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WIP audio in server
|
From: |
Jeremy Whiting |
|
Subject: |
WIP audio in server |
|
Date: |
Mon, 17 Aug 2015 13:36:40 -0600 |
Trevor,
On Mon, Aug 17, 2015 at 1:30 PM, Trevor Saunders <tbsaunde at tbsaunde.org>
wrote:
> On Thu, Aug 13, 2015 at 05:28:00PM -0600, Jeremy Whiting wrote:
>> Trevor,
>>
>> On Thu, Aug 13, 2015 at 4:43 PM, Trevor Saunders <tbsaunde at tbsaunde.org>
>> wrote:
>> > On Mon, Aug 10, 2015 at 08:21:00PM -0600, Jeremy Whiting wrote:
>> >> Ok, last patch of the day. Undone items:
>> >>
>> >> 1. Audio socket cleanup. Not sure what needs to be done here. Should
>> >> the socket files get deleted during shutdown, etc.
>> >
>> > is there some reason to use a socket instead of just pipe(2) then you
>> > wouldn't need to deal with this at all?
>>
>> I hadn't thought of that. I thought a unix socket would be easier
>> since in the near (maybe) future we will have the server spawn an
>> output module for each client when it is requested. I'm not sure if
>> using a pipe would make that easier or harder though.
>
> If you deal with audio in the main server process I don't think it would
> be any harder. The one thing I don't see how to handle is changing
> audio method while modules are running, but I'm not sure if that is or
> should be supported.
>
>> >> 2. Stopping audio (probably can be done from parse_stop in parse.c
>> >> 3. Play command use which is only used in generic.c
>> >
>> > iirc a couple other modules ivona and festival I think do there own
>> > thing where the synth is a separate process and does its own audio I
>> > think. I don't really have any great ideas here.
>>
>> Actually both are sending audio through module_tts_output which with
>> this patch sends it through the socket to the server to play.
>
> oh heh maybe I should have read the patch and stuff before talking.
>
>> >
>> > Have you considered using a process separate from the main server for
>> > audio output? I guess its not all that complicated, but it might allow
>> > us to sandbox a bit more of speech dispatcher.
>>
>> I haven't considered that. I'd rather not reinvent pulseaudio for our
>> own purposes if we can avoid it though.
>
> I'm not sure I'd call it reinventing Pulse, but it may be the case
> outputting audio is not complex enough to be worth it.
No worries. With the latest patch I sent all the audio is handled in
one thread. It's not very complex at all.
BR,
Jeremy
>
> Trev
>
>>
>> BR,
>> Jeremy
>>
>> >
>> > Trev
>> >
>> >>
>> >> BR,
>> >> Jeremy
>> >>
>> >> On Mon, Aug 10, 2015 at 7:05 PM, Jeremy Whiting <jpwhiting at kde.org>
>> >> wrote:
>> >> > Ok, another update. This time audio parameters are coming from the
>> >> > user's config (I think, GlobalFDSet getting initialized is a mystery
>> >> > to me so far because of macros and dotconf callbacks. Seems to work
>> >> > here though.
>> >> >
>> >> > BR,
>> >> > Jeremy
>> >> >
>> >> >
>> >> > On Mon, Aug 10, 2015 at 5:43 PM, Jeremy Whiting <jpwhiting at kde.org>
>> >> > wrote:
>> >> >> Ok, here's a working patch. A few things I'll fix before this is ready
>> >> >> for master though.
>> >> >>
>> >> >> 1. Audio initialization needs to come from the config files again.
>> >> >> 2. Audio socket cleanup.
>> >> >> 3. Documentation changes for this big change in how spd works.
>> >> >> 4. How to request the server stop playing audio (or maybe it knows
>> >> >> because it's telling the modules the same thing...
>> >> >> 5. Audio file playback in generic.c needs to open the file and send
>> >> >> the audio on the socket.
>> >> >>
>> >> >> BR,
>> >> >> Jeremy
>> >> >>
>> >> >>
>> >> >> On Wed, Aug 5, 2015 at 5:42 PM, Jeremy Whiting <jpwhiting at kde.org>
>> >> >> wrote:
>> >> >>> Ok, the audio in the server is in it's own thread now, and mostly all
>> >> >>> code is in server/audio.c to keep it separate from the other file
>> >> >>> descriptor handling for clients. I'm still getting pauses for some
>> >> >>> reason, but it is threaded now at least and works when run with -d
>> >> >>> also. I'll try to figure out where the pauses are coming from.
>> >> >>>
>> >> >>> BR,
>> >> >>> Jeremy
>> >> >>>
>> >> >>> On Fri, Jul 31, 2015 at 6:47 PM, Jeremy Whiting <jpwhiting at
>> >> >>> kde.org> wrote:
>> >> >>>> I've spent a bit of time wrestling with this code today and have
>> >> >>>> found
>> >> >>>> the following.
>> >> >>>>
>> >> >>>> If I don't initialize pulse by calling _pulse_open but only
>> >> >>>> initialize
>> >> >>>> once I have data to set the format it works mostly, though there are
>> >> >>>> pauses of silence in long phrases from espeak still.
>> >> >>>> If I do initialize pulse by calling _pulse_open in pulse_open and
>> >> >>>> also
>> >> >>>> reinitializing in pulse_play as required for audio rate changes etc.
>> >> >>>> it doesn't play anything somehow (it's getting stuck in
>> >> >>>> pa_simple_free
>> >> >>>> on a mutex somehow).
>> >> >>>>
>> >> >>>> Do I need to run a thread for every audio socket we attach in the
>> >> >>>> server? If so what should the thread's start_routine look like, just
>> >> >>>> while(1) and the callback audio_process_incoming is called when we
>> >> >>>> get
>> >> >>>> traffic on the fd ? Once this is working (and I clean up how we
>> >> >>>> initialize the audio to not hard coded values, but get the config
>> >> >>>> from
>> >> >>>> the config file) I can look at making the audio use the pulse async
>> >> >>>> api, but I want to get the proof of concept working as a first step.
>> >> >>>> My knowledge of pthreads seems to be blocking me at the moment
>> >> >>>> though.
>> >> >>>>
>> >> >>>> thanks,
>> >> >>>> Jeremy
>> >> >>>>
>> >> >>>>
>> >> >>>>
>> >> >>>> On Thu, Jul 30, 2015 at 7:42 PM, Jeremy Whiting <jpwhiting at
>> >> >>>> kde.org> wrote:
>> >> >>>>> Oops, I didn't see this until just now. At any rate I got that issue
>> >> >>>>> solved, now only playback failing (I verified I get samples in the
>> >> >>>>> server and I think it's initializing the AudioId properly since it's
>> >> >>>>> not NULL here. It seems to think it's playing from all the return
>> >> >>>>> values, but I hear no audio yet. I also wasn't sure what to do about
>> >> >>>>> sending the audio parameters just yet, so hard coded some based on
>> >> >>>>> my
>> >> >>>>> config here for now.
>> >> >>>>>
>> >> >>>>> BR,
>> >> >>>>> Jeremy
>> >> >>>>>
>> >> >>>>> On Thu, Jul 30, 2015 at 6:40 PM, Luke Yelavich
>> >> >>>>> <luke.yelavich at canonical.com> wrote:
>> >> >>>>>> On Fri, Jul 31, 2015 at 10:40:04AM AEST, Luke Yelavich wrote:
>> >> >>>>>>> On Fri, Jul 31, 2015 at 07:27:27AM AEST, Jeremy Whiting wrote:
>> >> >>>>>>> > Hey all,
>> >> >>>>>>> >
>> >> >>>>>>> > I'm implementing moving audio from the modules to the server
>> >> >>>>>>> > (and
>> >> >>>>>>> > modules will send audio data to the server on a unix socket).
>> >> >>>>>>> > I've got
>> >> >>>>>>> > the socket creation, and seem to have the ability to connect to
>> >> >>>>>>> > the
>> >> >>>>>>> > socket in the modules but it's hanging here when I try to run
>> >> >>>>>>> > spd-say
>> >> >>>>>>> > hello. Also I'm getting this in my speech-dispatcher.log as if
>> >> >>>>>>> > it's
>> >> >>>>>>> > trying to open a second audio connection from sd_espeak for some
>> >> >>>>>>> > reason when it hangs (and no log output after this):
>> >> >>>>>>> >
>> >> >>>>>>> > [Thu Jul 30 15:03:15 2015 : 829380] speechd: Adding audio
>> >> >>>>>>> > connection on socket 4
>> >> >>>>>>> > [Thu Jul 30 15:03:29 2015 : 105629] speechd: Adding module
>> >> >>>>>>> > on fd 28
>> >> >>>>>>> > [Thu Jul 30 15:03:29 2015 : 105654] speechd: Adding audio
>> >> >>>>>>> > connection on socket 4
>> >> >>>>>>> >
>> >> >>>>>>> > I'm probably doing something obviously wrong, but can't seem to
>> >> >>>>>>> > see
>> >> >>>>>>> > what at the moment though I've been beating my head against it
>> >> >>>>>>> > for a
>> >> >>>>>>> > while and debugging. Can you see anything obvious in my changes?
>> >> >>>>>>>
>> >> >>>>>>> Well, I wonder if what you have in speechd_audio_connection_new
>> >> >>>>>>> is correct. You make reference to module sockets and the server
>> >> >>>>>>> socket where clients connect, and not the audio socket.
>> >> >>>>>>
>> >> >>>>>> Helps if I attach the diff.
>> >> >>>>>>
>> >> >>>>>> Luke
>> >
>> >> From 51316a1237c09b50a1c84a44348a278c5982a056 Mon Sep 17 00:00:00 2001
>> >> From: Jeremy Whiting <jpwhiting at kde.org>
>> >> Date: Wed, 29 Jul 2015 18:28:03 -0600
>> >> Subject: [PATCH] Moving audio from modules to the server.
>> >>
>> >> Audio socket is created in the server.
>> >> Audio system is initialized in the server.
>> >> Modules connect to audio socket and send audio data (metadata first
>> >> separated by :, then samples).
>> >> Server receives AudioTrack deserializes the metadata and plays AudioTrack.
>> >> Updated speech-dispatcher.texi to remove section about AUDIO command
>> >> handling as it's gone.
>> >> ---
>> >> doc/speech-dispatcher.texi | 5 -
>> >> include/speechd_defines.h | 19 +-
>> >> src/Makefile.am | 2 +-
>> >> src/modules/Makefile.am | 30 +--
>> >> src/modules/espeak.c | 34 +--
>> >> src/modules/festival.c | 7 +-
>> >> src/modules/generic.c | 4 +-
>> >> src/modules/module_main.c | 16 +-
>> >> src/modules/module_utils.c | 184 +++++--------
>> >> src/modules/module_utils.h | 4 +-
>> >> src/modules/spd_audio.c | 321 ----------------------
>> >> src/modules/spd_audio.h | 46 ----
>> >> src/server/Makefile.am | 8 +-
>> >> src/server/audio.c | 647
>> >> +++++++++++++++++++++++++++++++++++++++++++++
>> >> src/server/audio.h | 60 +++++
>> >> src/server/module.c | 12 -
>> >> src/server/msg.h | 3 +-
>> >> src/server/output.c | 23 --
>> >> src/server/output.h | 1 -
>> >> src/server/server.c | 1 +
>> >> src/server/set.c | 14 +
>> >> src/server/set.h | 1 +
>> >> src/server/speechd.c | 21 +-
>> >> src/server/speechd.h | 9 +
>> >> 24 files changed, 877 insertions(+), 595 deletions(-)
>> >> delete mode 100644 src/modules/spd_audio.c
>> >> delete mode 100644 src/modules/spd_audio.h
>> >> create mode 100644 src/server/audio.c
>> >> create mode 100644 src/server/audio.h
>> >>
>> >> diff --git a/doc/speech-dispatcher.texi b/doc/speech-dispatcher.texi
>> >> index 88ca972..2d1a0ad 100755
>> >> --- a/doc/speech-dispatcher.texi
>> >> +++ b/doc/speech-dispatcher.texi
>> >> @@ -2752,11 +2752,6 @@ punctuation_some=NULL
>> >> 203 OK SETTINGS RECEIVED
>> >> @end example
>> >>
>> >> - at item AUDIO
>> >> -Audio has exactly the same structure as @code{SET}, but is transmitted
>> >> -only once immediatelly after @code{INIT} to transmit the requested audio
>> >> -parameters and tell the output module to open the audio device.
