From 3c0801cb92aaed698b88db617607534673e3efe7 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Rafa=C3=ABl=20Carr=C3=A9?= <funman@videolan.org>
Date: Mon, 13 Jun 2016 16:05:35 +0200
Subject: [PATCH] audio_filter: remove dtstospdif
Modified-By: Thomas Guillem <thomas@gllm.fr>
Signed-off-by: Thomas Guillem <thomas@gllm.fr>
---
extras/package/rpm/vlc.altlinux.spec | 1 -
modules/MODULES_LIST | 1 -
modules/audio_filter/Makefile.am | 4 +-
modules/audio_filter/converter/dtstospdif.c | 225 --------------------
po/POTFILES.in | 1 -
5 files changed, 1 insertion(+), 231 deletions(-)
delete mode 100644 modules/audio_filter/converter/dtstospdif.c
diff --git a/extras/package/rpm/vlc.altlinux.spec b/extras/package/rpm/vlc.altlinux.spec
index 9bf33de94982..9add728cc29c 100644
--- a/extras/package/rpm/vlc.altlinux.spec
+++ b/extras/package/rpm/vlc.altlinux.spec
@@ -1009,7 +1009,6 @@ strfile %buildroot%_gamesdatadir/fortune/vlc %buildroot%_gamesdatadir/fortune/vl
%dir %_vlc_pluginsdir/audio_filter
%_vlc_pluginsdir/audio_filter/libbandlimited_resampler_plugin.so*
%_vlc_pluginsdir/audio_filter/libdolby_surround_decoder_plugin.so*
-%_vlc_pluginsdir/audio_filter/libdtstospdif_plugin.so*
%_vlc_pluginsdir/audio_filter/libheadphone_channel_mixer_plugin.so*
%_vlc_pluginsdir/audio_filter/liblinear_resampler_plugin.so*
%_vlc_pluginsdir/audio_filter/libtrivial_channel_mixer_plugin.so*
diff --git a/modules/MODULES_LIST b/modules/MODULES_LIST
index c5a983ff9363..0c94d604b4f0 100644
--- a/modules/MODULES_LIST
+++ b/modules/MODULES_LIST
@@ -117,7 +117,6 @@ $Id$
* dsm: SMB access module
* dts: DTS basic parser/packetizer
* dtstofloat32: DTS Audio converter
- * dtstospdif: Audio converter that encapsulates DTS into S/PDIF
* dtv: DVB support (superseds bda module for Windows)
* dummy: dummy interface
* dv1394: Digital Video (Firewire/IEEE1394/I-Link) access module
diff --git a/modules/audio_filter/Makefile.am b/modules/audio_filter/Makefile.am
index da23f7e0800c..a436ed2af0f8 100644
--- a/modules/audio_filter/Makefile.am
+++ b/modules/audio_filter/Makefile.am
@@ -99,12 +99,10 @@ libaudio_format_plugin_la_CPPFLAGS = $(AM_CPPFLAGS)
libaudio_format_plugin_la_LIBADD = $(LIBM)
liba52tospdif_plugin_la_SOURCES = audio_filter/converter/a52tospdif.c
-libdtstospdif_plugin_la_SOURCES = audio_filter/converter/dtstospdif.c
audio_filter_LTLIBRARIES += \
liba52tospdif_plugin.la \
- libaudio_format_plugin.la \
- libdtstospdif_plugin.la
+ libaudio_format_plugin.la
# Resamplers
libbandlimited_resampler_plugin_la_SOURCES = \
diff --git a/modules/audio_filter/converter/dtstospdif.c b/modules/audio_filter/converter/dtstospdif.c
deleted file mode 100644
index 99c57bdd2c1f..000000000000
--- a/modules/audio_filter/converter/dtstospdif.c
+++ /dev/null
@@ -1,225 +0,0 @@
-/*****************************************************************************
- * dtstospdif.c : encapsulates DTS frames into S/PDIF packets
- *****************************************************************************
- * Copyright (C) 2003-2009 the VideoLAN team
- * $Id$
- *
- * Authors: Jon Lech Johansen <jon-vl@nanocrew.net>
- * Gildas Bazin
- * Derk-Jan Hartman
- * Pierre d'Herbemont
- * Rémi Denis-Courmont
- * Rafaël Carré
- *
- * 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 Street, Fifth Floor, Boston MA 02110-1301, USA.
