HomoFast-eSpeak-Persian/src/libespeak-ng/wave_sada.c

/*
 * Copyright (C) 2008, Sun Microsystems, Inc.
 * Copyright (C) 2015 Reece H. Dunn
 * eSpeak driver for Solaris Audio Device Architecture (SADA)
 * Written by Willie Walker, based on the eSpeak PulseAudio driver
 * from Gilles Casse
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 3 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, see: <http://www.gnu.org/licenses/>.
 */

#include "speech.h"

#include <errno.h>
#include <string.h>
#include <stropts.h>
#include <assert.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/audioio.h>

#include "wave.h"

#define SAMPLE_RATE 22050
#define SAMPLE_SIZE 16

#ifdef USE_SADA

static t_wave_callback *my_callback_is_output_enabled = NULL;

static const char *sun_audio_device = "/dev/audio";
static int sun_audio_fd = -1;

// The total number of 16-bit samples sent to be played via the
// wave_write method.
//
static uint32_t total_samples_sent;

// The total number of samples sent to be played via the wave_write
// method, but which were never played because of a call to
// wave_close.
//
static uint32_t total_samples_skipped;

// The last known playing index after a call to wave_close.
//
static uint32_t last_play_position = 0;

static uint32_t wave_samplerate;

// wave_open
//
// DESCRIPTION:
//
// initializes the audio subsytem.
//
// GLOBALS USED/MODIFIED:
//
// sun_audio_fd: modified to hold the file descriptor of the opened
// audio device.
//
void *wave_open(int srate, const char *device)
{
	if (device == NULL)
		device = sun_audio_device;

	audio_info_t ainfo;

	wave_samplerate = srate;

	if ((sun_audio_fd = open(device, O_WRONLY)) < 0) {
		fprintf(stderr, "wave_open() could not open: %s (%d)\n", device, sun_audio_fd);
		return NULL;
	}

	ioctl(sun_audio_fd, AUDIO_GETINFO, &ainfo);
	ainfo.play.encoding = AUDIO_ENCODING_LINEAR;
	ainfo.play.channels = 1;
	ainfo.play.sample_rate = wave_samplerate;
	ainfo.play.precision = SAMPLE_SIZE;

	if (ioctl(sun_audio_fd, AUDIO_SETINFO, &ainfo) == -1) {
		fprintf(stderr, "wave_open() failed to set audio params: %s\n", strerror(errno));
		close(sun_audio_fd);
		return NULL;
	}

	return (void *)sun_audio_fd;
}

// wave_write
//
// DESCRIPTION:
//
// Meant to be asynchronous, it supplies the wave sample to the lower
// audio layer and returns. The sample is played later on.  [[[WDW -
// we purposely do not open the audio device as non-blocking because
// managing that would be a pain.  So, we rely a lot upon fifo.cpp and
// event.cpp to not overload us, allowing us to get away with a
// blocking write.  event.cpp:polling_thread in particular appears to
// use get_remaining_time to prevent flooding.]]]
//
// PARAMETERS:
//
// theHandler: the audio device file descriptor
// theMono16BitsWaveBuffer: the audio data
// theSize: the number of bytes (not 16-bit samples)
//
// GLOBALS USED/MODIFIED:
//
// total_samples_sent: modified based upon 16-bit samples sent
//
// RETURNS:
//
// the number of bytes (not 16-bit samples) sent
//
size_t wave_write(void *theHandler,
                  char *theMono16BitsWaveBuffer,
                  size_t theSize)
{
	size_t num;
	if (my_callback_is_output_enabled && (0 == my_callback_is_output_enabled()))
		return 0;

#if defined(BYTE_ORDER) && BYTE_ORDER == BIG_ENDIAN
	// BIG-ENDIAN, swap the order of bytes in each sound sample
	int c;
	char *out_ptr;
	char *out_end;
	out_ptr = (char *)theMono16BitsWaveBuffer;
	out_end = out_ptr + theSize;
	while (out_ptr < out_end) {
		c = out_ptr[0];
		out_ptr[0] = out_ptr[1];
		out_ptr[1] = c;
		out_ptr += 2;
	}
#endif

	num = write((int)theHandler, theMono16BitsWaveBuffer, theSize);

	// Keep track of the total number of samples sent -- we use this in
	// wave_get_read_position and also use it to help calculate the
	// total_samples_skipped in wave_close.
	//
	total_samples_sent += num / 2;

	return num;
}

// wave_close
//
// DESCRIPTION:
//
// Does what SADA normally would call a flush, which means to cease
// all audio production in progress and throw any remaining audio
// away.  [[[WDW - see comment in wave_flush.]]]
//
// PARAMETERS:
//
// theHandler: the audio device file descriptor
//
// GLOBALS USED/MODIFIED:
//
// last_play_position: modified to reflect play position the last time
//                     this method was called
// total_samples_sent: used to help calculate total_samples_skipped
// total_samples_skipped: modified to hold the total number of 16-bit
//                        samples sent to wave_write, but which were
//                        never played
// sun_audio_fd: used because some calls to wave_close seem to
//               pass a NULL for theHandler for some odd reason
//
// RETURNS:
//
// The result of the ioctl call (non-0 means failure)
//
int wave_close(void *theHandler)
{
	int ret;
	audio_info_t ainfo;
	int audio_fd = (int)theHandler;
	if (!audio_fd)
		audio_fd = sun_audio_fd;
	// [[[WDW: maybe do a pause/resume ioctl???]]]
	ret = ioctl(audio_fd, I_FLUSH, FLUSHRW);
	ioctl(audio_fd, AUDIO_GETINFO, &ainfo);

