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[Gnash-commit] gnash ChangeLog Makefile.am configure.ac backen...
|
From: |
Tomas Groth |
|
Subject: |
[Gnash-commit] gnash ChangeLog Makefile.am configure.ac backen... |
|
Date: |
Sat, 07 Oct 2006 14:20:27 +0000 |
CVSROOT: /sources/gnash
Module name: gnash
Changes by: Tomas Groth <tgc> 06/10/07 14:20:27
Modified files:
. : ChangeLog Makefile.am configure.ac
backend : Makefile.am sound_handler_gst.cpp
sound_handler_sdl.cpp
macros : ffmpeg.m4 mad.m4
plugin : Makefile.am
server : gnash.h
server/swf : tag_loaders.cpp
utilities : Makefile.am
Log message:
The new soundbackend...
CVSWeb URLs:
http://cvs.savannah.gnu.org/viewcvs/gnash/ChangeLog?cvsroot=gnash&r1=1.1060&r2=1.1061
http://cvs.savannah.gnu.org/viewcvs/gnash/Makefile.am?cvsroot=gnash&r1=1.37&r2=1.38
http://cvs.savannah.gnu.org/viewcvs/gnash/configure.ac?cvsroot=gnash&r1=1.129&r2=1.130
http://cvs.savannah.gnu.org/viewcvs/gnash/backend/Makefile.am?cvsroot=gnash&r1=1.37&r2=1.38
http://cvs.savannah.gnu.org/viewcvs/gnash/backend/sound_handler_gst.cpp?cvsroot=gnash&r1=1.18&r2=1.19
http://cvs.savannah.gnu.org/viewcvs/gnash/backend/sound_handler_sdl.cpp?cvsroot=gnash&r1=1.20&r2=1.21
http://cvs.savannah.gnu.org/viewcvs/gnash/macros/ffmpeg.m4?cvsroot=gnash&r1=1.3&r2=1.4
http://cvs.savannah.gnu.org/viewcvs/gnash/macros/mad.m4?cvsroot=gnash&r1=1.10&r2=1.11
http://cvs.savannah.gnu.org/viewcvs/gnash/plugin/Makefile.am?cvsroot=gnash&r1=1.45&r2=1.46
http://cvs.savannah.gnu.org/viewcvs/gnash/server/gnash.h?cvsroot=gnash&r1=1.56&r2=1.57
http://cvs.savannah.gnu.org/viewcvs/gnash/server/swf/tag_loaders.cpp?cvsroot=gnash&r1=1.52&r2=1.53
http://cvs.savannah.gnu.org/viewcvs/gnash/utilities/Makefile.am?cvsroot=gnash&r1=1.26&r2=1.27
Patches:
Index: ChangeLog
===================================================================
RCS file: /sources/gnash/gnash/ChangeLog,v
retrieving revision 1.1060
retrieving revision 1.1061
diff -u -b -r1.1060 -r1.1061
--- ChangeLog 7 Oct 2006 12:29:35 -0000 1.1060
+++ ChangeLog 7 Oct 2006 14:20:27 -0000 1.1061
@@ -1,3 +1,13 @@
+2006-10-07 Tomas Groth Christensen <address@hidden>
+
+ * Makefile.am, configure.ac, backend/Makefile.am,
+ backend/sound_handler_gst.cpp, backend/sound_handler_sdl.cpp,
+ macros/ffmpeg.m4, macros/mad.m4, plugin/Makefile.am,
+ server/gnash.h, server/swf/tag_loaders.cpp,
+ utilities/Makefile.am: Commited the new soundbackend, based
+ on SDL and optionally ffmpeg or libmad. Also changed the
+ buildsystem to reflect the change.
+
2006-10-07 Bastiaan Jacques <address@hidden>
* configure.ac: Replace --enable-lotsa-warnings with --enable-strict,
@@ -9,7 +19,7 @@
* configure.ac: Fixed typo. '!=' -> '='.
* macros/sdl.m4: Fixed finding libpath.
-2006-10-07 Tomas Groth Christensen <tomas@>
+2006-10-07 Tomas Groth Christensen <address@hidden>
* configure.ac: Fixed typo, so that we check for OpenGL when
it is the selected renderer.
Index: Makefile.am
===================================================================
RCS file: /sources/gnash/gnash/Makefile.am,v
retrieving revision 1.37
retrieving revision 1.38
diff -u -b -r1.37 -r1.38
--- Makefile.am 3 Oct 2006 13:52:15 -0000 1.37
+++ Makefile.am 7 Oct 2006 14:20:27 -0000 1.38
@@ -133,8 +133,8 @@
@echo "X_LIBS is $(X_LIBS)"
@echo "DMALLOC_CFLAGS is $(DMALLOC_CFLAGS)"
@echo "DMALLOC_LIBS is $(DMALLOC_LIBS)"
- @echo "MP3_CFLAGS is $(MP3_CFLAGS)"
- @echo "MP3_LIBS is $(MP3_LIBS)"
+ @echo "MAD_CFLAGS is $(MAD_CFLAGS)"
+ @echo "MAD_LIBS is $(MAD_LIBS)"
@echo "PTHREAD_CFLAGS is $(PTHREAD_CFLAGS)"
@echo "PTHREAD_LIBS is $(PTHREAD_LIBS)"
@echo "CURL_CFLAGS is $(CURL_CFLAGS)"
Index: configure.ac
===================================================================
RCS file: /sources/gnash/gnash/configure.ac,v
retrieving revision 1.129
retrieving revision 1.130
diff -u -b -r1.129 -r1.130
--- configure.ac 7 Oct 2006 12:29:35 -0000 1.129
+++ configure.ac 7 Oct 2006 14:20:27 -0000 1.130
@@ -33,7 +33,7 @@
dnl also makes it possible to release a modified version which carries
dnl forward this exception.
-dnl $Id: configure.ac,v 1.129 2006/10/07 12:29:35 bjacques Exp $
+dnl $Id: configure.ac,v 1.130 2006/10/07 14:20:27 tgc Exp $
AC_PREREQ(2.50)
AC_INIT(gnash, 0.7.1-cvs)
@@ -223,30 +223,22 @@
sound,
[ --enable-sound=handler Enable sound support using the specified handler:
GST (Gstreamer)
- FFMPEG
- MAD (libmad)
- SDL (SDL_Mixer)],
+ SDL (SDL)],
[case "${enableval}" in
GST|gst) sound_handler=gst ;;
sdl|SDL) sound_handler=sdl ;;
- ffmpeg|FFMPEG) sound_handler=ffmpeg ;;
- mad|MAD) sound_handler=mad ;;
*) sound_handler=none ;;
esac],
[sound_handler=none]
)
case "${sound_handler}" in
gst) AC_DEFINE([SOUND_GST], [1], [Use GSTREAMER for sound]) ;;
- sdl) AC_DEFINE([SOUND_SDL], [1], [Use SDL-mixer for sound]) ;;
- ffmpeg) AC_DEFINE([SOUND_SDL], [1], [Use FFMPEG for sound]) ;;
- mad) AC_DEFINE([SOUND_SDL], [1], [Use libMAD for sound]) ;;
+ sdl) AC_DEFINE([SOUND_SDL], [1], [Use SDL for sound]) ;;
*)
esac
AM_CONDITIONAL(USE_SOUND_GST, test x$sound_handler = xgst)
AM_CONDITIONAL(USE_SOUND_SDL, test x$sound_handler = xsdl)
-AM_CONDITIONAL(USE_SOUND_MAD, test x$sound_handler = xsdl)
-AM_CONDITIONAL(USE_SOUND_FFMPEG, test x$sound_handler = xffmpeg)
dnl --------------------------------------------------------
dnl ...