- *****************************************************************************/
-
-/*****************************************************************************
- * Preamble
- *****************************************************************************/
-#ifdef HAVE_CONFIG_H
-# include "config.h"
-#endif
-
-#define VLC_MODULE_LICENSE VLC_LICENSE_GPL_2_PLUS
-#include <vlc_common.h>
-#include <vlc_plugin.h>
-
-#include <vlc_aout.h>
-#include <vlc_filter.h>
-
-/*****************************************************************************
- * Local structures
- *****************************************************************************/
-struct filter_sys_t
-{
- mtime_t start_date;
-
- /* 3 DTS frames (max 2048) have to be packed into an S/PDIF frame (6144).
- * We accumulate DTS frames from the decoder until we have enough to
- * send. */
- size_t i_frame_size;
- uint8_t *p_buf;
- unsigned i_frames;
-};
-
-/*****************************************************************************
- * Local prototypes
- *****************************************************************************/
-static int Create ( vlc_object_t * );
-static void Close ( vlc_object_t * );
-static block_t *DoWork( filter_t *, block_t * );
-
-/*****************************************************************************
- * Module descriptor
- *****************************************************************************/
-vlc_module_begin ()
- set_category( CAT_AUDIO )
- set_subcategory( SUBCAT_AUDIO_MISC )
- set_description( N_("Audio filter for DTS->S/PDIF encapsulation") )
- set_capability( "audio converter", 10 )
- set_callbacks( Create, Close )
-vlc_module_end ()
-
-/*****************************************************************************
- * Create:
- *****************************************************************************/
-static int Create( vlc_object_t *p_this )
-{
- filter_t * p_filter = (filter_t *)p_this;
- filter_sys_t *p_sys;
-
- if( p_filter->fmt_in.audio.i_format != VLC_CODEC_DTS ||
- ( p_filter->fmt_out.audio.i_format != VLC_CODEC_SPDIFL &&
- p_filter->fmt_out.audio.i_format != VLC_CODEC_SPDIFB ) )
- {
- return VLC_EGENERIC;
- }
-
- /* Allocate the memory needed to store the module's structure */
- p_sys = p_filter->p_sys = malloc( sizeof(*p_sys) );
- if( !p_sys )
- return VLC_ENOMEM;
-
- p_sys->p_buf = NULL;
- p_sys->i_frame_size = 0;
- p_sys->i_frames = 0;
-
- p_filter->pf_audio_filter = DoWork;
-
- return VLC_SUCCESS;
-}
-
-/*****************************************************************************
- * Close: free our resources
- *****************************************************************************/
-static void Close( vlc_object_t * p_this )
-{
- filter_t * p_filter = (filter_t *)p_this;
-
- free( p_filter->p_sys->p_buf );
- free( p_filter->p_sys );
-}
-
-/*****************************************************************************
- * DoWork: convert a buffer
- *****************************************************************************/
-static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
-{
- uint32_t i_ac5_spdif_type = 0;
- uint16_t i_fz = p_in_buf->i_nb_samples * 4;
- uint16_t i_frame, i_length = p_in_buf->i_buffer;
- static const uint8_t p_sync_le[6] = { 0x72, 0xF8, 0x1F, 0x4E, 0x00, 0x00 };
- static const uint8_t p_sync_be[6] = { 0xF8, 0x72, 0x4E, 0x1F, 0x00, 0x00 };
