X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fresample.c;h=e1d29f7e9289ef655ead8dd067e2cdf141d7a9cd;hb=5bebe94d860cb1a0819701213e961c8269aa3284;hp=86bed847c4b6ec78a7a9d4b43467deff2fac1609;hpb=4a899dd689defec022e8566465c873dd83b55cef;p=ffmpeg diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 86bed847c4b..e1d29f7e928 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -1,126 +1,56 @@ /* - * Sample rate convertion for both audio and video - * Copyright (c) 2000 Fabrice Bellard. + * samplerate conversion for both audio and video + * Copyright (c) 2000 Fabrice Bellard * - * This library is free software; you can redistribute it and/or + * 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 of the License, or (at your option) any later version. + * version 2.1 of the License, or (at your option) any later version. * - * This library is distributed in the hope that it will be useful, + * 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 this library; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * 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. + * @file libavcodec/resample.c + * samplerate conversion for both audio and video */ #include "avcodec.h" -#include "os_support.h" - -typedef struct { - /* fractional resampling */ - uint32_t incr; /* fractional increment */ - uint32_t frac; - int last_sample; - /* integer down sample */ - int iratio; /* integer divison ratio */ - int icount, isum; - int inv; -} ReSampleChannelContext; - -struct ReSampleContext { - ReSampleChannelContext channel_ctx[2]; - float ratio; - /* channel convert */ - int input_channels, output_channels, filter_channels; -}; - +#include "audioconvert.h" +#include "opt.h" -#define FRAC_BITS 16 -#define FRAC (1 << FRAC_BITS) - -static void init_mono_resample(ReSampleChannelContext *s, float ratio) -{ - ratio = 1.0 / ratio; - s->iratio = (int)floorf(ratio); - if (s->iratio == 0) - s->iratio = 1; - s->incr = (int)((ratio / s->iratio) * FRAC); - s->frac = FRAC; - s->last_sample = 0; - s->icount = s->iratio; - s->isum = 0; - s->inv = (FRAC / s->iratio); -} +struct AVResampleContext; -/* fractional audio resampling */ -static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) +static const char *context_to_name(void *ptr) { - unsigned int frac, incr; - int l0, l1; - short *q, *p, *pend; - - l0 = s->last_sample; - incr = s->incr; - frac = s->frac; - - p = input; - pend = input + nb_samples; - q = output; - - l1 = *p++; - for(;;) { - /* interpolate */ - *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS; - frac = frac + s->incr; - while (frac >= FRAC) { - frac -= FRAC; - if (p >= pend) - goto the_end; - l0 = l1; - l1 = *p++; - } - } - the_end: - s->last_sample = l1; - s->frac = frac; - return q - output; + return "audioresample"; } -static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) -{ - short *q, *p, *pend; - int c, sum; +static const AVOption options[] = {{NULL}}; +static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options }; - p = input; - pend = input + nb_samples; - q = output; - - c = s->icount; - sum = s->isum; - - for(;;) { - sum += *p++; - if (--c == 0) { - *q++ = (sum * s->inv) >> FRAC_BITS; - c = s->iratio; - sum = 0; - } - if (p >= pend) - break; - } - s->isum = sum; - s->icount = c; - return q - output; -} +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) @@ -194,119 +124,237 @@ static void stereo_mux(short *output, short *input1, short *input2, int n) } } -static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) +static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) { - short *buf1; - short *buftmp; - - buf1= (short*)av_malloc( nb_samples * sizeof(short) ); - - /* first downsample by an integer factor with averaging filter */ - if (s->iratio > 1) { - buftmp = buf1; - nb_samples = integer_downsample(s, buftmp, input, nb_samples); - } else { - buftmp = input; - } + int i; + short l,r; - /* then do a fractional resampling with linear interpolation */ - if (s->incr != FRAC) { - nb_samples = fractional_resample(s, output, buftmp, nb_samples); - } else { - memcpy(output, buftmp, nb_samples * sizeof(short)); + for(i=0;i 2 || input_channels > 2) + + 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; - for(i=0;ifilter_channels;i++) { - init_mono_resample(&s->channel_ctx[i], s->ratio); + 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 ! */ -/* XXX: do it with polyphase filters, since the quality here is - HORRIBLE. Return the number of samples available in output */ 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) { + 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 */ - bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) ); - bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) ); - + for(i=0; ifilter_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 */ - lenout= (int)(nb_samples * s->ratio) + 16; - bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); - bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); + bufout[0]= av_malloc( lenout * sizeof(short) ); + bufout[1]= av_malloc( lenout * sizeof(short) ); if (s->input_channels == 2 && s->output_channels == 1) { - buftmp2[0] = bufin[0]; buftmp3[0] = output; stereo_to_mono(buftmp2[0], input, nb_samples); - } else if (s->output_channels == 2 && s->input_channels == 1) { - buftmp2[0] = input; + } else if (s->output_channels >= 2 && s->input_channels == 1) { buftmp3[0] = bufout[0]; - } else if (s->output_channels == 2) { - buftmp2[0] = bufin[0]; - buftmp2[1] = bufin[1]; + 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 { - buftmp2[0] = input; 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++) { - nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); + 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; + } } - av_free(bufin[0]); - av_free(bufin[1]); + for(i=0; ifilter_channels; i++) + av_free(bufin[i]); av_free(bufout[0]); av_free(bufout[1]); @@ -315,5 +363,12 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl 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); }