>> >> -
>> >> @item QUIT
>> >> Terminates the output module. It should send the response, deallocate
>> >> all the resources, close all descriptors, terminate all child
>> >> diff --git a/include/speechd_defines.h b/include/speechd_defines.h
>> >> index 3a544f0..d540c29 100644
>> >> --- a/include/speechd_defines.h
>> >> +++ b/include/speechd_defines.h
>> >> @@ -22,14 +22,15 @@
>> >> #ifndef SPEECHD_DEFINES_H
>> >> #define SPEECHD_DEFINES_H
>> >>
>> >> -#define SPD_ALLCLIENTS "ALL"
>> >> -#define SPD_SELF "SELF"
>> >> -#define SPD_VOLUME "VOLUME"
>> >> -#define SPD_PITCH "PITCH"
>> >> -#define SPD_PITCH_RANGE "PITCH_RANGE"
>> >> -#define SPD_RATE "RATE"
>> >> -#define SPD_LANGUAGE "LANGUAGE"
>> >> -#define SPD_OUTPUT_MODULE "OUTPUT_MODULE"
>> >> -#define SPD_SYNTHESIS_VOICE "SYNTHESIS_VOICE"
>> >> +#define NEWLINE "\r\n"
>> >> +#define SPD_ALLCLIENTS "ALL"
>> >> +#define SPD_SELF "SELF"
>> >> +#define SPD_VOLUME "VOLUME"
>> >> +#define SPD_PITCH "PITCH"
>> >> +#define SPD_PITCH_RANGE "PITCH_RANGE"
>> >> +#define SPD_RATE "RATE"
>> >> +#define SPD_LANGUAGE "LANGUAGE"
>> >> +#define SPD_OUTPUT_MODULE "OUTPUT_MODULE"
>> >> +#define SPD_SYNTHESIS_VOICE "SYNTHESIS_VOICE"
>> >>
>> >> #endif /* not ifndef SPEECHD_DEFINES_H */
>> >> diff --git a/src/Makefile.am b/src/Makefile.am
>> >> index 81d0690..8690889 100644
>> >> --- a/src/Makefile.am
>> >> +++ b/src/Makefile.am
>> >> @@ -1,4 +1,4 @@
>> >> ## Process this file with automake to produce Makefile.in
>> >>
>> >> -SUBDIRS=common server audio modules api clients tests
>> >> +SUBDIRS=common audio server modules api clients tests
>> >>
>> >> diff --git a/src/modules/Makefile.am b/src/modules/Makefile.am
>> >> index 9012a4b..7010e39 100644
>> >> --- a/src/modules/Makefile.am
>> >> +++ b/src/modules/Makefile.am
>> >> @@ -1,84 +1,74 @@
>> >> ## Process this file with automake to produce Makefile.in
>> >>
>> >> inc_local = -I$(top_srcdir)/include
>> >> -audio_SOURCES = spd_audio.c spd_audio.h
>> >> common_SOURCES = module_main.c module_utils.c module_utils.h
>> >> common_LDADD = $(SNDFILE_LIBS) $(DOTCONF_LIBS) $(GLIB_LIBS)
>> >> $(GTHREAD_LIBS)
>> >>
>> >> AM_CFLAGS = $(ERROR_CFLAGS)
>> >> AM_CPPFLAGS = $(inc_local) -DDATADIR=\"$(snddatadir)\" -D_GNU_SOURCE \
>> >> - -DPLUGIN_DIR="\"$(audiodir)\"" \
>> >> $(DOTCONF_CFLAGS) $(GLIB_CFLAGS) $(GTHREAD_CFLAGS) \
>> >> $(ibmtts_include) $(SNDFILE_CFLAGS)
>> >>
>> >> modulebin_PROGRAMS = sd_dummy sd_generic sd_festival sd_cicero
>> >>
>> >> -sd_dummy_SOURCES = dummy.c $(audio_SOURCES) $(common_SOURCES) \
>> >> +sd_dummy_SOURCES = dummy.c $(common_SOURCES) \
>> >> module_utils_addvoice.c
>> >> sd_dummy_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> $(common_LDADD)
>> >> dist_snddata_DATA = dummy-message.wav
>> >>
>> >> sd_festival_SOURCES = festival.c festival_client.c festival_client.h \
>> >> - $(audio_SOURCES) $(common_SOURCES)
>> >> + $(common_SOURCES)
>> >> sd_festival_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> $(common_LDADD) $(EXTRA_SOCKET_LIBS)
>> >>
>> >> -sd_generic_SOURCES = generic.c $(audio_SOURCES) $(common_SOURCES) \
>> >> +sd_generic_SOURCES = generic.c $(common_SOURCES) \
>> >> module_utils_addvoice.c
>> >> sd_generic_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> $(common_LDADD)
>> >>
>> >> -sd_cicero_SOURCES = cicero.c $(audio_SOURCES) $(common_SOURCES)
>> >> +sd_cicero_SOURCES = cicero.c $(common_SOURCES)
>> >> sd_cicero_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> $(common_LDADD)
>> >>
>> >> if flite_support
>> >> modulebin_PROGRAMS += sd_flite
>> >> -sd_flite_SOURCES = flite.c $(audio_SOURCES) $(common_SOURCES)
>> >> +sd_flite_SOURCES = flite.c $(common_SOURCES)
>> >> sd_flite_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> $(flite_kal) $(flite_basic) \
>> >> $(common_LDADD)
>> >> endif
>> >>
>> >> if ibmtts_support
>> >> modulebin_PROGRAMS += sd_ibmtts
>> >> -sd_ibmtts_SOURCES = ibmtts.c $(audio_SOURCES) $(common_SOURCES) \
>> >> +sd_ibmtts_SOURCES = ibmtts.c $(common_SOURCES) \
>> >> module_utils_addvoice.c
>> >> sd_ibmtts_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> -libmeci \
>> >> $(common_LDADD)
>> >> endif
>> >>
>> >> if espeak_support
>> >> modulebin_PROGRAMS += sd_espeak
>> >> -sd_espeak_SOURCES = espeak.c $(audio_SOURCES) $(common_SOURCES)
>> >> +sd_espeak_SOURCES = espeak.c $(common_SOURCES)
>> >> sd_espeak_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> -lespeak $(EXTRA_ESPEAK_LIBS) \
>> >> $(common_LDADD)
>> >> endif
>> >>
>> >> if ivona_support
>> >> modulebin_PROGRAMS += sd_ivona
>> >> -sd_ivona_SOURCES = ivona.c ivona_client.c ivona_client.h
>> >> $(audio_SOURCES) \
>> >> +sd_ivona_SOURCES = ivona.c ivona_client.c ivona_client.h \
>> >> $(common_SOURCES)
>> >> sd_ivona_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) \
>> >> -ldumbtts \
>> >> $(common_LDADD)
>> >> endif
>> >>
>> >> if pico_support
>> >> modulebin_PROGRAMS += sd_pico
>> >> -sd_pico_SOURCES = pico.c $(audio_SOURCES) $(common_SOURCES)
>> >> +sd_pico_SOURCES = pico.c $(common_SOURCES)
>> >> sd_pico_LDADD = $(top_builddir)/src/common/libcommon.la \
>> >> - $(audio_dlopen_modules) -lttspico \
>> >> + -lttspico \
>> >> $(common_LDADD)
>> >> endif
>> >> diff --git a/src/modules/espeak.c b/src/modules/espeak.c
>> >> index 48fc743..6fb9ffe 100644
>> >> --- a/src/modules/espeak.c
>> >> +++ b/src/modules/espeak.c
>> >> @@ -43,7 +43,6 @@
>> >> #endif
>> >>
>> >> /* Speech Dispatcher includes. */
>> >> -#include "spd_audio.h"
>> >> #include <speechd_types.h>
>> >> #include "module_utils.h"
>> >>
>> >> @@ -558,22 +557,23 @@ static void *_espeak_stop_or_pause(void *nothing)
>> >> pthread_cond_broadcast(&playback_queue_condition);
>> >> pthread_mutex_unlock(&playback_queue_mutex);
>> >>
>> >> - if (module_audio_id) {
>> >> - DBG("Espeak: Stopping audio.");
>> >> - ret = spd_audio_stop(module_audio_id);
>> >> - DBG_WARN(ret == 0,
>> >> - "spd_audio_stop returned non-zero value.");
>> >> - while
>> >> (is_thread_busy(&espeak_play_suspended_mutex)) {
>> >> - ret = spd_audio_stop(module_audio_id);
>> >> - DBG_WARN(ret == 0,
>> >> - "spd_audio_stop returned non-zero
>> >> value.");
>> >> - g_usleep(5000);
>> >> - }
>> >> - } else {
>> >> - while
>> >> (is_thread_busy(&espeak_play_suspended_mutex)) {
>> >> - g_usleep(5000);
>> >> - }
>> >> - }
>> >> + /* TODO: Add a way to request the server stop audio
>> >> playback */
>> >> +// if (module_audio_id) {
>> >> +// DBG("Espeak: Stopping audio.");
>> >> +// ret = spd_audio_stop(module_audio_id);
>> >> +// DBG_WARN(ret == 0,
>> >> +// "spd_audio_stop returned non-zero value.");
>> >> +// while
>> >> (is_thread_busy(&espeak_play_suspended_mutex)) {
>> >> +// ret = spd_audio_stop(module_audio_id);
>> >> +// DBG_WARN(ret == 0,
>> >> +// "spd_audio_stop returned non-zero
>> >> value.");
>> >> +// g_usleep(5000);
>> >> +// }
>> >> +// } else {
>> >> +// while
>> >> (is_thread_busy(&espeak_play_suspended_mutex)) {
>> >> +// g_usleep(5000);
>> >> +// }
>> >> +// }
>> >>
>> >> DBG("Espeak: Waiting for synthesis to stop.");
>> >> ret = espeak_Cancel();
>> >> diff --git a/src/modules/festival.c b/src/modules/festival.c
>> >> index df6d58a..39ab49e 100644
>> >> --- a/src/modules/festival.c
>> >> +++ b/src/modules/festival.c
>> >> @@ -431,9 +431,10 @@ int module_stop(void)
>> >> if (!festival_stop) {
>> >> pthread_mutex_lock(&sound_output_mutex);
>> >> festival_stop = 1;
>> >> - if (festival_speaking && module_audio_id) {
>> >> - spd_audio_stop(module_audio_id);
>> >> - }
>> >> + // TODO: Add a way for modules to request speech
>> >> stop maybe ?
