diff options
Diffstat (limited to 'src/libxineadec')
| -rw-r--r-- | src/libxineadec/28k8.c | 555 | ||||
| -rw-r--r-- | src/libxineadec/Makefile.am | 27 | ||||
| -rw-r--r-- | src/libxineadec/adpcm.c | 1687 | ||||
| -rw-r--r-- | src/libxineadec/interplayaudio.c | 357 | ||||
| -rw-r--r-- | src/libxineadec/logpcm.c | 312 | ||||
| -rw-r--r-- | src/libxineadec/roqaudio.c | 242 |
6 files changed, 1 insertions, 3179 deletions
diff --git a/src/libxineadec/28k8.c b/src/libxineadec/28k8.c deleted file mode 100644 index f8c51d59c..000000000 --- a/src/libxineadec/28k8.c +++ /dev/null @@ -1,555 +0,0 @@ -/* - * Copyright (C) 2003 the xine project - * - * This file is part of xine, a free video player. - * - * xine 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. - * - * xine 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA - * - * RealAudio 28k8 Decoder by Ewald Snel <ewald@rambo.its.tudelft.nl> - * - * Based on public domain source code from: - * http://www.honeypot.net/audio/ - * - * $Id: 28k8.c,v 1.7 2004/01/12 17:35:17 miguelfreitas Exp $ - */ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <sys/types.h> -#include <unistd.h> -#include <math.h> - -#include "xine_internal.h" -#include "audio_out.h" -#include "buffer.h" -#include "xineutils.h" -#include "bswap.h" - -#define SUBBLOCK 38 -#define BLOCKSIZE 2736 - -typedef struct { - audio_decoder_class_t decoder_class; -} ra28k8_class_t; - -typedef struct ra28k8_decoder_s { - audio_decoder_t audio_decoder; - - xine_stream_t *stream; - - int output_open; - int decoder_ok; - - unsigned char buf[BLOCKSIZE]; - int size; - - float sb[41]; - float lhist[10]; - float output[40]; - float pr1[36]; - float st1a[111]; - float st1b[37]; - float st1[37]; - float pr2[10]; - float st2a[38]; - float st2b[11]; - float st2[11]; - -} ra28k8_decoder_t; - - -static const float ra28k8_amptable[8] = { - 0.515625, 0.90234375, 1.57910156, 2.76342773, - -0.515625,-0.90234375,-1.57910156,-2.76342773 -}; - -static const float ra28k8_codetable[128][5] = { - { 0.326171875, -1.4404296875, -0.6123046875, -0.8740234375, -1.24658203125 }, - { -2.45703125, -2.23486328125,-0.51025390625, 1.419921875, 1.6201171875 }, - { -1.37646484375,-1.30712890625,-0.462890625, -1.37939453125,-2.1728515625 }, - { -3.26123046875,-0.166015625, 0.7236328125, -0.623046875, 0.6162109375 }, - { -0.2744140625, -3.29931640625, 0.62548828125, 0.08740234375,-0.6220703125 }, - { -1.2265625, -3.4814453125, -2.40478515625, 3.37548828125, 1.17724609375 }, - { -1.2099609375, -0.076171875, 2.28662109375,-1.89111328125, 0 }, - { -4.0078125, 1.044921875, -0.2333984375, -1.35986328125, 0.26025390625 }, - { 0.92236328125, 1.34716796875, 0.67431640625,-3.39599609375,-2.88720703125 }, - { 2.4814453125, -1.201171875, -2.8212890625, 0.87744140625, 0.27734375 }, - { -1.078125, -1.61572265625,-2.20849609375,-3.044921875, -3.66455078125 }, - { -1.32763671875, 2.1279296875, -1.458984375, -0.56103515625, 1.30078125 }, - { 0.61474609375, 0.48583984375, 1.32373046875,-1.203125, -5.0732421875 }, - { 0.8408203125, -3.69580078125,-1.3388671875, 1.06005859375,-1.13720703125 }, - { 0.50390625, 0.36474609375,-0.4189453125, -3.8798828125, -6.27099609375 }, - { 1.5166015625, 2.37109375, -2.04736328125,-1.24072265625, 0.50537109375 }, - { 0.9091796875, -0.46875, -3.236328125, 0.2001953125, 2.8720703125 }, - { -1.21728515625,-1.283203125, -1.953125, -0.029296875, 3.5166015625 }, - { -1.3046875, 0.7060546875, 0.75, -1.87060546875, 0.60205078125 }, - { -2.5888671875, 3.375, 0.77587890625,-2.04443359375, 1.78955078125 }, - { -1.6875, -3.9892578125, -3.76416015625, 0.67578125, 2.2939453125 }, - { -2.29443359375,-3.03173828125,-5.45703125, 3.95703125, 8.2177734375 }, - { 0.4541015625, 3.419921875, 0.61962890625,-4.38330078125, 1.25341796875 }, - { 2.27001953125, 5.763671875, 1.68017578125,-2.76220703125, 0.58544921875 }, - { 1.2412109375, -0.08935546875,-4.32568359375,-3.89453125, 1.5771484375 }, - { -1.40234375, -0.98193359375,-4.74267578125,-4.09423828125, 6.33935546875 }, - { 1.5068359375, 1.044921875, -1.796875, -4.70849609375,-1.4140625 }, - { -3.71533203125, 3.18115234375,-1.11474609375,-1.2314453125, 3.091796875 }, - { -1.62744140625,-2.744140625, -4.4580078125, -5.43505859375, 2.70654296875 }, - { -0.19873046875,-3.28173828125,-8.5283203125, -1.41064453125, 5.6484375 }, - { 1.802734375, 3.318359375, -0.1279296875, -5.2958984375, -0.90625 }, - { 3.55224609375, 6.544921875, -1.45947265625,-5.17333984375, 2.41015625 }, - { 0.119140625, -1.08349609375, 1.296875, 1.84375, -2.642578125 }, - { -1.97412109375,-2.8974609375, 1.04052734375, 0.42138671875,-1.3994140625 }, - { -1.6123046875, 0.85107421875,-0.9794921875, -0.0625, -1.001953125 }, - { -3.10595703125, 1.6318359375, -0.77294921875,-0.01025390625, 0.5576171875 }, - { -1.87353515625,-0.89404296875, 3.12353515625, 1.24267578125,-1.390625 }, - { -4.556640625, -3.1875, 2.59228515625, 0.9697265625, -1.09619140625 }, - { -2.1923828125, 0.365234375, 0.94482421875,-1.47802734375,-0.24072265625 }, - { -4.51904296875, 2.6201171875, 1.55908203125,-2.19384765625, 0.87109375 }, - { 2.3359375, -0.1806640625, 0.9111328125, 0.51611328125,-0.92236328125 }, - { 3.5849609375, -1.3134765625, -1.25830078125, 0.330078125, -0.29833984375 }, - { -0.2451171875, 1.09130859375,-0.9033203125, -0.86767578125,-1.00048828125 }, - { 0.49365234375, 1.89453125, -1.20361328125, 1.07861328125,-0.07421875 }, - { 1.265625, 1.38134765625, 2.728515625, 1.38623046875,-3.5673828125 }, - { -1.48876953125,-2.4013671875, 2.90771484375, 4.49267578125,-2.17138671875 }, - { 0.34033203125, 1.908203125, 2.8310546875, -2.17333984375,-2.267578125 }, - { -1.03564453125, 2.658203125, -1.2548828125, 0.15673828125,-0.5869140625 }, - { 1.3896484375, -1.0185546875, 1.724609375, 0.2763671875, -0.345703125 }, - { -2.08935546875, 0.4638671875, 2.431640625, 1.83056640625, 0.220703125 }, - { -1.212890625, 1.7099609375, 0.83935546875,-0.0830078125, 0.1162109375 }, - { -1.67724609375, 0.12841796875, 1.0322265625, -0.97900390625, 1.15283203125 }, - { -3.5830078125, -0.58984375, 4.56396484375,-0.59375, -1.95947265625 }, - { -6.5908203125, -0.21435546875, 3.919921875, -2.06640625, 0.17626953125 }, - { -1.82080078125, 2.65283203125, 0.978515625, -2.30810546875,-0.61474609375 }, - { -1.9462890625, 3.78076171875, 4.11572265625,-1.80224609375,-0.48193359375 }, - { 2.5380859375, -0.20654296875, 0.5615234375, -0.62548828125, 0.3984375 }, - { 3.61767578125, 2.00634765625,-1.92822265625, 1.3134765625, 0.0146484384313 }, - { 0.6083984375, 1.49169921875,-0.01708984375,-0.6689453125, -0.1201171875 }, - { -0.72705078125, 2.75146484375,-0.3310546875, -1.28271484375, 1.5478515625 }, - { 2.3583984375, -2.23876953125, 0.98046875, -0.5185546875, 0.39013671875 }, - { -0.06298828125, 0.35009765625, 2.2431640625, 7.29345703125, 5.2275390625 }, - { 0.20361328125, 1.34716796875, 0.9033203125, -2.46923828125,-0.56298828125 }, - { -1.89794921875, 3.59423828125,-2.81640625, 2.09228515625, 0.3251953125 }, - { 0.70458984375,-0.4580078125, 0.009765625, -1.03466796875,-0.82861328125 }, - { -1.8125, -1.6611328125, -1.080078125, 0.0537109375, 1.04296875 }, - { -1.44140625, 0.005859375, -0.765625, -1.708984375, -0.90576171875 }, - { -0.64208984375,-0.84521484375, 0.56640625, -0.2724609375, 0.83447265625 }, - { 0.04296875, -2.23095703125, 0.0947265625, -0.2216796875, -1.44384765625 }, - { -1.38623046875,-0.8134765625, -0.13330078125, 1.017578125, -0.07568359375 }, - { -0.09228515625,-1.16015625, 0.81201171875,-0.5078125, -1.19580078125 }, - { -1.3876953125, -0.66845703125, 0.310546875, -0.12109375, -1.30712890625 }, - { 0.74072265625, 0.03857421875,-1.47119140625,-1.79150390625,-0.47509765625 }, - { 0.93408203125,-1.21728515625,-2.59375, -0.36572265625, 0.62060546875 }, - { -1.41748046875,-1.623046875, -1.833984375, -1.8017578125, -0.89306640625 }, - { -1.42236328125,-0.75537109375,-1.34765625, -0.6865234375, 0.548828125 }, - { 0.900390625, -0.8955078125, 0.22265625, 0.3447265625, -2.0859375 }, - { 0.22802734375,-2.078125, -0.93212890625, 0.74267578125, 0.5537109375 }, - { -0.06201171875,-0.4853515625, -0.31103515625,-0.72802734375,-3.1708984375 }, - { 0.42626953125,-0.99853515625,-1.869140625, -1.36328125, -0.2822265625 }, - { 1.12841796875,-0.88720703125, 1.28515625, -1.490234375, 0.9609375 }, - { 0.31298828125, 0.5830078125, 0.92431640625, 2.00537109375, 3.0966796875 }, - { -0.02197265625, 0.5849609375, 1.0546875, -0.70751953125, 1.07568359375 }, - { -0.978515625, 0.83642578125, 1.7177734375, 1.294921875, 2.07568359375 }, - { 1.43359375, -1.9375, 0.625, 0.06396484375,-0.720703125 }, - { 1.38037109375, 0.00390625, -0.94140625, 1.2978515625, 1.71533203125 }, - { 1.56201171875,-0.3984375, 1.31201171875,-0.85009765625,-0.68701171875 }, - { 1.439453125, 1.96728515625, 0.1923828125, -0.12353515625, 0.6337890625 }, - { 2.0927734375, 0.02490234375,-2.20068359375,-0.015625, -0.32177734375 }, - { 1.90576171875, 2.7568359375, -2.728515625, -1.265625, 2.78662109375 }, - { -0.2958984375, 0.6025390625, -0.78466796875,-2.53271484375, 0.32421875 }, - { -0.25634765625, 1.767578125, -1.0703125, -1.23388671875, 0.83349609375 }, - { 2.09814453125,-1.58740234375,-1.11474609375, 0.396484375, -1.10546875 }, - { 2.81494140625, 0.2578125, -1.60498046875, 0.66015625, 0.81640625 }, - { 1.33544921875, 0.60595703125,-0.53857421875,-1.59814453125,-1.66357421875 }, - { 1.96923828125, 0.8046875, -1.44775390625,-0.5732421875, 0.705078125 }, - { 0.0361328125, 0.4482421875, 0.97607421875, 0.44677734375,-0.5009765625 }, - { -1.21875, -0.78369140625, 0.9931640625, 1.4404296875, 0.11181640625 }, - { -1.05859375, 0.99462890625, 0.00732421921566,-0.6171875, -0.1015625 }, - { -1.734375, 0.7470703125, 0.28369140625, 0.72802734375, 0.4697265625 }, - { -1.27587890625,-1.1416015625, 1.76806640625,-0.7265625, -1.06689453125 }, - { -0.85302734375, 0.03955078125, 2.7041015625, 0.69921875, -1.10205078125 }, - { -0.49755859375, 0.42333984375, 0.1044921875, -1.115234375, -0.7373046875 }, - { -0.822265625, 1.375, -0.11181640625, 1.24560546875,-0.67822265625 }, - { 1.32177734375, 0.24609375, 0.23388671875, 1.35888671875,-0.49267578125 }, - { 1.22900390625,-0.72607421875,-0.779296875, 0.30322265625, 0.94189453125 }, - { -0.072265625, 1.0771484375, -2.09375, 0.630859375, -0.68408203125 }, - { -0.25732421875, 0.60693359375,-1.33349609375, 0.93212890625, 0.625 }, - { 1.04931640625,-0.73291015625, 1.80078125, 0.2978515625, -2.24169921875 }, - { 1.6142578125, -1.64501953125, 0.91552734375, 1.775390625, -0.59423828125 }, - { 1.2568359375, 1.22705078125, 0.70751953125,-1.5009765625, -2.43115234375 }, - { 0.3974609375, 0.8916015625, -1.21923828125, 2.0673828125, -1.99072265625 }, - { 0.8125, -0.107421875, 1.6689453125, 0.4892578125, 0.54443359375 }, - { 0.38134765625, 0.8095703125, 1.91357421875, 2.9931640625, 1.533203125 }, - { 0.560546875, 1.98486328125, 0.740234375, 0.39794921875, 0.09716796875 }, - { 0.58154296875, 1.21533203125, 1.25048828125, 1.18212890625, 1.19287109375 }, - { 0.3759765625, -2.88818359375, 2.69287109375,-0.1796875, -1.56201171875 }, - { 0.5810546875, 0.51123046875, 1.8271484375, 3.38232421875,-1.02001953125 }, - { 0.142578125, 1.51318359375, 2.103515625, -0.3701171875, -1.19873046875 }, - { 0.25537109375, 1.91455078125, 1.974609375, 0.6767578125, 0.04150390625 }, - { 2.13232421875, 0.4912109375, -0.611328125, -0.7158203125, -0.67529296875 }, - { 1.880859375, 0.77099609375,-0.03759765625, 1.0078125, 0.423828125 }, - { 2.49462890625, 1.42529296875,-0.0986328125, 0.17529296875,-0.24853515625 }, - { 1.7822265625, 1.5654296875, 1.12451171875, 0.82666015625, 0.6328125 }, - { 1.41845703125,-1.90771484375, 0.11181640625,-0.583984375, -1.138671875 }, - { 2.91845703125,-1.75048828125, 0.39306640625, 1.86767578125,-1.5322265625 }, - { 1.8291015625, -0.2958984375, 0.02587890625,-0.13134765625,-1.61181640625 }, - { 0.2958984375, 0.9853515625, -0.642578125, 1.984375, 0.1943359375 } -}; - -static const float ra28k8_table1[111] = { - 0.576690972, 0.580838025, 0.585013986, 0.589219987, 0.59345597, 0.597723007, - 0.602020264, 