mirror of
https://github.com/rhasspy/espeak-ng.git
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[1.36.12]
lang=hr: adjust phonemes [S tS tS; Z dZ dZ;] git-svn-id: https://espeak.svn.sourceforge.net/svnroot/espeak/trunk@173 d46cf337-b52f-0410-862d-fd96e6ae7743
This commit is contained in:
jonsd
Executable
+576
@@ -0,0 +1,576 @@
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/***************************************************************************
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* Copyright (C) 2008, Sun Microsystems, Inc. *
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* eSpeak driver for Solaris Audio Device Architecture (SADA) *
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* Written by Willie Walker, based on the eSpeak PulseAudio driver *
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* from Gilles Casse *
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* *
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* This program is free software; you can redistribute it and/or modify *
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* it under the terms of the GNU General Public License as published by *
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* the Free Software Foundation; either version 3 of the License, or *
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* (at your option) any later version. *
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* *
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* This program is distributed in the hope that it will be useful, *
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* but WITHOUT ANY WARRANTY; without even the implied warranty of *
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the *
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* GNU General Public License for more details. *
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* *
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* You should have received a copy of the GNU General Public License *
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* along with this program; if not, write to the *
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* Free Software Foundation, Inc., *
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* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. *
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***************************************************************************/
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#include "speech.h"
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#ifdef USE_ASYNC
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// This source file is only used for asynchronious modes
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#include <errno.h>
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#include <string.h>
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#include <stropts.h>
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#include <assert.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <fcntl.h>
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#include <sys/audioio.h>
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#include "wave.h"
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#include "debug.h"
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enum {ONE_BILLION=1000000000};
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#define SAMPLE_RATE 22050
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#define SAMPLE_SIZE 16
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#ifdef USE_SADA
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static t_wave_callback* my_callback_is_output_enabled=NULL;
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static const char *sun_audio_device = "/dev/audio";
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static int sun_audio_fd = -1;
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// The total number of 16-bit samples sent to be played via the
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// wave_write method.
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//
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static uint32_t total_samples_sent;
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// The total number of samples sent to be played via the wave_write
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// method, but which were never played because of a call to
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// wave_close.
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//
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static uint32_t total_samples_skipped;
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// The last known playing index after a call to wave_close.
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//
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static uint32_t last_play_position=0;
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//>
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// wave_init
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//
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// DESCRIPTION:
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//
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// initializes the audio subsytem.
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//
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// GLOBALS USED/MODIFIED:
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//
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// sun_audio_fd: modified to hold the file descriptor of the opened
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// audio device.
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//
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//<wave_init
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void wave_init() {
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ENTER("wave_init");
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audio_info_t ainfo;
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char *audio_device = NULL;
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audio_device = getenv("AUDIODEV");
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if (audio_device != NULL) {
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if ((sun_audio_fd = open(audio_device, O_WRONLY)) < 0) {
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SHOW("wave_init() could not open: %s (%d)\n",
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audio_device, sun_audio_fd);
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}
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}
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if (sun_audio_fd < 0) {
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if ((sun_audio_fd = open(sun_audio_device, O_WRONLY)) < 0) {
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SHOW("wave_init() could not open: %s (%d)\n",
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sun_audio_device, sun_audio_fd);
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}
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}
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SHOW("wave_init() sun_audio_fd: %d\n", sun_audio_fd);
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if (sun_audio_fd < 0) {
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return;
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}
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ioctl(sun_audio_fd, AUDIO_GETINFO, &ainfo);
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SHOW("wave_init() play buffer size: %d\n", ainfo.play.buffer_size);
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ainfo.play.encoding = AUDIO_ENCODING_LINEAR;
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ainfo.play.channels = 1;
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ainfo.play.sample_rate = SAMPLE_RATE;
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ainfo.play.precision = SAMPLE_SIZE;
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if (ioctl(sun_audio_fd, AUDIO_SETINFO, &ainfo) == -1) {
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SHOW("wave_init() failed to set audio params: %s\n", strerror(errno));
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close(sun_audio_fd);
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return;
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}
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}
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//>
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// wave_open
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//
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// DESCRIPTION:
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//
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// opens the audio subsystem given a specific API (e.g., "alsa",
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// "oss", ...). We ignore the_api and just return the sun_audio_fd we
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// opened in wave_init. This return value will be passed in as the
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// theHandler parameter in all other methods.