>> >> +// if (festival_speaking && module_audio_id) {
>> >> +// spd_audio_stop(module_audio_id);
>> >> +// }
>> >> pthread_mutex_unlock(&sound_output_mutex);
>> >> }
>> >> }
>> >> diff --git a/src/modules/generic.c b/src/modules/generic.c
>> >> index 172b6ec..0360f0b 100644
>> >> --- a/src/modules/generic.c
>> >> +++ b/src/modules/generic.c
>> >> @@ -388,8 +388,8 @@ void *_generic_speak(void *nothing)
>> >> homedir =
>> >>
>> >> g_strdup("UNKNOWN_HOME_DIRECTORY");
>> >>
>> >> - play_command =
>> >> - spd_audio_get_playcmd(module_audio_id);
>> >> +// play_command =
>> >> +// spd_audio_get_playcmd(module_audio_id);
>> >> if (play_command == NULL) {
>> >> DBG("This audio backend has no
>> >> default play command; using \"play\"\n");
>> >> play_command = "play";
>> >> diff --git a/src/modules/module_main.c b/src/modules/module_main.c
>> >> index f53b053..49b73bc 100644
>> >> --- a/src/modules/module_main.c
>> >> +++ b/src/modules/module_main.c
>> >> @@ -31,7 +31,6 @@
>> >> #include <pthread.h>
>> >> #include <glib.h>
>> >> #include <dotconf.h>
>> >> -#include <ltdl.h>
>> >>
>> >> #include <spd_utils.h>
>> >> #include "module_utils.h"
>> >> @@ -77,10 +76,7 @@ int main(int argc, char *argv[])
>> >> char *configfilename = NULL;
>> >> char *status_info = NULL;
>> >>
>> >> - /* Initialize ltdl's list of preloaded audio backends. */
>> >> - LTDL_SET_PRELOADED_SYMBOLS();
>> >> module_num_dc_options = 0;
>> >> - module_audio_id = 0;
>> >>
>> >> if (argc >= 2) {
>> >> configfilename = g_strdup(argv[1]);
>> >> @@ -149,6 +145,16 @@ int main(int argc, char *argv[])
>> >> exit(1);
>> >> }
>> >>
>> >> + ret = module_audio_init(&status_info);
>> >> +
>> >> + if (ret != 0) {
>> >> + printf("399-%s\n", status_info);
>> >> + printf("%s\n", "399 ERR CANT INIT AUDIO");
>> >> + g_free(status_info);
>> >> + module_close();
>> >> + exit(1);
>> >> + }
>> >> +
>> >> printf("299-%s\n", status_info);
>> >> ret = printf("%s\n", "299 OK LOADED SUCCESSFULLY");
>> >>
>> >> @@ -190,8 +196,6 @@ int main(int argc, char *argv[])
>> >> else
>> >> PROCESS_CMD(SET, do_set)
>> >> else
>> >> - PROCESS_CMD(AUDIO, do_audio)
>> >> - else
>> >> PROCESS_CMD(LOGLEVEL, do_loglevel)
>> >> else
>> >> PROCESS_CMD_W_ARGS(DEBUG, do_debug)
>> >> diff --git a/src/modules/module_utils.c b/src/modules/module_utils.c
>> >> index 298274a..bde2257 100644
>> >> --- a/src/modules/module_utils.c
>> >> +++ b/src/modules/module_utils.c
>> >> @@ -26,17 +26,22 @@
>> >> #endif
>> >>
>> >> #include <sndfile.h>
>> >> +#include <sys/types.h>
>> >> +#include <sys/socket.h>
>> >> +#include <sys/un.h>
>> >>
>> >> #include <fdsetconv.h>
>> >> #include <spd_utils.h>
>> >> #include "module_utils.h"
>> >> -
>> >> -static char *module_audio_pars[10];
>> >> +#include <speechd_defines.h>
>> >>
>> >> extern char *module_index_mark;
>> >>
>> >> pthread_mutex_t module_stdout_mutex = PTHREAD_MUTEX_INITIALIZER;
>> >>
>> >> +/* Socket for sending audio data to */
>> >> +int audio_socket;
>> >> +
>> >> char *do_message(SPDMessageType msgtype)
>> >> {
>> >> int ret;
>> >> @@ -95,11 +100,6 @@ char *do_message(SPDMessageType msgtype)
>> >> msg_settings_old.voice_type = -1;
>> >> }
>> >>
>> >> - /* Volume is controlled by the synthesizer. Always play at normal
>> >> on audio device. */
>> >> - if (spd_audio_set_volume(module_audio_id, 85) < 0) {
>> >> - DBG("Can't set volume. audio not initialized?");
>> >> - }
>> >> -
>> >> ret = module_speak(msg->str, strlen(msg->str), msgtype);
>> >>
>> >> g_string_free(msg, 1);
>> >> @@ -267,91 +267,12 @@ char *do_set(void)
>> >> return g_strdup("401 ERROR INTERNAL"); /* Can't be reached */
>> >> }
>> >>
>> >> -#define SET_AUDIO_STR(name,idx) \
>> >> - if(!strcmp(cur_item, #name)){ \
>> >> - g_free(module_audio_pars[idx]); \
>> >> - if(!strcmp(cur_value, "NULL")) module_audio_pars[idx] =
>> >> NULL; \
>> >> - else module_audio_pars[idx] = g_strdup(cur_value); \
>> >> - }
>> >> -
>> >> -char *do_audio(void)
>> >> -{
>> >> - char *cur_item = NULL;
>> >> - char *cur_value = NULL;
>> >> - char *line = NULL;
>> >> - int ret;
>> >> - size_t n;
>> >> - int err = 0; /* Error status */
>> >> - char *status = NULL;
>> >> - char *msg;
>> >> -
>> >> - printf("207 OK RECEIVING AUDIO SETTINGS\n");
>> >> - fflush(stdout);
>> >> -
>> >> - while (1) {
>> >> - line = NULL;
>> >> - n = 0;
>> >> - ret = spd_getline(&line, &n, stdin);
>> >> - if (ret == -1) {
>> >> - err = 1;
>> >> - break;
>> >> - }
>> >> - if (!strcmp(line, ".\n")) {
>> >> - g_free(line);
>> >> - break;
>> >> - }
>> >> - if (!err) {
>> >> - cur_item = strtok(line, "=");
>> >> - if (cur_item == NULL) {
>> >> - err = 1;
>> >> - continue;
>> >> - }
>> >> - cur_value = strtok(NULL, "\n");
>> >> - if (cur_value == NULL) {
>> >> - err = 1;
>> >> - continue;
>> >> - }
>> >> -
>> >> - SET_AUDIO_STR(audio_output_method, 0)
>> >> - else
>> >> - SET_AUDIO_STR(audio_oss_device, 1)
>> >> - else
>> >> - SET_AUDIO_STR(audio_alsa_device, 2)
>> >> - else
>> >> - SET_AUDIO_STR(audio_nas_server, 3)
>> >> - else
>> >> - SET_AUDIO_STR(audio_pulse_server, 4)
>> >> - else
>> >> - SET_AUDIO_STR(audio_pulse_min_length, 5)
>> >> - else
>> >> - err = 2; /* Unknown parameter */
>> >> - }
>> >> - g_free(line);
>> >> - }
>> >> -
>> >> - if (err == 1)
>> >> - return g_strdup("302 ERROR BAD SYNTAX");
>> >> - if (err == 2)
>> >> - return g_strdup("303 ERROR INVALID PARAMETER OR VALUE");
>> >> -
>> >> - err = module_audio_init(&status);
>> >> -
>> >> - if (err == 0)
>> >> - msg = g_strdup_printf("203 OK AUDIO INITIALIZED");
>> >> - else
>> >> - msg = g_strdup_printf("300-%s\n300 UNKNOWN ERROR", status);
>> >> -
>> >> - g_free(status);
>> >> - return msg;
>> >> -}
>> >> -
>> >> #define SET_LOGLEVEL_NUM(name, cond) \
>> >> if(!strcmp(cur_item, #name)){ \
>> >> number = strtol(cur_value, &tptr, 10); \
>> >> if(!(cond)){ err = 2; continue; } \
>> >> if (tptr == cur_value){ err = 2; continue; } \
>> >> log_level = number; \
>> >> - spd_audio_set_loglevel(module_audio_id, number); \
>> >> }
>> >>
>> >> char *do_loglevel(void)
>> >> @@ -516,9 +437,6 @@ void do_quit(void)
>> >> printf("210 OK QUIT\n");
>> >> fflush(stdout);
>> >>
>> >> - spd_audio_close(module_audio_id);
>> >> - module_audio_id = NULL;
>> >> -
>> >> module_close();
>> >> return;
>> >> }
>> >> @@ -988,50 +906,74 @@ void *module_get_ht_option(GHashTable * hash_table,
>> >> const char *key)
>> >> return option;
>> >> }
>> >>
>> >> -int module_audio_init(char **status_info)
>> >> +/* Determine address for the unix socket */
>> >> +static char *_get_default_audio_unix_socket_name(void)
>> >> {
>> >> - char *error = 0;
>> >> - gchar **outputs;
>> >> - int i = 0;
>> >> + GString *socket_filename;
>> >> + char *h;
>> >> + const char *rundir = g_get_user_runtime_dir();
>> >> + socket_filename = g_string_new("");
>> >> + g_string_printf(socket_filename, "%s/speech-dispatcher/audio.sock",
>> >> + rundir);
>> >> + // Do not return glib string, but glibc string...
>> >> + h = strdup(socket_filename->str);
>> >> + g_string_free(socket_filename, 1);
>> >> + return h;
>> >> +}
>> >>
>> >> - DBG("Openning audio output system");
>> >> - if (NULL == module_audio_pars[0]) {
>> >> - *status_info =
>> >> +int module_audio_init(char **status_info)
>> >> +{
>> >> + /* Open connection to audio socket */
>> >> + char *str;
>> >> + char *socket_filename = _get_default_audio_unix_socket_name();
>> >> + int len;
>> >> + struct sockaddr_un server;
>> >> +
>> >> + if ((audio_socket = socket(AF_UNIX, SOCK_STREAM, 0)) == -1) {
>> >> + *status_info =
>> >> g_strdup
>> >> - ("Sound output method specified in configuration not
>> >> supported. "
>> >> - "Please choose 'oss', 'alsa', 'nas', 'libao' or
>> >> 'pulse'.");
>> >> + ("Unable to create socket to send audio data");
>> >> return -1;
>> >> }
>> >> -
>> >> - outputs = g_strsplit(module_audio_pars[0], ",", 0);
>> >> - while (NULL != outputs[i]) {
>> >> - module_audio_id =
>> >> - spd_audio_open(outputs[i], (void
>> >> **)&module_audio_pars[1],
>> >> - &error);
>> >> - if (module_audio_id) {
>> >> - DBG("Using %s audio output method", outputs[i]);
>> >> - g_strfreev(outputs);
>> >> - *status_info =
>> >> - g_strdup("audio initialized successfully.");
>> >> - return 0;
>> >> - }
>> >> - i++;
>> >> +
>> >> + server.sun_family = AF_UNIX;
>> >> + strcpy(server.sun_path, socket_filename);
>> >> + len = strlen(server.sun_path) + sizeof(server.sun_family);
>> >> + if (connect(audio_socket, (struct sockaddr *)&server, len) == -1) {
>> >> + *status_info =
>> >> + g_strdup_printf
>> >> + ("Unable to connect to server socket at %s",
>> >> socket_filename);
>> >> + return -1;
>> >> }
>> >>
>> >> - *status_info =
>> >> - g_strdup_printf("Opening sound device failed. Reason: %s. ",
>> >> error);
>> >> - g_free(error); /* g_malloc'ed, in spd_audio_open. */
>> >> -
>> >> - g_strfreev(outputs);
>> >> - return -1;
>> >> + str = g_strdup("ACK"NEWLINE);
>> >> + if (send(audio_socket, str, strlen(str), 0) == -1) {
>> >> + g_free (str);
>> >> + *status_info =
>> >> + g_strdup_printf
>> >> + ("Unable to send ACK on audio socket %s",
>> >> socket_filename);
>> >> + return -1;
>> >> + }
>> >> + g_free(str);
>> >>
>> >> + return 0;
>> >> }
>> >>
>> >> int module_tts_output(AudioTrack track, AudioFormat format)
>> >> {
>> >> -
>> >> - if (spd_audio_play(module_audio_id, track, format) < 0) {
>> >> - DBG("Can't play track for unknown reason.");
>> >> + /* Send audiotrack data to the socket */
>> >> + char *metadata = g_strdup_printf("%d:%d:%d:%d:%d"NEWLINE,
>> >> + format,
>> >> + track.bits,
>> >> + track.num_channels,
>> >> + track.sample_rate,
>> >> + track.num_samples);
>> >> + if (send(audio_socket, metadata, strlen(metadata), 0) == -1) {
>> >> + DBG("Can't send audiotrack metadata for some reason.");
>> >> + return -1;
>> >> + }
>> >> + if (send(audio_socket, track.samples, track.num_samples *
>> >> sizeof(signed short), 0) == -1) {
>> >> + DBG("Can't send audio samples for some reason.");
>> >> return -1;
>> >> }
>> >> return 0;
>> >> diff --git a/src/modules/module_utils.h b/src/modules/module_utils.h
>> >> index 7483930..895db80 100644
>> >> --- a/src/modules/module_utils.h
>> >> +++ b/src/modules/module_utils.h
>> >> @@ -41,12 +41,10 @@
>> >> #include <sys/ipc.h>
>> >>
>> >> #include <speechd_types.h>
>> >> -#include "spd_audio.h"
>> >> +#include <spd_audio_plugin.h>
>> >>
>> >> int log_level;
>> >>
>> >> -AudioID *module_audio_id;
>> >> -
>> >> SPDMsgSettings msg_settings;
>> >> SPDMsgSettings msg_settings_old;
>> >>
>> >> diff --git a/src/modules/spd_audio.c b/src/modules/spd_audio.c
>> >> deleted file mode 100644
>> >> index 9bf8e37..0000000
>> >> --- a/src/modules/spd_audio.c
>> >> +++ /dev/null
>> >> @@ -1,321 +0,0 @@
>> >> -
>> >> -/*
>> >> - * spd_audio.c -- Spd Audio Output Library
>> >> - *
>> >> - * Copyright (C) 2004, 2006 Brailcom, o.p.s.
