From 0ea721f7ce81357bc4ec6eea609cd50482c3d15b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Diego=20=27Flameeyes=27=20Petten=C3=B2?= Date: Sat, 2 Dec 2006 01:19:48 +0000 Subject: Start working on a branch where FFmpeg is not copied, patched and carved to be built with automake but instead imported inline and built using its own build system. This is an import of a slightly modified FFmpeg current tree. xine-lib builds, install and run fine with it, but there are of course plenty of things that needs to be fixed before it can even be considered for a 1.2.x series. Work will continue in the next days of course. CVS patchset: 8397 CVS date: 2006/12/02 01:19:48 --- contrib/ffmpeg/libavcodec/resample.c | 249 +++++++++++++++++++++++++++++++++++ 1 file changed, 249 insertions(+) create mode 100644 contrib/ffmpeg/libavcodec/resample.c (limited to 'contrib/ffmpeg/libavcodec/resample.c') diff --git a/contrib/ffmpeg/libavcodec/resample.c b/contrib/ffmpeg/libavcodec/resample.c new file mode 100644 index 000000000..043e812c8 --- /dev/null +++ b/contrib/ffmpeg/libavcodec/resample.c @@ -0,0 +1,249 @@ +/* + * Sample rate convertion for both audio and video + * Copyright (c) 2000 Fabrice Bellard. + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file resample.c + * Sample rate convertion for both audio and video. + */ + +#include "avcodec.h" + +struct AVResampleContext; + +struct ReSampleContext { + struct AVResampleContext *resample_context; + short *temp[2]; + int temp_len; + float ratio; + /* channel convert */ + int input_channels, output_channels, filter_channels; +}; + +/* n1: number of samples */ +static void stereo_to_mono(short *output, short *input, int n1) +{ + short *p, *q; + int n = n1; + + p = input; + q = output; + while (n >= 4) { + q[0] = (p[0] + p[1]) >> 1; + q[1] = (p[2] + p[3]) >> 1; + q[2] = (p[4] + p[5]) >> 1; + q[3] = (p[6] + p[7]) >> 1; + q += 4; + p += 8; + n -= 4; + } + while (n > 0) { + q[0] = (p[0] + p[1]) >> 1; + q++; + p += 2; + n--; + } +} + +/* n1: number of samples */ +static void mono_to_stereo(short *output, short *input, int n1) +{ + short *p, *q; + int n = n1; + int v; + + p = input; + q = output; + while (n >= 4) { + v = p[0]; q[0] = v; q[1] = v; + v = p[1]; q[2] = v; q[3] = v; + v = p[2]; q[4] = v; q[5] = v; + v = p[3]; q[6] = v; q[7] = v; + q += 8; + p += 4; + n -= 4; + } + while (n > 0) { + v = p[0]; q[0] = v; q[1] = v; + q += 2; + p += 1; + n--; + } +} + +/* XXX: should use more abstract 'N' channels system */ +static void stereo_split(short *output1, short *output2, short *input, int n) +{ + int i; + + for(i=0;i 2) + { + av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported."); + return NULL; + } + + s = av_mallocz(sizeof(ReSampleContext)); + if (!s) + { + av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context."); + return NULL; + } + + s->ratio = (float)output_rate / (float)input_rate; + + s->input_channels = input_channels; + s->output_channels = output_channels; + + s->filter_channels = s->input_channels; + if (s->output_channels < s->filter_channels) + s->filter_channels = s->output_channels; + +/* + * ac3 output is the only case where filter_channels could be greater than 2. + * input channels can't be greater than 2, so resample the 2 channels and then + * expand to 6 channels after the resampling. + */ + if(s->filter_channels>2) + s->filter_channels = 2; + + s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0); + + return s; +} + +/* resample audio. 'nb_samples' is the number of input samples */ +/* XXX: optimize it ! */ +int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) +{ + int i, nb_samples1; + short *bufin[2]; + short *bufout[2]; + short *buftmp2[2], *buftmp3[2]; + int lenout; + + if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { + /* nothing to do */ + memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); + return nb_samples; + } + + /* XXX: move those malloc to resample init code */ + for(i=0; ifilter_channels; i++){ + bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); + memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); + buftmp2[i] = bufin[i] + s->temp_len; + } + + /* make some zoom to avoid round pb */ + lenout= (int)(nb_samples * s->ratio) + 16; + bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); + bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); + + if (s->input_channels == 2 && + s->output_channels == 1) { + buftmp3[0] = output; + stereo_to_mono(buftmp2[0], input, nb_samples); + } else if (s->output_channels >= 2 && s->input_channels == 1) { + buftmp3[0] = bufout[0]; + memcpy(buftmp2[0], input, nb_samples*sizeof(short)); + } else if (s->output_channels >= 2) { + buftmp3[0] = bufout[0]; + buftmp3[1] = bufout[1]; + stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); + } else { + buftmp3[0] = output; + memcpy(buftmp2[0], input, nb_samples*sizeof(short)); + } + + nb_samples += s->temp_len; + + /* resample each channel */ + nb_samples1 = 0; /* avoid warning */ + for(i=0;ifilter_channels;i++) { + int consumed; + int is_last= i+1 == s->filter_channels; + + nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last); + s->temp_len= nb_samples - consumed; + s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short)); + memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short)); + } + + if (s->output_channels == 2 && s->input_channels == 1) { + mono_to_stereo(output, buftmp3[0], nb_samples1); + } else if (s->output_channels == 2) { + stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); + } else if (s->output_channels == 6) { + ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); + } + + for(i=0; ifilter_channels; i++) + av_free(bufin[i]); + + av_free(bufout[0]); + av_free(bufout[1]); + return nb_samples1; +} + +void audio_resample_close(ReSampleContext *s) +{ + av_resample_close(s->resample_context); + av_freep(&s->temp[0]); + av_freep(&s->temp[1]); + av_free(s); +} -- cgit v1.2.3