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//
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// PARAMETERS:
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//
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// the_api: "alsa", "oss" (ignored)
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//
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// GLOBALS USED/MODIFIED:
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//
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// sun_audio_fd: used as return value
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//
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// RETURNS:
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//
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// sun_audio_fd opened in wave_init, which is passed in as theHandler
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// parameter in all other methods
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//
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//<wave_open
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void* wave_open(const char* the_api)
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{
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ENTER("wave_open");
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return((void*) sun_audio_fd);
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}
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//>
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// wave_write
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//
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// DESCRIPTION:
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//
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// Meant to be asynchronous, it supplies the wave sample to the lower
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// audio layer and returns. The sample is played later on. [[[WDW -
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// we purposely do not open the audio device as non-blocking because
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// managing that would be a pain. So, we rely a lot upon fifo.cpp and
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// event.cpp to not overload us, allowing us to get away with a
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// blocking write. event.cpp:polling_thread in particular appears to
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// use get_remaining_time to prevent flooding.]]]
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//
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// PARAMETERS:
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//
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// theHandler: the audio device file descriptor
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// theMono16BitsWaveBuffer: the audio data
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// theSize: the number of bytes (not 16-bit samples)
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//
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// GLOBALS USED/MODIFIED:
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//
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// total_samples_sent: modified based upon 16-bit samples sent
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//
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// RETURNS:
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//
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// the number of bytes (not 16-bit samples) sent
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//
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//<wave_write
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size_t wave_write(void* theHandler,
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char* theMono16BitsWaveBuffer,
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size_t theSize)
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{
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size_t num;
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ENTER("wave_write");
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if (my_callback_is_output_enabled && (0==my_callback_is_output_enabled())) {
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SHOW_TIME("wave_write > my_callback_is_output_enabled: no!");
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return 0;
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}
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num = write((int) theHandler, theMono16BitsWaveBuffer, theSize);
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// Keep track of the total number of samples sent -- we use this in
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// wave_get_read_position and also use it to help calculate the
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// total_samples_skipped in wave_close.
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//
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total_samples_sent += num / 2;
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if (num < theSize) {
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SHOW("ERROR: wave_write only wrote %d of %d bytes\n", num, theSize);
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} else {
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SHOW("wave_write wrote %d bytes\n", theSize);
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}
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SHOW_TIME("wave_write > LEAVE");
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return num;
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}
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//>
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// wave_close
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//
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// DESCRIPTION:
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//
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// Does what SADA normally would call a flush, which means to cease
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// all audio production in progress and throw any remaining audio
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// away. [[[WDW - see comment in wave_flush.]]]
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//
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// PARAMETERS:
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//
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// theHandler: the audio device file descriptor
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//
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// GLOBALS USED/MODIFIED:
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//
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// last_play_position: modified to reflect play position the last time
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// this method was called
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// total_samples_sent: used to help calculate total_samples_skipped
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// total_samples_skipped: modified to hold the total number of 16-bit
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// samples sent to wave_write, but which were
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// never played
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// sun_audio_fd: used because some calls to wave_close seem to
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// pass a NULL for theHandler for some odd reason
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//
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// RETURNS:
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//
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// The result of the ioctl call (non-0 means failure)
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//
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//<wave_close
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int wave_close(void* theHandler)
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{
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int ret;
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audio_info_t ainfo;
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int audio_fd = (int) theHandler;
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if (!audio_fd) {
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audio_fd = sun_audio_fd;
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}
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ENTER("wave_close");
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// [[[WDW: maybe do a pause/resume ioctl???]]]
