X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fresample.c;h=ba8ce8910c082b3ccb7073e5bdc69f08a03735f5;hb=9d25f1f6194dba9cfd60c0596aa59ad145d61382;hp=63371b0a1d8b9b0fc41d2d033be5f6a69e503853;hpb=b9d2085ba14aa733503ff02d966204992f46ff00;p=ffmpeg diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 63371b0a1d8..ba8ce8910c0 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -1,38 +1,60 @@ /* - * 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 Libav. + * + * Libav 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, + * Libav 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 Libav; 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 + * samplerate conversion for both audio and video */ #include "avcodec.h" +#include "audioconvert.h" +#include "libavutil/opt.h" +#include "libavutil/samplefmt.h" + +#define MAX_CHANNELS 8 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, LIBAVUTIL_VERSION_INT +}; + struct ReSampleContext { struct AVResampleContext *resample_context; - short *temp[2]; + short *temp[MAX_CHANNELS]; int temp_len; float ratio; /* channel convert */ int input_channels, output_channels, filter_channels; + AVAudioConvert *convert_ctx[2]; + enum AVSampleFormat 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 */ @@ -86,82 +108,118 @@ static void mono_to_stereo(short *output, short *input, int n1) } } -/* XXX: should use more abstract 'N' channels system */ -static void stereo_split(short *output1, short *output2, short *input, int n) +static void deinterleave(short **output, short *input, int channels, int samples) { - int i; + int i, j; - for(i=0;i 2) - { - av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported."); - return NULL; - } + + if (input_channels > MAX_CHANNELS) { + av_log(NULL, AV_LOG_ERROR, + "Resampling with input channels greater than %d is unsupported.\n", + MAX_CHANNELS); + return NULL; + } + if (output_channels != input_channels && + (input_channels > 2 || + output_channels > 2 && + !(output_channels == 6 && input_channels == 2))) { + av_log(NULL, AV_LOG_ERROR, + "Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n"); + return NULL; + } s = av_mallocz(sizeof(ReSampleContext)); - if (!s) - { - av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context."); - return NULL; - } + 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; -/* - * 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->sample_fmt[0] = sample_fmt_in; + s->sample_fmt[1] = sample_fmt_out; + s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]); + s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]); + + if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) { + if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_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", + av_get_sample_fmt_name(s->sample_fmt[0])); + av_free(s); + return NULL; + } + } + + if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) { + if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1, + AV_SAMPLE_FMT_S16, 1, NULL, 0))) { + av_log(s, AV_LOG_ERROR, + "Cannot convert s16 sample format to %s sample format\n", + av_get_sample_fmt_name(s->sample_fmt[1])); + av_audio_convert_free(s->convert_ctx[0]); + av_free(s); + return NULL; + } + } + + 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; - s->resample_context= av_resample_init(output_rate, input_rate); - return s; } @@ -170,9 +228,10 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels, 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 *bufin[MAX_CHANNELS]; + short *bufout[MAX_CHANNELS]; + short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS]; + short *output_bak = NULL; int lenout; if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { @@ -181,68 +240,131 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl return nb_samples; } + if (s->sample_fmt[0] != AV_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] != AV_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; ifilter_channels; i++){ - bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); + 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; + bufout[i] = av_malloc(lenout * sizeof(short)); } - - /* 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) { + 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); + memcpy(buftmp2[0], input, nb_samples * sizeof(short)); + } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) { + for (i = 0; i < s->input_channels; i++) { + buftmp3[i] = bufout[i]; + } + deinterleave(buftmp2, input, s->input_channels, nb_samples); } else { buftmp3[0] = output; - memcpy(buftmp2[0], input, nb_samples*sizeof(short)); + 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++) { + for (i = 0; i < s->filter_channels; i++) { int consumed; - int is_last= i+1 == s->filter_channels; + 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)); + 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) { + } else if (s->output_channels == 6 && s->input_channels == 2) { ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); + } else if (s->output_channels == s->input_channels && s->input_channels >= 2) { + interleave(output, buftmp3, s->output_channels, nb_samples1); } - for(i=0; ifilter_channels; i++) + if (s->sample_fmt[1] != AV_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[i]); + } - av_free(bufout[0]); - av_free(bufout[1]); return nb_samples1; } void audio_resample_close(ReSampleContext *s) { + int i; av_resample_close(s->resample_context); - av_freep(&s->temp[0]); - av_freep(&s->temp[1]); + for (i = 0; i < s->filter_channels; i++) + av_freep(&s->temp[i]); + 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); }