@@ -399,12 +391,12 @@
AC_PATH_PROG(DOXYGEN, doxygen)
AM_CONDITIONAL(ENABLE_DOXYGEN, [ test x"$DOXYGEN" != x ])
-GNASH_PATH_MP3
+GNASH_PATH_MAD
GNASH_PATH_OGG
-if test x$gui = xsdl; then
+if test x$gui = xsdl -o x$sound_handler = xsdl; then
GNASH_PATH_SDL
fi
-GNASH_PATH_SDL_MIXER
+
AM_CONDITIONAL(HAS_SDL, [ test x$has_sdl = xyes ])
GNASH_PATH_BOOST
@@ -449,7 +441,10 @@
AM_CONDITIONAL(HAS_GTK2, [ test x$has_gtk2 = xyes ])
+if test x"$mad" = x"no"; then
GNASH_PATH_FFMPEG
+fi
+
AM_CONDITIONAL(HAVE_FFMPEG, [test x"${FFMPEG_LIBS}" != x])
GNASH_PATH_GSTREAMER
@@ -622,10 +617,10 @@
echo " Intel 810 LOD bias hack disabled (default). Use
--enable-i810-lod-bias to enable."
fi
-if test x"$mp3" = x"yes"; then
+if test x"$mad" = x"yes"; then
echo " MP3 (libmad) support enabled"
else
- echo " MP3 (libmad) support disabled (default). Use --enable-mp3 to
enable"
+ echo " MP3 (libmad) support disabled (default). Use --enable-mad to
enable"
fi
if test x"$pthreads" = x"yes"; then
@@ -970,15 +965,15 @@
fi
if test x"$sound_handler" = x"sdl"; then
- if test x"$SDL_MIXER_LIBS" != x; then
- echo " SDL_Mixer flags are: $SDL_MIXER_CFLAGS"
- echo " SDL_Mixer libs are: $SDL_MIXER_LIBS"
- else
- echo "ERROR: sdl sound requested, but no SDL_Mixer development package
installed!\
- You need to have the SDL Mixer development package installed\
- if you wish to have sdl-based MP3 support for Gnash.\
+ if test x"$SDL_LIBS" != x; then
+ echo " SDL flags are: $SDL_CFLAGS"
+ echo " SDL libs are: $SDL_LIBS"
+ else
+ echo "ERROR: sdl sound requested, but no SDL development package
installed!\
+ You need to have the SDL development package installed\
+ if you wish to have sdl-based sound support for Gnash.\
To compile this project with SDL sound support, install\
- libsdl-mixer1.2-dev (using apt-get) or SDL_mixer-devel (using yum).
+ libsdl-1.2-dev (using apt-get) or SDL-devel (using yum).
In alternative build with --enable-sound=gst for gstreamer use"
nogo=true
fi
Index: backend/Makefile.am
===================================================================
RCS file: /sources/gnash/gnash/backend/Makefile.am,v
retrieving revision 1.37
retrieving revision 1.38
diff -u -b -r1.37 -r1.38
--- backend/Makefile.am 7 Oct 2006 09:43:33 -0000 1.37
+++ backend/Makefile.am 7 Oct 2006 14:20:27 -0000 1.38
@@ -20,11 +20,6 @@
AUTOMAKE_OPTIONS =
-if MP3
-MP3_CFLAGS = $(MAD_CFLAGS)
-MP3_LIBS = $(MAD_LIBS)
-endif
-
AM_CPPFLAGS = # -Wall
## WARNING: make sure GLIB_LIBS appears first
@@ -36,12 +31,10 @@
$(OGG_LIBS) \
$(ZLIB_LIBS) \
$(SDL_LIBS) \
- $(SDL_MIXER_LIBS) \
$(PANGO_LIBS) \
$(ATK_LIBS) \
$(X_LIBS) \
$(DMALLOC_LIBS) \
- $(MP3_LIBS) \
$(PTHREAD_LIBS) \
$(NULL)
@@ -54,14 +47,12 @@
-I$(top_srcdir)/libgeometry \
$(PTHREAD_CFLAGS) \
$(SDL_CFLAGS) \
- $(SDL_MIXER_CFLAGS) \
$(PANGO_CFLAGS) \
$(GLIB_CFLAGS) \
$(ATK_CFLAGS) \
$(DMALLOC_CFLAGS) \
$(PNG_CFLAGS) \
$(JPEG_CFLAGS) \
- $(MP3_CFLAGS) \
$(OGG_CFLAGS) \
$(NULL)
@@ -93,8 +84,13 @@
if USE_SOUND_SDL
SOUND_SOURCES = sound_handler_sdl.cpp
-if MP3
-SOUND_SOURCES += sound_handler_mp3.cpp
+if HAVE_FFMPEG
+AM_LDFLAGS += $(FFMPEG_LIBS)
+INCLUDES += $(FFMPEG_CFLAGS)
+endif
+if HAVE_MAD
+AM_LDFLAGS += $(MAD_LIBS)
+INCLUDES += $(MAD_CFLAGS)
endif
endif
Index: backend/sound_handler_gst.cpp
===================================================================
RCS file: /sources/gnash/gnash/backend/sound_handler_gst.cpp,v
retrieving revision 1.18
retrieving revision 1.19
diff -u -b -r1.18 -r1.19
--- backend/sound_handler_gst.cpp 20 Sep 2006 06:29:41 -0000 1.18
+++ backend/sound_handler_gst.cpp 7 Oct 2006 14:20:27 -0000 1.19
@@ -18,7 +18,7 @@
// Based on sound_handler_sdl.cpp by Thatcher Ulrich http://tulrich.com 2003
// which has been donated to the Public Domain.
-/* $Id: sound_handler_gst.cpp,v 1.18 2006/09/20 06:29:41 nihilus Exp $ */
+/* $Id: sound_handler_gst.cpp,v 1.19 2006/10/07 14:20:27 tgc Exp $ */
#ifdef HAVE_CONFIG_H
#include "config.h"
@@ -28,7 +28,7 @@
// (HAVE_GST_GST_H seems broken atm, specifically when an older glib
// install is around)
//
-//#ifdef HAVE_GST_GST_H
+
#ifdef SOUND_GST
#include "gnash.h"
#include "container.h"
@@ -256,7 +256,7 @@
// this gets called when a stream gets more data
- virtual long fill_stream_data(void* data, int data_bytes, int
handle_id)
+ virtual long fill_stream_data(void* data, int data_bytes, int
/*sample_count*/, int handle_id)
{
// @@ does a negative handle_id have any meaning ?
@@ -648,6 +648,16 @@
}
+ virtual void get_info(int sound_handle, int* format, bool* stereo) {
+
+ // Check if the sound exists.
+ if (sound_handle >= 0 && (unsigned int) sound_handle <
m_sound_data.size())
+ {
+ *format = m_sound_data[sound_handle]->format;
+ *stereo = m_sound_data[sound_handle]->stereo;
+ }
+
+ }
};
Index: backend/sound_handler_sdl.cpp
===================================================================
RCS file: /sources/gnash/gnash/backend/sound_handler_sdl.cpp,v
retrieving revision 1.20
retrieving revision 1.21
diff -u -b -r1.20 -r1.21
--- backend/sound_handler_sdl.cpp 20 Sep 2006 06:08:18 -0000 1.20
+++ backend/sound_handler_sdl.cpp 7 Oct 2006 14:20:27 -0000 1.21
@@ -1,106 +1,160 @@
-// sound_handler_sdl.cpp -- Thatcher Ulrich http://tulrich.com 2003
+// Copyright (C) 2005, 2006 Free Software Foundation, Inc.
