data)) ; + + data.magic = SRC_MAGIC ; + data.sf.magic = SNDFILE_MAGIC ; + + if ((data.sf.sndfile = sf_open (filename, SFM_READ, &data.sf.sfinfo)) == NULL) + { puts (sf_strerror (NULL)) ; + exit (1) ; + } ; + + /* Initialize the sample rate converter. */ + if ((data.src_state = src_callback_new (src_input_callback, converter, data.sf.sfinfo.channels, &error, &data.sf)) == NULL) + { printf ("\n\nError : src_new() failed : %s.\n\n", src_strerror (error)) ; + exit (1) ; + } ; + + printf ( + + " Playing : %s\n" + " Converter : %s\n" + "\n" + " Press <control-c> to exit.\n" + "\n", + filename, src_get_name (converter)) ; + + if ((audio_out = audio_open (data.sf.sfinfo.channels, data.sf.sfinfo.samplerate)) == NULL) + { printf ("\n\nError : audio_open () failed.\n") ; + exit (1) ; + } ; + + /* Pass the data and the callbacl function to audio_play */ + audio_play ((get_audio_callback_t) varispeed_get_data, audio_out, &data) ; + + /* Cleanup */ + audio_close (audio_out) ; + sf_close (data.sf.sndfile) ; + src_delete (data.src_state) ; + +} /* varispeed_play */ + +static long +src_input_callback (void *cb_data, float **audio) +{ SNDFILE_CB_DATA * data = (SNDFILE_CB_DATA *) cb_data ; + const int input_frames = ARRAY_LEN (data->buffer) / data->sfinfo.channels ; + int read_frames ; + + if (data->magic != SNDFILE_MAGIC) + { printf ("\n\n%s:%d Eeeek, something really bad happened!\n", __FILE__, __LINE__) ; + exit (1) ; + } ; + + for (read_frames = 0 ; read_frames < input_frames ; ) + { sf_count_t position ; + + read_frames += (int) sf_readf_float (data->sndfile, data->buffer + read_frames * data->sfinfo.channels, input_frames - read_frames) ; + + position = sf_seek (data->sndfile, 0, SEEK_CUR) ; + + if (position < 0 || position == data->sfinfo.frames) + sf_seek (data->sndfile, 0, SEEK_SET) ; + } ; + + *audio = & (data->buffer [0]) ; + + return input_frames ; +} /* src_input_callback */ + + +/*============================================================================== +*/ + +static int +varispeed_get_data (SRC_CB_DATA *data, float *samples, int out_frames) +{ float *output ; + int rc, out_frame_count ; + + if (data->magic != SRC_MAGIC) + { printf ("\n\n%s:%d Eeeek, something really bad happened!\n", __FILE__, __LINE__) ; + exit (1) ; + } ; + + for (out_frame_count = 0 ; out_frame_count < out_frames ; out_frame_count += VARISPEED_BLOCK_LEN) + { double src_ratio = 1.0 - 0.5 * sin (data->freq_point * 2 * M_PI / 20000) ; + + data->freq_point ++ ; + + output = samples + out_frame_count * data->sf.sfinfo.channels ; + + if ((rc = src_callback_read (data->src_state, src_ratio, VARISPEED_BLOCK_LEN, output)) < VARISPEED_BLOCK_LEN) + { printf ("\nError : src_callback_read short output (%d instead of %d)\n\n", rc, VARISPEED_BLOCK_LEN) ; + exit (1) ; + } ; + } ; + + return out_frames ; +} /* varispeed_get_data */ + +/*============================================================================== +*/ + +#else /* (HAVE_SNFILE == 0) */ + +/* Alternative main function when libsndfile is not available. */ + +int +main (void) +{ puts ( + "\n" + "****************************************************************\n" + " This example program was compiled without libsndfile \n" + " (http://www.zip.com.au/~erikd/libsndfile/).\n" + " It is therefore completely broken and non-functional.\n" + "****************************************************************\n" + "\n" + ) ; + + return 0 ; +} /* main */ + +#endif |