0.606384277, 0.610748291, 0.615142822, 0.619598389, 0.624084473, - 0.628570557, 0.633117676, 0.637695313, 0.642272949, 0.646911621, 0.651580811, - 0.656280518, 0.66104126, 0.665802002, 0.670593262, 0.675445557, 0.680328369, - 0.685241699, 0.690185547, 0.695159912, 0.700164795, 0.705230713, 0.710327148, - 0.715454102, 0.720611572, 0.725830078, 0.731048584, 0.736328125, 0.741638184, - 0.747009277, 0.752380371, 0.7578125, 0.763305664, 0.768798828, 0.774353027, - 0.779937744, 0.785583496, 0.791229248, 0.796936035, 0.802703857, 0.808502197, - 0.814331055, 0.820220947, 0.826141357, 0.832092285, 0.838104248, 0.844146729, - 0.850250244, 0.856384277, 0.862548828, 0.868774414, 0.875061035, 0.881378174, - 0.88772583, 0.894134521, 0.900604248, 0.907104492, 0.913635254, 0.920227051, - 0.926879883, 0.933563232, 0.940307617, 0.94708252, 0.953918457, 0.96081543, - 0.96774292, 0.974731445, 0.981781006, 0.988861084, 0.994842529, 0.998565674, - 0.999969482, 0.99911499, 0.996002197, 0.990600586, 0.982910156, 0.973022461, - 0.960876465, 0.946533203, 0.930053711, 0.911437988, 0.89074707, 0.868041992, - 0.843322754, 0.816680908, 0.788208008, 0.757904053, 0.725891113, 0.692199707, - 0.656921387, 0.620178223, 0.582000732, 0.542480469, 0.501739502, 0.459838867, - 0.416900635, 0.373016357, 0.328277588, 0.282775879, 0.236663818, 0.189971924, - 0.142852783, 0.0954284668,0.0477600098 -}; - -static const float ra28k8_table2[38] = { - 0.505699992, 0.524200022, 0.54339999, 0.563300014, 0.583953857, 0.60534668, - 0.627502441, 0.650482178, 0.674316406, 0.699005127, 0.724578857, 0.75112915, - 0.778625488, 0.807128906, 0.836669922, 0.86730957, 0.899078369, 0.932006836, - 0.961486816, 0.982757568, 0.995635986, 1, 0.995819092, 0.983154297, - 0.96206665, 0.932769775, 0.895507813, 0.850585938, 0.798400879, 0.739379883, - 0.674072266, 0.602996826, 0.526763916, 0.446014404, 0.361480713, 0.273834229, - 0.183868408, 0.0923461914 -}; - -static const float ra28k8_table1a[36] = { - 0.654184103, 0.66194123, 0.669790328, 0.677732527, 0.685768902, 0.693900526, - 0.702128589, 0.710454226, 0.718878567, 0.727402806, 0.736028135, 0.744755745, - 0.753586829, 0.762522638, 0.771564424, 0.780713439, 0.789970934, 0.799338162, - 0.808816493, 0.818407178, 0.828111589, 0.837931097, 0.847867012, 0.857920766, - 0.868093729, 0.878387332, 0.888803005, 0.899342179, 0.910006344, 0.920796931, - 0.931715488, 0.942763507, 0.953942537, 0.965254128, 0.976699829, 0.98828125 -}; - -static const float ra28k8_table2a[10] = { - 0.373657227, 0.41229248, 0.454956055, 0.50201416, 0.553955078, - 0.61126709, 0.674499512, 0.74432373, 0.821289063, 0.90625 -}; - - -static inline float sum (float *p1, float *p2, int len) { - float s = 0; - - while (len--) { - s += (*(p1++)) * (*(p2++)); - } - - return s; -} - -static inline void colmul (float *out, float *in, const float *table, int len) { - - while (len--) { - *(out++) = (*(in++)) * (*(table++)); - } -} - -static inline float clip (float a, float amin, float amax) { - if (a < amin) - return amin; - else if (a > amax) - return amax; - else - return a; -} - -static inline int predict (float *in, float *tgt, int n) { - int x, y; - float f0, f1, f2, temp; - - if (in[n] == 0 || in[0] <= 0) { - return 0; - } - - f0 = in[0]; - - for (x=n; (x > 0) && (f0 > 0); x--) { - f1 = in[(n - x) + 1]; - - for (y=1; y <= (n - x); y++) { - f1 += in[y] * tgt[x + y]; - } - - f2 = -f1 / f0; - f0 += f1 * f2; - - tgt[x] = f2; - - for (y=0; y < ((n - x) + 1)>>1; y++) { - temp = tgt[n - y] + tgt[x + y + 1] * f2; - tgt[x + y + 1] += tgt[n - y] * f2; - tgt[n - y] = temp; - } - } - - return (f0 > 0); -} - -static void decode_28k8_block (ra28k8_decoder_t *this, uint8_t *in, int16_t *out) { - int i, j, x, z, phase, phasep, code; - float buffer[5], work[111], temp1[37], temp2[11], d; - - for (x=0, z=0; x < 2304; x++) { - phasep = 5*(phase = (x & 7)); - - /* unpack */ - code = ((in[0] >> z) | (in[1] << (8 - z))) & (1023 >> (~x & 1)); - - if (z == 7 && (x & 1)) { - code |= (in[2] & 1) << 9; - } - - z += (9 + (x & 1)); - in += (z >> 3); - z &= 7; - - /* decode */ - memmove (this->sb, &this->sb[5], 36*sizeof(float)); - - for (i=0; i < 5; i++) { - this->sb[36 + i] = -sum (&this->sb[i], this->pr1, 36); - } - - /* convert log and do rms */ - d = clip (32 - sum (this->pr2, this->lhist, 10), 0, 60); - d = pow(10, d / 20) * ra28k8_amptable[code & 7]; - - for (i=0; i < 5; i++) { - buffer[i] = ra28k8_codetable[code >> 3][i] * d; - } - - d = sum (buffer, buffer, 5) / 5; - d = (d < 1) ? 0 : 10 * log10(d); - - /* shift and store */ - memmove (this->lhist, &this->lhist[1], 9*sizeof(float)); - this->lhist[9] = d - 32; - - for (i=1; i < 5; i++) { - for(j=0; j < i; j++) { - buffer[i] -= this->pr1[35 - j] * buffer[i - j - 1]; - } - } - - /* output */ - for (i=0; i < 5; i++) { - float f = clip (this->sb[36 + i] + buffer[i], -4095, 4095); - this->output[phasep + i] = this->sb[36 + i] = f; - *(out++) = (8 * f); - } - - if (phase == 3) { - - /* rotate and multiply */ - memmove (this->st1a, &this->st1a[40], 71*sizeof(float)); - memcpy (&this->st1a[71], &this->output[20], 20*sizeof(float)); - memcpy (&this->st1a[91], this->output, 20*sizeof(float)); - memmove (this->st2a, &this->st2a[8], 30*sizeof(float)); - memcpy (&this->st2a[30], &this->lhist[2], 4*sizeof(float)); - memcpy (&this->st2a[34], &this->lhist[6], 4*sizeof(float)); - - colmul (work, this->st1a, ra28k8_table1, 111); - - for (i=36; i >= 0; i--) { - this->st1b[i] = this->st1b[i] * (0.5625) + sum (&work[36 - i], &work[36], 40); - temp1[i] = this->st1b[i] + sum (&work[76 - i], &work[76], 35); - } - - colmul (work, this->st2a, ra28k8_table2, 38); - - for (i=10; i >= 0; i--) { - this->st2b[i] = this->st2b[i] * (0.5625) + sum (&work[10 - i], &work[10], 8); - temp2[i] = this->st2b[i] + sum (&work[18 - i], &work[18], 20); - } - - /* to prevent clipping */ - temp1[0] *= 1.00390625; - temp2[0] *= 1.00390625; - - if (predict (temp1, this->st1, 36)) { - colmul (this->pr1, &this->st1[1], ra28k8_table1a, 36); - } - if (predict (temp2, this->st2, 10)) { - colmul (this->pr2, &this->st2[1], ra28k8_table2a, 10); - } - } - } -} - -static void ra28k8_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { - ra28k8_decoder_t *this = (ra28k8_decoder_t *) this_gen; - - if (buf->decoder_flags & BUF_FLAG_PREVIEW) { - - } else if (buf->decoder_flags & BUF_FLAG_HEADER) { - - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, "Real 28.8"); - this->decoder_ok = 1; - if (!this->decoder_ok) - _x_stream_info_set(this->stream, XINE_STREAM_INFO_AUDIO_HANDLED, 0); - - } else if (this->decoder_ok ) { - audio_buffer_t *audio_buffer; - int i, j; - - if (!this->output_open) { - this->output_open = this->stream->audio_out->open (this->stream->audio_out, - this->stream, 16, 8000, AO_CAP_MODE_MONO); - } - - /* copy and deinterleave sub-blocks */ - i = 0; - j = this->size; - - while (i < buf->size || j == BLOCKSIZE) { - if (j == BLOCKSIZE) { - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - - decode_28k8_block (this, this->buf, (int16_t *) audio_buffer->mem); - - audio_buffer->vpts = buf->pts; - audio_buffer->num_frames = 11520; - - this->stream->audio_out->put_buffer (this->stream->audio_out, - audio_buffer, this->stream); - - j = 0; - } else if ((buf->size - i) >= SUBBLOCK) { - memcpy (&this->buf[j], &buf->content[i], SUBBLOCK); - - if ((j += 12*SUBBLOCK) >= BLOCKSIZE) { - if (j == (BLOCKSIZE + 11*SUBBLOCK)) - j = BLOCKSIZE; - else - j -= (BLOCKSIZE - SUBBLOCK); - } - - i += SUBBLOCK; - } else { - memcpy (&this->buf[j], &buf->content[i], (buf->size - i)); - - j += (buf->size - i); - break; - } - } - - this->size = j; - } -} - -static void ra28k8_reset (audio_decoder_t *this_gen) { - ra28k8_decoder_t *this = (ra28k8_decoder_t *) this_gen; - - this->size = 0; - - memset (this->sb, 0, sizeof(this->sb)); - memset (this->lhist, 0, sizeof(this->lhist)); - memset (this->output, 0, sizeof(this->lhist)); - memset (this->pr1, 0, sizeof(this->pr1)); - memset (this->st1a, 0, sizeof(this->st1a)); - memset (this->st1b, 0, sizeof(this->st1b)); - memset (this->st1, 0, sizeof(this->st1)); - memset (this->pr2, 0, sizeof(this->pr2)); - memset (this->st2a, 0, sizeof(this->st2a)); - memset (this->st2b, 0, sizeof(this->st2b)); - memset (this->st2, 0, sizeof(this->st2)); -} - -static void ra28k8_discontinuity (audio_decoder_t *this_gen) { -} - -static void ra28k8_dispose (audio_decoder_t *this_gen) { - - ra28k8_decoder_t *this = (ra28k8_decoder_t *) this_gen; - - if (this->output_open) - this->stream->audio_out->close (this->stream->audio_out, this->stream); - this->output_open = 0; - - free (this_gen); -} - -static audio_decoder_t *open_plugin (audio_decoder_class_t *class_gen, xine_stream_t *stream) { - - ra28k8_decoder_t *this ; - - this = (ra28k8_decoder_t *) xine_xmalloc (sizeof (ra28k8_decoder_t)); - - this->audio_decoder.decode_data = ra28k8_decode_data; - this->audio_decoder.reset = ra28k8_reset; - this->audio_decoder.discontinuity = ra28k8_discontinuity; - this->audio_decoder.dispose = ra28k8_dispose; - - this->size = 0; - this->stream = stream; - - return &this->audio_decoder; -} - -static char *get_identifier (audio_decoder_class_t *this) { - return "28k8"; -} - -static char *get_description (audio_decoder_class_t *this) { - return "RealAudio 28k8 decoder plugin"; -} - -static void dispose_class (audio_decoder_class_t *this) { - free (this); -} - -static void *init_plugin (xine_t *xine, void *data) { - - ra28k8_class_t *this ; - - this = (ra28k8_class_t *) xine_xmalloc (sizeof (ra28k8_class_t)); - - this->decoder_class.open_plugin = open_plugin; - this->decoder_class.get_identifier = get_identifier; - this->decoder_class.get_description = get_description; - this->decoder_class.dispose = dispose_class; - - return this; -} - -static uint32_t audio_types[] = { - BUF_AUDIO_28_8, 0 -}; - -static decoder_info_t dec_info_audio = { - audio_types, /* supported types */ - 9 /* priority */ -}; - -plugin_info_t xine_plugin_info[] = { - /* type, API, "name", version, special_info, init_function */ - { PLUGIN_AUDIO_DECODER, 15, "28k8", XINE_VERSION_CODE, &dec_info_audio, init_plugin }, - { PLUGIN_NONE, 0, "", 0, NULL, NULL } -}; diff --git a/src/libxineadec/Makefile.am b/src/libxineadec/Makefile.am index f25dfe29d..4bad809de 100644 --- a/src/libxineadec/Makefile.am +++ b/src/libxineadec/Makefile.am @@ -9,25 +9,8 @@ AM_CFLAGS = -DNSF_PLAYER SUBDIRS = gsm610 nosefart lib_LTLIBRARIES = \ - xineplug_decode_adpcm.la \ - xineplug_decode_logpcm.la \ - xineplug_decode_roqaudio.la \ xineplug_decode_gsm610.la \ - xineplug_decode_interplayaudio.la \ - xineplug_decode_nsf.la \ - xineplug_decode_28k8.la - -xineplug_decode_roqaudio_la_SOURCES = roqaudio.c -xineplug_decode_roqaudio_la_LIBADD = $(XINE_LIB) -xineplug_decode_roqaudio_la_LDFLAGS = -avoid-version -module @XINE_PLUGIN_MIN_SYMS@ - -xineplug_decode_adpcm_la_SOURCES = adpcm.c -xineplug_decode_adpcm_la_LIBADD = $(XINE_LIB) -xineplug_decode_adpcm_la_LDFLAGS = -avoid-version -module @XINE_PLUGIN_MIN_SYMS@ - -xineplug_decode_logpcm_la_SOURCES = logpcm.c -xineplug_decode_logpcm_la_LIBADD = $(XINE_LIB) -xineplug_decode_logpcm_la_LDFLAGS = -avoid-version -module @XINE_PLUGIN_MIN_SYMS@ + xineplug_decode_nsf.la xineplug_decode_gsm610_la_SOURCES = gsm610.c xineplug_decode_gsm610_la_LDFLAGS = -avoid-version -module @XINE_PLUGIN_MIN_SYMS@ @@ -35,10 +18,6 @@ xineplug_decode_gsm610_la_LIBADD = \ $(XINE_LIB) \ $(top_builddir)/src/libxineadec/gsm610/libgsm610.la -xineplug_decode_interplayaudio_la_SOURCES = interplayaudio.c -xineplug_decode_interplayaudio_la_LIBADD = $(top_builddir)/src/xine-engine/libxine.la -xineplug_decode_interplayaudio_la_LDFLAGS = -avoid-version -module @XINE_PLUGIN_MIN_SYMS@ - xineplug_decode_nsf_la_SOURCES = nsf.c #xineplug_decode_nsf_la_CFLAGS = -DNSF_PLAYER xineplug_decode_nsf_la_LDFLAGS = -avoid-version -module @XINE_PLUGIN_MIN_SYMS@ @@ -46,8 +25,4 @@ xineplug_decode_nsf_la_LIBADD = \ $(XINE_LIB) \ $(top_builddir)/src/libxineadec/nosefart/libnosefart.la -xineplug_decode_28k8_la_SOURCES = 28k8.c -xineplug_decode_28k8_la_LIBADD = $(top_builddir)/src/xine-engine/libxine.la -xineplug_decode_28k8_la_LDFLAGS = -avoid-version -module @XINE_PLUGIN_MIN_SYMS@ - # noinst_HEADERS = diff --git a/src/libxineadec/adpcm.c b/src/libxineadec/adpcm.c deleted file mode 100644 index 85d7804f1..000000000 --- a/src/libxineadec/adpcm.c +++ /dev/null @@ -1,1687 +0,0 @@ -/* - * Copyright (C) 2000-2001 the xine project - * - * This file is part of xine, a free video player. - * - * xine is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * xine is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA - * - * ADPCM Decoders by Mike Melanson (melanson@pcisys.net) - * - * This file is in charge of decoding all of the various ADPCM data - * formats that various entities have created. Details about the data - * formats can be found here: - * http://www.pcisys.net/~melanson/codecs/ - * CD-ROM/XA ADPCM decoder by Stuart Caie (kyzer@4u.net) - * - based on information in the USENET post by Jac Goudsmit (jac@codim.nl) - * <01bbc34c$dbf64020$f9c8a8c0@cray.codim.nl> - * - tested for correctness using Jon Atkins's CDXA software: - * http://jonatkins.org/cdxa/ - * this is also useful for extracting streams from Playstation discs - * - * - * $Id: adpcm.c,v 1.38 2004/01/12 17:35:18 miguelfreitas Exp $ - */ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <sys/types.h> -#include <unistd.h> - -#define LOG_MODULE "adpcm_audio_decoder" -#define LOG_VERBOSE -/* -#define LOG -*/ - -#include "xine_internal.h" -#include "video_out.h" -#include "audio_out.h" -#include "buffer.h" -#include "xineutils.h" -#include "bswap.h" - -/* pertinent tables */ -static int ima_adpcm_step[89] = { - 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, - 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, - 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, - 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, - 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, - 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, - 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, - 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, - 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 -}; - -static int dialogic_ima_step[49] = { - 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, - 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, - 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, - 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552 -}; - -static int ima_adpcm_index[16] = { - -1, -1, -1, -1, 2, 4, 6, 8, - -1, -1, -1, -1, 2, 4, 6, 8 -}; - -static int ms_adapt_table[] = { - 230, 230, 230, 230, 307, 409, 512, 614, - 768, 614, 512, 409, 307, 230, 230, 230 -}; - -static int ms_adapt_coeff1[] = { - 256, 512, 0, 192, 240, 460, 392 -}; - -static int ms_adapt_coeff2[] = { - 0, -256, 0, 64, 0, -208, -232 -}; - -static int ea_adpcm_table[] = { - 0, 240, 460, 392, 0, 0, -208, -220, 0, 1, - 3, 4, 7, 8, 10, 11, 0, -1, -3, -4 -}; - -static int xa_adpcm_table[] = { - 0, 240, 460, 392, 0, 0, -208, -220 -}; - -#define QT_IMA_ADPCM_PREAMBLE_SIZE 2 -#define QT_IMA_ADPCM_BLOCK_SIZE 0x22 -#define QT_IMA_ADPCM_SAMPLES_PER_BLOCK \ - ((QT_IMA_ADPCM_BLOCK_SIZE - QT_IMA_ADPCM_PREAMBLE_SIZE) * 2) - -#define MS_ADPCM_PREAMBLE_SIZE 7 -#define MS_IMA_ADPCM_PREAMBLE_SIZE 4 -#define DK4_ADPCM_PREAMBLE_SIZE 4 -#define DK3_ADPCM_PREAMBLE_SIZE 16 - -/* useful macros */ -/* clamp a number between 0 and 88 */ -#define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88; -/* clamp a number within a signed 16-bit range */ -#define CLAMP_S16(x) if (x < -32768) x = -32768; \ - else if (x > 32767) x = 32767; -/* clamp a number above 16 */ -#define CLAMP_ABOVE_16(x) if (x < 16) x = 16; -/* sign extend a 16-bit value */ -#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; -/* sign extend a 4-bit value */ -#define SE_4BIT(x) if (x & 0x8) x -= 0x10; - -#define AUDIOBUFSIZE 128*1024 - -typedef struct { - audio_decoder_class_t decoder_class; -} adpcm_class_t; - -typedef struct adpcm_decoder_s { - audio_decoder_t audio_decoder; - - xine_stream_t *stream; - - uint32_t rate; - uint32_t bits_per_sample; - uint32_t channels; - uint32_t ao_cap_mode; - - unsigned int buf_type; - int output_open; - - unsigned char *buf; - int bufsize; - int size; - - /* these fields are used for decoding ADPCM data transported in MS file */ - unsigned short *decode_buffer; - unsigned int in_block_size; - unsigned int out_block_size; /* size in samples (2 bytes/sample) */ - - int xa_mode; /* 1 for mode A, 0 for mode B or mode C */ - int xa_p_l; /* previous sample, left/mono channel */ - int xa_p_r; /* previous sample, right channel */ - int xa_pp_l; /* 2nd-previous sample, left/mono channel */ - int xa_pp_r; /* 2nd-previous sample, right channel */ - -} adpcm_decoder_t; - -/* - * decode_ima_nibbles - * - * So many different audio encoding formats leverage off of the IMA - * ADPCM algorithm that it makes sense to create a function that takes - * care of handling the common decoding portion. - * - * This function takes a buffer of ADPCM nibbles that are stored in an - * array of signed 16-bit numbers. The function then decodes the nibbles - * in place so that the buffer contains the decoded audio when the function - * is finished. - * - * The addresses of the initial predictor and index values are passed, - * rather than their values, so that the function can return the final - * predictor and index values after decoding. This is done in case the - * calling function cares (in the case of IMA ADPCM from Westwood Studios' - * VQA files, the values are initialized to 0 at the beginning of the file - * and maintained throughout all of the IMA blocks). - */ -static void decode_ima_nibbles(unsigned short *output, - int output_size, int channels, - int *predictor_l, int *index_l, - int *predictor_r, int *index_r) { - - int step[2]; - int predictor[2]; - int index[2]; - int diff; - int i; - int sign; - int delta; - int channel_number = 0; - - /* take care of the left */ - step[0] = ima_adpcm_step[*index_l]; - predictor[0] = *predictor_l; - index[0] = *index_l; - - /* only handle the right if non-NULL pointers */ - if (index_r) { - step[1] = ima_adpcm_step[*index_r]; - predictor[1] = *predictor_r; - index[1] = *index_r; - } - - for (i = 0; i < output_size; i++) { - delta = output[i]; - - index[channel_number] += ima_adpcm_index[delta]; - CLAMP_0_TO_88(index[channel_number]); - - sign = delta & 8; - delta = delta & 7; - - diff = step[channel_number] >> 3; - if (delta & 4) diff += step[channel_number]; - if (delta & 2) diff += step[channel_number] >> 1; - if (delta & 1) diff += step[channel_number] >> 2; - - if (sign) - predictor[channel_number] -= diff; - else - predictor[channel_number] += diff; - - CLAMP_S16(predictor[channel_number]); - output[i] = predictor[channel_number]; - step[channel_number] = ima_adpcm_step[index[channel_number]]; - - /* toggle channel */ - channel_number ^= channels - 1; - } - - /* save the index and predictor values in case the calling function cares */ - *predictor_l = predictor[0]; - *index_l = index[0]; - - /* only save the right channel information if pointers are non-NULL */ - if (predictor_r) { - *predictor_r = predictor[1]; - *index_r = index[1]; - } -} - -#define DK3_GET_NEXT_NIBBLE() \ - if (decode_top_nibble_next) \ - { \ - nibble = (last_byte >> 4) & 0x0F; \ - decode_top_nibble_next = 0; \ - } \ - else \ - { \ - last_byte = this->buf[i + j++]; \ - if (j > this->in_block_size) break; \ - nibble = last_byte & 0x0F; \ - decode_top_nibble_next = 1; \ - } - -static void dk3_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - - int i, j; - audio_buffer_t *audio_buffer; - int bytes_to_send; - - int sum_pred; - int diff_pred; - int sum_index; - int diff_index; - int diff_channel; - int out_ptr; - - unsigned char last_byte = 0; - unsigned char nibble; - int decode_top_nibble_next = 0; - - /* ADPCM work variables */ - int sign; - int delta; - int step; - int diff; - - /* make sure the input size checks out */ - if ((this->size % this->in_block_size) != 0) { - lprintf ("received DK3 ADPCM block that does not line up\n"); - this->size = 0; - return; - } - - /* iterate through each block in the in buffer */ - for (i = 0; i < this->size; i += this->in_block_size) { - - sum_pred = LE_16(&this->buf[i + 10]); - diff_pred = LE_16(&this->buf[i + 12]); - SE_16BIT(sum_pred); - SE_16BIT(diff_pred); - diff_channel = diff_pred; - sum_index = this->buf[i + 14]; - diff_index = this->buf[i + 15]; - - j = DK3_ADPCM_PREAMBLE_SIZE; /* start past the preamble */ - out_ptr = 0; - last_byte = 0; - decode_top_nibble_next = 0; - while (j < this->in_block_size) { - - /* process the first predictor of the sum channel */ - DK3_GET_NEXT_NIBBLE(); - - step = ima_adpcm_step[sum_index]; - - sign = nibble & 8; - delta = nibble & 7; - - diff = step >> 3; - if (delta & 4) diff += step; - if (delta & 2) diff += step >> 1; - if (delta & 1) diff += step >> 2; - - if (sign) - sum_pred -= diff; - else - sum_pred += diff; - - CLAMP_S16(sum_pred); - - sum_index += ima_adpcm_index[nibble]; - CLAMP_0_TO_88(sum_index); - - /* process the diff channel predictor */ - DK3_GET_NEXT_NIBBLE(); - - step = ima_adpcm_step[diff_index]; - - sign = nibble & 8; - delta = nibble & 7; - - diff = step >> 3; - if (delta & 4) diff += step; - if (delta & 2) diff += step >> 1; - if (delta & 1) diff += step >> 2; - - if (sign) - diff_pred -= diff; - else - diff_pred += diff; - - CLAMP_S16(diff_pred); - - diff_index += ima_adpcm_index[nibble]; - CLAMP_0_TO_88(diff_index); - - /* output the first pair of stereo PCM samples */ - diff_channel = (diff_channel + diff_pred) / 2; - this->decode_buffer[out_ptr++] = sum_pred + diff_channel; - this->decode_buffer[out_ptr++] = sum_pred - diff_channel; - - /* process the second predictor of the sum channel */ - DK3_GET_NEXT_NIBBLE(); - - step = ima_adpcm_step[sum_index]; - - sign = nibble & 8; - delta = nibble & 7; - - diff = step >> 3; - if (delta & 4) diff += step; - if (delta & 2) diff += step >> 1; - if (delta & 1) diff += step >> 2; - - if (sign) - sum_pred -= diff; - else - sum_pred += diff; - - CLAMP_S16(sum_pred); - - sum_index += ima_adpcm_index[nibble]; - CLAMP_0_TO_88(sum_index); - - /* output the second pair of stereo PCM samples */ - this->decode_buffer[out_ptr++] = sum_pred + diff_channel; - this->decode_buffer[out_ptr++] = sum_pred - diff_channel; - } - - /* dispatch the decoded audio */ - j = 0; - while (j < out_ptr) { - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* out_ptr and j are sample counts, mem_size is a byte count */ - if (((out_ptr - j) * 2) > audio_buffer->mem_size) - bytes_to_send = audio_buffer->mem_size; - else - bytes_to_send = (out_ptr - j) * 2; - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[j], - bytes_to_send); - /* byte count / 2 (bytes / sample) / channels */ - audio_buffer->num_frames = bytes_to_send / 2 / this->channels; - - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - j += bytes_to_send / 2; /* 2 bytes per sample */ - } - } - - /* reset buffer */ - this->size = 0; -} - -static void dk4_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - - int predictor_l = 0; - int predictor_r = 0; - int index_l = 0; - int index_r = 0; - - int i, j; - unsigned int out_ptr = 0; - audio_buffer_t *audio_buffer; - int bytes_to_send; - - /* make sure the input size checks out */ - if ((this->size % this->in_block_size) != 0) { - lprintf ("received DK4 ADPCM block that does not line up\n"); - this->size = 0; - return; - } - - /* iterate through each block in the in buffer */ - for (i = 0; i < this->size; i += this->in_block_size) { - - out_ptr = 0; - - /* the first predictor value goes straight to the output */ - predictor_l = this->decode_buffer[0] = LE_16(&this->buf[i + 0]); - SE_16BIT(predictor_l); - index_l = this->buf[i + 2]; - if (this->channels == 2) { - predictor_r = this->decode_buffer[1] = LE_16(&this->buf[i + 4]); - SE_16BIT(predictor_r); - index_r = this->buf[i + 6]; - } - - /* break apart the ADPCM nibbles */ - out_ptr = this->channels; - for (j = DK4_ADPCM_PREAMBLE_SIZE * this->channels; - j < this->in_block_size; j++) { - this->decode_buffer[out_ptr++] = this->buf[i + j] >> 4; - this->decode_buffer[out_ptr++] = this->buf[i + j] & 0x0F; - } - - /* process the nibbles */ - decode_ima_nibbles(&this->decode_buffer[this->channels], - out_ptr - this->channels, - this->channels, - &predictor_l, &index_l, - &predictor_r, &index_r); - - /* dispatch the decoded audio */ - j = 0; - while (j < out_ptr) { - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* out_ptr and j are sample counts, mem_size is a byte count */ - if (((out_ptr - j) * 2) > audio_buffer->mem_size) - bytes_to_send = audio_buffer->mem_size; - else - bytes_to_send = (out_ptr - j) * 2; - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[j], - bytes_to_send); - /* byte count / 2 (bytes / sample) / channels */ - audio_buffer->num_frames = bytes_to_send / 2 / this->channels; - - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - j += bytes_to_send / 2; /* 2 bytes per sample */ - } - } - - /* reset buffer */ - this->size = 0; -} - -static void ms_ima_adpcm_decode_block(adpcm_decoder_t *this, - buf_element_t *buf) { - - int predictor_l = 0; - int predictor_r = 0; - int index_l = 0; - int index_r = 0; - int channel_counter; - int channel_index; - int channel_index_l; - int channel_index_r; - - int i, j; - audio_buffer_t *audio_buffer; - int bytes_to_send; - - /* check the size */ - if ((this->size % this->in_block_size) != 0) { - lprintf ("received MS IMA block that does not line up\n"); - this->size = 0; - return; - } - - /* iterate through each block in the in buffer */ - for (i = 0; i < this->size; i += this->in_block_size) { - - /* initialize algorithm for this block */ - predictor_l = LE_16(&this->buf[i]); - SE_16BIT(predictor_l); - index_l = this->buf[i + 2]; - if (this->channels == 2) { - predictor_r = LE_16(&this->buf[i + MS_IMA_ADPCM_PREAMBLE_SIZE]); - SE_16BIT(predictor_r); - index_r = this->buf[i + MS_IMA_ADPCM_PREAMBLE_SIZE + 2]; - } - - /* break apart all of the nibbles in the block */ - if (this->channels == 1) { - for (j = 0; - j < (this->in_block_size - MS_IMA_ADPCM_PREAMBLE_SIZE) / 2; j++) { - this->decode_buffer[j * 2 + 0] = - this->buf[i + MS_IMA_ADPCM_PREAMBLE_SIZE + j] & 0x0F; - this->decode_buffer[j * 2 + 1] = - this->buf[i + MS_IMA_ADPCM_PREAMBLE_SIZE + j] >> 4; - } - } else { - /* encoded as 8 nibbles (4 bytes) per channel; switch channel every - * 4th byte */ - channel_counter = 0; - channel_index_l = 0; - channel_index_r = 1; - channel_index = channel_index_l; - for (j = 0; - j < (this->in_block_size - MS_IMA_ADPCM_PREAMBLE_SIZE * 2); j++) { - this->decode_buffer[channel_index + 0] = - this->buf[i + MS_IMA_ADPCM_PREAMBLE_SIZE * 2 + j] & 0x0F; - this->decode_buffer[channel_index + 2] = - this->buf[i + MS_IMA_ADPCM_PREAMBLE_SIZE * 2 + j] >> 4; - channel_index += 4; - channel_counter++; - if (channel_counter == 4) { - channel_index_l = channel_index; - channel_index = channel_index_r; - } else if (channel_counter == 8) { - channel_index_r = channel_index; - channel_index = channel_index_l; - channel_counter = 0; - } - } - } - - /* process the nibbles */ - decode_ima_nibbles(this->decode_buffer, - this->out_block_size, - this->channels, - &predictor_l, &index_l, - &predictor_r, &index_r); - - /* dispatch the decoded audio */ - j = 0; - while (j < this->out_block_size) { - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* out_block_size and j are sample counts, mem_size is a byte count */ - if (((this->out_block_size - j) * 2) > audio_buffer->mem_size) - bytes_to_send = audio_buffer->mem_size; - else - bytes_to_send = (this->out_block_size - j) * 2; - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[j], - bytes_to_send); - /* byte count / 2 (bytes / sample) / channels */ - audio_buffer->num_frames = bytes_to_send / 2 / this->channels; - - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - j += bytes_to_send / 2; /* 2 bytes per sample */ - } - } - - /* reset buffer */ - this->size = 0; - -} - -static void qt_ima_adpcm_decode_block(adpcm_decoder_t *this, - buf_element_t *buf) { - - int initial_predictor_l = 0; - int initial_predictor_r = 0; - int initial_index_l = 0; - int initial_index_r = 0; - - int i, j; - unsigned short *output; - unsigned int out_ptr; - audio_buffer_t *audio_buffer; - - /* check the size */ - if ((this->size % (QT_IMA_ADPCM_BLOCK_SIZE * this->channels) != 0)) { - lprintf ("received QT IMA block that does not line up\n"); - this->size = 0; - return; - } - - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - output = (unsigned short *)audio_buffer->mem; - out_ptr = 0; - - /* iterate through the blocks (and there are 2 bytes/sample) */ - for (i = 0; i < this->size; i+= - (QT_IMA_ADPCM_BLOCK_SIZE * this->channels)) { - - /* send the buffer if it gets full */ - if ((audio_buffer->mem_size / 2) <= - out_ptr + (QT_IMA_ADPCM_SAMPLES_PER_BLOCK * this->channels)) { - - audio_buffer->vpts = buf->pts; - buf->pts = 0; - audio_buffer->num_frames = out_ptr / this->channels; - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - /* get a new audio buffer */ - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - output = (unsigned short *)audio_buffer->mem; - out_ptr = 0; - } - - /* get the left (or mono) channel preamble bytes */ - initial_predictor_l = BE_16(&this->buf[i]); - initial_index_l = initial_predictor_l; - - /* mask, sign-extend, and clamp the predictor portion */ - initial_predictor_l &= 0xFF80; - SE_16BIT(initial_predictor_l); - CLAMP_S16(initial_predictor_l); - - /* mask and clamp the index portion */ - initial_index_l &= 0x7F; - CLAMP_0_TO_88(initial_index_l); - - /* if stereo, handle the right channel too */ - if (this->channels > 1) { - initial_predictor_r = BE_16(&this->buf[i + QT_IMA_ADPCM_BLOCK_SIZE]); - initial_index_r = initial_predictor_r; - - /* mask, sign-extend, and clamp the predictor portion */ - initial_predictor_r &= 0xFF80; - SE_16BIT(initial_predictor_r); - CLAMP_S16(initial_predictor_r); - - /* mask and clamp the index portion */ - initial_index_r &= 0x7F; - CLAMP_0_TO_88(initial_index_r); - } - - /* break apart all of the nibbles in the block */ - if (this->channels == 1) - for (j = 0; j < QT_IMA_ADPCM_SAMPLES_PER_BLOCK / 2; j++) { - output[out_ptr + j * 2 + 0] = this->buf[i + 2 + j] & 0x0F; - output[out_ptr + j * 2 + 1] = this->buf[i + 2 + j] >> 4; - } - else - for (j = 0; j < QT_IMA_ADPCM_SAMPLES_PER_BLOCK / 2 * 2; j++) { - output[out_ptr + j * 4 + 0] = this->buf[i + 2 + j] & 0x0F; - output[out_ptr + j * 4 + 1] = - this->buf[i + 2 + QT_IMA_ADPCM_BLOCK_SIZE + j] & 0x0F; - output[out_ptr + j * 4 + 2] = this->buf[i + 2 + j] >> 4; - output[out_ptr + j * 4 + 3] = - this->buf[i + 2 + QT_IMA_ADPCM_BLOCK_SIZE + j] >> 4; - } - - /* process the nibbles */ - decode_ima_nibbles(&output[out_ptr], - QT_IMA_ADPCM_SAMPLES_PER_BLOCK * this->channels, - this->channels, - &initial_predictor_l, &initial_index_l, - &initial_predictor_r, &initial_index_r); - - out_ptr += QT_IMA_ADPCM_SAMPLES_PER_BLOCK * this->channels; - } - - audio_buffer->vpts = buf->pts; - audio_buffer->num_frames = out_ptr / this->channels; - - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - this->size = 0; -} - -static void ms_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - - int i, j; - unsigned int out_ptr = 0; - audio_buffer_t *audio_buffer; - int bytes_to_send; - - int current_channel = 0; - int idelta[2]; - int sample1[2]; - int sample2[2]; - int coeff1[2]; - int coeff2[2]; - int upper_nibble = 1; - int nibble; - int snibble; /* signed nibble */ - int predictor; - - /* make sure the input size checks out */ - if ((this->size % this->in_block_size) != 0) { - lprintf ("received MS ADPCM block that does not line up\n"); - this->size = 0; - return; - } - - /* iterate through each block in the in buffer */ - for (i = 0; i < this->size; i += this->in_block_size) { - - /* fetch the header information, in stereo if both channels are present */ - j = i; - upper_nibble = 1; - current_channel = 0; - out_ptr = 0; -#ifdef LOG - if (this->buf[j] > 6) - printf("MS ADPCM: coefficient (%d) out of range (should be [0..6])\n", - this->buf[j]); -#endif - coeff1[0] = ms_adapt_coeff1[this->buf[j]]; - coeff2[0] = ms_adapt_coeff2[this->buf[j]]; - j++; - if (this->channels == 2) { - if (this->buf[j] > 6) { - lprintf("MS ADPCM: coefficient (%d) out of range (should be [0..6])\n", - this->buf[j]); - } - coeff1[1] = ms_adapt_coeff1[this->buf[j]]; - coeff2[1] = ms_adapt_coeff2[this->buf[j]]; - j++; - } - - idelta[0] = LE_16(&this->buf[j]); - j += 2; - SE_16BIT(idelta[0]); - if (this->channels == 2) { - idelta[1] = LE_16(&this->buf[j]); - j += 2; - SE_16BIT(idelta[1]); - } - - sample1[0] = LE_16(&this->buf[j]); - j += 2; - SE_16BIT(sample1[0]); - if (this->channels == 2) { - sample1[1] = LE_16(&this->buf[j]); - j += 2; - SE_16BIT(sample1[1]); - } - - sample2[0] = LE_16(&this->buf[j]); - j += 2; - SE_16BIT(sample2[0]); - if (this->channels == 2) { - sample2[1] = LE_16(&this->buf[j]); - j += 2; - SE_16BIT(sample2[1]); - } - - /* first 2 samples go directly to the output */ - if (this->channels == 1) { - this->decode_buffer[out_ptr++] = sample2[0]; - this->decode_buffer[out_ptr++] = sample1[0]; - } else { - this->decode_buffer[out_ptr++] = sample2[0]; - this->decode_buffer[out_ptr++] = sample2[1]; - this->decode_buffer[out_ptr++] = sample1[0]; - this->decode_buffer[out_ptr++] = sample1[1]; - } - - j = MS_ADPCM_PREAMBLE_SIZE * this->channels; - while (j < this->in_block_size) { - /* get the next nibble */ - if (upper_nibble) - nibble = snibble = this->buf[i + j] >> 4; - else - nibble = snibble = this->buf[i + j++] & 0x0F; - upper_nibble ^= 1; - SE_4BIT(snibble); - - predictor = ( - ((sample1[current_channel] * coeff1[current_channel]) + - (sample2[current_channel] * coeff2[current_channel])) / 256) + - (snibble * idelta[current_channel]); - CLAMP_S16(predictor); - sample2[current_channel] = sample1[current_channel]; - sample1[current_channel] = predictor; - this->decode_buffer[out_ptr++] = predictor; - - /* compute the next adaptive scale factor (a.k.a. the variable idelta) */ - idelta[current_channel] = - (ms_adapt_table[nibble] * idelta[current_channel]) / 256; - CLAMP_ABOVE_16(idelta[current_channel]); - - /* toggle the channel */ - current_channel ^= this->channels - 1; - } - - /* dispatch the decoded audio */ - j = 0; - while (j < out_ptr) { - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* out_ptr and j are sample counts, mem_size is a byte count */ - if (((out_ptr - j) * 2) > audio_buffer->mem_size) - bytes_to_send = audio_buffer->mem_size; - else - bytes_to_send = (out_ptr - j) * 2; - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[j], - bytes_to_send); - /* byte count / 2 (bytes / sample) / channels */ - audio_buffer->num_frames = bytes_to_send / 2 / this->channels; - - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - j += bytes_to_send / 2; /* 2 bytes per sample */ - } - } - - /* reset buffer */ - this->size = 0; -} - -static void smjpeg_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - - unsigned int block_size; - int predictor = 0; - int index = 0; - - int i; - unsigned int out_ptr = 0; - audio_buffer_t *audio_buffer; - int bytes_to_send; - - /* fetch the size for this block and check if the decode buffer needs - * to increase */ - block_size = buf->size - 4; /* compensate for preamble */ - block_size *= 2; /* 2 samples / byte */ - if (block_size > this->out_block_size) { - this->out_block_size = block_size; - if (this->decode_buffer) { - free(this->decode_buffer); - } - this->decode_buffer = xine_xmalloc(this->out_block_size * 2); - } - - out_ptr = 0; - predictor = BE_16(&this->buf[0]); - index = this->buf[2]; - - /* break apart the ADPCM nibbles (iterate through each byte in block) */ - for (i = 0; i < block_size / 2; i++) { - this->decode_buffer[out_ptr++] = this->buf[i + 4] & 0x0F; - this->decode_buffer[out_ptr++] = this->buf[i + 4] >> 4; - } - - /* process the nibbles */ - decode_ima_nibbles(this->decode_buffer, - out_ptr, - 