+//
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+// GNU General Public License for more details.
+
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+//
-// This source code has been donated to the Public Domain. Do
-// whatever you want with it.
-
-// A sound_handler that uses SDL_mixer for output
-
-/* $Id: sound_handler_sdl.cpp,v 1.20 2006/09/20 06:08:18 nihilus Exp $ */
+// Based on sound_handler_sdl.cpp by Thatcher Ulrich http://tulrich.com 2003
+// which has been donated to the Public Domain.
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
-//#ifdef HAVE_SDL_MIXER_H
#ifdef SOUND_SDL
#include "gnash.h"
#include "container.h"
-#include "SDL_mixer.h"
#include "log.h"
#include "types.h" // for IF_VERBOSE_* macros
+#include <pthread.h>
+#include <cmath>
+#include <vector>
+
+#ifdef USE_FFMPEG
+#include <ffmpeg/avcodec.h>
+#include <ffmpeg/avformat.h>
+#elif defined(USE_MAD)
+#include <mad.h>
+#endif
+
+#include <SDL/SDL.h>
+#include <SDL/SDL_audio.h>
+
+void sdl_audio_callback(void *udata, Uint8 *stream, int len); // SDL C audio
handler
+
+// Used to hold the info about active sounds
+typedef struct
+{
+#ifdef USE_FFMPEG
+ // ffmpeg stuff
+ AVCodec *codec;
+ AVCodecContext *cc;
+#elif defined(USE_MAD)
+ // mad stuff
+ mad_stream stream;
+ mad_frame frame;
+ mad_synth synth;
+#endif
+
+ // data size
+ long data_size;
+
+ // position in the stream
+ long position;
+
+ // The compressed data
+ Uint8* data;
+
+ // data size
+ long raw_data_size;
+
+ // position in the raw stream
+ long raw_position;
+
+ // The decompressed data
+ Uint8* raw_data;
+
+ long loop_count;
+
+} active_sound;
+
+
+// Used to hold the sounddata when doing on-demand-decoding
+typedef struct
+{
+ // The (un)compressed data
+ Uint8* data;
+
+ // data format
+ int format;
+
+ // data size
+ long data_size;
+
+ // stereo or not
+ bool stereo;
+
+ // number of samples
+ int sample_count;
+
+ // sample rate
+ int sample_rate;
-using namespace gnash;
+ // Volume, SWF range: 0-100, SDL range 0-128
+ // It's the SWF range that is represented here
+ int volume;
-// Use SDL_mixer to handle sounds.
+ // active sounds being playes
+ std::vector<active_sound*> m_active_sounds;
+
+} sound_data;
+
+// Use SDL and ffmpeg/mad/nothing to handle sounds.
class SDL_sound_handler : public gnash::sound_handler
{
public:
- bool m_opened;
- bool m_stereo;
- int m_sample_rate;
- uint16_t m_format;
- std::vector<Mix_Chunk*> m_samples;
-
- #define SAMPLE_RATE 44100
- #define MIX_CHANNELS 8
- #define CHANNELS 2 //stereo - 2, mono - 1
- #define BUFSIZE 4096 // for 44100 bufsize 1024 is small
+ // Sound data.
+ std::vector<sound_data*> m_sound_data;
+
+ // Is sound device opened?
+ bool soundOpened;
+
+ // SDL_audio specs
+ SDL_AudioSpec audioSpec;
+
+ // Keeps track of numbers of playing sounds
+ int soundsPlaying;
+
+ // mutex for making sure threads doesn't mess things up
+ pthread_mutex_t mutex;
SDL_sound_handler()
- :
- m_opened(false),
- m_stereo(false),
- m_sample_rate(0),
- m_format(0)
+ : soundOpened(false),
+ soundsPlaying(0)
{
- // !!! some drivers on Linux always open audio with channels=2
- if (Mix_OpenAudio(SAMPLE_RATE, AUDIO_S16SYS, CHANNELS, BUFSIZE)
!= 0)
- {
- gnash::log_error("can't open SDL_mixer: %s\n",
Mix_GetError());
- }
- else
- {
- m_opened = true;
- Mix_AllocateChannels(MIX_CHANNELS);
- Mix_Volume(-1, MIX_MAX_VOLUME);
-
- // get and print the audio format in use
- int channels;
- int num_times_opened = Mix_QuerySpec(&m_sample_rate,
&m_format, &channels);
- UNUSED(num_times_opened);
- m_stereo = channels == 2 ? true : false;
-
- const char *format_str = "Unknown";
- switch (m_format)
- {
- case AUDIO_U8: format_str = "U8"; break;
- case AUDIO_S8: format_str = "S8"; break;
- case AUDIO_U16LSB: format_str = "U16LSB"; break;
- case AUDIO_S16LSB: format_str = "S16LSB"; break;
- case AUDIO_U16MSB: format_str = "U16MSB"; break;
- case AUDIO_S16MSB: format_str = "S16MSB"; break;
- }
-
- // @@ Usually I don't care about this. Need a
IF_VERBOSE_SOUND or something.
- // IF_VERBOSE_DEBUG(
- // gnash::log_msg("SDL_mixer: opened %d times,
frequency=%dHz, format=%s, stereo=%s\n",
- // num_times_opened,
- // m_sample_rate,
- // format_str,
- // m_stereo ? "yes" : "no"));
- // char name[32];
- // IF_VERBOSE_DEBUG(
- // gnash::log_msg("Using audio driver: %s\n",
SDL_AudioDriverName(name, 32)));
- }
+ // Init mutex
+ pthread_mutex_init(&mutex , NULL);
+
+ // This is our sound settings
+ audioSpec.freq = 44100;
+ audioSpec.format = AUDIO_S16SYS; // AUDIO_S8 AUDIO_U8;
+ audioSpec.channels = 2;
+ audioSpec.callback = sdl_audio_callback;
+ audioSpec.userdata = this;
+ audioSpec.samples = 512;
}
~SDL_sound_handler()
{
- if (m_opened)
- {
- Mix_CloseAudio();
- for (int i = m_samples.size() - 1; i >= 0; i--)
//Optimized
- {
- if (m_samples[i])
- {
- Mix_FreeChunk(m_samples[i]);
- }
- }
- }
- else
- {
- assert(m_samples.size() == 0);
+
+ for (size_t i= m_sound_data.size(); i > 0; i--) {
+ stop_sound(i);
+ delete_sound(i);
}
+ if (soundOpened) SDL_CloseAudio();
+ pthread_mutex_destroy(&mutex);
+
}
@@ -112,145 +166,366 @@
int sample_rate,
bool stereo)
// Called to create a sample. We'll return a sample ID that
- // can use for playing it.
- {
- if (m_opened == false)
+ // can be use for playing it.