-
- if( p_in_buf->i_buffer != p_filter->p_sys->i_frame_size )
- {
- /* Frame size changed, reset everything */
- msg_Warn( p_filter, "Frame size changed from %zu to %zu, "
- "resetting everything.",
- p_filter->p_sys->i_frame_size, p_in_buf->i_buffer );
-
- p_filter->p_sys->i_frame_size = p_in_buf->i_buffer;
- p_filter->p_sys->p_buf = xrealloc( p_filter->p_sys->p_buf,
- p_in_buf->i_buffer * 3 );
- p_filter->p_sys->i_frames = 0;
- }
-
- /* Backup frame */
- /* TODO: keeping the blocks in a list would save one memcpy */
- memcpy( p_filter->p_sys->p_buf + p_in_buf->i_buffer *
- p_filter->p_sys->i_frames,
- p_in_buf->p_buffer, p_in_buf->i_buffer );
-
- p_filter->p_sys->i_frames++;
-
- if( p_filter->p_sys->i_frames < 3 )
- {
- if( p_filter->p_sys->i_frames == 1 )
- /* We'll need the starting date */
- p_filter->p_sys->start_date = p_in_buf->i_pts;
-
- /* Not enough data */
- block_Release( p_in_buf );
- return NULL;
- }
-
- p_filter->p_sys->i_frames = 0;
- block_t *p_out_buf = block_Alloc( 12 * p_in_buf->i_nb_samples );
- if( !p_out_buf )
- goto out;
-
- for( i_frame = 0; i_frame < 3; i_frame++ )
- {
- uint16_t i_length_padded = i_length;
- uint8_t * p_out = p_out_buf->p_buffer + (i_frame * i_fz);
- uint8_t * p_in = p_filter->p_sys->p_buf + (i_frame * i_length);
-
- switch( p_in_buf->i_nb_samples )
- {
- case 512: i_ac5_spdif_type = 0x0B; break;
- case 1024: i_ac5_spdif_type = 0x0C; break;
- case 2048: i_ac5_spdif_type = 0x0D; break;
- }
-
- /* Copy the S/PDIF headers. */
- if( p_filter->fmt_out.audio.i_format == VLC_CODEC_SPDIFB )
- {
- memcpy( p_out, p_sync_be, 6 );
- p_out[5] = i_ac5_spdif_type;
- SetWBE( p_out + 6, i_length << 3 );
- }
- else
- {
- memcpy( p_out, p_sync_le, 6 );
- p_out[4] = i_ac5_spdif_type;
- SetWLE( p_out + 6, i_length << 3 );
- }
-
- if( ( (p_in[0] == 0x1F || p_in[0] == 0x7F) && p_filter->fmt_out.audio.i_format == VLC_CODEC_SPDIFL ) ||
- ( (p_in[0] == 0xFF || p_in[0] == 0xFE) && p_filter->fmt_out.audio.i_format == VLC_CODEC_SPDIFB ) )
- {
- /* We are dealing with a big endian bitstream and a little endian output
- * or a little endian bitstream and a big endian output.
- * Byteswap the stream */
- swab( p_in, p_out + 8, i_length );
-
- /* If i_length is odd, we have to adjust swapping a bit.. */
- if( i_length & 1 )
- {
- p_out[8+i_length-1] = 0;
- p_out[8+i_length] = p_in[i_length-1];
- i_length_padded++;
- }
- }
- else
- {
- memcpy( p_out + 8, p_in, i_length );
- }
-
- if( i_fz > i_length + 8 )
- {
- memset( p_out + 8 + i_length_padded, 0,
- i_fz - i_length_padded - 8 );
- }
- }
-
- p_out_buf->i_pts = p_filter->p_sys->start_date;
- p_out_buf->i_nb_samples = p_in_buf->i_nb_samples * 3;
- p_out_buf->i_buffer = p_out_buf->i_nb_samples * 4;
-out:
- block_Release( p_in_buf );
- return p_out_buf;
-}
diff --git a/po/POTFILES.in b/po/POTFILES.in
index 6451cc20907e..5d205f5a51fc 100644
--- a/po/POTFILES.in
+++ b/po/POTFILES.in
@@ -287,7 +287,6 @@ modules/audio_filter/compressor.c
modules/audio_filter/converter/a52tofloat32.c
modules/audio_filter/converter/a52tospdif.c
modules/audio_filter/converter/dtstofloat32.c
-modules/audio_filter/converter/dtstospdif.c
modules/audio_filter/converter/format.c
modules/audio_filter/converter/mpgatofixed32.c
modules/audio_filter/equalizer.c
--
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