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ret = ioctl(audio_fd, I_FLUSH, FLUSHRW);
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ioctl(audio_fd, AUDIO_GETINFO, &ainfo);
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// Calculate the number of samples that won't get
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// played. We also keep track of the last_play_position
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// because wave_close can be called multiple times
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// before another call to wave_write.
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//
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if (last_play_position != ainfo.play.samples) {
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last_play_position = ainfo.play.samples;
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total_samples_skipped = total_samples_sent - last_play_position;
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}
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SHOW_TIME("wave_close > LEAVE");
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return ret;
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}
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//>
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// wave_is_busy
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//
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// DESCRIPTION:
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//
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// Returns a non-0 value if audio is being played.
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//
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// PARAMETERS:
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//
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// theHandler: the audio device file descriptor
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//
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// GLOBALS USED/MODIFIED:
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//
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// sun_audio_fd: used because some calls to wave_is_busy seem to
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// pass a NULL for theHandler for some odd reason
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//
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// RETURNS:
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//
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// A non-0 value if audio is being played
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//
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//<wave_is_busy
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int wave_is_busy(void* theHandler)
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{
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audio_info_t ainfo;
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int audio_fd = (int) theHandler;
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if (!audio_fd) {
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audio_fd = sun_audio_fd;
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}
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ENTER("wave_is_busy");
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ioctl(audio_fd, AUDIO_GETINFO, &ainfo);
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SHOW("wave_is_busy: active=%d", ainfo.play.active);
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SHOW_TIME("wave_is_busy > LEAVE");
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return ainfo.play.active;
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}
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//>
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// wave_terminate
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//
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// DESCRIPTION:
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//
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// Used to end our session with eSpeak.
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//
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// GLOBALS USED/MODIFIED:
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//
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// sun_audio_fd: modified - closed and set to -1
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//
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//<wave_terminate
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void wave_terminate()
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{
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ENTER("wave_terminate");
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close(sun_audio_fd);
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sun_audio_fd = -1;
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SHOW_TIME("wave_terminate > LEAVE");
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}
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//>
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// wave_flush
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//
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// DESCRIPTION:
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//
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// Appears to want to tell the audio subsystem to make sure it plays
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// the audio. In our case, the system is already doing this, so this
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// is basically a no-op. [[[WDW - if you do a drain, you block, so
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// don't do that. In addition the typical SADA notion of flush is
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// currently handled by wave_close. I think this is most likely just
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// terminology conflict between eSpeak and SADA.]]]
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//
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// PARAMETERS:
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//
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// theHandler: the audio device file descriptor
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//
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//<wave_flush
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void wave_flush(void* theHandler)
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{
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ENTER("wave_flush");
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//ioctl((int) theHandler, AUDIO_DRAIN, 0);
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SHOW_TIME("wave_flush > LEAVE");
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}
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//>
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// wave_set_callback_is_output_enabled
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//
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// DESCRIPTION:
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//
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// Sets the callback to call from wave_write before it sends data to
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// be played. It helps wave_write determine if the data should be
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// thrown away or not.
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//
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// PARAMETERS:
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//
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// cb: the callback to call from wave_write
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//
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//<wave_set_callback_is_output_enabled
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void wave_set_callback_is_output_enabled(t_wave_callback* cb)
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{
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my_callback_is_output_enabled = cb;
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}
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||||
//>
|
||||
// wave_test_get_write_buffer
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//
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||||
// DESCRIPTION:
|
||||
//
|
||||
// Unnecessary and is used for debug output from
|
||||
// speak_lib.cpp:dispatch_audio.
|
||||
//
|
||||
// RETURNS:
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||||
//
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// NULL
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||||
//
|
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//<wave_test_get_write_buffer
|
||||
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||||
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.
|
||||
//
|
||||
//<wave_get_read_position
|
||||
|
||||
uint32_t wave_get_read_position(void* theHandler)
|
||||
{
|
||||
audio_info_t ainfo;
|
||||
ENTER("wave_get_read_position");
|
||||
ioctl((int) theHandler, AUDIO_GETINFO, &ainfo);
|
||||
SHOW("wave_get_read_position: %d\n", ainfo.play.samples);
|
||||
SHOW_TIME("wave_get_read_position > LEAVE");
|
||||
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.]]]