1, - &predictor, &index, - 0, 0); - - /* dispatch the decoded audio */ - i = 0; - while (i < out_ptr) { - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* out_ptr and i are sample counts, mem_size is a byte count */ - if (((out_ptr - i) * 2) > audio_buffer->mem_size) - bytes_to_send = audio_buffer->mem_size; - else - bytes_to_send = (out_ptr - i) * 2; - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[i], - bytes_to_send); - /* byte count / 2 (bytes / sample) / channels */ - audio_buffer->num_frames = bytes_to_send / 2 / this->channels; - - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - i += bytes_to_send / 2; /* 2 bytes per sample */ - } - - /* reset buffer */ - this->size = 0; -} - -static void vqa_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - - /* VQA IMA blocks do not have a preamble with an initial index and - * predictor; there is one master index and predictor pair per channel that - * is initialized to 0 and maintained throughout all of the VQA IMA - * blocks. (That is why the following variables are static.) */ - static int index_l = 0; - static int index_r = 0; - static int predictor_l = 0; - static int predictor_r = 0; - - int out_ptr = 0; - int i; - audio_buffer_t *audio_buffer; - int bytes_to_send; - - /* break apart the ADPCM nibbles */ - for (i = 0; i < this->size; i++) { - if (this->channels == 1) { - this->decode_buffer[out_ptr++] = this->buf[i] & 0x0F; - this->decode_buffer[out_ptr++] = (this->buf[i] >> 4) & 0x0F; - } else { - if ((i & 0x1) == 0) { - /* left channel */ - this->decode_buffer[out_ptr + 0] = this->buf[i] & 0x0F; - this->decode_buffer[out_ptr + 2] = (this->buf[i] >> 4) & 0x0F; - } else { - /* right channel */ - this->decode_buffer[out_ptr + 1] = this->buf[i] & 0x0F; - this->decode_buffer[out_ptr + 3] = (this->buf[i] >> 4) & 0x0F; - out_ptr += 4; - } - } - } - - /* process the nibbles */ - decode_ima_nibbles(this->decode_buffer, - out_ptr, - this->channels, - &predictor_l, &index_l, - &predictor_r, &index_r); - - /* dispatch the decoded audio */ - i = 0; - while (i < out_ptr) { - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* out_ptr and i are sample counts, mem_size is a byte count */ - if (((out_ptr - i) * 2) > audio_buffer->mem_size) - bytes_to_send = audio_buffer->mem_size; - else - bytes_to_send = (out_ptr - i) * 2; - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[i], - bytes_to_send); - /* byte count / 2 (bytes / sample) / channels */ - audio_buffer->num_frames = bytes_to_send / 2 / this->channels; - - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - i += bytes_to_send / 2; /* 2 bytes per sample */ - } - - /* reset buffer */ - this->size = 0; -} - -static void ea_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - uint32_t samples_in_chunk; - int32_t previous_left_sample, previous_right_sample; - int32_t current_left_sample, current_right_sample; - int32_t next_left_sample, next_right_sample; - int32_t coeff1l, coeff2l, coeff1r, coeff2r; - uint8_t shift_left, shift_right; - - int count1, count2, i = 0, j = 0; - - samples_in_chunk = ALE_32(&this->buf[i]); - i += 4; - current_left_sample = (int16_t)ALE_16(&this->buf[i]); - i += 2; - previous_left_sample = (int16_t)ALE_16(&this->buf[i]); - i += 2; - current_right_sample = (int16_t)ALE_16(&this->buf[i]); - i += 2; - previous_right_sample = (int16_t)ALE_16(&this->buf[i]); - i += 2; - - if (samples_in_chunk * 4 > this->out_block_size) { - this->out_block_size = samples_in_chunk * 4; - if (this->decode_buffer) { - free(this->decode_buffer); - } - this->decode_buffer = xine_xmalloc(this->out_block_size); - } - - for (count1 = 0; count1 < samples_in_chunk/28;count1++) { - coeff1l = ea_adpcm_table[(this->buf[i] >> 4) & 0x0F]; - coeff2l = ea_adpcm_table[((this->buf[i] >> 4) & 0x0F) + 4]; - coeff1r = ea_adpcm_table[this->buf[i] & 0x0F]; - coeff2r = ea_adpcm_table[(this->buf[i] & 0x0F) + 4]; - i++; - - shift_left = ((this->buf[i] >> 4) & 0x0F) + 8; - shift_right = (this->buf[i] & 0x0F) + 8; - i++; - - for (count2 = 0; count2 < 28; count2++) { - next_left_sample = (((this->buf[i] & 0xF0) << 24) >> shift_left); - next_right_sample = (((this->buf[i] & 0x0F) << 28) >> shift_right); - i++; - - next_left_sample = (next_left_sample + (current_left_sample * coeff1l) + (previous_left_sample * coeff2l) + 0x80) >> 8; - next_right_sample = (next_right_sample + (current_right_sample * coeff1r) + (previous_right_sample * coeff2r) + 0x80) >> 8; - CLAMP_S16(next_left_sample); - CLAMP_S16(next_right_sample); - - previous_left_sample = current_left_sample; - current_left_sample = next_left_sample; - previous_right_sample = current_right_sample; - current_right_sample = next_right_sample; - this->decode_buffer[j] = (unsigned short)current_left_sample; - j++; - this->decode_buffer[j] = (unsigned short)current_right_sample; - j++; - } - } - - i = 0; - while (i < j) { - audio_buffer_t *audio_buffer; - int bytes_to_send; - - audio_buffer = this->stream->audio_out->get_buffer(this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - if (((j - i) * 2) > audio_buffer->mem_size) { - bytes_to_send = audio_buffer->mem_size; - } - else { - bytes_to_send = (j - i) * 2; - } - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[i], bytes_to_send); - - audio_buffer->num_frames = (bytes_to_send / 4); - audio_buffer->vpts = buf->pts; - buf->pts = 0; - this->stream->audio_out->put_buffer(this->stream->audio_out, audio_buffer, this->stream); - - i += bytes_to_send / 2; - } - - this->size = 0; -} - -/* clamp a number between 0 and 48 */ -#define CLAMP_0_TO_48(x) if (x < 0) x = 0; else if (x > 48) x = 48; -/* clamp a number within a signed 12-bit range */ -#define CLAMP_S12(x) if (x < -2048) x = -2048; \ - else if (x > 2048) x = 2048; -static void dialogic_ima_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - - int i; - unsigned int out_ptr = 0; - audio_buffer_t *audio_buffer; - unsigned int block_size; - - /* IMA ADPCM work variables */ - /* the predictor and index values are initialized to 0 and maintained - * throughout the entire stream */ - static int predictor = 0; - static int index = 16; - int step = index; - int diff; - int sign; - int delta; - - /* fetch the size for this block and check if the decode buffer needs - * to increase */ - block_size = buf->size * 2; /* 2 samples / byte */ - if (block_size > this->out_block_size) { - this->out_block_size = block_size; - if (this->decode_buffer) { - free(this->decode_buffer); - } - this->decode_buffer = xine_xmalloc(this->out_block_size * 2); - } - - /* break apart the nibbles */ - for (i = 0; i < this->size; i++) { - this->decode_buffer[out_ptr++] = this->buf[i] >> 4; - this->decode_buffer[out_ptr++] = this->buf[i] & 0xF; - } - - /* decode the nibbles in place using an alternate IMA step table */ - for (i = 0; i < out_ptr; i++) { - - delta = this->decode_buffer[i]; - index += ima_adpcm_index[delta]; - CLAMP_0_TO_48(index); - - sign = delta & 8; - delta = delta & 7; - - diff = step >> 3; - if (delta & 4) diff += step; - if (delta & 2) diff += step >> 1; - if (delta & 1) diff += step >> 2; - - if (sign) - predictor -= diff; - else - predictor += diff; - - CLAMP_S12(predictor); - this->decode_buffer[i] = predictor << 4; - step = dialogic_ima_step[index]; - } - - /* dispatch the decoded audio */ - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - audio_buffer->vpts = buf->pts; - audio_buffer->num_frames = out_ptr; - xine_fast_memcpy(audio_buffer->mem, this->decode_buffer, out_ptr * 2); - - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - /* reset buffer */ - this->size = 0; -} - - -static void xa_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { - int32_t p_l, pp_l, coeff_p_l, coeff_pp_l, range_l; - int32_t p_r, pp_r, coeff_p_r, coeff_pp_r, range_r; - int32_t snd_group, snd_unit, snd_data, samp, i, j; - uint8_t *inp; - - /* restore decoding history */ - p_l = this->xa_p_l; pp_l = this->xa_pp_l; - p_r = this->xa_p_r; pp_r = this->xa_pp_r; - - inp = &this->buf[0]; - j = 0; - - if (this->xa_mode) { - if (this->channels == 2) { - /* mode A (8 bits per sample / 4 sound units) stereo - * - sound units 0,2 are left channel, 1,3 are right channel - * - sound data (8 bits) is shifted left to 16-bit border, then - * shifted right by the range parameter, therefore it's shifted - * (8-range) bits left. - * - two coefficients tables (4 entries each) are merged into one - * - coefficients are multiples of 1/256, so '>> 8' is applied - * after multiplication to get correct answer. - */ - for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { - for (snd_unit = 0; snd_unit < 4; snd_unit += 2) { - /* get left channel coeffs and range */ - coeff_p_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3)]; - coeff_pp_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3) + 4]; - range_l = 8 - (inp[snd_unit] & 0xF); - - /* get right channel coeffs and range */ - coeff_p_r = xa_adpcm_table[((inp[snd_unit+1] >> 4) & 0x3)]; - coeff_pp_r = xa_adpcm_table[((inp[snd_unit+1] >> 4) & 0x3) + 4]; - range_r = 8 - (inp[snd_unit+1] & 0xF); - - for (snd_data = 0; snd_data < 28; snd_data++) { - /* left channel */ - samp = ((signed char *)inp)[16 + (snd_data << 2) + snd_unit]; - samp <<= range_l; - samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; - CLAMP_S16(samp); - pp_l = p_l; - p_l = samp; - this->decode_buffer[j++] = (unsigned short) samp; - - /* right channel */ - samp = ((signed char *)inp)[16 + (snd_data << 2) + snd_unit+1]; - samp <<= range_r; - samp += (coeff_p_r * p_r + coeff_pp_r * pp_r) >> 8; - CLAMP_S16(samp); - pp_r = p_r; - p_r = samp; - this->decode_buffer[j++] = (unsigned short) samp; - } - } - } - } - else { - /* mode A (8 bits per sample / 4 sound units) mono - * - other details as before - */ - for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { - for (snd_unit = 0; snd_unit < 4; snd_unit++) { - /* get coeffs and range */ - coeff_p_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3)]; - coeff_pp_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3) + 4]; - range_l = 8 - (inp[snd_unit] & 0xF); - - for (snd_data = 0; snd_data < 28; snd_data++) { - samp = ((signed char *)inp)[16 + (snd_data << 2) + snd_unit]; - samp <<= range_l; - samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; - CLAMP_S16(samp); - pp_l = p_l; p_l = samp; - this->decode_buffer[j++] = (unsigned short) samp; - } - } - } - } - } - else { - if (this->channels == 2) { - /* mode B/C (4 bits per sample / 8 sound units) stereo - * - sound units 0,2,4,6 are left channel, 1,3,5,7 are right channel - * - sound parameters 0-7 are stored as 16 bytes in the order - * "0123012345674567", so inp[x+4] gives sound parameter x while - * inp[x] doesn't. - * - sound data (4 bits) is shifted left to 16-bit border, then - * shifted right by the range parameter, therefore it's shifted - * (12-range) bits left. - * - other details as before - */ - for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { - for (snd_unit = 0; snd_unit < 8; snd_unit += 2) { - /* get left channel coeffs and range */ - coeff_p_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3)]; - coeff_pp_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3) + 4]; - range_l = 12 - (inp[snd_unit+4] & 0xF); - - /* get right channel coeffs and range */ - coeff_p_r = xa_adpcm_table[((inp[snd_unit+5] >> 4) & 0x3)]; - coeff_pp_r = xa_adpcm_table[((inp[snd_unit+5] >> 4) & 0x3) + 4]; - range_r = 12 - (inp[snd_unit+5] & 0xF); - - for (snd_data = 0; snd_data < 28; snd_data++) { - /* left channel */ - samp = (inp[16 + (snd_data << 2) + (snd_unit >> 1)]) & 0xF; - SE_4BIT(samp); - samp <<= range_l; - samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; - CLAMP_S16(samp); - pp_l = p_l; - p_l = samp; - this->decode_buffer[j++] = (unsigned short) samp; - - /* right channel */ - samp = (inp[16 + (snd_data << 2) + (snd_unit >> 1)] >> 4) & 0xF; - SE_4BIT(samp); - samp <<= range_r; - samp += (coeff_p_r * p_r + coeff_pp_r * pp_r) >> 8; - CLAMP_S16(samp); - pp_r = p_r; - p_r = samp; - this->decode_buffer[j++] = (unsigned