{
- return 0;
+
+ sound_data *sounddata = new sound_data;
+ if (!sounddata) {
+ gnash::log_error("could not allocate memory for
sounddata !\n");
+ return -1;
}
+ sounddata->format = format;
+ sounddata->data_size = data_bytes;
+ sounddata->stereo = stereo;
+ sounddata->sample_count = sample_count;
+ sounddata->sample_rate = sample_rate;
+ sounddata->volume = 100;
+
int16_t* adjusted_data = 0;
int adjusted_size = 0;
- Mix_Chunk* sample = 0;
+
+ pthread_mutex_lock(&mutex);
switch (format)
{
case FORMAT_RAW:
+
+ if (data_bytes > 0) {
convert_raw_data(&adjusted_data, &adjusted_size, data,
sample_count, 1, sample_rate, stereo);
+ if (!adjusted_data) {
+ gnash::log_error("Some kind of error
with raw sound data\n");
+ pthread_mutex_unlock(&mutex);
+ return -1;
+ }
+ sounddata->data_size = adjusted_size;
+ sounddata->data = (Uint8*) adjusted_data;
+ }
break;
case FORMAT_NATIVE16:
+
+ if (data_bytes > 0) {
convert_raw_data(&adjusted_data, &adjusted_size, data,
sample_count, 2, sample_rate, stereo);
+ if (!adjusted_data) {
+ gnash::log_error("Some kind of error
with adpcm sound data\n");
+ pthread_mutex_unlock(&mutex);
+ return -1;
+ }
+ sounddata->data_size = adjusted_size;
+ sounddata->data = (Uint8*) adjusted_data;
+ }
break;
case FORMAT_MP3:
-#ifdef HAVE_MAD_H
- extern void convert_mp3_data(int16_t **adjusted_data,
int *adjusted_size, void *data, const int sample_count, const int sample_size,
const int sample_rate, const bool stereo);
- if (1) {
- int16_t* x_adjusted_data = 0;
- int x_adjusted_size = 0;
-// Mix_Chunk* x_sample = 0;
- convert_mp3_data(&x_adjusted_data,
&x_adjusted_size, data, sample_count, 0, sample_rate, stereo);
- /* convert_mp3_data doesn't ACTUALLY convert
- samplerate, so... */
- convert_raw_data(&adjusted_data,
&adjusted_size, x_adjusted_data, sample_count, 0, sample_rate, stereo);
- if (x_adjusted_data) {
- delete x_adjusted_data;
- }
- } else {
- convert_mp3_data(&adjusted_data,
&adjusted_size, data, sample_count, 0, sample_rate, stereo);
- }
-#else
- log_error("SDL sound handler: mp3 sound requested, but
MAD support was not compiled in\n");
+ //case FORMAT_VORBIS:
+#ifndef USE_FFMPEG
+#ifndef USE_MAD
+ gnash::log_warning("gnash has not been compiled to
handle mp3 audio\n");
+ pthread_mutex_unlock(&mutex);
+ return -1;
#endif
- break;
+#endif
+ sounddata->data = new Uint8[data_bytes];
+ if (!sounddata->data) {
+ gnash::log_error("could not allocate space for
data in soundhandler\n");
+ pthread_mutex_unlock(&mutex);
+ return -1;
+ }
+ memcpy(sounddata->data, data, data_bytes);
+ break;
default:
// Unhandled format.
- log_error("SDL sound handler: unknown format sound
requested\n");
- break;
+ gnash::log_error("unknown format sound requested; gnash
does not handle it\n");
+ pthread_mutex_unlock(&mutex);
+ return -1; // Unhandled format, set to NULL.
}
- if (adjusted_data)
- {
- sample = Mix_QuickLoad_RAW((unsigned char*)
adjusted_data, adjusted_size);
- Mix_VolumeChunk(sample, MIX_MAX_VOLUME); // full
volume by default
- }
+ m_sound_data.push_back(sounddata);
+ int sound_id = m_sound_data.size()-1;
+
+ pthread_mutex_unlock(&mutex);
+
+ return sound_id;
- m_samples.push_back(sample);
- return m_samples.size() - 1;
}
- virtual void play_sound(int sound_handle, int loop_count, int
secondOffset, long start_position)
- // Play the index'd sample.
- {
- if (m_opened && (sound_handle >= 0) && (unsigned int) sound_handle <
m_samples.size())
+ // this gets called when a stream gets more data
+ virtual long fill_stream_data(void* data, int data_bytes, int
sample_count, int handle_id)
{
- if (m_samples[sound_handle])
+
+ pthread_mutex_lock(&mutex);
+ // @@ does a negative handle_id have any meaning ?
+ // should we change it to unsigned instead ?
+ if (handle_id < 0 || (unsigned int) handle_id+1 >
m_sound_data.size()) {
+ pthread_mutex_unlock(&mutex);
+ return 1;
+ }
+ int start_size = 0;
+
+ // Handling of the sound data
+ if (m_sound_data[handle_id]->format == FORMAT_NATIVE16)
{
- // Play this sample on the first available
channel.
- Mix_PlayChannel(-1, m_samples[sound_handle],
loop_count);
+ int16_t* adjusted_data = 0;
+ int adjusted_size = 0;
+
+ convert_raw_data(&adjusted_data, &adjusted_size, data,
sample_count, 2, m_sound_data[handle_id]->sample_rate,
m_sound_data[handle_id]->stereo);
+ if (!adjusted_data || adjusted_size < 1) {
+ gnash::log_error("Some kind of error with
re-formating sound data\n");
+ pthread_mutex_unlock(&mutex);
+ return -1;
+ }
+ adjusted_data = (int16_t*)data;
+ adjusted_size = data_bytes;
+
+ // Reallocate the required memory.
+ Uint8* tmp_data = new Uint8[adjusted_size +
m_sound_data[handle_id]->data_size];
+ memcpy(tmp_data, m_sound_data[handle_id]->data,
m_sound_data[handle_id]->data_size);
+ memcpy(tmp_data + m_sound_data[handle_id]->data_size,
adjusted_data, adjusted_size);
+ if (m_sound_data[handle_id]->data_size > 0) delete []
m_sound_data[handle_id]->data;
+ m_sound_data[handle_id]->data = tmp_data;
+
+ start_size = m_sound_data[handle_id]->data_size;
+ m_sound_data[handle_id]->data_size += adjusted_size;
+
+ for(uint32_t i=0; i <
m_sound_data[handle_id]->m_active_sounds.size(); i++) {
+
m_sound_data[handle_id]->m_active_sounds[i]->data =
m_sound_data[handle_id]->data;
+
m_sound_data[handle_id]->m_active_sounds[i]->data_size =
m_sound_data[handle_id]->data_size;
+
m_sound_data[handle_id]->m_active_sounds[i]->position =
m_sound_data[handle_id]->data_size;
+
m_sound_data[handle_id]->m_active_sounds[i]->raw_data =
m_sound_data[handle_id]->data;
+ }
+ } else if (m_sound_data[handle_id]->format == FORMAT_MP3) {
+
+ // Reallocate the required memory.
+ Uint8* tmp_data = new Uint8[data_bytes +
m_sound_data[handle_id]->data_size];
+ memcpy(tmp_data, m_sound_data[handle_id]->data,
m_sound_data[handle_id]->data_size);
+ memcpy(tmp_data + m_sound_data[handle_id]->data_size,
data, data_bytes);
+ if (m_sound_data[handle_id]->data_size > 0) delete []
m_sound_data[handle_id]->data;
+ m_sound_data[handle_id]->data = tmp_data;
+
+ start_size = m_sound_data[handle_id]->data_size;
+ m_sound_data[handle_id]->data_size += data_bytes;
+
+ // If playback has already started, we also update the
active sounds
+ for(uint32_t i=0; i <
m_sound_data[handle_id]->m_active_sounds.size(); i++) {
+
m_sound_data[handle_id]->m_active_sounds[i]->data =
m_sound_data[handle_id]->data;
+
m_sound_data[handle_id]->m_active_sounds[i]->data_size =
m_sound_data[handle_id]->data_size;
+#ifdef USE_MAD
+ uint64_t this_frame = (const
Uint8*)m_sound_data[handle_id]->m_active_sounds[i]->stream.this_frame -
m_sound_data[handle_id]->m_active_sounds[i]->data;
+ uint64_t next_frame = (const
Uint8*)m_sound_data[handle_id]->m_active_sounds[i]->stream.next_frame -
m_sound_data[handle_id]->m_active_sounds[i]->data;
+
mad_stream_buffer(&m_sound_data[handle_id]->m_active_sounds[i]->stream,
m_sound_data[handle_id]->data, m_sound_data[handle_id]->data_size);
+
m_sound_data[handle_id]->m_active_sounds[i]->stream.this_frame += this_frame;
+
m_sound_data[handle_id]->m_active_sounds[i]->stream.next_frame += next_frame;
+#endif
}
+
+ } else {
+ gnash::log_error("Behavior for this codec is unknown.