|
||||
//
|
||||
//<wave_get_write_position
|
||||
|
||||
uint32_t wave_get_write_position(void* theHandler)
|
||||
{
|
||||
ENTER("wave_get_write_position");
|
||||
SHOW("wave_get_write_position: %d\n", total_samples_sent);
|
||||
SHOW_TIME("wave_get_write_position > LEAVE");
|
||||
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.
|
||||
//
|
||||
//<wave_get_remaining_time
|
||||
|
||||
int wave_get_remaining_time(uint32_t sample, uint32_t* time)
|
||||
{
|
||||
uint32_t a_time=0;
|
||||
uint32_t actual_index;
|
||||
|
||||
audio_info_t ainfo;
|
||||
ENTER("wave_get_remaining_time");
|
||||
if (!time) {
|
||||
return(-1);
|
||||
SHOW_TIME("wave_get_remaining_time > LEAVE");
|
||||
}
|
||||
|
||||
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) / SAMPLE_RATE;
|
||||
*time = (uint32_t) a_time;
|
||||
}
|
||||
SHOW("wave_get_remaining_time for %d: %d\n", sample, *time);
|
||||
SHOW_TIME("wave_get_remaining_time > LEAVE");
|
||||
return 0;
|
||||
}
|
||||
|
||||
#else
|
||||
// notdef USE_SADA
|
||||
|
||||
void wave_init() {}
|
||||
void* wave_open(const char* the_api) {return (void *)1;}
|
||||
size_t wave_write(void* theHandler, char* theMono16BitsWaveBuffer, size_t theSize) {return theSize;}
|
||||
int wave_close(void* theHandler) {return 0;}
|
||||
int wave_is_busy(void* theHandler) {return 0;}
|
||||
void wave_terminate() {}
|
||||
uint32_t wave_get_read_position(void* theHandler) {return 0;}
|
||||
uint32_t wave_get_write_position(void* theHandler) {return 0;}
|
||||
void wave_flush(void* theHandler) {}
|
||||
typedef int (t_wave_callback)(void);
|
||||
void wave_set_callback_is_output_enabled(t_wave_callback* cb) {}
|
||||
extern void* wave_test_get_write_buffer() {return NULL;}
|
||||
|
||||
int wave_get_remaining_time(uint32_t sample, uint32_t* time)
|
||||
{
|
||||
if (!time) return(-1);
|
||||
*time = (uint32_t)0;
|
||||
return 0;
|
||||
}
|
||||
|
||||
#endif // of USE_PORTAUDIO
|
||||
|
||||
//>
|
||||
//<clock_gettime2, add_time_in_ms
|
||||
|
||||
void clock_gettime2(struct timespec *ts)
|
||||
{
|
||||
struct timeval tv;
|
||||
|
||||
if (!ts)
|
||||
{
|
||||
return;
|
||||
}
|
||||
|
||||
assert (gettimeofday(&tv, NULL) != -1);
|
||||
ts->tv_sec = tv.tv_sec;
|
||||
ts->tv_nsec = tv.tv_usec*1000;
|
||||
}
|
||||
|
||||
void add_time_in_ms(struct timespec *ts, int time_in_ms)
|
||||
{
|
||||
if (!ts)
|
||||
{
|
||||
return;
|
||||
}
|
||||
|
||||
uint64_t t_ns = (uint64_t)ts->tv_nsec + 1000000 * (uint64_t)time_in_ms;
|
||||
while(t_ns >= ONE_BILLION)
|
||||
{
|
||||
SHOW("event > add_time_in_ms ns: %d sec %Lu nsec \n", ts->tv_sec, t_ns);
|
||||
ts->tv_sec += 1;
|
||||
t_ns -= ONE_BILLION;
|
||||
}
|
||||
ts->tv_nsec = (long int)t_ns;
|
||||
}
|
||||
|
||||
#endif // USE_ASYNC
|
||||
|
||||
//>
|
||||
Reference in New Issue
Block a user