Mercurial > libavcodec.hg
view resample.c @ 3643:ae1dfbbb25bb libavcodec
fix low bitrate mp3
author | michael |
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date | Sun, 27 Aug 2006 11:42:40 +0000 |
parents | 0b546eab515d |
children | c8c591fe26f8 |
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/* * Sample rate convertion for both audio and video * Copyright (c) 2000 Fabrice Bellard. * * This library 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 of the License, or (at your option) any later version. * * This library 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 this library; 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<n;i++) { *output1++ = *input++; *output2++ = *input++; } } static void stereo_mux(short *output, short *input1, short *input2, int n) { int i; for(i=0;i<n;i++) { *output++ = *input1++; *output++ = *input2++; } } static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) { int i; short l,r; for(i=0;i<n;i++) { l=*input1++; r=*input2++; *output++ = l; /* left */ *output++ = (l/2)+(r/2); /* center */ *output++ = r; /* right */ *output++ = 0; /* left surround */ *output++ = 0; /* right surroud */ *output++ = 0; /* low freq */ } } ReSampleContext *audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate) { ReSampleContext *s; if ( input_channels > 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; i<s->filter_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;i<s->filter_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; i<s->filter_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); }