Please send this SWF to the developers\n");
}
+
+ pthread_mutex_unlock(&mutex);
+ return start_size;
+
+
}
- virtual void stop_all_sounds()
- {
- if (m_opened)
- {
- for (int i = 0; i < MIX_CHANNELS; i++)
+
+ virtual void play_sound(int sound_handle, int loop_count, int
/*offset*/, long start_position)
+ // Play the index'd sample.
{
- if (Mix_Playing(i))
+ pthread_mutex_lock(&mutex);
+
+ // Check if the sound exists.
+ if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size())
{
- Mix_HaltChannel(i);
+ // Invalid handle.
+ pthread_mutex_unlock(&mutex);
+ return;
+ }
+
+ // If this is called from a streamsoundblocktag, we only start
if this
+ // sound isn't already playing. If a active_sound-struct is
existing we
+ // assume it is also playing.
+ if (start_position > 0 &&
m_sound_data[sound_handle]->m_active_sounds.size() > 0) {
+ pthread_mutex_unlock(&mutex);
+ return;
+ }
+
+ // Make a "active_sound" for this sound which is later placed
on the vector of instances of this sound being played
+ active_sound* sound = new active_sound;
+
+ // Copy data-info to the active_sound
+ sound->data_size = m_sound_data[sound_handle]->data_size;
+ sound->data = m_sound_data[sound_handle]->data;
+ sound->position = start_position;
+ if (start_position < 0) sound->position = 0;
+
+
+ // Set number of loop we should do. -1 is infinte loop, 0 plays
it once, 1 twice etc.
+ sound->loop_count = loop_count;
+
+ if (m_sound_data[sound_handle]->format == FORMAT_MP3) {
+
+#ifdef USE_FFMPEG
+ // Init the avdecoder-decoder
+ avcodec_init();
+ avcodec_register_all();// change this to only register
mp3?
+ sound->codec = avcodec_find_decoder(CODEC_ID_MP3);
+
+ if (!sound->codec) {
+ gnash::log_error("Your FFMPEG can't decode
MP3?!\n");
+ pthread_mutex_unlock(&mutex);
+ return;
+ }
+
+ sound->raw_data = new
uint8[AVCODEC_MAX_AUDIO_FRAME_SIZE];
+ memset((void*)sound->raw_data, 0,
AVCODEC_MAX_AUDIO_FRAME_SIZE);
+ sound->raw_position = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ sound->raw_data_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+ sound->cc = avcodec_alloc_context();
+ avcodec_open(sound->cc, sound->codec);
+#elif defined(USE_MAD)
+ // Init the mad decoder
+ mad_stream_init(&sound->stream);
+ mad_frame_init(&sound->frame);
+ mad_synth_init(&sound->synth);
+ mad_stream_buffer(&sound->stream, sound->data,
sound->data_size);
+
+ sound->raw_data = 0;
+ sound->raw_data_size = 0;
+ sound->raw_position = 0;
+#endif
+ } else {
+ sound->raw_data_size =
m_sound_data[sound_handle]->data_size;
+ sound->raw_data = m_sound_data[sound_handle]->data;
+ sound->raw_position = 0;
+ sound->position = m_sound_data[sound_handle]->data_size;
+
+ }
+
+ if (!soundOpened) {
+ if (SDL_OpenAudio(&audioSpec, NULL) < 0 ) {
+ gnash::log_error("Unable to START SOUND: %s\n",
SDL_GetError());
+ pthread_mutex_unlock(&mutex);
+ return;
}
+ soundOpened = true;
+
+
}
+
+ ++soundsPlaying;
+ m_sound_data[sound_handle]->m_active_sounds.push_back(sound);
+
+ if (soundsPlaying == 1) {
+ SDL_PauseAudio(0);
}
+
+ pthread_mutex_unlock(&mutex);
+
}
- virtual int get_volume(int sound_handle)
+
+ virtual void stop_sound(int sound_handle)
{
- int previous_volume = 100;
- if (m_opened && (sound_handle >= 0) && (unsigned int)
sound_handle < m_samples.size())
+ pthread_mutex_lock(&mutex);
+
+ // Check if the sound exists.
+ if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size())
{
- // if you passed a negative value for volume then
- // this volume is still the current volume for the
chunk
- previous_volume =
Mix_VolumeChunk(m_samples[sound_handle], -1);
+ // Invalid handle.
+ } else {
+
+ for (uint32_t i = 0; i <
m_sound_data[sound_handle]->m_active_sounds.size(); i++) {
+
m_sound_data[sound_handle]->m_active_sounds[i]->position =
m_sound_data[sound_handle]->m_active_sounds[i]->data_size;
+
m_sound_data[sound_handle]->m_active_sounds[i]->raw_position =
m_sound_data[sound_handle]->m_active_sounds[i]->raw_data_size;
+
m_sound_data[sound_handle]->m_active_sounds[i]->loop_count = 0;
+ }
}
- return previous_volume;
+ pthread_mutex_unlock(&mutex);
+
}
- virtual void set_volume(int sound_handle, int volume)
+
+ virtual void delete_sound(int sound_handle)
+ // this gets called when it's done with a sample.
{
- if (m_opened && sound_handle >= 0 && (unsigned int)
sound_handle < m_samples.size())
+ pthread_mutex_lock(&mutex);
+
+ if (sound_handle >= 0 && (unsigned int) sound_handle <
m_sound_data.size())
{
- int vol = (MIX_MAX_VOLUME / 100) * volume;
- Mix_VolumeChunk(m_samples[sound_handle], vol);
+ delete[] m_sound_data[sound_handle]->data;
}
+ pthread_mutex_unlock(&mutex);
+
}
- virtual void stop_sound(int sound_handle)
- {
- if (m_opened && sound_handle >= 0 && (unsigned int)
sound_handle < m_samples.size())
- {
- for (int i = 0; i < MIX_CHANNELS; i++)
+ // This will stop all sounds playing. Will cause problems if the
soundhandler is made static
+ // and supplys sound_handling for many SWF's, since it will stop all
sounds with no regard
+ // for what sounds is associated with what SWF.
+ virtual void stop_all_sounds()
{
- Mix_Chunk* playing_chunk = Mix_GetChunk(i);
- if (playing_chunk == m_samples[sound_handle])
+ int num_sounds = m_sound_data.size();
+ for (size_t i = num_sounds; i > 0; i--) //Optimized
+ stop_sound(i);
+ }
+
+
+ // returns the sound volume level as an integer from 0 to 100,
+ // where 0 is off and 100 is full volume. The default setting is
100.
+ virtual int get_volume(int sound_handle) {
+
+ pthread_mutex_lock(&mutex);
+
+ int ret;
+ // Check if the sound exists.
+ if (sound_handle >= 0 && (unsigned int) sound_handle <
m_sound_data.size())
{
- // Stop this channel.