	// Calculate the number of samples that won't get
	// played.  We also keep track of the last_play_position
	// because wave_close can be called multiple times
	// before another call to wave_write.
	//
	if (last_play_position != ainfo.play.samples) {
		last_play_position = ainfo.play.samples;
		total_samples_skipped = total_samples_sent - last_play_position;
	}
	return ret;
}

// wave_is_busy
//
// DESCRIPTION:
//
// Returns a non-0 value if audio is being played.
//
// PARAMETERS:
//
// theHandler: the audio device file descriptor
//
// GLOBALS USED/MODIFIED:
//
// sun_audio_fd: used because some calls to wave_is_busy seem to
//               pass a NULL for theHandler for some odd reason
//
// RETURNS:
//
// A non-0 value if audio is being played
//
int wave_is_busy(void *theHandler)
{
	(void)theHandler; // unused

	uint32_t time;
	if (total_samples_sent >= 1)
		wave_get_remaining_time(total_samples_sent - 1, &time);
	else
		time = 0;
	return time != 0;
}

// wave_terminate
//
// DESCRIPTION:
//
// Used to end our session with eSpeak.
//
// GLOBALS USED/MODIFIED:
//
// sun_audio_fd: modified - closed and set to -1
//
void wave_terminate()
{
	close(sun_audio_fd);
	sun_audio_fd = -1;
}

// wave_flush
//
// DESCRIPTION:
//
// Appears to want to tell the audio subsystem to make sure it plays
// the audio.  In our case, the system is already doing this, so this
// is basically a no-op.  [[[WDW - if you do a drain, you block, so
// don't do that.  In addition the typical SADA notion of flush is
// currently handled by wave_close.  I think this is most likely just
// terminology conflict between eSpeak and SADA.]]]
//
// PARAMETERS:
//
// theHandler: the audio device file descriptor
//
void wave_flush(void *theHandler)
{
	(void)theHandler; // unused
}

// wave_set_callback_is_output_enabled
//
// DESCRIPTION:
//
// Sets the callback to call from wave_write before it sends data to
// be played.  It helps wave_write determine if the data should be
// thrown away or not.
//
// PARAMETERS:
//
// cb: the callback to call from wave_write
//
void wave_set_callback_is_output_enabled(t_wave_callback *cb)
{
	my_callback_is_output_enabled = cb;
}

// wave_test_get_write_buffer
//
// DESCRIPTION:
//
// Unnecessary and is used for debug output from
// speak_lib.cpp:dispatch_audio.
//
// RETURNS:
//
// NULL
//
void *wave_test_get_write_buffer()
{
	return NULL;
}

// wave_get_read_position
//
// DESCRIPTION:
//
// Concerns the sample which is currently played by the audio layer,
// where 'sample' is a small buffer of synthesized wave data,
// identified so that the user callback could be called when the
// 'sample' is really played. The identifier is returned by
// wave_get_write_position.  This method is unused.
//
// PARAMETERS:
//
// theHandler: the audio device file descriptor
//
// RETURNS:
//
// The total number of 16-bit samples played by the audio system
// so far.
//
uint32_t wave_get_read_position(void *theHandler)
{
	audio_info_t ainfo;
	ioctl((int)theHandler, AUDIO_GETINFO, &ainfo);
	return ainfo.play.samples;
}

// wave_get_write_position
//
// DESCRIPTION:
//
// Returns an identifier for a new sample, where 'sample' is a small
// buffer of synthesized wave data, identified so that the user
// callback could be called when the 'sample' is really played.  This
// implementation views the audio as one long continuous stream of
// 16-bit samples.
//
// PARAMETERS:
//
// theHandler: the audio device file descriptor
//
// GLOBALS USED/MODIFIED:
//
// total_samples_sent: used as the return value
//
// RETURNS:
//
// total_samples_sent, which is the index for the end of this long
// continuous stream.  [[[WDW: with a unit32_t managing 16-bit
// samples at 22050Hz, we have about 54 hours of play time before
// the index wraps back to 0.  We don't handle that wrapping, so
// the behavior after 54 hours of play time is undefined.]]]
//
uint32_t wave_get_write_position(void *theHandler)
{
	(void)theHandler; // unused

	return total_samples_sent;
}

// wave_get_remaining_time
//
// DESCRIPTION:
//
// Returns the remaining time (in ms) before the sample is played.
// The sample in this case is a return value from a previous call to
// wave_get_write_position.
//
// PARAMETERS:
//
// sample: an index returned from wave_get_write_position representing
//         an index into the long continuous stream of 16-bit samples
// time: a return value representing the delay in milliseconds until
//       sample is played.  A value of 0 means the sample is either
//       currently being played or it has already been played.
//
// GLOBALS USED/MODIFIED:
//
// sun_audio_fd: used to determine total number of samples played by
//               the audio system
// total_samples_skipped: used in remaining time calculation
//
// RETURNS:
//
// Time in milliseconds before the sample is played or 0 if the sample
// is currently playing or has already been played.
//
int wave_get_remaining_time(uint32_t sample, uint32_t *time)
{
	uint32_t a_time = 0;
	uint32_t actual_index;

	audio_info_t ainfo;
	if (!time)
		return -1;

	ioctl(sun_audio_fd, AUDIO_GETINFO, &ainfo);

	// See if this sample has already been played or is currently
	// playing.
	//
	actual_index = sample - total_samples_skipped;
	if ((sample < total_samples_skipped) ||
	    (actual_index <= ainfo.play.samples))
		*time = 0;
	else {
		a_time = ((actual_index - ainfo.play.samples) * 1000) / wave_samplerate;
		*time = (uint32_t)a_time;
	}
	return 0;
}

#endif