>> >> - *
>> >> - * This is free software; you can redistribute it and/or modify it under
>> >> the
>> >> - * terms of the GNU Lesser General Public License as published by the
>> >> Free
>> >> - * Software Foundation; either version 2.1, or (at your option) any later
>> >> - * version.
>> >> - *
>> >> - * This software 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 Lesser General Public
>> >> License
>> >> - * along with this package; see the file COPYING. If not, write to the
>> >> Free
>> >> - * Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
>> >> - * 02110-1301, USA.
>> >> - *
>> >> - * $Id: spd_audio.c,v 1.21 2008-06-09 10:29:12 hanke Exp $
>> >> - */
>> >> -
>> >> -/*
>> >> - * spd_audio is a simple realtime audio output library with the
>> >> capability of
>> >> - * playing 8 or 16 bit data, immediate stop and synchronization. This
>> >> library
>> >> - * currently provides OSS, NAS, ALSA and PulseAudio backend. The
>> >> available backends are
>> >> - * specified at compile-time using the directives WITH_OSS, WITH_NAS,
>> >> WITH_ALSA,
>> >> - * WITH_PULSE, WITH_LIBAO but the user program is allowed to switch
>> >> between them at run-time.
>> >> - */
>> >> -
>> >> -#ifdef HAVE_CONFIG_H
>> >> -#include <config.h>
>> >> -#endif
>> >> -
>> >> -#include "spd_audio.h"
>> >> -
>> >> -#include <stdio.h>
>> >> -#include <string.h>
>> >> -#include <fcntl.h>
>> >> -#include <sys/ioctl.h>
>> >> -#include <sys/time.h>
>> >> -#include <time.h>
>> >> -#include <unistd.h>
>> >> -#include <errno.h>
>> >> -
>> >> -#include <pthread.h>
>> >> -
>> >> -#include <glib.h>
>> >> -#include <ltdl.h>
>> >> -
>> >> -static int spd_audio_log_level;
>> >> -static lt_dlhandle lt_h;
>> >> -
>> >> -/* Dynamically load a library with RTLD_GLOBAL set.
>> >> -
>> >> - This is needed when a dynamically-loaded library has its own plugins
>> >> - that call into the parent library.
>> >> - Most of the credit for this function goes to Gary Vaughan.
>> >> -*/
>> >> -static lt_dlhandle my_dlopenextglobal(const char *filename)
>> >> -{
>> >> - lt_dlhandle handle = NULL;
>> >> - lt_dladvise advise;
>> >> -
>> >> - if (lt_dladvise_init(&advise))
>> >> - return handle;
>> >> -
>> >> - if (!lt_dladvise_ext(&advise) && !lt_dladvise_global(&advise))
>> >> - handle = lt_dlopenadvise(filename, advise);
>> >> -
>> >> - lt_dladvise_destroy(&advise);
>> >> - return handle;
>> >> -}
>> >> -
>> >> -/* Open the audio device.
>> >> -
>> >> - Arguments:
>> >> - type -- The requested device. Currently AudioOSS or AudioNAS.
>> >> - pars -- and array of pointers to parameters to pass to
>> >> - the device backend, terminated by a NULL pointer.
>> >> - See the source/documentation of each specific backend.
>> >> - error -- a pointer to the string where error description is
>> >> - stored in case of failure (returned AudioID == NULL).
>> >> - Otherwise will contain NULL.
>> >> -
>> >> - Return value:
>> >> - Newly allocated AudioID structure that can be passed to
>> >> - all other spd_audio functions, or NULL in case of failure.
>> >> -
>> >> -*/
>> >> -AudioID *spd_audio_open(char *name, void **pars, char **error)
>> >> -{
>> >> - AudioID *id;
>> >> - spd_audio_plugin_t const *p;
>> >> - spd_audio_plugin_t *(*fn) (void);
>> >> - gchar *libname;
>> >> - int ret;
>> >> -
>> >> - /* now check whether dynamic plugin is available */
>> >> - ret = lt_dlinit();
>> >> - if (ret != 0) {
>> >> - *error = (char *)g_strdup_printf("lt_dlinit() failed");
>> >> - return (AudioID *) NULL;
>> >> - }
>> >> -
>> >> - ret = lt_dlsetsearchpath(PLUGIN_DIR);
>> >> - if (ret != 0) {
>> >> - *error = (char *)g_strdup_printf("lt_dlsetsearchpath()
>> >> failed");
>> >> - return (AudioID *) NULL;
>> >> - }
>> >> -
>> >> - libname = g_strdup_printf(SPD_AUDIO_LIB_PREFIX "%s", name);
>> >> - lt_h = my_dlopenextglobal(libname);
>> >> - g_free(libname);
>> >> - if (NULL == lt_h) {
>> >> - *error =
>> >> - (char *)g_strdup_printf("Cannot open plugin %s. error:
>> >> %s",
>> >> - name, lt_dlerror());
>> >> - return (AudioID *) NULL;
>> >> - }
>> >> -
>> >> - fn = lt_dlsym(lt_h, SPD_AUDIO_PLUGIN_ENTRY_STR);
>> >> - if (NULL == fn) {
>> >> - *error = (char *)g_strdup_printf("Cannot find symbol %s",
>> >> -
>> >> SPD_AUDIO_PLUGIN_ENTRY_STR);
>> >> - return (AudioID *) NULL;
>> >> - }
>> >> -
>> >> - p = fn();
>> >> - if (p == NULL || p->name == NULL) {
>> >> - *error = (char *)g_strdup_printf("plugin %s not found",
>> >> name);
>> >> - return (AudioID *) NULL;
>> >> - }
>> >> -
>> >> - id = p->open(pars);
>> >> - if (id == NULL) {
>> >> - *error =
>> >> - (char *)g_strdup_printf("Couldn't open %s plugin",
>> >> name);
>> >> - return (AudioID *) NULL;
>> >> - }
>> >> -
>> >> - id->function = p;
>> >> -#if defined(BYTE_ORDER) && (BYTE_ORDER == BIG_ENDIAN)
>> >> - id->format = SPD_AUDIO_BE;
>> >> -#else
>> >> - id->format = SPD_AUDIO_LE;
>> >> -#endif
>> >> -
>> >> - *error = NULL;
>> >> -
>> >> - return id;
>> >> -}
>> >> -
>> >> -/* Play a track on the audio device (blocking).
>> >> -
>> >> - Arguments:
>> >> - id -- the AudioID* of the device returned by spd_audio_open
>> >> - track -- a track to play (see spd_audio.h)
>> >> -
>> >> - Return value:
>> >> - 0 if everything is ok, a non-zero value in case of failure.
>> >> - See the particular backend documentation or source for the
>> >> - meaning of these non-zero values.
>> >> -
>> >> - Comment:
>> >> - spd_audio_play() is a blocking function. It returns exactly
>> >> - when the given track stopped playing. However, it's possible
>> >> - to safely interrupt it using spd_audio_stop() described below.
>> >> - (spd_audio_stop() needs to be called from another thread, obviously.)
>> >> -
>> >> -*/
>> >> -int spd_audio_play(AudioID * id, AudioTrack track, AudioFormat format)
>> >> -{
>> >> - int ret;
>> >> -
>> >> - if (id && id->function->play) {
>> >> - /* Only perform byte swapping if the driver in use has
>> >> given us audio in
>> >> - an endian format other than what the running CPU
>> >> supports. */
>> >> - if (format != id->format) {
>> >> - unsigned char *out_ptr, *out_end, c;
>> >> - out_ptr = (unsigned char *)track.samples;
>> >> - out_end =
>> >> - out_ptr +
>> >> - track.num_samples * 2 * track.num_channels;
>> >> - while (out_ptr < out_end) {
>> >> - c = out_ptr[0];
>> >> - out_ptr[0] = out_ptr[1];
>> >> - out_ptr[1] = c;
>> >> - out_ptr += 2;
>> >> - }
>> >> - }
>> >> - ret = id->function->play(id, track);
>> >> - } else {
>> >> - fprintf(stderr, "Play not supported on this device\n");
>> >> - return -1;
>> >> - }
>> >> -
>> >> - return ret;
>> >> -}
>> >> -
>> >> -/* Stop playing the current track on device id
>> >> -
>> >> -Arguments:
>> >> - id -- the AudioID* of the device returned by spd_audio_open
>> >> -
>> >> -Return value:
>> >> - 0 if everything is ok, a non-zero value in case of failure.
>> >> - See the particular backend documentation or source for the
>> >> - meaning of these non-zero values.
>> >> -
>> >> -Comment:
>> >> - spd_audio_stop() safely interrupts spd_audio_play() when called
>> >> - from another thread. It shouldn't cause any clicks or unwanted
>> >> - effects in the sound output.
>> >> -
>> >> - It's safe to call spd_audio_stop() even if the device isn't playing
>> >> - any track. In that case, it does nothing. However, there is a danger
>> >> - when using spd_audio_stop(). Since you must obviously do it from
>> >> - another thread than where spd_audio_play is running, you must make
>> >> - yourself sure that the device is still open and the id you pass it
>> >> - is valid and will be valid until spd_audio_stop returns. In other
>> >> words,
>> >> - you should use some mutex or other synchronization device to be sure
>> >> - spd_audio_close isn't called before or during spd_audio_stop
>> >> execution.
>> >> -*/
>> >> -
>> >> -int spd_audio_stop(AudioID * id)
>> >> -{
>> >> - int ret;
>> >> - if (id && id->function->stop) {
>> >> - ret = id->function->stop(id);
>> >> - } else {
>> >> - fprintf(stderr, "Stop not supported on this device\n");
>> >> - return -1;
>> >> - }
>> >> - return ret;
>> >> -}
>> >> -
>> >> -/* Close the audio device id
>> >> -
>> >> -Arguments:
>> >> - id -- the AudioID* of the device returned by spd_audio_open
>> >> -
>> >> -Return value:
>> >> - 0 if everything is ok, a non-zero value in case of failure.
>> >> -
>> >> -Comments:
>> >> -
>> >> - Please make sure no other spd_audio function with this device id
>> >> - is running in another threads. See spd_audio_stop() for detailed
>> >> - description of possible problems.
>> >> -*/
>> >> -int spd_audio_close(AudioID * id)
>> >> -{
>> >> - int ret = 0;
>> >> - if (id && id->function->close) {
>> >> - ret = (id->function->close(id));
>> >> - }
>> >> -
>> >> - if (NULL != lt_h) {
>> >> - lt_dlclose(lt_h);
>> >> - lt_h = NULL;
>> >> - lt_dlexit();
>> >> - }
>> >> -
>> >> - return ret;
>> >> -}
>> >> -
>> >> -/* Set volume for playing tracks on the device id
>> >> -
>> >> -Arguments:
>> >> - id -- the AudioID* of the device returned by spd_audio_open
>> >> - volume -- a value in the range <-100:100> where -100 means the
>> >> - least volume (probably silence), 0 the default volume
>> >> - and +100 the highest volume possible to make on that
>> >> - device for a single flow (i.e. not using mixer).