- Mix_HaltChannel(i);
+ ret = m_sound_data[sound_handle]->volume;
+ } else {
+ ret = 0; // Invalid handle
}
+ pthread_mutex_unlock(&mutex);
+ return ret;
}
+
+
+ // A number from 0 to 100 representing a volume level.
+ // 100 is full volume and 0 is no volume. The default setting is
100.
+ virtual void set_volume(int sound_handle, int volume) {
+
+ pthread_mutex_lock(&mutex);
+
+ // Check if the sound exists.
+ if (sound_handle < 0 || (unsigned int) sound_handle >=
m_sound_data.size())
+ {
+ // Invalid handle.
+ } else {
+
+ // Set volume for this sound. Should this only apply to
the active sounds?
+ m_sound_data[sound_handle]->volume = volume;
}
+ pthread_mutex_unlock(&mutex);
+
+
}
+ virtual void get_info(int sound_handle, int* format, bool* stereo) {
- virtual void delete_sound(int sound_handle)
- // this gets called this when it's done with a sample.
- {
- if (sound_handle >= 0 && (unsigned int) sound_handle <
m_samples.size())
- {
- Mix_Chunk* chunk = m_samples[sound_handle];
- if (chunk)
+ pthread_mutex_lock(&mutex);
+
+ // Check if the sound exists.
+ if (sound_handle >= 0 && (unsigned int) sound_handle <
m_sound_data.size())
{
- delete [] (chunk->abuf);
- Mix_FreeChunk(chunk);
- m_samples[sound_handle] = 0;
- }
+ *format = m_sound_data[sound_handle]->format;
+ *stereo = m_sound_data[sound_handle]->stereo;
}
+
+ pthread_mutex_unlock(&mutex);
}
+
virtual void convert_raw_data(
int16_t** adjusted_data,
int* adjusted_size,
@@ -274,6 +549,9 @@
// }
// // xxxxx
+ bool m_stereo = (audioSpec.channels == 2 ? true : false);
+ int m_sample_rate = audioSpec.freq;
+
// simple hack to handle dup'ing mono to stereo
if ( !stereo && m_stereo)
{
@@ -334,6 +612,8 @@
}
}
+
+
};
@@ -343,8 +623,232 @@
return new SDL_sound_handler;
}
+// The callback function which refills the buffer with data
+void sdl_audio_callback (void *udata, Uint8 *stream, int buffer_length)
+{
+ // We run through all of the sounds, and mix all of the active sounds
+ // into the stream given by the callback.
+ // If a sound is looping it will be decoded from the beginning again.
+ // When a sound is finished the empty space in the stream will be set
to 0.
+
+ SDL_sound_handler* handler = static_cast<SDL_sound_handler*>(udata);
+
+ // If nothing to play there is no reason to play
+ // Is this a potential deadlock problem?
+ if (handler->soundsPlaying == 0) {
+ SDL_PauseAudio(1);
+ return;
+ }
+
+ pthread_mutex_lock(&handler->mutex);
+
+ // Mixed sounddata buffer
+ Uint8* buffer = stream; //new Uint8[len];
+ memset(buffer, 0, buffer_length);
+
+ for(uint32_t i=0; i < handler->m_sound_data.size(); i++) {
+ for(uint32_t j = 0; j <
handler->m_sound_data[i]->m_active_sounds.size(); j++) {
+
+ active_sound* sound =
handler->m_sound_data[i]->m_active_sounds[j];
+
+ // When the current sound dont have enough decoded data
to fill the buffer,
+ // we first mix what is alreadt decoded, then decode
some more data, and
+ // mix some more until the buffer is full. If a sound
loops the magic
+ // happens here ;)
+ if (sound->raw_data_size - sound->raw_position <
buffer_length
+ && (sound->position < sound->data_size ||
sound->loop_count != 0)) {
+
+ // First we mix what is decoded
+ int index = 0;
+ if (sound->raw_data_size - sound->raw_position
> 0) {
+ SDL_MixAudio(stream, (const
Uint8*)(sound->raw_data + sound->raw_position),
+ sound->raw_data_size -
sound->raw_position,
+ SDL_MIX_MAXVOLUME);
+ index = sound->raw_data_size -
sound->raw_position;
+ }
+ sound->raw_position += index;
+
+ // Then we decode some data
+ int outsize = 0;
+#ifdef USE_FFMPEG
+ // If we need to loop, we reset the data pointer
+ if (sound->data_size == sound->position &&
sound->loop_count != 0) {
+ sound->loop_count--;
+ sound->position = 0;
+ }
+
+ memset(sound->raw_data, 0,
sound->raw_data_size);
+ int bytes_decoded =
avcodec_decode_audio(sound->cc,
+ (int16_t *)sound->raw_data,
+ &outsize,
+ (uint8_t *)(sound->data +
sound->position),
+ //AVCODEC_MAX_AUDIO_FRAME_SIZE);
+ sound->data_size -
sound->position);
+
+ if (bytes_decoded < 0) {
+ gnash::log_error("Error while decoding
MP3-stream\n");
+ // TODO: Remove the sound from the
active list
+ sound->position = sound->data_size;
+ continue;
+ }
+
+ sound->position += bytes_decoded;
+
+#elif defined(USE_MAD)
+ // If we need to loop, we reset the data
pointer, and tell mad about it
+ if (sound->data_size == sound->position &&
sound->loop_count != 0) {
+ sound->position = 0;
+ mad_stream_buffer(&sound->stream,
sound->data, sound->data_size);
+ sound->loop_count--;
+ }
+
+ int ret;
+ while(true) {
+ ret = mad_frame_decode(&sound->frame,
&sound->stream);
+
+ // There is always some junk in front
of the data,
+ // so we continue until we get past it.
+ if (ret && sound->stream.error ==
MAD_ERROR_LOSTSYNC) continue;
+
+ break;
+ }
+
+ if (ret == -1 && sound->stream.error !=
MAD_ERROR_BUFLEN) {
+ gnash::log_error("Error while decoding
MP3-stream, MAD error: %d", sound->stream.error);
+ continue;
+ } else if (ret == -1 && sound->stream.error ==
MAD_ERROR_BUFLEN) {
+ // the buffer is empty, no more to
decode!
+ sound->position = sound->data_size;
+ } else {
+ sound->position +=
sound->stream.next_frame - sound->stream.buffer - sound->position;
+ }
+
+ mad_synth_frame (&sound->synth, &sound->frame);
+
+ outsize = sound->synth.pcm.length *
((handler->m_sound_data[i]->stereo == true) ? 4 : 2);
+
+ if (sound->raw_data) delete[] sound->raw_data;
+ sound->raw_data = new Uint8[outsize];
+ int sample;
+
+ int16_t* dst = (int16_t*) sound->raw_data;
+
+ // transfer the decoded samples into the
sound-struct, and do some
+ // scaling while we're at it.
+ for(int f = 0; f < sound->synth.pcm.length; f++)
+ {
+ for (int e = 0; e <
((handler->m_sound_data[i]->stereo == true) ? 2 : 1); e++){ // channels
(stereo/mono)
+
+ mad_fixed_t mad_sample =
sound->synth.pcm.samples[e][f];
+
+ // round
+ mad_sample += (1L <<
(MAD_F_FRACBITS - 16));
+
+ // clip
+ if (mad_sample >= MAD_F_ONE)
mad_sample = MAD_F_ONE - 1;
+ else if (mad_sample <
-MAD_F_ONE) mad_sample = -MAD_F_ONE;
+
+ // quantize
+ sample = mad_sample >>
(MAD_F_FRACBITS + 1 - 16);
+
+ if ( sample != (int16_t)sample
) sample = sample < 0 ? -32768 : 32767;
+
+ *dst++ = sample;
+ }
+ }
+
+#endif
+ // If we need to convert samplerate...
