Mercurial > libavcodec.hg
view resample.c @ 10467:212a837ebd27 libavcodec
Split synth filter out of dca.c.
author | michael |
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date | Wed, 28 Oct 2009 10:51:51 +0000 |
parents | ff9ae6249d18 |
children | 7dd2a45249a9 |
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/* * samplerate conversion 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 libavcodec/resample.c * samplerate conversion for both audio and video */ #include "avcodec.h" #include "audioconvert.h" #include "opt.h" struct AVResampleContext; static const char *context_to_name(void *ptr) { return "audioresample"; } static const AVOption options[] = {{NULL}}; static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options }; struct ReSampleContext { struct AVResampleContext *resample_context; short *temp[2]; int temp_len; float ratio; /* channel convert */ int input_channels, output_channels, filter_channels; AVAudioConvert *convert_ctx[2]; enum SampleFormat sample_fmt[2]; ///< input and output sample format unsigned sample_size[2]; ///< size of one sample in sample_fmt short *buffer[2]; ///< buffers used for conversion to S16 unsigned buffer_size[2]; ///< sizes of allocated buffers }; /* 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 *av_audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate, enum SampleFormat sample_fmt_out, enum SampleFormat sample_fmt_in, int filter_length, int log2_phase_count, int linear, double cutoff) { ReSampleContext *s; if ( input_channels > 2) { av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n"); return NULL; } s = av_mallocz(sizeof(ReSampleContext)); if (!s) { av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); 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; s->sample_fmt [0] = sample_fmt_in; s->sample_fmt [1] = sample_fmt_out; s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3; s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3; if (s->sample_fmt[0] != SAMPLE_FMT_S16) { if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1, s->sample_fmt[0], 1, NULL, 0))) { av_log(s, AV_LOG_ERROR, "Cannot convert %s sample format to s16 sample format\n", avcodec_get_sample_fmt_name(s->sample_fmt[0])); av_free(s); return NULL; } } if (s->sample_fmt[1] != SAMPLE_FMT_S16) { if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1, SAMPLE_FMT_S16, 1, NULL, 0))) { av_log(s, AV_LOG_ERROR, "Cannot convert s16 sample format to %s sample format\n", avcodec_get_sample_fmt_name(s->sample_fmt[1])); av_audio_convert_free(s->convert_ctx[0]); av_free(s); return NULL; } } /* * AC-3 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; #define TAPS 16 s->resample_context= av_resample_init(output_rate, input_rate, filter_length, log2_phase_count, linear, cutoff); *(const AVClass**)s->resample_context = &audioresample_context_class; return s; } #if LIBAVCODEC_VERSION_MAJOR < 53 ReSampleContext *audio_resample_init(int output_channels, int input_channels, int output_rate, int input_rate) { return av_audio_resample_init(output_channels, input_channels, output_rate, input_rate, SAMPLE_FMT_S16, SAMPLE_FMT_S16, TAPS, 10, 0, 0.8); } #endif /* 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]; short *output_bak = NULL; 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; } if (s->sample_fmt[0] != SAMPLE_FMT_S16) { int istride[1] = { s->sample_size[0] }; int ostride[1] = { 2 }; const void *ibuf[1] = { input }; void *obuf[1]; unsigned input_size = nb_samples*s->input_channels*2; if (!s->buffer_size[0] || s->buffer_size[0] < input_size) { av_free(s->buffer[0]); s->buffer_size[0] = input_size; s->buffer[0] = av_malloc(s->buffer_size[0]); if (!s->buffer[0]) { av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); return 0; } } obuf[0] = s->buffer[0]; if (av_audio_convert(s->convert_ctx[0], obuf, ostride, ibuf, istride, nb_samples*s->input_channels) < 0) { av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n"); return 0; } input = s->buffer[0]; } lenout= 4*nb_samples * s->ratio + 16; if (s->sample_fmt[1] != SAMPLE_FMT_S16) { output_bak = output; if (!s->buffer_size[1] || s->buffer_size[1] < lenout) { av_free(s->buffer[1]); s->buffer_size[1] = lenout; s->buffer[1] = av_malloc(s->buffer_size[1]); if (!s->buffer[1]) { av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n"); return 0; } } output = s->buffer[1]; } /* XXX: move those malloc to resample init code */ for(i=0; i<s->filter_channels; i++){ bufin[i]= 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 */ bufout[0]= av_malloc( lenout * sizeof(short) ); bufout[1]= 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); } if (s->sample_fmt[1] != SAMPLE_FMT_S16) { int istride[1] = { 2 }; int ostride[1] = { s->sample_size[1] }; const void *ibuf[1] = { output }; void *obuf[1] = { output_bak }; if (av_audio_convert(s->convert_ctx[1], obuf, ostride, ibuf, istride, nb_samples1*s->output_channels) < 0) { av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n"); return 0; } } 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_freep(&s->buffer[0]); av_freep(&s->buffer[1]); av_audio_convert_free(s->convert_ctx[0]); av_audio_convert_free(s->convert_ctx[1]); av_free(s); }