>> >> -
>> >> -Return value:
>> >> - 0 if everything is ok, a non-zero value in case of failure.
>> >> - See the particular backend documentation or source for the
>> >> - meaning of these non-zero values.
>> >> -
>> >> -Comments:
>> >> -
>> >> - In case of /dev/dsp, it's not possible to set volume for
>> >> - the particular flow. For that reason, the value 0 means
>> >> - the volume the track was recorded on and each smaller value
>> >> - means less volume (since this works by deviding the samples
>> >> - in the track by a constant).
>> >> -*/
>> >> -int spd_audio_set_volume(AudioID * id, int volume)
>> >> -{
>> >> - if ((volume > 100) || (volume < -100)) {
>> >> - fprintf(stderr, "Requested volume out of range");
>> >> - return -1;
>> >> - }
>> >> - if (id == NULL) {
>> >> - fprintf(stderr, "audio id is NULL in
>> >> spd_audio_set_volume\n");
>> >> - return -1;
>> >> - }
>> >> - id->volume = volume;
>> >> - return 0;
>> >> -}
>> >> -
>> >> -void spd_audio_set_loglevel(AudioID * id, int level)
>> >> -{
>> >> - if (level) {
>> >> - spd_audio_log_level = level;
>> >> - if (id != 0 && id->function != 0)
>> >> - id->function->set_loglevel(level);
>> >> - }
>> >> -}
>> >> -
>> >> -char const *spd_audio_get_playcmd(AudioID * id)
>> >> -{
>> >> - if (id != 0 && id->function != 0) {
>> >> - return id->function->get_playcmd();
>> >> - }
>> >> - return NULL;
>> >> -}
>> >> diff --git a/src/modules/spd_audio.h b/src/modules/spd_audio.h
>> >> deleted file mode 100644
>> >> index f9452e8..0000000
>> >> --- a/src/modules/spd_audio.h
>> >> +++ /dev/null
>> >> @@ -1,46 +0,0 @@
>> >> -
>> >> -/*
>> >> - * spd_audio.h -- The SPD Audio Library Header
>> >> - *
>> >> - * Copyright (C) 2004 Brailcom, o.p.s.
>> >> - *
>> >> - * This is free software; you can redistribute it and/or modify it under
>> >> the
>> >> - * terms of the GNU Lesser General Public License as published by the
>> >> Free
>> >> - * Software Foundation; either version 2.1, or (at your option) any later
>> >> - * version.
>> >> - *
>> >> - * This software 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 Lesser General Public
>> >> License
>> >> - * along with this package; see the file COPYING. If not, write to the
>> >> Free
>> >> - * Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
>> >> - * 02110-1301, USA.
>> >> - *
>> >> - * $Id: spd_audio.h,v 1.21 2008-10-15 17:28:17 hanke Exp $
>> >> - */
>> >> -
>> >> -#ifndef __SPD_AUDIO_H
>> >> -#define __SPD_AUDIO_H
>> >> -
>> >> -#include <spd_audio_plugin.h>
>> >> -
>> >> -#define SPD_AUDIO_LIB_PREFIX "spd_"
>> >> -
>> >> -AudioID *spd_audio_open(char *name, void **pars, char **error);
>> >> -
>> >> -int spd_audio_play(AudioID * id, AudioTrack track, AudioFormat format);
>> >> -
>> >> -int spd_audio_stop(AudioID * id);
>> >> -
>> >> -int spd_audio_close(AudioID * id);
>> >> -
>> >> -int spd_audio_set_volume(AudioID * id, int volume);
>> >> -
>> >> -void spd_audio_set_loglevel(AudioID * id, int level);
>> >> -
>> >> -char const *spd_audio_get_playcmd(AudioID * id);
>> >> -
>> >> -#endif /* ifndef #__SPD_AUDIO_H */
>> >> diff --git a/src/server/Makefile.am b/src/server/Makefile.am
>> >> index 607cf8e..bad5a61 100644
>> >> --- a/src/server/Makefile.am
>> >> +++ b/src/server/Makefile.am
>> >> @@ -9,12 +9,14 @@ speech_dispatcher_SOURCES = speechd.c speechd.h
>> >> server.c server.h \
>> >> parse.c parse.h set.c set.h msg.h alloc.c alloc.h \
>> >> compare.c compare.h speaking.c speaking.h options.c options.h \
>> >> output.c output.h sem_functions.c sem_functions.h \
>> >> - index_marking.c index_marking.h
>> >> + index_marking.c index_marking.h audio.c audio.h
>> >> speech_dispatcher_CFLAGS = $(ERROR_CFLAGS)
>> >> speech_dispatcher_CPPFLAGS = $(inc_local) $(DOTCONF_CFLAGS)
>> >> $(GLIB_CFLAGS) \
>> >> $(GMODULE_CFLAGS) $(GTHREAD_CFLAGS) -DSYS_CONF=\"$(spdconfdir)\" \
>> >> -DSND_DATA=\"$(snddatadir)\" -DMODULEBINDIR=\"$(modulebindir)\" \
>> >> - -D_GNU_SOURCE -DDEFAULT_AUDIO_METHOD=\"$(default_audio_method)\"
>> >> + -D_GNU_SOURCE -DDEFAULT_AUDIO_METHOD=\"$(default_audio_method)\" \
>> >> + -DPLUGIN_DIR=\"$(audiodir)\"
>> >> speech_dispatcher_LDFLAGS = $(RDYNAMIC)
>> >> speech_dispatcher_LDADD = $(lib_common) $(DOTCONF_LIBS) $(GLIB_LIBS) \
>> >> - $(GMODULE_LIBS) $(GTHREAD_LIBS) $(EXTRA_SOCKET_LIBS)
>> >> + $(GMODULE_LIBS) $(GTHREAD_LIBS) $(EXTRA_SOCKET_LIBS) \
>> >> + $(audio_dlopen_modules)
>> >> diff --git a/src/server/audio.c b/src/server/audio.c
>> >> new file mode 100644
>> >> index 0000000..08a6567
>> >> --- /dev/null
>> >> +++ b/src/server/audio.c
>> >> @@ -0,0 +1,647 @@
>> >> +
>> >> +/*
>> >> + * spd_audio.c -- Spd Audio Output Library
>> >> + *
>> >> + * Copyright (C) 2004, 2006 Brailcom, o.p.s.
>> >> + *
>> >> + * This is free software; you can redistribute it and/or modify it under
>> >> the
>> >> + * terms of the GNU Lesser General Public License as published by the
>> >> Free
>> >> + * Software Foundation; either version 2.1, or (at your option) any later
>> >> + * version.
>> >> + *
>> >> + * This software 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 Lesser General Public
>> >> License
>> >> + * along with this package; see the file COPYING. If not, write to the
>> >> Free
>> >> + * Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
>> >> + * 02110-1301, USA.
>> >> + *
>> >> + * $Id: spd_audio.c,v 1.21 2008-06-09 10:29:12 hanke Exp $
>> >> + */
>> >> +
>> >> +/*
>> >> + * spd_audio is a simple realtime audio output library with the
>> >> capability of
>> >> + * playing 8 or 16 bit data, immediate stop and synchronization. This
>> >> library
>> >> + * currently provides OSS, NAS, ALSA and PulseAudio backend. The
>> >> available backends are
>> >> + * specified at compile-time using the directives WITH_OSS, WITH_NAS,
>> >> WITH_ALSA,
>> >> + * WITH_PULSE, WITH_LIBAO but the user program is allowed to switch
>> >> between them at run-time.
>> >> + */
>> >> +
>> >> +#ifdef HAVE_CONFIG_H
>> >> +#include <config.h>
>> >> +#endif
>> >> +
>> >> +#include "audio.h"
>> >> +
>> >> +#include <stdio.h>
>> >> +#include <string.h>
>> >> +#include <fcntl.h>
>> >> +#include <sys/ioctl.h>
>> >> +#include <sys/time.h>
>> >> +#include <time.h>
>> >> +#include <unistd.h>
>> >> +#include <errno.h>
>> >> +
>> >> +#include <pthread.h>
>> >> +
>> >> +#include <glib.h>
>> >> +#include <glib/gstdio.h>
>> >> +#include <ltdl.h>
>> >> +
>> >> +#include "speechd.h"
>> >> +#include "speechd_defines.h"
>> >> +#include "set.h"
>> >> +
>> >> +static int spd_audio_log_level;
>> >> +static lt_dlhandle lt_h;
>> >> +
>> >> +/* Server audio socket file descriptor */
>> >> +int audio_server_socket;
>> >> +
>> >> +AudioID *audio_id;
>> >> +static char *audio_pars[10]; /* Audio module parameters */
>> >> +
>> >> +static pthread_t audio_thread;
>> >> +static sem_t audio_play_semaphore;
>> >> +
>> >> +static gboolean audio_close_requested = FALSE;
>> >> +
>> >> +/* Dynamically load a library with RTLD_GLOBAL set.
>> >> +
>> >> + This is needed when a dynamically-loaded library has its own plugins
>> >> + that call into the parent library.
>> >> + Most of the credit for this function goes to Gary Vaughan.
>> >> +*/
>> >> +static lt_dlhandle my_dlopenextglobal(const char *filename)
>> >> +{
>> >> + lt_dlhandle handle = NULL;
>> >> + lt_dladvise advise;
>> >> +
>> >> + if (lt_dladvise_init(&advise))
>> >> + return handle;
>> >> +
>> >> + if (!lt_dladvise_ext(&advise) && !lt_dladvise_global(&advise))
>> >> + handle = lt_dlopenadvise(filename, advise);
>> >> +
>> >> + lt_dladvise_destroy(&advise);
>> >> + return handle;
>> >> +}
>> >> +
>> >> +/* Open the audio device.
>> >> +
>> >> + Arguments:
>> >> + type -- The requested device. Currently AudioOSS or AudioNAS.
>> >> + pars -- and array of pointers to parameters to pass to
>> >> + the device backend, terminated by a NULL pointer.
>> >> + See the source/documentation of each specific backend.
>> >> + error -- a pointer to the string where error description is
>> >> + stored in case of failure (returned AudioID == NULL).
>> >> + Otherwise will contain NULL.
>> >> +
>> >> + Return value:
>> >> + Newly allocated AudioID structure that can be passed to
>> >> + all other spd_audio functions, or NULL in case of failure.
>> >> +
>> >> +*/
>> >> +AudioID *spd_audio_open(char *name, void **pars, char **error)
>> >> +{
>> >> + MSG(5, "spd_audio_open called with name %s", name);
>> >> + AudioID *id;
>> >> + spd_audio_plugin_t const *p;
>> >> + spd_audio_plugin_t *(*fn) (void);
>> >> + gchar *libname;
>> >> + int ret;
>> >> +
>> >> + /* now check whether dynamic plugin is available */
>> >> + ret = lt_dlinit();
>> >> + if (ret != 0) {
>> >> + *error = (char *)g_strdup_printf("lt_dlinit() failed");
>> >> + return (AudioID *) NULL;
>> >> + }
>> >> +
>> >> + ret = lt_dlsetsearchpath(PLUGIN_DIR);
>> >> + if (ret != 0) {
>> >> + *error = (char *)g_strdup_printf("lt_dlsetsearchpath()
>> >> failed");
>> >> + return (AudioID *) NULL;
>> >> + }
>> >> +
>> >> + libname = g_strdup_printf(SPD_AUDIO_LIB_PREFIX "%s", name);
>> >> + lt_h = my_dlopenextglobal(libname);
>> >> + g_free(libname);
>> >> + if (NULL == lt_h) {
>> >> + *error =
>> >> + (char *)g_strdup_printf("Cannot open plugin %s. error:
>> >> %s",
>> >> + name, lt_dlerror());
>> >> + return (AudioID *) NULL;
>> >> + }
>> >> +
>> >> + fn = lt_dlsym(lt_h, SPD_AUDIO_PLUGIN_ENTRY_STR);
>> >> + if (NULL == fn) {
>> >> + *error = (char *)g_strdup_printf("Cannot find symbol %s",
>> >> +
>> >> SPD_AUDIO_PLUGIN_ENTRY_STR);
>> >> + return (AudioID *) NULL;
>> >> + }
>> >> +
>> >> + MSG(5, "calling init function");
>> >> + p = fn();
>> >> + if (p == NULL || p->name == NULL) {
>> >> + *error = (char *)g_strdup_printf("plugin %s not found",
>> >> name);
>> >> + return (AudioID *) NULL;
>> >> + }
>> >> +
>> >> + MSG(5, "calling open function");
>> >> + id = p->open(pars);
>> >> + if (id == NULL) {
>> >> + *error =
>> >> + (char *)g_strdup_printf("Couldn't open %s plugin",
>> >> name);
>> >> + return (AudioID *) NULL;
>> >> + }
>> >> +
>> >> + id->function = p;
>> >> +#if defined(BYTE_ORDER) && (BYTE_ORDER == BIG_ENDIAN)
>> >> + id->format = SPD_AUDIO_BE;
>> >> +#else
>> >> + id->format = SPD_AUDIO_LE;
>> >> +#endif
>> >> +
>> >> + *error = NULL;
>> >> +
>> >> + return id;
>> >> +}
>> >> +
>> >> +/* Play a track on the audio device (blocking).
