Version bump.

(Portage version: 2.1.4)

1 files changed, 333 insertions, 0 deletions

diff --git a/app-accessibility/flite/files/flite-1.3-alsa-support.patch b/app-accessibility/flite/files/flite-1.3-alsa-support.patch
new file mode 100644
index 000000000000..9d3415e3fe4b
--- /dev/null
+++ b/app-accessibility/flite/files/flite-1.3-alsa-support.patch
@@ -0,0 +1,333 @@
+diff -uNr flite-1.3-release/configure.in flite-1.3-release-mod/configure.in
+--- flite-1.3-release/configure.in	2005-08-13 13:43:21.000000000 +0200
++++ flite-1.3-release-mod/configure.in	2006-11-13 21:16:27.000000000 +0200
+@@ -206,10 +206,10 @@
+ AC_CHECK_HEADER(sys/audioio.h,
+               [AUDIODRIVER="sun"
+                AUDIODEFS=-DCST_AUDIO_SUNOS])
+-dnl AC_CHECK_HEADER(sys/asoundlib.h,
+-dnl              [AUDIODRIVER="alsa"
+-dnl	       AUDIODEFS=-DCST_AUDIO_ALSA
+-dnl               AUDIOLIBS=-lasound])
++AC_CHECK_HEADER(alsa/asoundlib.h,
++              [AUDIODRIVER="alsa"
++	       AUDIODEFS=-DCST_AUDIO_ALSA
++               AUDIOLIBS=-lasound])
+ AC_CHECK_HEADER(mmsystem.h,
+ 	      [AUDIODRIVER="wince"
+ 	       AUDIODEFS=-DCST_AUDIO_WINCE
+diff -uNr flite-1.3-release/src/audio/au_alsa.c flite-1.3-release-mod/src/audio/au_alsa.c
+--- flite-1.3-release/src/audio/au_alsa.c	1970-01-01 02:00:00.000000000 +0200
++++ flite-1.3-release-mod/src/audio/au_alsa.c	2006-11-13 21:16:54.000000000 +0200
+@@ -0,0 +1,311 @@
++/*************************************************************************/
++/*                                                                       */
++/*                  Language Technologies Institute                      */
++/*                     Carnegie Mellon University                        */
++/*                        Copyright (c) 2000                             */
++/*                        All Rights Reserved.                           */
++/*                                                                       */
++/*  Permission is hereby granted, free of charge, to use and distribute  */
++/*  this software and its documentation without restriction, including   */
++/*  without limitation the rights to use, copy, modify, merge, publish,  */
++/*  distribute, sublicense, and/or sell copies of this work, and to      */
++/*  permit persons to whom this work is furnished to do so, subject to   */
++/*  the following conditions:                                            */
++/*   1. The code must retain the above copyright notice, this list of    */
++/*      conditions and the following disclaimer.                         */
++/*   2. Any modifications must be clearly marked as such.                */
++/*   3. Original authors' names are not deleted.                         */
++/*   4. The authors' names are not used to endorse or promote products   */
++/*      derived from this software without specific prior written        */
++/*      permission.                                                      */
++/*                                                                       */
++/*  CARNEGIE MELLON UNIVERSITY AND THE CONTRIBUTORS TO THIS WORK         */
++/*  DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING      */
++/*  ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT   */
++/*  SHALL CARNEGIE MELLON UNIVERSITY NOR THE CONTRIBUTORS BE LIABLE      */
++/*  FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES    */
++/*  WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN   */
++/*  AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION,          */
++/*  ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF       */
++/*  THIS SOFTWARE.                                                       */
++/*                                                                       */