+ if (outsize > 0 &&
handler->m_sound_data[i]->sample_rate != handler->audioSpec.freq) {
+ int16_t* adjusted_data = 0;
+ int adjusted_size = 0;
+ int sample_count = outsize /
((handler->m_sound_data[i]->stereo == true) ? 4 : 2);
+
+ // Convert to needed samplerate
+
handler->convert_raw_data(&adjusted_data, &adjusted_size, sound->raw_data,
sample_count, 0,
+
handler->m_sound_data[i]->sample_rate, handler->m_sound_data[i]->stereo);
+
+ // Hopefully this wont happen
+ if (!adjusted_data) {
+ continue;
+ }
+
+ // Move the new data to the sound-struct
+ if (sound->raw_data) delete[]
sound->raw_data;
+ sound->raw_data = new
Uint8[adjusted_size];
+ memcpy(sound->raw_data, adjusted_data,
adjusted_size);
+ sound->raw_data_size = adjusted_size;
+ delete[] adjusted_data;
+
+ } else {
+ sound->raw_data_size = outsize;
+ }
+
+ sound->raw_position = 0;
+
+
+ // Then we mix the newly decoded data
+ SDL_MixAudio((Uint8*)(stream+index),(const
Uint8*) sound->raw_data,
+ buffer_length - index,
+ SDL_MIX_MAXVOLUME);
+ sound->raw_position = buffer_length - index;
+
+ // When the current sound has enough decoded data to
fill
+ // the buffer, we do just that.
+ } else if (sound->raw_data_size - sound->raw_position
>= buffer_length ) {
+
+ // Mix the raw data
+ SDL_MixAudio((Uint8*)(stream),(const Uint8*)
(sound->raw_data + sound->raw_position),
+ buffer_length,
+ SDL_MIX_MAXVOLUME);
+ sound->raw_position += buffer_length;
+
+ // When the current sound doesn't have anymore data to
decode,
+ // and doesn't loop (anymore), but still got unplayed
data,
+ // we put the last data on the stream
+ } else if (sound->raw_data_size - sound->raw_position <
buffer_length && sound->raw_data_size > sound->raw_position) {
+
+ // Mix the remaining data
+ SDL_MixAudio((Uint8*)(stream),(const Uint8*)
(sound->raw_data + sound->raw_position),
+ sound->raw_data_size -
sound->raw_position,
+ SDL_MIX_MAXVOLUME);
+ sound->raw_position = sound->raw_data_size;
+
+ } else {
+ }
+
+ // Sound is done, remove it from the active list (mp3)
+ if (sound->position == sound->data_size &&
sound->loop_count == 0 && handler->m_sound_data[i]->format == 2) {
+#ifdef USE_FFMPEG
+ avcodec_close(sound->cc);
+#elif defined(USE_MAD)
+ mad_synth_finish(&sound->synth);
+ mad_frame_finish(&sound->frame);
+ mad_stream_finish(&sound->stream);
+#endif
+ delete[] sound->raw_data;
+
handler->m_sound_data[i]->m_active_sounds.erase(handler->m_sound_data[i]->m_active_sounds.begin()
+ j);
+ handler->soundsPlaying--;
+
+
+ // Sound is done, remove it from the active list
(adpcm/native16)
+ } else if (sound->loop_count == 0 &&
handler->m_sound_data[i]->format == 7 && sound->raw_position >=
sound->raw_data_size && sound->raw_data_size != 0) {
+
handler->m_sound_data[i]->m_active_sounds.erase(handler->m_sound_data[i]->m_active_sounds.begin()
+ j);
+ handler->soundsPlaying--;
+ } else if (sound->raw_position == 0 &&
sound->raw_data_size == 0) {
+
handler->m_sound_data[i]->m_active_sounds.erase(handler->m_sound_data[i]->m_active_sounds.begin()
+ j);
+ handler->soundsPlaying--;
+ }
+
+ }
+ }
+ pthread_mutex_unlock(&handler->mutex);
+
+}
+
#endif
// Local Variables:
// mode: C++
// End:
+
Index: macros/ffmpeg.m4
===================================================================
RCS file: /sources/gnash/gnash/macros/ffmpeg.m4,v
retrieving revision 1.3
retrieving revision 1.4
diff -u -b -r1.3 -r1.4
--- macros/ffmpeg.m4 7 Oct 2006 02:54:24 -0000 1.3
+++ macros/ffmpeg.m4 7 Oct 2006 14:20:27 -0000 1.4
@@ -131,6 +131,8 @@
FFMPEG_LIBS="${ac_cv_path_ffmpeg_lib} -lavcodec -lavutil"
fi
+ AC_DEFINE([USE_FFMPEG], [1], [Use FFMPEG for MP3 decoding])
+
AC_SUBST(FFMPEG_LIBS)
])
Index: macros/mad.m4
===================================================================
RCS file: /sources/gnash/gnash/macros/mad.m4,v
retrieving revision 1.10
retrieving revision 1.11
diff -u -b -r1.10 -r1.11
--- macros/mad.m4 9 Aug 2006 01:59:07 -0000 1.10
+++ macros/mad.m4 7 Oct 2006 14:20:27 -0000 1.11
@@ -35,16 +35,16 @@
dnl
dnl
-AC_DEFUN([GNASH_PATH_MP3],
+AC_DEFUN([GNASH_PATH_MAD],
[
- AC_ARG_ENABLE(mp3, [ --enable-mp3 Enable support for playing
mp3s],
+ AC_ARG_ENABLE(mad, [ --enable-mad Enable support for decoding
mp3 with mad],
[case "${enableval}" in
- yes) mp3=yes ;;
- no) mp3=no ;;
- *) AC_MSG_ERROR([bad value ${enableval} for enable-mp3 option]) ;;
- esac], mp3=no)
+ yes) mad=yes ;;
+ no) mad=no ;;
+ *) AC_MSG_ERROR([bad value ${enableval} for enable-mad option]) ;;
+ esac], mad=no)
- if test x"$mp3" = x"yes"; then
+ if test x"$mad" = x"yes"; then
dnl Look for the header
AC_ARG_WITH(mad_incl, [ --with-mad_incl directory where libmad
header is], with_mad_incl=${withval})
AC_CACHE_VAL(ac_cv_path_mad_incl,[
@@ -142,9 +142,10 @@
if test x"${ac_cv_path_mad_lib}" != x ; then
MAD_LIBS="${ac_cv_path_mad_lib}"
+ AC_DEFINE([USE_MAD], [1], [Use MAD for MP3 decoding])
fi
- AM_CONDITIONAL(MP3, [test x$mp3 = xyes])
+ AM_CONDITIONAL(HAVE_MAD, [test x$mad = xyes])
AC_SUBST(MAD_CFLAGS)
AC_SUBST(MAD_LIBS)
Index: plugin/Makefile.am
===================================================================
RCS file: /sources/gnash/gnash/plugin/Makefile.am,v
retrieving revision 1.45
retrieving revision 1.46
diff -u -b -r1.45 -r1.46
--- plugin/Makefile.am 4 Oct 2006 05:13:00 -0000 1.45
+++ plugin/Makefile.am 7 Oct 2006 14:20:27 -0000 1.46
@@ -43,11 +43,6 @@
endif
SUBDIRS = $(KLASH) # test mozilla-sdk
-if MP3
-MP3_CFLAGS = $(MAD_CFLAGS)
-MP3_LIBS = $(MAD_LIBS)
-endif
-
# if DMALLOC
# MALL_CFLAGS = $(DMALLOC_CFLAGS)
# MALL_LIBS = $(DMALLOC_LIBS)
@@ -71,10 +66,8 @@
$(CAIRO_CFLAGS) \
$(ATK_CFLAGS) \
$(SDL_CFLAGS) \
- $(SDL_MIXER_CFLAGS) \
$(X_CFLAGS) \
$(DMALLOC_CFLAGS) \
- $(MP3_CFLAGS) \
$(OGG_CFLAGS)
Index: server/gnash.h
===================================================================
RCS file: /sources/gnash/gnash/server/gnash.h,v
retrieving revision 1.56
retrieving revision 1.57
diff -u -b -r1.56 -r1.57
--- server/gnash.h 2 Oct 2006 11:33:08 -0000 1.56
+++ server/gnash.h 7 Oct 2006 14:20:27 -0000 1.57
@@ -35,7 +35,7 @@
//
//
-/* $Id: gnash.h,v 1.56 2006/10/02 11:33:08 nihilus Exp $ */
+/* $Id: gnash.h,v 1.57 2006/10/07 14:20:27 tgc Exp $ */
/// \mainpage
///
@@ -432,12 +432,14 @@
int sample_rate, /* one of 5512, 11025, 22050,
44100 */
bool stereo
) = 0;
-#ifdef SOUND_GST
+
// gnash calls this to fill up soundstreams data
- virtual long fill_stream_data(void* data, int data_bytes, int
handle_id) = 0;
-#endif
+ virtual long fill_stream_data(void* data, int data_bytes, int
sample_count, int handle_id) = 0;
+
+ // Gives info about the format, samplerate and stereo of the sound
in question;
+ virtual void get_info(int sound_handle, int* format, bool* stereo) = 0;
+
// gnash calls this when it wants you to play the defined sound.