>> >> +
>> >> + Arguments:
>> >> + id -- the AudioID* of the device returned by spd_audio_open
>> >> + track -- a track to play (see spd_audio.h)
>> >> +
>> >> + Return value:
>> >> + 0 if everything is ok, a non-zero value in case of failure.
>> >> + See the particular backend documentation or source for the
>> >> + meaning of these non-zero values.
>> >> +
>> >> + Comment:
>> >> + spd_audio_play() is a blocking function. It returns exactly
>> >> + when the given track stopped playing. However, it's possible
>> >> + to safely interrupt it using spd_audio_stop() described below.
>> >> + (spd_audio_stop() needs to be called from another thread, obviously.)
>> >> +
>> >> +*/
>> >> +int spd_audio_play(AudioID * id, AudioTrack track, AudioFormat format)
>> >> +{
>> >> + int ret;
>> >> +
>> >> + if (id && id->function->play) {
>> >> + /* Only perform byte swapping if the driver in use has
>> >> given us audio in
>> >> + an endian format other than what the running CPU
>> >> supports. */
>> >> + if (format != id->format) {
>> >> + unsigned char *out_ptr, *out_end, c;
>> >> + out_ptr = (unsigned char *)track.samples;
>> >> + out_end =
>> >> + out_ptr +
>> >> + track.num_samples * 2 * track.num_channels;
>> >> + while (out_ptr < out_end) {
>> >> + c = out_ptr[0];
>> >> + out_ptr[0] = out_ptr[1];
>> >> + out_ptr[1] = c;
>> >> + out_ptr += 2;
>> >> + }
>> >> + }
>> >> + MSG(5, "playing audio on audio_id %d", id);
>> >> + ret = id->function->play(id, track);
>> >> + } else {
>> >> + fprintf(stderr, "Play not supported on this device\n");
>> >> + return -1;
>> >> + }
>> >> +
>> >> + return ret;
>> >> +}
>> >> +
>> >> +/* Stop playing the current track on device id
>> >> +
>> >> +Arguments:
>> >> + id -- the AudioID* of the device returned by spd_audio_open
>> >> +
>> >> +Return value:
>> >> + 0 if everything is ok, a non-zero value in case of failure.
>> >> + See the particular backend documentation or source for the
>> >> + meaning of these non-zero values.
>> >> +
>> >> +Comment:
>> >> + spd_audio_stop() safely interrupts spd_audio_play() when called
>> >> + from another thread. It shouldn't cause any clicks or unwanted
>> >> + effects in the sound output.
>> >> +
>> >> + It's safe to call spd_audio_stop() even if the device isn't playing
>> >> + any track. In that case, it does nothing. However, there is a danger
>> >> + when using spd_audio_stop(). Since you must obviously do it from
>> >> + another thread than where spd_audio_play is running, you must make
>> >> + yourself sure that the device is still open and the id you pass it
>> >> + is valid and will be valid until spd_audio_stop returns. In other
>> >> words,
>> >> + you should use some mutex or other synchronization device to be sure
>> >> + spd_audio_close isn't called before or during spd_audio_stop
>> >> execution.
>> >> +*/
>> >> +
>> >> +int spd_audio_stop(AudioID * id)
>> >> +{
>> >> + int ret;
>> >> + if (id && id->function->stop) {
>> >> + ret = id->function->stop(id);
>> >> + } else {
>> >> + fprintf(stderr, "Stop not supported on this device\n");
>> >> + return -1;
>> >> + }
>> >> + return ret;
>> >> +}
>> >> +
>> >> +/* Close the audio device id
>> >> +
>> >> +Arguments:
>> >> + id -- the AudioID* of the device returned by spd_audio_open
>> >> +
>> >> +Return value:
>> >> + 0 if everything is ok, a non-zero value in case of failure.
>> >> +
>> >> +Comments:
>> >> +
>> >> + Please make sure no other spd_audio function with this device id
>> >> + is running in another threads. See spd_audio_stop() for detailed
>> >> + description of possible problems.
>> >> +*/
>> >> +int spd_audio_close(AudioID * id)
>> >> +{
>> >> + int ret = 0;
>> >> + if (id && id->function->close) {
>> >> + ret = (id->function->close(id));
>> >> + }
>> >> +
>> >> + if (NULL != lt_h) {
>> >> + lt_dlclose(lt_h);
>> >> + lt_h = NULL;
>> >> + lt_dlexit();
>> >> + }
>> >> +
>> >> + return ret;
>> >> +}
>> >> +
>> >> +/* Set volume for playing tracks on the device id
>> >> +
>> >> +Arguments:
>> >> + id -- the AudioID* of the device returned by spd_audio_open
>> >> + volume -- a value in the range <-100:100> where -100 means the
>> >> + least volume (probably silence), 0 the default volume
>> >> + and +100 the highest volume possible to make on that
>> >> + device for a single flow (i.e. not using mixer).
>> >> +
>> >> +Return value:
>> >> + 0 if everything is ok, a non-zero value in case of failure.
>> >> + See the particular backend documentation or source for the
>> >> + meaning of these non-zero values.
>> >> +
>> >> +Comments:
>> >> +
>> >> + In case of /dev/dsp, it's not possible to set volume for
>> >> + the particular flow. For that reason, the value 0 means
>> >> + the volume the track was recorded on and each smaller value
>> >> + means less volume (since this works by deviding the samples
>> >> + in the track by a constant).
>> >> +*/
>> >> +int spd_audio_set_volume(AudioID * id, int volume)
>> >> +{
>> >> + if ((volume > 100) || (volume < -100)) {
>> >> + fprintf(stderr, "Requested volume out of range");
>> >> + return -1;
>> >> + }
>> >> + if (id == NULL) {
>> >> + fprintf(stderr, "audio id is NULL in
>> >> spd_audio_set_volume\n");
>> >> + return -1;
>> >> + }
>> >> + id->volume = volume;
>> >> + return 0;
>> >> +}
>> >> +
>> >> +void spd_audio_set_loglevel(AudioID * id, int level)
>> >> +{
>> >> + if (level) {
>> >> + spd_audio_log_level = level;
>> >> + if (id != 0 && id->function != 0)
>> >> + id->function->set_loglevel(level);
>> >> + }
>> >> +}
>> >> +
>> >> +char const *spd_audio_get_playcmd(AudioID * id)
>> >> +{
>> >> + if (id != 0 && id->function != 0) {
>> >> + return id->function->get_playcmd();
>> >> + }
>> >> + return NULL;
>> >> +}
>> >> +
>> >> +void speechd_audio_socket_init(void)
>> >> +{
>> >> + /* For now use unix socket for audio. Maybe later we can add inet
>> >> socket support */
>> >> + GString *audio_socket_filename;
>> >> + audio_socket_filename = g_string_new("");
>> >> + if (SpeechdOptions.runtime_speechd_dir) {
>> >> + g_string_printf(audio_socket_filename, "%s/audio.sock",
>> >> + SpeechdOptions.runtime_speechd_dir);
>> >> + } else {
>> >> + FATAL
>> >> + ("Socket name file not set and user has no runtime
>> >> directory");
>> >> + }
>> >> + g_free(SpeechdOptions.audio_socket_path);
>> >> + SpeechdOptions.audio_socket_path =
>> >> g_strdup(audio_socket_filename->str);
>> >> + g_string_free(audio_socket_filename, 1);
>> >> +
>> >> + MSG(1, "Creating audio socket at %s",
>> >> SpeechdOptions.audio_socket_path);
>> >> +
>> >> + /* Audio data is only using unix sockets for now, possibly adapt to
>> >> use
>> >> + * inet sockets also later? */
>> >> + if (g_file_test(SpeechdOptions.audio_socket_path,
>> >> G_FILE_TEST_EXISTS))
>> >> + if (g_unlink(SpeechdOptions.audio_socket_path) == -1)
>> >> + FATAL
>> >> + ("Local socket file for audio exists but
>> >> impossible to delete. Wrong permissions?");
>> >> + /* Connect and start listening on local unix socket */
>> >> + audio_server_socket =
>> >> make_local_socket(SpeechdOptions.audio_socket_path);
>> >> +}
>> >> +
>> >> +/* play the audio data on _fd_ if we got some activity. */
>> >> +int play_audio(int fd)
>> >> +{
>> >> + size_t bytes = 0; /* Number of bytes we got */
>> >> + int buflen = BUF_SIZE;
>> >> + char *buf = (char *)g_malloc(buflen + 1);
>> >> + AudioTrack track;
>> >> + AudioFormat format;
>> >> + gchar ** metadata;
>> >> + int bytes_read;
>> >> +
>> >> + /* Read data from socket */
>> >> + /* Read exactly one complete line, the `parse' routine relies on it
>> >> */
>> >> + {
>> >> + while (1) {
>> >> + int n = read(fd, buf + bytes, 1);
>> >> + if (n <= 0) {
>> >> + MSG(5, "ERROR: Read 0 bytes from fd");
>> >> + g_free(buf);
>> >> + return -1;
>> >> + }
>> >> + /* Note, bytes is a 0-based index into buf. */
>> >> + if ((buf[bytes] == '\n')
>> >> + && (bytes >= 1) && (buf[bytes - 1] == '\r')) {
>> >> + buf[++bytes] = '\0';
>> >> + break;
>> >> + }
>> >> + if (buf[bytes] == '\0')
>> >> + buf[bytes] = '?';
>> >> + if ((++bytes) == buflen) {
>> >> + buflen *= 2;
>> >> + buf = g_realloc(buf, buflen + 1);
>> >> + }
>> >> + }
>> >> + }
>> >> +
>> >> + /* Parse the data and read the reply */
>> >> + MSG2(5, "protocol", "%d:DATA:|%s| (%d)", fd, buf, bytes);
>> >> + if (strcmp(buf, "ACK"NEWLINE) == 0) {
>> >> + g_free(buf);
>> >> + return 0;
>> >> + }
>> >> + /* parse the AudioTrack information from buf */
>> >> + metadata = g_strsplit(buf, ":", 5);
>> >> + if (metadata == NULL || metadata[0] == NULL
>> >> + || metadata[1] == NULL || metadata[2] == NULL
>> >> + || metadata[3] == NULL || metadata[4] == NULL) {