++/*********************************************************************** */
++/*             Author:  Lukas Loehrer (                                  */
++/*               Date:  January 2005                                     */
++/*************************************************************************/
++/*                                                                       */
++/*  Native access to alsa audio devices on Linux                         */
++/*  Tested with libasound version 1.0.10                                 */
++/*************************************************************************/
++
++#include <stdlib.h>
++#include <unistd.h>
++#include <sys/types.h>
++#include <assert.h>
++#include <errno.h>
++
++#include "cst_string.h"
++#include "cst_wave.h"
++#include "cst_audio.h"
++
++#include <alsa/asoundlib.h>
++
++
++/*static char *pcm_dev_name = "hw:0,0"; */
++static char *pcm_dev_name ="default";
++
++static inline void print_pcm_state(snd_pcm_t *handle, char *msg)
++{
++  fprintf(stderr, "PCM state at %s = %s\n", msg,
++		  snd_pcm_state_name(snd_pcm_state(handle)));
++}
++
++cst_audiodev *audio_open_alsa(int sps, int channels, cst_audiofmt fmt)
++{
++  cst_audiodev *ad;
++  unsigned 	int real_rate;
++  int err;
++
++  /* alsa specific stuff */
++  snd_pcm_t *pcm_handle;          
++  snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
++  snd_pcm_hw_params_t *hwparams;
++  snd_pcm_format_t format;
++  snd_pcm_access_t access = SND_PCM_ACCESS_RW_INTERLEAVED;
++
++  /* Allocate the snd_pcm_hw_params_t structure on the stack. */
++  snd_pcm_hw_params_alloca(&hwparams);
++
++  /* Open pcm device */
++  err = snd_pcm_open(&pcm_handle, pcm_dev_name, stream, 0);
++  if (err < 0) 
++  {
++	cst_errmsg("audio_open_alsa: failed to open audio device %s. %s\n",
++			   pcm_dev_name, snd_strerror(err));
++	return NULL;
++  }
++
++  /* Init hwparams with full configuration space */
++  err = snd_pcm_hw_params_any(pcm_handle, hwparams);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to get hardware parameters from audio device. %s\n", snd_strerror(err));
++	return NULL;
++  }
++
++  /* Set access mode */
++  err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, access);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set access mode. %s.\n", snd_strerror(err));
++	return NULL;
++  }
++
++  /* Determine matching alsa sample format */
++  /* This could be implemented in a more */
++  /* flexible way (byte order conversion). */
++  switch (fmt)
++  {
++  case CST_AUDIO_LINEAR16:
++	if (CST_LITTLE_ENDIAN)
++	  format = SND_PCM_FORMAT_S16_LE;
++	else
++	  format = SND_PCM_FORMAT_S16_BE;
++	break;
++  case CST_AUDIO_LINEAR8:
++	format = SND_PCM_FORMAT_U8;
++	break;
++  case CST_AUDIO_MULAW:
++	format = SND_PCM_FORMAT_MU_LAW;
++	break;
++  default:
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to find suitable format.\n");
++	return NULL;
++	break;
++  }
++
++  /* Set samble format */
++  err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, format);
++  if (err <0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set format. %s.\n", snd_strerror(err));
++	return NULL;
++  }
++
++  /* Set sample rate near the disired rate */
++  real_rate = sps;
++  err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &real_rate, 0);
++  if (err < 0)   
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set sample rate near %d. %s.\n", sps, snd_strerror(err));
++	return NULL;
++  }
++  /*FIXME:  This is probably too strict */
++  assert(sps == real_rate);
++
++  /* Set number of channels */
++  assert(channels >0);
++  err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, channels);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set number of channels to %d. %s.\n", channels, snd_strerror(err));
++	return NULL;
++  }
++