short) samp; - } - } - } - } - else { - /* mode B or C (4 bits per sample / 8 sound units) mono - * - other details as before - */ - for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { - for (snd_unit = 0; snd_unit < 8; snd_unit++) { - /* get coeffs and range */ - coeff_p_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3)]; - coeff_pp_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3) + 4]; - range_l = 12 - (inp[snd_unit+4] & 0xF); - - for (snd_data = 0; snd_data < 28; snd_data++) { - samp = inp[16 + (snd_data << 2) + (snd_unit >> 1)]; - if (snd_unit & 1) samp >>= 4; samp &= 0xF; - SE_4BIT(samp); - samp <<= range_l; - samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; - CLAMP_S16(samp); - pp_l = p_l; - p_l = samp; - this->decode_buffer[j++] = (unsigned short) samp; - } - } - } - } - } - - /* store decoding history */ - this->xa_p_l = p_l; this->xa_pp_l = pp_l; - this->xa_p_r = p_r; this->xa_pp_r = pp_r; - - /* despatch the decoded audio */ - i = 0; - while (i < j) { - audio_buffer_t *audio_buffer; - int bytes_to_send; - - audio_buffer= this->stream->audio_out->get_buffer(this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - lprintf ("Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - if (((j - i) * 2) > audio_buffer->mem_size) { - bytes_to_send = audio_buffer->mem_size; - } - else { - bytes_to_send = (j - i) * 2; - } - - xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[i], - bytes_to_send); - - audio_buffer->num_frames = bytes_to_send / (2 * this->channels); - audio_buffer->vpts = buf->pts; - buf->pts = 0; - this->stream->audio_out->put_buffer(this->stream->audio_out, - audio_buffer, this->stream); - - i += bytes_to_send / 2; - } - - /* reset input buffer */ - this->size = 0; -} - -static void adpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { - adpcm_decoder_t *this = (adpcm_decoder_t *) this_gen; - - if (buf->decoder_flags & BUF_FLAG_STDHEADER) { - xine_waveformatex *audio_header; - - this->rate = buf->decoder_info[1]; - this->channels = buf->decoder_info[3]; - this->ao_cap_mode = - (this->channels == 2) ? AO_CAP_MODE_STEREO : AO_CAP_MODE_MONO; - - this->buf = xine_xmalloc(AUDIOBUFSIZE); - this->bufsize = AUDIOBUFSIZE; - this->size = 0; - - /* load the stream information */ - switch (buf->type & 0xFFFF0000) { - - case BUF_AUDIO_MSADPCM: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "Microsoft ADPCM"); - break; - - case BUF_AUDIO_MSIMAADPCM: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "Microsoft IMA ADPCM"); - break; - - case BUF_AUDIO_QTIMAADPCM: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "QT IMA ADPCM"); - break; - - case BUF_AUDIO_DK3ADPCM: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "Duck DK3 ADPCM"); - break; - - case BUF_AUDIO_DK4ADPCM: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "Duck DK4 ADPCM"); - break; - - case BUF_AUDIO_SMJPEG_IMA: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "SMJPEG IMA ADPCM"); - break; - - case BUF_AUDIO_VQA_IMA: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "VQA IMA ADPCM"); - break; - - case BUF_AUDIO_EA_ADPCM: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "EA ADPCM"); - break; - - case BUF_AUDIO_DIALOGIC_IMA: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "Dialogic IMA ADPCM"); - break; - - case BUF_AUDIO_XA_ADPCM: - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "CD-ROM/XA ADPCM"); - break; - - } - - /* if the data was transported in an MS-type file (packet size will be - * non-0 indicating an audio header), create a decode buffer */ - if (buf->size) { - audio_header = (xine_waveformatex *)buf->content; - this->in_block_size = audio_header->nBlockAlign; - - switch(buf->type) { - case BUF_AUDIO_MSADPCM: - this->out_block_size = - (this->in_block_size - - ((MS_ADPCM_PREAMBLE_SIZE - 2) * this->channels)) * 2; - break; - - case BUF_AUDIO_DK4ADPCM: - /* A DK4 ADPCM block has 4 preamble bytes per channel and the - * initial predictor is also the first output sample (hence - * the +1) */ - this->out_block_size = - (this->in_block_size - (4 * this->channels)) * 2 + this->channels; - break; - - case BUF_AUDIO_DK3ADPCM: - /* A DK3 ADPCM block as 16 preamble bytes. A set of 3 nibbles, - * or 1.5 bytes, decodes to 4 PCM samples, so 6 nibbles, or 3 - * bytes, decode to 8 PCM samples. */ - this->out_block_size = - (this->in_block_size - DK3_ADPCM_PREAMBLE_SIZE) * 8 / 3; - break; - - case BUF_AUDIO_MSIMAADPCM: - /* a block of IMA ADPCM stored in an MS-type file has 4 - * preamble bytes per channel. */ - this->out_block_size = - (this->in_block_size - - (MS_IMA_ADPCM_PREAMBLE_SIZE * this->channels)) * 2; - break; - - default: - this->out_block_size = 0; - } - - /* allocate 2 bytes per sample */ - this->decode_buffer = xine_xmalloc(this->out_block_size * 2); - } - - /* the decoder will not know the size of the output buffer until - * an audio packet comes through */ - if ((buf->type == BUF_AUDIO_SMJPEG_IMA) || - (buf->type == BUF_AUDIO_EA_ADPCM) || - (buf->type == BUF_AUDIO_DIALOGIC_IMA)) { - this->in_block_size = this->out_block_size = 0; - this->decode_buffer = NULL; - } - - /* make this decode buffer large enough to hold a second of decoded - * audio */ - if (buf->type == BUF_AUDIO_VQA_IMA) { - this->out_block_size = this->rate * this->channels; - /* allocate 2 bytes per sample */ - this->decode_buffer = xine_xmalloc(this->out_block_size * 2); - } - - /* XA blocks are always 2304 bytes of input data. For output, there - * are 18 sound groups. These sound groups have 4 sound units (mode A) - * or 8 sound units (mode B or mode C). The sound units have 28 sound - * data samples. So, either 18*4*28=2016 or 18*8*28=4032 samples per - * sector. 2 bytes per sample means 4032 or 8064 bytes per sector. - */ - if ((buf->type & 0xFFFF0000) == BUF_AUDIO_XA_ADPCM) { - /* initialise decoder state */ - this->xa_mode = buf->decoder_info[2]; - this->xa_p_l = this->xa_pp_l = this->xa_p_r = this->xa_pp_r = 0; - /* allocate 2 bytes per sample */ - this->decode_buffer = xine_xmalloc((this->xa_mode) ? 4032 : 8064); - } - - return; - } - - if (!this->output_open) { - lprintf ("opening audio output (%d Hz sampling rate, mode=%d)\n", - this->rate, this->ao_cap_mode); - this->output_open = this->stream->audio_out->open (this->stream->audio_out, - this->stream, this->bits_per_sample, this->rate, this->ao_cap_mode); - } - - /* if the audio still isn't open, bail */ - if (!this->output_open) - return; - - /* accumulate compressed audio data */ - if( this->size + buf->size > this->bufsize ) { - this->bufsize = this->size + 2 * buf->size; - lprintf("increasing source buffer to %d to avoid overflow.\n", - this->bufsize); - this->buf = realloc( this->buf, this->bufsize ); - } - - xine_fast_memcpy (&this->buf[this->size], buf->content, buf->size); - this->size += buf->size; - - /* time to decode a frame */ - if (buf->decoder_flags & BUF_FLAG_FRAME_END) { - - switch(buf->type & 0xFFFF0000) { - - case BUF_AUDIO_MSADPCM: - ms_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_MSIMAADPCM: - ms_ima_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_QTIMAADPCM: - qt_ima_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_DK3ADPCM: - dk3_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_DK4ADPCM: - dk4_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_SMJPEG_IMA: - smjpeg_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_VQA_IMA: - vqa_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_EA_ADPCM: - ea_adpcm_decode_block(this, buf); - break; - - case BUF_AUDIO_DIALOGIC_IMA: - dialogic_ima_decode_block(this, buf); - break; - - case BUF_AUDIO_XA_ADPCM: - xa_adpcm_decode_block(this, buf); - break; - } - } -} - -static void adpcm_reset (audio_decoder_t *this_gen) { - - /* adpcm_decoder_t *this = (adpcm_decoder_t *) this_gen; */ - -} - -static void adpcm_discontinuity (audio_decoder_t *this_gen) { - - /* adpcm_decoder_t *this = (adpcm_decoder_t *) this_gen; */ - -} - -static void adpcm_dispose (audio_decoder_t *this_gen) { - - adpcm_decoder_t *this = (adpcm_decoder_t *) this_gen; - - if (this->output_open) - this->stream->audio_out->close (this->stream->audio_out, this->stream); - this->output_open = 0; - - if (this->decode_buffer) - free(this->decode_buffer); - if (this->buf) - free(this->buf); - - free (this_gen); -} - -/* - * ADPCM decoder class code - */ - -static audio_decoder_t *open_plugin (audio_decoder_class_t *class_gen, xine_stream_t *stream) { - - adpcm_decoder_t *this ; - - this = (adpcm_decoder_t *) xine_xmalloc (sizeof (adpcm_decoder_t)); - - this->audio_decoder.decode_data = adpcm_decode_data; - this->audio_decoder.reset = adpcm_reset; - this->audio_decoder.discontinuity = adpcm_discontinuity; - this->audio_decoder.dispose = adpcm_dispose; - - this->output_open = 0; - this->rate = 0; - this->bits_per_sample = 16; /* these codecs always output 16-bit PCM */ - this->channels = 0; - this->ao_cap_mode = 0; - this->decode_buffer = NULL; - this->stream = stream; - - return &this->audio_decoder; -} - -static char *get_identifier (audio_decoder_class_t *this) { - return "ADPCM"; -} - -static char *get_description (audio_decoder_class_t *this) { - return "Multiple ADPCM audio format decoder plugin"; -} - -static void dispose_class (audio_decoder_class_t *this) { - free (this); -} - -static void *init_plugin (xine_t *xine, void *data) { - - adpcm_class_t *this ; - - this = (adpcm_class_t *) xine_xmalloc (sizeof (adpcm_class_t)); - - this->decoder_class.open_plugin = open_plugin; - this->decoder_class.get_identifier = get_identifier; - this->decoder_class.get_description = get_description; - this->decoder_class.dispose = dispose_class; - - return this; -} - -static uint32_t audio_types[] = { - BUF_AUDIO_MSADPCM, BUF_AUDIO_MSIMAADPCM, - BUF_AUDIO_QTIMAADPCM, BUF_AUDIO_DK3ADPCM, - BUF_AUDIO_DK4ADPCM, BUF_AUDIO_SMJPEG_IMA, - BUF_AUDIO_VQA_IMA, BUF_AUDIO_EA_ADPCM, - BUF_AUDIO_DIALOGIC_IMA, BUF_AUDIO_XA_ADPCM, - 0 - }; - -static decoder_info_t dec_info_audio = { - audio_types, /* supported types */ - 9 /* priority */ -}; - -plugin_info_t xine_plugin_info[] = { - /* type, API, "name", version, special_info, init_function */ - { PLUGIN_AUDIO_DECODER, 15, "adpcm", XINE_VERSION_CODE, &dec_info_audio, init_plugin }, - { PLUGIN_NONE, 0, "", 0, NULL, NULL } -}; diff --git a/src/libxineadec/interplayaudio.c b/src/libxineadec/interplayaudio.c deleted file mode 100644 index 7d05f41ac..000000000 --- a/src/libxineadec/interplayaudio.c +++ /dev/null @@ -1,357 +0,0 @@ -/* - * Copyright (C) 2000-2001 the xine project - * - * This file is part of xine, a free video player. - * - * xine 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. - * - * xine 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA - * - * Interplay DPCM Audio Decoder by Mike Melanson (melanson@pcisys.net) - * For more information regarding the Interplay MVE file format, visit: - * http://www.pcisys.net/~melanson/codecs/ - * - * $Id: interplayaudio.c,v 1.8 2004/01/12 17:35:18 miguelfreitas Exp $ - */ - -#include <stdio.h> -#include <stdlib.h> -#include <string.h> -#include <sys/types.h> -#include <unistd.h> - -#include "xine_internal.h" -#include "audio_out.h" -#include "buffer.h" -#include "xineutils.h" -#include "bswap.h" - -#define AUDIOBUFSIZE 128*1024 - -typedef struct { - audio_decoder_class_t decoder_class; -} interplay_class_t; - -typedef struct interplay_decoder_s { - audio_decoder_t audio_decoder; - - xine_stream_t *stream; - - int sample_rate; /* audio sample rate */ - int bits_per_sample; /* bits/sample, usually 8 or 16 */ - int channels; /* 1 or 2, usually */ - - int output_open; /* flag to indicate audio is ready */ - - unsigned char *buf; /* data accumulation buffer */ - int bufsize; /* maximum size of buf */ - int size; /* size of accumulated data in buf */ - - int last_left_delta; - int last_right_delta; - -} interplay_decoder_t; - -/************************************************************************** - * Interplay DPCM specific decode functions - *************************************************************************/ - -int interplay_delta_table[] = { - 0, 1, 2, 3, 4, 5, 6, 7, - 8, 9, 10, 11, 12, 13, 14, 15, - 16, 17, 18, 19, 20, 21, 22, 23, - 24, 25, 26, 27, 28, 29, 30, 31, - 32, 33, 34, 35, 36, 37, 38, 39, - 40, 41, 42, 43, 47, 51, 56, 61, - 66, 72, 79, 86, 94, 102, 112, 122, - 133, 145, 158, 173, 189, 206, 225, 245, - 267, 292, 318, 348, 379, 414, 452, 493, - 538, 587, 640, 699, 763, 832, 908, 991, - 1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993, - 2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008, - 4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059, - 8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206, - 17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589, - -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1, - 1, 1, 5481, 10503, 15105, 19322, 23186, 26728, - 29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298, - -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597, - -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772, - -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373, - -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180, - -1081, -991, -908, -832, -763, -699, -640, -587, - -538, -493, -452, -414, -379, -348, -318, -292, - -267, -245, -225, -206, -189, -173, -158, -145, - -133, -122, -112, -102, -94, -86, -79, -72, - -66, -61, -56, -51, -47, -43, -42, -41, - -40, -39, -38, -37, -36, -35, -34, -33, - -32, -31, -30, -29, -28, -27, -26, -25, - -24, -23, -22, -21, -20, -19, -18, -17, - -16, -15, -14, -13, -12, -11, -10, -9, - -8, -7, -6, -5, -4, -3, -2, -1 - -}; - -/************************************************************************** - * xine audio plugin functions - *************************************************************************/ - -/* clamp a number within a signed 16-bit range */ -#define CLAMP_S16(x) if (x < -32768) x = -32768; \ - else if (x > 32767) x = 32767; - -static void interplay_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { - - interplay_decoder_t *this = (interplay_decoder_t *) this_gen; - audio_buffer_t *audio_buffer; - int i; - int samples_to_send; - int delta[2]; - int stream_ptr = 0; - int sequence_number; - int channel_number = 0; - - if (buf->decoder_flags & BUF_FLAG_STDHEADER) { - - /* When the engine sends a BUF_FLAG_HEADER flag, it is time to initialize - * the decoder. The buffer element type has 4 decoder_info fields, - * 0..3. Field 1 is the sample rate. Field 2 is the bits/sample. Field - * 3 is the number of channels. */ - this->sample_rate = buf->decoder_info[1]; - this->bits_per_sample = buf->decoder_info[2]; - this->channels = buf->decoder_info[3]; - - /* initialize the data accumulation buffer */ - this->buf = xine_xmalloc(AUDIOBUFSIZE); - this->bufsize = AUDIOBUFSIZE; - this->size = 0; - - /* take this opportunity to initialize stream/meta information */ - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "Interplay MVE DPCM"); - - return; - } - - /* if the audio output is not open yet, open the audio output */ - if (!this->output_open) { - this->output_open = this->stream->audio_out->open( - this->stream->audio_out, - this->stream, - 16, - this->sample_rate, - (this->channels == 2) ? AO_CAP_MODE_STEREO : AO_CAP_MODE_MONO); - } - - /* if the audio still isn't open, do not go any further with the decode */ - if (!this->output_open) - return; - - /* accumulate the data passed through the buffer element type; increase - * the accumulator buffer size as necessary */ - if( this->size + buf->size > this->bufsize ) { - this->bufsize = this->size + 2 * buf->size; - xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG, - "interplayaudio: increasing source buffer to %d to avoid overflow.\n", this->bufsize); - this->buf = realloc( this->buf, this->bufsize ); - } - xine_fast_memcpy (&this->buf[this->size], buf->content, buf->size); - this->size += buf->size; - - /* When a buffer element type has the BUF_FLAG_FRAME_END flag set, it is - * time to decode the data in the buffer. */ - if (buf->decoder_flags & BUF_FLAG_FRAME_END) { - - /* if this is the first frame, fetch the first deltas from the stream */ - sequence_number = LE_16(&this->buf[stream_ptr]); - stream_ptr += 6; /* skip over the stream mask and stream length */ - if (sequence_number == 1) { - delta[0] = LE_16(&this->buf[stream_ptr]); - stream_ptr += 2; - if (this->channels == 2) { - delta[1] = LE_16(&this->buf[stream_ptr]); - stream_ptr += 2; - } - } else { - delta[0] = this->last_left_delta; - if (this->channels == 2) - delta[1] = this->last_right_delta; - } - - /* iterate through each 8-bit delta in the input buffer */ - while (stream_ptr < this->size) { - - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - xprintf (this->stream->xine, XINE_VERBOSITY_DEBUG, - "interplayaudio: Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* this->size and stream_ptr are sample counts, mem_size is a byte count */ - if ((this->size - stream_ptr) > (audio_buffer->mem_size / 2)) - samples_to_send = audio_buffer->mem_size / 2; - else - samples_to_send = this->size - stream_ptr; - - /* fill up this buffer */ - for (i = 0; i < samples_to_send; i++) { - delta[channel_number] += interplay_delta_table[this->buf[stream_ptr++]]; - CLAMP_S16(delta[channel_number]); - audio_buffer->mem[i] = delta[channel_number]; - - /* toggle channel */ - channel_number ^= this->channels - 1; - } - - audio_buffer->num_frames = samples_to_send / this->channels; - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - } - - /* stash away the deltas for the next block */ - this->last_left_delta = delta[0]; - this->last_right_delta = delta[1]; - - /* reset data accumulation buffer */ - this->size = 0; - } -} - -/* This function resets the state of the audio decoder. This usually - * entails resetting the data accumulation buffer. */ -static void interplay_reset (audio_decoder_t *this_gen) { - - interplay_decoder_t *this = (interplay_decoder_t *) this_gen; - - this->size = 0; -} - -/* This function resets the last pts value of the audio decoder. */ -static void interplay_discontinuity (audio_decoder_t *this_gen) { -} - -/* This function closes the audio output and frees the private audio decoder - * structure. */ -static void interplay_dispose (audio_decoder_t *this_gen) { - - interplay_decoder_t *this = (interplay_decoder_t *) this_gen; - - /* close the audio output */ - if (this->output_open) - this->stream->audio_out->close (this->stream->audio_out, this->stream); - this->output_open = 0; - - /* free anything that was allocated during operation */ - free(this->buf); - free(this); -} - -/* This function allocates, initializes, and returns a private audio - * decoder structure. */ -static audio_decoder_t *open_plugin (audio_decoder_class_t *class_gen, xine_stream_t *stream) { - - interplay_decoder_t *this ; - - this = (interplay_decoder_t *) xine_xmalloc (sizeof (interplay_decoder_t)); - - /* connect the member functions */ - this->audio_decoder.decode_data = interplay_decode_data; - this->audio_decoder.reset = interplay_reset; - this->audio_decoder.discontinuity = interplay_discontinuity; - this->audio_decoder.dispose = interplay_dispose; - - /* connect the stream */ - this->stream = stream; - - /* audio output is not open at the start */ - this->output_open = 0; - - /* initialize the basic audio parameters */ - this->channels = 0; - this->sample_rate = 0; - this->bits_per_sample = 0; - - /* initialize the data accumulation buffer */ - this->buf = NULL; - this->bufsize = 0; - this->size = 0; - - /* return the newly-initialized audio decoder */ - return &this->audio_decoder; -} - -/* This function returns a brief string that describes (usually with the - * decoder's most basic name) the audio decoder plugin. */ -static char *get_identifier (audio_decoder_class_t *this) { - return "Interplay DPCM Audio"; -} - -/* This function returns a slightly longer string describing the audio - * decoder plugin. */ -static char *get_description (audio_decoder_class_t *this) { - return "Interplay DPCM audio decoder plugin"; -} - -/* This function frees the audio decoder class and any other memory that was - * allocated. */ -static void dispose_class (audio_decoder_class_t *this_gen) { - - interplay_class_t *this = (interplay_class_t *)this_gen; - - free (this); -} - -/* This function allocates a private audio decoder class and initializes - * the class's member functions. */ -static void *init_plugin (xine_t *xine, void *data) { - - interplay_class_t *this ; - - this = (interplay_class_t *) xine_xmalloc (sizeof (interplay_class_t)); - - this->decoder_class.open_plugin = open_plugin; - this->decoder_class.get_identifier = get_identifier; - this->decoder_class.get_description = get_description; - this->decoder_class.dispose = dispose_class; - - return this; -} - -/* This is a list of all of the internal xine audio buffer types that - * this decoder is able to handle. Check src/xine-engine/buffer.h for a - * list of valid buffer types (and add a new one if the one you need does - * not exist). Terminate the list with a 0. */ -static uint32_t audio_types[] = { - BUF_AUDIO_INTERPLAY, - 0 -}; - -/* This data structure combines the list of supported xine buffer types and - * the priority that the plugin should be given with respect to other - * plugins that handle the same buffer type. A plugin with priority (n+1) - * will be used instead of a plugin with priority (n). */ -static decoder_info_t dec_info_audio = { - audio_types, /* supported types */ - 5 /* priority */ -}; - -/* The plugin catalog entry. This is the only information that this plugin - * will export to the public. */ -plugin_info_t xine_plugin_info[] = { - /* { type, API version, "name", version, special_info, init_function }, */ - { PLUGIN_AUDIO_DECODER, 15, "interplayaudio", XINE_VERSION_CODE, &dec_info_audio, &init_plugin }, - { PLUGIN_NONE, 0, "", 0, NULL, NULL } -}; - diff --git a/src/libxineadec/logpcm.c b/src/libxineadec/logpcm.c deleted file mode 100644 index 3e2ea1a25..000000000 --- a/src/libxineadec/logpcm.c +++ /dev/null @@ -1,312 +0,0 @@ -/* - * Copyright (C) 2000-2003 the xine project - * - * This file is part of xine, a free video player. - * - * xine 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. - * - * xine 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA - * - * Logarithmic PCM Decoder - * This audio decoder handles the mu-law and A-law logarithmic coding - * standards. - * - * mu-law -> PCM conversion routine found at this site: - * http://www.speech.cs.cmu.edu/comp.speech/Section2/Q2.7.html - * and credited to this person: - * Craig Reese: IDA/Supercomputing Research Center - * - * A-law -> PCM conversion routine lifted from SoX Sound eXchange: - * http://sox.sourceforge.net/ - * which listed the code as being lifted from Sun Microsystems. - * - * $Id: logpcm.c,v 1.19 2004/01/12 17:35:18 miguelfreitas Exp $ - * - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <stdlib.h> -#include <string.h> - -#include "buffer.h" -#include "xine_internal.h" -#include "video_out.h" -#include "audio_out.h" -#include "xineutils.h" -#include "bswap.h" - -#define AUDIOBUFSIZE 128*1024 -#define LOGPCM_BITS_PER_SAMPLE 16 - -typedef struct { - audio_decoder_class_t decoder_class; -} logpcm_class_t; - -typedef struct logpcm_decoder_s { - audio_decoder_t audio_decoder; - - xine_stream_t *stream; - - int64_t pts; - - int output_open; - int output_channels; - int samplerate; - - unsigned char *buf; - int bufsize; - int size; - - short logtable[256]; -} logpcm_decoder_t; - -/* -** This routine converts from ulaw to 16 bit linear. -** -** Craig Reese: IDA/Supercomputing Research Center -** 29 September 1989 -** -** References: -** 1) CCITT Recommendation G.711 (very difficult to follow) -** 2) MIL-STD-188-113,"Interoperability and Performance Standards -** for Analog-to_Digital Conversion Techniques," -** 17 February 1987 -** -** Input: 8 bit ulaw sample -** Output: signed 16 bit linear sample -*/ - -static int mulaw2linear(unsigned char mulawbyte) { - - static int