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ca6a88a8fc
Original commit message from CVS: * gst/multifile/gstmultifilesink.c: (gst_multi_file_sink_class_init): * gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init): * gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer), (gst_mve_video_palette), (gst_mve_video_code_map), (gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create), (gst_mve_demux_chain): * gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk): * gst/mve/mveaudioenc.c: (mve_compress_audio): * gst/mve/mvevideodec16.c: (ipvideo_copy_block): * gst/mve/mvevideodec8.c: (ipvideo_copy_block): * gst/mve/mvevideoenc16.c: (mve_encode_frame16): * gst/mve/mvevideoenc8.c: (mve_encode_frame8): Use proper print statements. Fixes build on mac os x. <wingo> oo look at me my name is edward i'm hacking on macos wooo
120 lines
4.3 KiB
C
120 lines
4.3 KiB
C
/*
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* Interplay MVE audio compressor
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* Copyright (C) 2003, 2004 Alexander Belyakov <abel@krasu.ru>
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* Copyright (C) 2006 Jens Granseuer <jensgr@gmx.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <math.h>
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#include <gst/gst.h>
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static const gint32 dec_table[256] =
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{
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0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15,
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16, 17, 18, 19,
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20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31,
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32, 33, 34, 35, 36, 37,
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38, 39, 40, 41, 42, 43, 47, 51, 56, 61,
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66, 72, 79, 86, 94, 102, 112,
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122, 133, 145, 158, 173, 189, 206, 225, 245,
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267, 292, 318, 348, 379,
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414, 452, 493, 538, 587, 640, 699, 763, 832, 908, 991,
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1081, 1180, 1288,
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1405, 1534, 1673, 1826, 1993, 2175, 2373, 2590, 2826, 3084, 3365, 3672,
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4008,
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4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059, 8794, 9597, 10472,
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11428, 12471, 13609, 14851, 16206,
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17685, 19298, 21060, 22981, 25078,
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27367, 29864, 32589, 35563, 38808, 42350, 46214, 50431, 55033, 60055,
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65535,
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1, -65535, -60055, -55033, -50431, -46214, -42350, -38808, -35563,
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-32589, -29864, -27367, -25078, -22981, -21060, -19298,
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-17685, -16206,
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-14851, -13609, -12471, -11428, -10472, -9597, -8794, -8059, -7385, -6767,
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-6202, -5683, -5208, -4772,
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-4373, -4008, -3672, -3365, -3084, -2826,
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-2590, -2373, -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
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-1081, -991, -908, -832, -763, -699, -640, -587, -538, -493, -452, -414,
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-379, -348, -318, -292,
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-267, -245, -225, -206, -189, -173, -158, -145,
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-133, -122, -112, -102, -94, -86, -79, -72,
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-66, -61, -56, -51, -47, -43,
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-42, -41, -40, -39, -38, -37, -36, -35, -34, -33,
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-32, -31, -30, -29,
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-28, -27, -26, -25, -24, -23, -22, -21, -20, -19, -18, -17,
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-16, -15,
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-14, -13, -12, -11, -10, -9, -8, -7, -6, -5, -4, -3, -2, -1
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};
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/* This value could be non-optimal. Without knowledge of the value
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distribution in the real signal, the actual optimum cannot be evaluated.
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Should be somewhere between 11.458 and 11.542. */
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static const gdouble DPCM_SCALE = 11.5131;
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static gint8
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mve_enc_delta (guint n)
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{
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if (n < 44)
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return n;
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return floor (DPCM_SCALE * log (n));
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}
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gint
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mve_compress_audio (guint8 * dest, const guint8 * src, guint16 len,
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guint8 channels)
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{
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gint16 prev[2], s;
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gint delta, real_res;
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gint cur_chan;
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guint8 v;
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for (cur_chan = 0; cur_chan < channels; ++cur_chan) {
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prev[cur_chan] = GST_READ_UINT16_LE (src);
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GST_WRITE_UINT16_LE (dest, prev[cur_chan]);
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src += 2;
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dest += 2;
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len -= 2;
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}
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cur_chan = 0;
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while (len > 0) {
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s = GST_READ_UINT16_LE (src);
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src += 2;
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delta = s - prev[cur_chan];
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if (delta >= 0)
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v = mve_enc_delta (delta);
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else
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v = 256 - mve_enc_delta (-delta);
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real_res = dec_table[v] + prev[cur_chan];
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if (real_res < -32768 || real_res > 32767) {
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/* correct overflow */
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/* GST_DEBUG ("co:%d + %d = %d -> new v:%d, dec_table:%d will be %d",
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prev[cur_chan], dec_table[v], real_res,
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v, dec_table[v], prev[cur_chan]+dec_table[v]); */
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if (s > 0) {
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if (real_res > 32767)
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--v;
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} else {
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if (real_res < -32768)
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++v;
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}
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real_res = dec_table[v] + prev[cur_chan];
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}
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if (G_UNLIKELY (abs (real_res - s) > 32767)) {
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GST_ERROR ("sign loss left unfixed in audio stream, deviation:%d",
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real_res - s);
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return -1;
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}
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*dest++ = v;
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--len;
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/* use previous output instead of input. That way output will not go too far from input. */
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prev[cur_chan] += dec_table[v];
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cur_chan = channels - 1 - cur_chan;
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}
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return 0;
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}
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