>> >> + MSG(5, "Error: Unable to read Audiotrack metadata!");
>> >> + return -1;
>> >> + }
>> >> + format = strtol(metadata[0], NULL, 10);
>> >> + track.bits = strtol(metadata[1], NULL, 10);
>> >> + track.num_channels = strtol(metadata[2], NULL, 10);
>> >> + track.sample_rate = strtol(metadata[3], NULL, 10);
>> >> + track.num_samples = strtol(metadata[4], NULL, 10);
>> >> +
>> >> + MSG(5, "Track num samples is %d", track.num_samples);
>> >> +
>> >> + if (track.num_samples <= 0) {
>> >> + MSG(5, "Error: num_samples is invalid");
>> >> + return -1;
>> >> + }
>> >> + /* then free buf */
>> >> + g_free(buf);
>> >> + /* Get the rest of the data */
>> >> + track.samples = g_malloc0_n(track.num_samples, sizeof(signed
>> >> short));
>> >> + bytes_read = read(fd, track.samples, track.num_samples *
>> >> sizeof(signed short));
>> >> +
>> >> + if (bytes_read != track.num_samples * sizeof(signed short)) {
>> >> + MSG(5, "Error: num_samples %d doesn't match bytes read %d",
>> >> track.num_samples, bytes_read);
>> >> + return -1;
>> >> + }
>> >> +
>> >> + MSG(5, "Going to play audio on audio with id %d", audio_id);
>> >> +
>> >> + /* And play the AudioTrack */
>> >> + if (spd_audio_play(audio_id, track, format) < 0) {
>> >> + MSG(5, "Error: unable to play audio");
>> >> + return -1;
>> >> + }
>> >> +
>> >> + return 0;
>> >> +}
>> >> +
>> >> +static gboolean audio_socket_process_incoming (gint fd,
>> >> + GIOCondition condition,
>> >> + gpointer data)
>> >> +{
>> >> + int ret;
>> >> + ret = speechd_audio_connection_new(fd);
>> >> + if (ret != 0) {
>> >> + MSG(2, "Error: Failed to add new module audio!");
>> >> + if (SPEECHD_DEBUG) {
>> >> + FATAL("Failed to add new module audio!");
>> >> + }
>> >> + }
>> >> +
>> >> + return TRUE;
>> >> +}
>> >> +
>> >> +static gboolean audio_process_incoming (gint fd,
>> >> + GIOCondition condition,
>> >> + gpointer data)
>> >> +{
>> >> + MSG(5, "audio_process_incoming called for fd %d", fd);
>> >> + int nread;
>> >> +
>> >> + ioctl(fd, FIONREAD, &nread);
>> >> +
>> >> + if (nread == 0) {
>> >> + /* module has gone */
>> >> + MSG(2, "Info: Module has gone.");
>> >> + return FALSE;
>> >> + }
>> >> +
>> >> + MSG(5, "read %d bytes from fd %d", nread, fd);
>> >> +
>> >> + /* client sends some commands or data */
>> >> + if (play_audio(fd) == -1) {
>> >> + MSG(2, "Error: Failed to serve client on fd %d!", fd);
>> >> + }
>> >> +
>> >> + return TRUE;
>> >> +}
>> >> +
>> >> +/* Playback thread. */
>> >> +static void *_speechd_play(void *nothing)
>> >> +{
>> >> + char *error = 0;
>> >> + gchar **outputs;
>> >> + int i = 0;
>> >> + gboolean found_audio_module = FALSE;
>> >> +
>> >> + MSG(1, "Playback thread starting.......");
>> >> +
>> >> + /* TODO: Use real values from config rather than these hard coded
>> >> test values */
>> >> + if (GlobalFDSet.audio_oss_device != NULL)
>> >> + audio_pars[1] = g_strdup(GlobalFDSet.audio_oss_device);
>> >> + else
>> >> + audio_pars[1] = NULL;
>> >> +
>> >> + if (GlobalFDSet.audio_alsa_device != NULL)
>> >> + audio_pars[2] = g_strdup(GlobalFDSet.audio_alsa_device);
>> >> + else
>> >> + audio_pars[2] = NULL;
>> >> +
>> >> +
>> >> + if (GlobalFDSet.audio_nas_server != NULL)
>> >> + audio_pars[3] = g_strdup(GlobalFDSet.audio_nas_server);
>> >> + else
>> >> + audio_pars[3] = NULL;
>> >> +
>> >> + if (GlobalFDSet.audio_pulse_server != NULL)
>> >> + audio_pars[4] = g_strdup(GlobalFDSet.audio_pulse_server);
>> >> + else
>> >> + audio_pars[4] = NULL;
>> >> +
>> >> + if (GlobalFDSet.audio_pulse_min_length != NULL)
>> >> + audio_pars[5] = g_strdup_printf("%d",
>> >> GlobalFDSet.audio_pulse_min_length);
>> >> + else
>> >> + audio_pars[5] = NULL;
>> >> +
>> >> + MSG(1, "Openning audio output system");
>> >> + if (GlobalFDSet.audio_output_method == NULL) {
>> >> + MSG(1, "Sound output method specified in configuration not
>> >> supported. "
>> >> + "Please choose 'oss', 'alsa', 'nas', 'libao' or
>> >> 'pulse'.");
>> >> + return 0;
>> >> + }
>> >> +
>> >> + outputs = g_strsplit(GlobalFDSet.audio_output_method, ",", 0);
>> >> + while (NULL != outputs[i]) {
>> >> + audio_id =
>> >> + spd_audio_open(outputs[i], (void **)&audio_pars[1],
>> >> + &error);
>> >> + if (audio_id) {
>> >> + spd_audio_set_loglevel(audio_id,
>> >> SpeechdOptions.log_level);
>> >> + MSG(5, "Using %s audio output method with log level
>> >> %d", outputs[i], SpeechdOptions.log_level);
>> >> +
>> >> + /* Volume is controlled by the synthesizer. Always
>> >> play at normal on audio device. */
>> >> + if (spd_audio_set_volume(audio_id, 85) < 0) {
>> >> + MSG(2, "Can't set volume. audio not
>> >> initialized?");
>> >> + }
>> >> +
>> >> + g_strfreev(outputs);
>> >> + MSG(5, "audio initialized successfully.");
>> >> + found_audio_module = TRUE;
>> >> + break;
>> >> + }
>> >> + i++;
>> >> + }
>> >> +
>> >> + if (!found_audio_module) {
>> >> + MSG(1, "Opening sound device failed. Reason: %s. ", error);
>> >> + g_free(error); /* g_malloc'ed, in spd_audio_open.
>> >> */
>> >> + }
>> >> +
>> >> + /* Connect to the server socket */
>> >> + g_unix_fd_add(audio_server_socket, G_IO_IN,
>> >> + audio_socket_process_incoming, NULL);
>> >> +
>> >> + /* Block all signals to this thread. */
>> >> +// set_speaking_thread_parameters();
>> >> +
>> >> + while (!audio_close_requested) {
>> >> + /* If semaphore not set, set suspended lock and suspend
>> >> until it is signaled. */
>> >> + if (0 != sem_trywait(&audio_play_semaphore)) {
>> >> + sem_wait(&audio_play_semaphore);
>> >> + }
>> >> + MSG(5, "Playback semaphore on.");
>> >> + if (audio_close_requested)
>> >> + break;
>> >> + }
>> >> +
>> >> + MSG(1, "Playback thread ended.......");
>> >> + return 0;
>> >> +}
>> >> +
>> >> +void speechd_audio_init()
>> >> +{
>> >> + int ret = 0;
>> >> +
>> >> + audio_id = 0;
>> >> + sem_init(&audio_play_semaphore, 0, 0);
>> >> +
>> >> + ret = pthread_create(&audio_thread, NULL, _speechd_play, NULL);
>> >> + if (ret != 0)
>> >> + FATAL("Audio thread failed!\n");
>> >> +}
>> >> +
>> >> +/* activity is on audio_server_socket (request for a new connection) */
>> >> +int speechd_audio_connection_new(int audio_server_socket)
>> >> +{
>> >> + MSG(5, "Adding audio connection on socket %d", audio_server_socket);
>> >> + TAudioFDSetElement *new_fd_set;
>> >> + struct sockaddr_in module_address;
>> >> + unsigned int module_len = sizeof(module_address);
>> >> + int module_socket;
>> >> +
>> >> + module_socket =
>> >> + accept(audio_server_socket, (struct sockaddr *)&module_address,
>> >> + &module_len);
>> >> +
>> >> + if (module_socket == -1) {
>> >> + MSG(2,
>> >> + "Error: Can't handle connection request of a module for
>> >> audio");
>> >> + return -1;
>> >> + }
>> >> +
>> >> + /* We add the associated client_socket to the descriptor set. */
>> >> + if (module_socket > SpeechdStatus.max_fd)
>> >> + SpeechdStatus.max_fd = module_socket;
>> >> + MSG(4, "Adding module on fd %d", module_socket);
>> >> +
>> >> + /* Create a record in fd_settings */
>> >> + new_fd_set = (TAudioFDSetElement *) default_audio_fd_set();
>> >> + if (new_fd_set == NULL) {
>> >> + MSG(2,
>> >> + "Error: Failed to create a record in fd_settings for
>> >> the module for audio");
>> >> + if (SpeechdStatus.max_fd == module_socket)
>> >> + SpeechdStatus.max_fd--;
>> >> + return -1;
>> >> + }
>> >> + new_fd_set->fd = module_socket;
>> >> + new_fd_set->fd_source = g_unix_fd_add(module_socket, G_IO_IN,
>> >> audio_process_incoming, NULL);
>> >> +
>> >> + return 0;
>> >> +}
>> >> +
>> >> +void speechd_audio_cleanup(void)
>> >> +{
>> >> + if (close(audio_server_socket) == -1)
>> >> + MSG(2, "close() audio server socket failed: %s",
>> >> strerror(errno));
>> >> +
>> >> + MSG(2, "Closing audio output...");
>> >> + spd_audio_close(audio_id);
>> >> + audio_id = NULL;
>> >> +}
>> >> diff --git a/src/server/audio.h b/src/server/audio.h
>> >> new file mode 100644
>> >> index 0000000..d371c50
>> >> --- /dev/null
>> >> +++ b/src/server/audio.h
>> >> @@ -0,0 +1,60 @@
>> >> +
>> >> +/*
>> >> + * spd_audio.h -- The SPD Audio Library Header
>> >> + *
>> >> + * Copyright (C) 2004 Brailcom, o.p.s.
>> >> + *
>> >> + * This is free software; you can redistribute it and/or modify it under
>> >> the
>> >> + * terms of the GNU Lesser General Public License as published by the
>> >> Free
>> >> + * Software Foundation; either version 2.1, or (at your option) any later
>> >> + * version.
>> >> + *
>> >> + * This software 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 Lesser General Public
>> >> License
>> >> + * along with this package; see the file COPYING. If not, write to the
>> >> Free
>> >> + * Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
>> >> + * 02110-1301, USA.