- //
// loop_count == 0 means play the sound once (1 means play it twice,
etc)
virtual void play_sound(int sound_handle, int loop_count, int
secondOffset, long start) = 0;
Index: server/swf/tag_loaders.cpp
===================================================================
RCS file: /sources/gnash/gnash/server/swf/tag_loaders.cpp,v
retrieving revision 1.52
retrieving revision 1.53
diff -u -b -r1.52 -r1.53
--- server/swf/tag_loaders.cpp 2 Oct 2006 17:14:41 -0000 1.52
+++ server/swf/tag_loaders.cpp 7 Oct 2006 14:20:27 -0000 1.53
@@ -36,7 +36,7 @@
//
//
-/* $Id: tag_loaders.cpp,v 1.52 2006/10/02 17:14:41 strk Exp $ */
+/* $Id: tag_loaders.cpp,v 1.53 2006/10/07 14:20:27 tgc Exp $ */
#ifdef HAVE_CONFIG_H
#include "config.h"
@@ -1684,8 +1684,6 @@
// If we don't have a sound_handler registered stop here
if (!s_sound_handler) return;
-#ifdef SOUND_GST
-
// FIXME:
// no character id for soundstreams... so we make one up...
// This only works if there is only one stream in the movie...
@@ -1723,6 +1721,9 @@
return;
}
+ // Since the ADPCM is converted to NATIVE16, the format is set to
that...
+ if (format == sound_handler::FORMAT_ADPCM) format =
sound_handler::FORMAT_NATIVE16;
+
int handler_id = s_sound_handler->create_sound(
NULL,
data_bytes,
@@ -1732,15 +1733,6 @@
stereo);
m->set_loading_sound_stream_id(handler_id);
-#else
- static bool already_warned=false;
- if ( ! already_warned )
- {
- log_warning("Only Gstreamer sound backend "
- "supports SoundStreamHead tag");
- already_warned=true;
- }
-#endif
}
@@ -1752,9 +1744,6 @@
assert(tag == SWF::SOUNDSTREAMBLOCK); // 19
-#ifdef SOUND_GST
-
-
// discard garbage data
//int garbage = in->read_u32();
in->skip_bytes(4);
@@ -1763,6 +1752,8 @@
// If we don't have a sound_handler registered stop here
if (!s_sound_handler) return;
+ int handle_id = m->get_loading_sound_stream_id();
+
// store the data with the appropiate sound.
int data_bytes = 0;
@@ -1770,18 +1761,59 @@
data_bytes = in->get_tag_end_position() - in->get_position();
if (data_bytes <= 0) return;
-
unsigned char *data = new unsigned char[data_bytes];
+
+
+ int format = 0;
+ bool stereo = true;
+ int sample_count = -1;
+
+ s_sound_handler->get_info(handle_id, &format, &stereo);
+
+ if (format == sound_handler::FORMAT_ADPCM)
+ {
+ // Uncompress the ADPCM before handing data to host.
+ sample_count = data_bytes / (stereo ? 4 : 2);
+ data_bytes = sample_count * (stereo ? 4 : 2);
+ data = new unsigned char[data_bytes];
+ sound_handler::adpcm_expand(data, in, sample_count, stereo);
+ format = sound_handler::FORMAT_NATIVE16;
+ } else if (format == sound_handler::FORMAT_NATIVE16)
+ {
+ // Raw data
+ sample_count = data_bytes / (stereo ? 4 : 2);
for (int i = 0; i < data_bytes; i++)
{
data[i] = in->read_u8();
}
- int handle_id = m->get_loading_sound_stream_id();
+ } else {
+
+ for (int i = 0; i < data_bytes; i++)
+ {
+ data[i] = in->read_u8();
+ }
+
+ // Swap bytes on behalf of the host, to make it easier for the
handler.
+ // @@ I'm assuming this is a good idea? Most sound handlers
will prefer native endianness?
+ /*if (format == sound_handler::FORMAT_UNCOMPRESSED &&
sample_16bit)
+ {
+ #ifndef _TU_LITTLE_ENDIAN_
+ // Swap sample bytes to get big-endian format.
+ for (int i = 0; i < data_bytes - 1; i += 2)
+ {
+ swap(&data[i], &data[i+1]);
+ }
+ #endif // not _TU_LITTLE_ENDIAN_
+
+ format = sound_handler::FORMAT_NATIVE16;
+ }*/
+ }
+
// Fill the data on the apropiate sound, and receives the starting point
// for later "start playing from this frame" events.
- long start = s_sound_handler->fill_stream_data(data, data_bytes,
handle_id);
+ long start = s_sound_handler->fill_stream_data(data, data_bytes,
sample_count, handle_id);
delete [] data;
@@ -1790,15 +1822,6 @@
// @@ who's going to delete the start_stream_sound_tag ??
-#else
- static bool already_warned=false;
- if ( ! already_warned )
- {
- log_warning("Only Gstreamer sound backend "
- "supports SoundStreamBlock tag");
- already_warned=true;
- }
-#endif
}
Index: utilities/Makefile.am
===================================================================
RCS file: /sources/gnash/gnash/utilities/Makefile.am,v
retrieving revision 1.26
retrieving revision 1.27
diff -u -b -r1.26 -r1.27
--- utilities/Makefile.am 3 Oct 2006 15:40:50 -0000 1.26
+++ utilities/Makefile.am 7 Oct 2006 14:20:27 -0000 1.27
@@ -20,11 +20,6 @@
AUTOMAKE_OPTIONS = # dejagnu
-if MP3
-MP3_CFLAGS = $(MAD_CFLAGS)
-MP3_LIBS = $(MAD_LIBS)
-endif
-
# @@ shouldn't we drop package_LIBS and leave deps
# on the gnash libraries side instead ?
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