exp_lut[8] = {0,132,396,924,1980,4092,8316,16764}; - int sign, exponent, mantissa, sample; - - mulawbyte = ~mulawbyte; - sign = (mulawbyte & 0x80); - exponent = (mulawbyte >> 4) & 0x07; - mantissa = mulawbyte & 0x0F; - sample = exp_lut[exponent] + (mantissa << (exponent + 3)); - if (sign != 0) sample = -sample; - - return(sample); -} - -/* - * The following A-law -> PCM conversion function came from SoX which in - * turn came from Sun Microsystems. - */ -#define SIGN_BIT (0x80) /* Sign bit for a A-law byte. */ -#define QUANT_MASK (0xf) /* Quantization field mask. */ -#define SEG_SHIFT (4) /* Left shift for segment number. */ -#define SEG_MASK (0x70) /* Segment field mask. */ - -/* - * alaw2linear() - Convert an A-law value to 16-bit signed linear PCM - * - */ -static int alaw2linear(unsigned char a_val) { - - int t; - int seg; - - a_val ^= 0x55; - - t = (a_val & QUANT_MASK) << 4; - seg = ((unsigned)a_val & SEG_MASK) >> SEG_SHIFT; - switch (seg) { - case 0: - t += 8; - break; - - case 1: - t += 0x108; - break; - - default: - t += 0x108; - t <<= seg - 1; - } - return ((a_val & SIGN_BIT) ? t : -t); -} - - -static void logpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { - - logpcm_decoder_t *this = (logpcm_decoder_t *) this_gen; - audio_buffer_t *audio_buffer; - int in; - int i; - int bytes_to_send; - - if (buf->decoder_flags & BUF_FLAG_STDHEADER) { - - this->samplerate = buf->decoder_info[1]; - this->output_channels = buf->decoder_info[3]; - this->buf = xine_xmalloc(AUDIOBUFSIZE); - this->bufsize = AUDIOBUFSIZE; - this->size = 0; - - /* stream/meta info */ - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - (buf->type == BUF_AUDIO_MULAW) ? "mu-law log PCM" : - "A-law log PCM" ); - - /* pre-calculate the possible log values */ - if (buf->type == BUF_AUDIO_MULAW) - for (i = 0; i < 256; i++) - this->logtable[i] = (short)mulaw2linear(i); - else if (buf->type == BUF_AUDIO_ALAW) - for (i = 0; i < 256; i++) - this->logtable[i] = (short)alaw2linear(i); - - return; - } - - if (!this->output_open) { - this->output_open = this->stream->audio_out->open(this->stream->audio_out, - this->stream, LOGPCM_BITS_PER_SAMPLE, this->samplerate, - (this->output_channels == 2) ? AO_CAP_MODE_STEREO : AO_CAP_MODE_MONO); - } - - /* if the audio still isn't open, bail */ - if (!this->output_open) - return; - - if( this->size + buf->size > this->bufsize ) { - this->bufsize = this->size + 2 * buf->size; - xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG, - "logpcm: increasing source buffer to %d to avoid overflow.\n", this->bufsize); - this->buf = realloc( this->buf, this->bufsize ); - } - - xine_fast_memcpy (&this->buf[this->size], buf->content, buf->size); - this->size += buf->size; - - if (buf->decoder_flags & BUF_FLAG_FRAME_END) { /* time to decode a frame */ - - /* iterate through each 8-bit sample in the input buffer */ - in = 0; - while (in < this->size) { - - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - if (audio_buffer->mem_size == 0) { - xprintf (this->stream->xine, XINE_VERBOSITY_DEBUG, - "logpcm: Help! Allocated audio buffer with nothing in it!\n"); - return; - } - - /* this->size and in are sample counts, mem_size is a byte count */ - if ((this->size - in) > (audio_buffer->mem_size / 2)) - bytes_to_send = audio_buffer->mem_size / 2; - else - bytes_to_send = this->size - in; - - /* fill up this buffer */ - for (i = 0; i < bytes_to_send; i++) - audio_buffer->mem[i] = this->logtable[this->buf[in++]]; - - audio_buffer->num_frames = bytes_to_send / this->output_channels; - audio_buffer->vpts = buf->pts; - buf->pts = 0; /* only first buffer gets the real pts */ - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - } - - /* reset internal accumulation buffer */ - this->size = 0; - } -} - -static void logpcm_reset (audio_decoder_t *this_gen) { -} - -static void logpcm_discontinuity (audio_decoder_t *this_gen) { - logpcm_decoder_t *this = (logpcm_decoder_t *) this_gen; - - this->pts = 0; -} - -static void logpcm_dispose (audio_decoder_t *this_gen) { - - logpcm_decoder_t *this = (logpcm_decoder_t *) this_gen; - - if (this->output_open) - this->stream->audio_out->close (this->stream->audio_out, this->stream); - this->output_open = 0; - - if (this->buf) - free(this->buf); - - free (this_gen); -} - -static audio_decoder_t *open_plugin (audio_decoder_class_t *class_gen, xine_stream_t *stream) { - - logpcm_decoder_t *this ; - - this = (logpcm_decoder_t *) xine_xmalloc (sizeof (logpcm_decoder_t)); - - this->audio_decoder.decode_data = logpcm_decode_data; - this->audio_decoder.reset = logpcm_reset; - this->audio_decoder.discontinuity = logpcm_discontinuity; - this->audio_decoder.dispose = logpcm_dispose; - - this->output_open = 0; - this->output_channels = 0; - this->stream = stream; - this->buf = NULL; - this->size = 0; - - return &this->audio_decoder; -} - -static char *get_identifier (audio_decoder_class_t *this) { - return "Log PCM"; -} - -static char *get_description (audio_decoder_class_t *this) { - return "Logarithmic PCM audio format decoder plugin"; -} - -static void dispose_class (audio_decoder_class_t *this) { - free (this); -} - -static void *init_plugin (xine_t *xine, void *data) { - - logpcm_class_t *this ; - - this = (logpcm_class_t *) xine_xmalloc (sizeof (logpcm_class_t)); - - this->decoder_class.open_plugin = open_plugin; - this->decoder_class.get_identifier = get_identifier; - this->decoder_class.get_description = get_description; - this->decoder_class.dispose = dispose_class; - - return this; -} - -static uint32_t audio_types[] = { BUF_AUDIO_MULAW, BUF_AUDIO_ALAW, 0 }; - -static decoder_info_t dec_info_audio = { - audio_types, /* supported types */ - 9 /* priority */ -}; - -plugin_info_t xine_plugin_info[] = { - /* type, API, "name", version, special_info, init_function */ - { PLUGIN_AUDIO_DECODER, 15, "logpcm", XINE_VERSION_CODE, &dec_info_audio, &init_plugin }, - { PLUGIN_NONE, 0, "", 0, NULL, NULL } -}; diff --git a/src/libxineadec/roqaudio.c b/src/libxineadec/roqaudio.c deleted file mode 100644 index 6bc713a98..000000000 --- a/src/libxineadec/roqaudio.c +++ /dev/null @@ -1,242 +0,0 @@ -/* - * Copyright (C) 2000-2003 the xine project - * - * This file is part of xine, a free video player. - * - * xine 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. - * - * xine 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA - * - * RoQ DPCM Audio Decoder by Mike Melanson (melanson@pcisys.net) - * For more information regarding the RoQ file format, visit: - * http://www.csse.monash.edu.au/~timf/ - * - * $Id: roqaudio.c,v 1.22 2004/01/12 17:35:18 miguelfreitas Exp $ - * - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include <stdlib.h> -#include <string.h> - -#include "xine_internal.h" -#include "audio_out.h" -#include "buffer.h" -#include "xineutils.h" -#include "bswap.h" - -#define RoQ_AUDIO_SAMPLE_RATE 22050 -#define RoQ_AUDIO_BITS_PER_SAMPLE 16 - -#define AUDIOBUFSIZE 128*1024 - -#define CLAMP_S16(x) if (x < -32768) x = -32768; \ - else if (x > 32767) x = 32767; -#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; - -typedef struct { - audio_decoder_class_t decoder_class; -} roqaudio_class_t; - -typedef struct roqaudio_decoder_s { - audio_decoder_t audio_decoder; - - xine_stream_t *stream; - - int64_t pts; - - int output_open; - int output_channels; - - unsigned char *buf; - int bufsize; - int size; - - short square_array[256]; -} roqaudio_decoder_t; - -static void roqaudio_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { - roqaudio_decoder_t *this = (roqaudio_decoder_t *) this_gen; - audio_buffer_t *audio_buffer; - int in, out; - int predictor[2]; - int channel_number = 0; - - if (buf->decoder_flags & BUF_FLAG_STDHEADER) { - this->output_channels = buf->decoder_info[3]; - - this->buf = xine_xmalloc(AUDIOBUFSIZE); - this->bufsize = AUDIOBUFSIZE; - this->size = 0; - - /* stream/meta info */ - _x_meta_info_set(this->stream, XINE_META_INFO_AUDIOCODEC, - "RoQ DPCM Audio"); - - return; - } - - if (!this->output_open) { - this->output_open = this->stream->audio_out->open(this->stream->audio_out, - this->stream, RoQ_AUDIO_BITS_PER_SAMPLE, RoQ_AUDIO_SAMPLE_RATE, - (this->output_channels == 2) ? AO_CAP_MODE_STEREO : AO_CAP_MODE_MONO); - } - - /* if the audio still isn't open, bail */ - if (!this->output_open) - return; - - if( this->size + buf->size > this->bufsize ) { - this->bufsize = this->size + 2 * buf->size; - xprintf(this->stream->xine, XINE_VERBOSITY_DEBUG, - "RoQ: increasing source buffer to %d to avoid overflow.\n", this->bufsize); - this->buf = realloc( this->buf, this->bufsize ); - } - - xine_fast_memcpy (&this->buf[this->size], buf->content, buf->size); - this->size += buf->size; - - if (buf->decoder_flags & BUF_FLAG_FRAME_END) { /* time to decode a frame */ - audio_buffer = this->stream->audio_out->get_buffer (this->stream->audio_out); - - out = 0; - - /* prepare the initial predictors */ - if (this->output_channels == 1) - predictor[0] = LE_16(&this->buf[6]); - else - { - predictor[0] = this->buf[7] << 8; - predictor[1] = this->buf[6] << 8; - } - SE_16BIT(predictor[0]); - SE_16BIT(predictor[1]); - - /* decode the samples */ - for (in = 8; in < this->size; in++) - { - predictor[channel_number] += this->square_array[this->buf[in]]; - CLAMP_S16(predictor[channel_number]); - audio_buffer->mem[out++] = predictor[channel_number]; - - /* toggle channel */ - channel_number ^= this->output_channels - 1; - } - - audio_buffer->vpts = buf->pts; - audio_buffer->num_frames = - (buf->size - 8) / this->output_channels; - - this->stream->audio_out->put_buffer (this->stream->audio_out, audio_buffer, this->stream); - - this->size = 0; - } -} - -static void roqaudio_reset (audio_decoder_t *this_gen) { -} - -static void roqaudio_discontinuity (audio_decoder_t *this_gen) { - - roqaudio_decoder_t *this = (roqaudio_decoder_t *) this_gen; - - this->pts = 0; -} - -static void roqaudio_dispose (audio_decoder_t *this_gen) { - - roqaudio_decoder_t *this = (roqaudio_decoder_t *) this_gen; - - if (this->output_open) - this->stream->audio_out->close (this->stream->audio_out, this->stream); - this->output_open = 0; - - if (this->buf) - free(this->buf); - - free (this_gen); -} - -static audio_decoder_t *open_plugin (audio_decoder_class_t *class_gen, xine_stream_t *stream) { - - roqaudio_decoder_t *this; - int i; - short square; - - this = (roqaudio_decoder_t *) xine_xmalloc (sizeof (roqaudio_decoder_t)); - - this->audio_decoder.decode_data = roqaudio_decode_data; - this->audio_decoder.reset = roqaudio_reset; - this->audio_decoder.discontinuity = roqaudio_discontinuity; - this->audio_decoder.dispose = roqaudio_dispose; - this->size = 0; - - this->stream = stream; - - this->buf = NULL; - this->output_open = 0; - this->output_channels = 0; - - /* initialize tables of squares */ - for (i = 0; i < 128; i++) { - square = i * i; - this->square_array[i] = square; - this->square_array[i + 128] = -square; - } - - return &this->audio_decoder; -} - -static char *get_identifier (audio_decoder_class_t *this) { - return "RoQ Audio"; -} - -static char *get_description (audio_decoder_class_t *this) { - return "Id Roq audio decoder plugin"; -} - -static void dispose_class (audio_decoder_class_t *this) { - free (this); -} - -static void *init_plugin (xine_t *xine, void *data) { - - roqaudio_class_t *this; - - this = (roqaudio_class_t *) xine_xmalloc (sizeof (roqaudio_class_t)); - - this->decoder_class.open_plugin = open_plugin; - this->decoder_class.get_identifier = get_identifier; - this->decoder_class.get_description = get_description; - this->decoder_class.dispose = dispose_class; - - return this; -} - -static uint32_t audio_types[] = { - BUF_AUDIO_ROQ, 0 -}; - -static decoder_info_t dec_info_audio = { - audio_types, /* supported types */ - 5 /* priority */ -}; - -plugin_info_t xine_plugin_info[] = { - /* type, API, "name", version, special_info, init_function */ - { PLUGIN_AUDIO_DECODER, 15, "roqaudio", XINE_VERSION_CODE, &dec_info_audio, init_plugin }, - { PLUGIN_NONE, 0, "", 0, NULL, NULL } -}; |