>> >> + *
>> >> + * $Id: spd_audio.h,v 1.21 2008-10-15 17:28:17 hanke Exp $
>> >> + */
>> >> +
>> >> +#ifndef __SPD_AUDIO_H
>> >> +#define __SPD_AUDIO_H
>> >> +
>> >> +#include <spd_audio_plugin.h>
>> >> +
>> >> +#define SPD_AUDIO_LIB_PREFIX "spd_"
>> >> +
>> >> +AudioID *spd_audio_open(char *name, void **pars, char **error);
>> >> +
>> >> +int spd_audio_play(AudioID * id, AudioTrack track, AudioFormat format);
>> >> +
>> >> +int spd_audio_stop(AudioID * id);
>> >> +
>> >> +int spd_audio_close(AudioID * id);
>> >> +
>> >> +int spd_audio_set_volume(AudioID * id, int volume);
>> >> +
>> >> +void spd_audio_set_loglevel(AudioID * id, int level);
>> >> +
>> >> +char const *spd_audio_get_playcmd(AudioID * id);
>> >> +
>> >> +/* Speech dispatcher server functions */
>> >> +
>> >> +/* speechd_play_audio() reads audio data and plays it */
>> >> +int speechd_play_audio(int fd);
>> >> +
>> >> +int speechd_audio_connection_new(int audio_socket);
>> >> +/* Initialize the audio socket */
>> >> +void speechd_audio_socket_init(void);
>> >> +/* Initialize the audio backend based on user's settings in a new thread
>> >> */
>> >> +void speechd_audio_init(void);
>> >> +
>> >> +/* Clean up audio socket and module */
>> >> +void speechd_audio_cleanup(void);
>> >> +
>> >> +#endif /* ifndef #__SPD_AUDIO_H */
>> >> diff --git a/src/server/module.c b/src/server/module.c
>> >> index 3932e6d..b27727d 100644
>> >> --- a/src/server/module.c
>> >> +++ b/src/server/module.c
>> >> @@ -313,18 +313,6 @@ OutputModule *load_output_module(char *mod_name,
>> >> char *mod_prog,
>> >> output_module_debug(module);
>> >> }
>> >>
>> >> - /* Initialize audio settings */
>> >> - ret = output_send_audio_settings(module);
>> >> - if (ret != 0) {
>> >> - MSG(1,
>> >> - "ERROR: Can't initialize audio in output module, see
>> >> reason above.");
>> >> - module->working = 0;
>> >> - kill(module->pid, 9);
>> >> - waitpid(module->pid, NULL, WNOHANG);
>> >> - destroy_module(module);
>> >> - return NULL;
>> >> - }
>> >> -
>> >> /* Send log level configuration setting */
>> >> ret = output_send_loglevel_setting(module);
>> >> if (ret != 0) {
>> >> diff --git a/src/server/msg.h b/src/server/msg.h
>> >> index 67c07a9..a8f5048 100644
>> >> --- a/src/server/msg.h
>> >> +++ b/src/server/msg.h
>> >> @@ -24,7 +24,8 @@
>> >> #ifndef MSG_H
>> >> #define MSG_H
>> >>
>> >> -#define NEWLINE
>> >> "\r\n"
>> >> +#include <speechd_defines.h>
>> >> +
>> >> #define OK_LANGUAGE_SET "201 OK
>> >> LANGUAGE SET" NEWLINE
>> >> #define OK_PRIORITY_SET "202 OK
>> >> PRIORITY SET" NEWLINE
>> >> #define OK_RATE_SET "203 OK
>> >> RATE SET" NEWLINE
>> >> diff --git a/src/server/output.c b/src/server/output.c
>> >> index 0d3e40b..cfa02bc 100644
>> >> --- a/src/server/output.c
>> >> +++ b/src/server/output.c
>> >> @@ -482,29 +482,6 @@ int output_send_settings(TSpeechDMessage * msg,
>> >> OutputModule * output)
>> >> g_string_append_printf(set_str, #name"=NULL\n"); \
>> >> }
>> >>
>> >> -int output_send_audio_settings(OutputModule * output)
>> >> -{
>> >> - GString *set_str;
>> >> - int err;
>> >> -
>> >> - MSG(4, "Module set parameters.");
>> >> - set_str = g_string_new("");
>> >> - ADD_SET_STR(audio_output_method);
>> >> - ADD_SET_STR(audio_oss_device);
>> >> - ADD_SET_STR(audio_alsa_device);
>> >> - ADD_SET_STR(audio_nas_server);
>> >> - ADD_SET_STR(audio_pulse_server);
>> >> - ADD_SET_INT(audio_pulse_min_length);
>> >> -
>> >> - SEND_CMD_N("AUDIO");
>> >> - SEND_DATA_N(set_str->str);
>> >> - SEND_CMD_N(".");
>> >> -
>> >> - g_string_free(set_str, 1);
>> >> -
>> >> - return 0;
>> >> -}
>> >> -
>> >> int output_send_loglevel_setting(OutputModule * output)
>> >> {
>> >> GString *set_str;
>> >> diff --git a/src/server/output.h b/src/server/output.h
>> >> index 10bbe80..d48602b 100644
>> >> --- a/src/server/output.h
>> >> +++ b/src/server/output.h
>> >> @@ -41,7 +41,6 @@ GString *output_read_reply(OutputModule * output);
>> >> char *output_read_reply2(OutputModule * output);
>> >> int output_send_data(char *cmd, OutputModule * output, int wfr);
>> >> int output_send_settings(TSpeechDMessage * msg, OutputModule * output);
>> >> -int output_send_audio_settings(OutputModule * output);
>> >> int output_send_loglevel_setting(OutputModule * output);
>> >> int output_module_is_speaking(OutputModule * output, char **index_mark);
>> >> int waitpid_with_timeout(pid_t pid, int *status_ptr, int options,
>> >> diff --git a/src/server/server.c b/src/server/server.c
>> >> index 6bdc78e..cf72880 100644
>> >> --- a/src/server/server.c
>> >> +++ b/src/server/server.c
>> >> @@ -31,6 +31,7 @@
>> >> #include "speaking.h"
>> >> #include "sem_functions.h"
>> >> #include "history.h"
>> >> +#include "speechd_defines.h"
>> >>
>> >> int last_message_id = 0;
>> >>
>> >> diff --git a/src/server/set.c b/src/server/set.c
>> >> index d0c2305..23e2a9c 100644
>> >> --- a/src/server/set.c
>> >> +++ b/src/server/set.c
>> >> @@ -537,6 +537,20 @@ TFDSetElement *default_fd_set(void)
>> >> return (new);
>> >> }
>> >>
>> >> +TAudioFDSetElement *default_audio_fd_set(void)
>> >> +{
>> >> + TAudioFDSetElement *new;
>> >> +
>> >> + new = (TAudioFDSetElement *) g_malloc(sizeof(TAudioFDSetElement));
>> >> +
>> >> + /* Fill with the global settings values */
>> >> + /* We can't use global_fdset copy as this
>> >> + returns static structure and we need dynamic */
>> >> + new->output_module = g_strdup(GlobalFDSet.output_module);
>> >> +
>> >> + return (new);
>> >> +}
>> >> +
>> >> int get_client_uid_by_fd(int fd)
>> >> {
>> >> int *uid;
>> >> diff --git a/src/server/set.h b/src/server/set.h
>> >> index 08866ab..2b7f488 100644
>> >> --- a/src/server/set.h
>> >> +++ b/src/server/set.h
>> >> @@ -94,6 +94,7 @@ int set_debug_all(int debug);
>> >> int set_debug_destination_all(char *debug_destination);
>> >>
>> >> TFDSetElement *default_fd_set(void);
>> >> +TAudioFDSetElement *default_audio_fd_set(void);
>> >>
>> >> char *set_param_str(char *parameter, char *value);
>> >>
>> >> diff --git a/src/server/speechd.c b/src/server/speechd.c
>> >> index d790c3f..21ca5c1 100644
>> >> --- a/src/server/speechd.c
>> >> +++ b/src/server/speechd.c
>> >> @@ -31,8 +31,10 @@
>> >> #include <sys/stat.h>
>> >> #include <sys/socket.h>
>> >> #include <sys/un.h>
>> >> +#include <ltdl.h>
>> >>
>> >> #include "speechd.h"
>> >> +#include "audio.h"
>> >>
>> >> /* Declare dotconf functions and data structures*/
>> >> #include "configuration.h"
>> >> @@ -76,6 +78,10 @@ static gboolean client_process_incoming (gint
>> >> fd,
>> >> GIOCondition condition,
>> >> gpointer data);
>> >>
>> >> +static gboolean audio_process_incoming (gint fd,
>> >> + GIOCondition condition,
>> >> + gpointer data);
>> >> +
>> >> void check_client_count(void);
>> >>
>> >> #ifdef __SUNPRO_C
>> >> @@ -867,6 +873,7 @@ int make_local_socket(const char *filename)
>> >> FATAL("listen() failed for local socket");
>> >> }
>> >>
>> >> + MSG(5, "Successfully opened local socket at %s", filename);
>> >> return sock;
>> >> }
>> >>
>> >> @@ -911,6 +918,7 @@ int make_inet_socket(const int port)
>> >> ("listen() failed for inet socket, another Speech
>> >> Dispatcher running?");
>> >> }
>> >>
>> >> + MSG(5, "Successfully opened inet socket on port %d", port);
>> >> return server_socket;
>> >> }
>> >>
>> >> @@ -981,6 +989,7 @@ int main(int argc, char *argv[])
>> >> char *spawn_communication_method = NULL;
>> >> int spawn_port = 0;
>> >> char *spawn_socket_path = NULL;
>> >> + char *status_info;
>> >>
>> >> /* Strip all permisions for 'others' from the files created */
>> >> umask(007);
>> >> @@ -991,6 +1000,9 @@ int main(int argc, char *argv[])
>> >> custom_logfile = NULL;
>> >> custom_log_kind = NULL;
>> >>
>> >> + /* Initialize ltdl's list of preloaded audio backends. */
>> >> + LTDL_SET_PRELOADED_SYMBOLS();
>> >> +
>> >> /* initialize i18n support */
>> >> i18n_init();
>> >>
>> >> @@ -1116,8 +1128,11 @@ int main(int argc, char *argv[])
>> >> exit(1);
>> >> }
>> >>
>> >> + /* We need this first since modules will connect to it */
>> >> + speechd_audio_socket_init();
>> >> +
>> >> speechd_init();
>> >> -
>> >> +
>> >> /* Handle socket_path 'default' */
>> >> // TODO: This is a hack, we should do that at appropriate places...
>> >> if (!strcmp(SpeechdOptions.socket_path, "default")) {
>> >> @@ -1237,6 +1252,8 @@ int main(int argc, char *argv[])
>> >> g_unix_signal_add(SIGUSR1, speechd_reload_dead_modules, NULL);
>> >> (void)signal(SIGPIPE, SIG_IGN);
>> >>
>> >> + speechd_audio_init();
>> >> +
>> >> MSG(4, "Creating new thread for speak()");
>> >> ret = pthread_create(&speak_thread, NULL, speak, NULL);
>> >> if (ret != 0)
>> >> @@ -1280,6 +1297,8 @@ int main(int argc, char *argv[])
>> >> if (close(server_socket) == -1)
>> >> MSG(2, "close() failed: %s", strerror(errno));
>> >>
>> >> + speechd_audio_cleanup();
>> >> +
>> >> MSG(4, "Removing pid file");
>> >> destroy_pid_file();
>> >>
>> >> diff --git a/src/server/speechd.h b/src/server/speechd.h
>> >> index e5e620b..0e80d6f 100644
>> >> --- a/src/server/speechd.h
>> >> +++ b/src/server/speechd.h
>> >> @@ -111,6 +111,12 @@ typedef struct {
>> >> } TFDSetElement;
>> >>
>> >> typedef struct {
>> >> + int fd; /* File descriptor the module is on. */
>> >> + guint fd_source; /* Used to store the GSource ID for
>> >> watching fd activity in the main loop */
>> >> + char *output_module; /* Output module name. (e.g. "festival",
>> >> "flite", "apollo", ...) */
>> >> +} TAudioFDSetElement;
>> >> +
>> >> +typedef struct {
>> >> char *pattern;
>> >> TFDSetElement val;
>> >> } TFDSetClientSpecific;
>> >> @@ -153,6 +159,7 @@ struct {
>> >> char *communication_method;
>> >> int communication_method_set;
>> >> char *socket_path;
>> >> + char *audio_socket_path;
>> >> int socket_path_set;
>> >> int port, port_set;
>> >> int localhost_access_only, localhost_access_only_set;
>> >> @@ -256,6 +263,8 @@ int isanum(const char *str);
>> >> absolute (starting with slash) or relative. */
>> >> char *spd_get_path(char *filename, char *startdir);
>> >>
>> >> +int make_local_socket(const char *filename);
>> >> +
>> >> /* Functions used in speechd.c only */
>> >> int speechd_connection_new(int server_socket);
>> >> int speechd_connection_destroy(int fd);
>> >> --
>> >> 2.5.0
>> >>
>> >
>> >> _______________________________________________
>> >> Speechd mailing list
>> >> Speechd at lists.freebsoft.org
>> >> http://lists.freebsoft.org/mailman/listinfo/speechd
>> >
>> >
>> > _______________________________________________
>> > Speechd mailing list
>> > Speechd at lists.freebsoft.org
>> > http://lists.freebsoft.org/mailman/listinfo/speechd
>>
>> _______________________________________________
>> Speechd mailing list
>> Speechd at lists.freebsoft.org
>> http://lists.freebsoft.org/mailman/listinfo/speechd
>
> _______________________________________________
> Speechd mailing list
> Speechd at lists.freebsoft.org
> http://lists.freebsoft.org/mailman/listinfo/speechd