++  /* Commit hardware parameters */
++  err = snd_pcm_hw_params(pcm_handle, hwparams);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set hw parameters. %s.\n", snd_strerror(err));
++	return NULL;
++  }
++
++  /* Make sure the device is ready to accept data */
++  assert(snd_pcm_state(pcm_handle) == SND_PCM_STATE_PREPARED);
++
++  /* Write hardware parameters to flite audio device data structure */
++  ad = cst_alloc(cst_audiodev, 1);
++  assert(ad != NULL);
++  ad->real_sps = ad->sps = sps;
++  ad->real_channels = ad->channels = channels;
++  ad->real_fmt = ad->fmt = fmt;
++  ad->platform_data = (void *) pcm_handle;
++
++  return ad;
++}
++
++int audio_close_alsa(cst_audiodev *ad)
++{
++  int result;
++  snd_pcm_t *pcm_handle;
++
++  if (ad == NULL)
++	return 0;
++
++  pcm_handle = (snd_pcm_t *) ad->platform_data;
++  result = snd_pcm_close(pcm_handle);
++  if (result < 0)
++  {
++	cst_errmsg("audio_close_alsa: Error: %s.\n", snd_strerror(result));
++  }
++  cst_free(ad);
++  return result;
++}
++
++/* Returns zero if recovery was successful. */
++static int recover_from_error(snd_pcm_t *pcm_handle, ssize_t res)
++{
++  if (res == -EPIPE) /* xrun */
++  {
++	res = snd_pcm_prepare(pcm_handle);
++	if (res < 0) 
++	{
++	  /* Failed to recover from xrun */
++	  cst_errmsg("recover_from_write_error: failed to recover from xrun. %s\n.", snd_strerror(res));
++	  return res;
++	}
++  } 
++  else if (res == -ESTRPIPE) /* Suspend */
++  {
++	while ((res = snd_pcm_resume(pcm_handle)) == -EAGAIN) 
++	{
++	  snd_pcm_wait(pcm_handle, 1000);
++	}
++	if (res < 0) 
++	{
++	  res = snd_pcm_prepare(pcm_handle);
++	  if (res <0) 
++	  {
++		/* Resume failed */
++		cst_errmsg("audio_recover_from_write_error: failed to resume after suspend. %s\n.", snd_strerror(res));
++		return res;
++	  }
++	}
++  } 
++  else if (res < 0) 
++  {
++	/* Unknown failure */
++	cst_errmsg("audio_recover_from_write_error: %s.\n", snd_strerror(res));
++	return res;
++  }
++  return 0;
++}
++
++int audio_write_alsa(cst_audiodev *ad, void *samples, int num_bytes)
++{
++  size_t frame_size;
++  ssize_t num_frames, res;
++  snd_pcm_t *pcm_handle;
++  char *buf = (char *) samples;
++
++  /* Determine frame size in bytes */
++  frame_size  = audio_bps(ad->real_fmt) * ad->real_channels;
++  /* Require that only complete frames are handed in */
++  assert((num_bytes % frame_size) == 0);
++  num_frames = num_bytes / frame_size;
++  pcm_handle = (snd_pcm_t *) ad->platform_data;
++
++  while (num_frames > 0) 
++  {
++	res = snd_pcm_writei(pcm_handle, buf, num_frames);
++	if (res != num_frames) 
++	{
++	  if (res == -EAGAIN || (res > 0 && res < num_frames)) 
++	  {
++		snd_pcm_wait(pcm_handle, 100);
++	  }
++	  else if (recover_from_error(pcm_handle, res) < 0) 
++	  {
++		return -1;
++	  }
++	}
++
++	if (res >0) 
++	{
++	  num_frames -= res;
++	  buf += res * frame_size;
++	}
++  }
++  return num_bytes;
++}
++
++int audio_flush_alsa(cst_audiodev *ad)
++{
++  int result;
++  result = snd_pcm_drain((snd_pcm_t *) ad->platform_data);
++  if (result < 0)
++  {
++	cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result));
++  }
++	/* Prepare device for more data */
++  result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data);
++if (result < 0)
++  {
++	cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result));
++  }
++  return result;
++}
++
++int audio_drain_alsa(cst_audiodev *ad)
++{
++  int result;
++  result = snd_pcm_drop((snd_pcm_t *) ad->platform_data);
++  if (result < 0)
++  {
++	cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result));
++  }
++/* Prepare device for more data */
++  result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data);
++if (result < 0)
++  {
++	cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result));
++  }
++  return result;
++}