af_resample.c 8.46 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33
/*
 *
 * 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 2.1 of the License, or (at your option) any later version.
 *
 * 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 Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file
 * sample format and channel layout conversion audio filter
 */

#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"

#include "libavresample/avresample.h"

#include "audio.h"
#include "avfilter.h"
34
#include "formats.h"
35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59
#include "internal.h"

typedef struct ResampleContext {
    AVAudioResampleContext *avr;

    int64_t next_pts;
} ResampleContext;

static av_cold void uninit(AVFilterContext *ctx)
{
    ResampleContext *s = ctx->priv;

    if (s->avr) {
        avresample_close(s->avr);
        avresample_free(&s->avr);
    }
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterLink *inlink  = ctx->inputs[0];
    AVFilterLink *outlink = ctx->outputs[0];

    AVFilterFormats        *in_formats      = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
    AVFilterFormats        *out_formats     = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
60 61 62 63
    AVFilterFormats        *in_samplerates  = ff_all_samplerates();
    AVFilterFormats        *out_samplerates = ff_all_samplerates();
    AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
    AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();
64 65 66 67

    avfilter_formats_ref(in_formats,  &inlink->out_formats);
    avfilter_formats_ref(out_formats, &outlink->in_formats);

68 69 70 71 72 73
    avfilter_formats_ref(in_samplerates,  &inlink->out_samplerates);
    avfilter_formats_ref(out_samplerates, &outlink->in_samplerates);

    ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
    ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121
    return 0;
}

static int config_output(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    AVFilterLink *inlink = ctx->inputs[0];
    ResampleContext   *s = ctx->priv;
    char buf1[64], buf2[64];
    int ret;

    if (s->avr) {
        avresample_close(s->avr);
        avresample_free(&s->avr);
    }

    if (inlink->channel_layout == outlink->channel_layout &&
        inlink->sample_rate    == outlink->sample_rate    &&
        inlink->format         == outlink->format)
        return 0;

    if (!(s->avr = avresample_alloc_context()))
        return AVERROR(ENOMEM);

    av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
    av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
    av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
    av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
    av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
    av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);

    /* if both the input and output formats are s16 or u8, use s16 as
       the internal sample format */
    if (av_get_bytes_per_sample(inlink->format)  <= 2 &&
        av_get_bytes_per_sample(outlink->format) <= 2)
        av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);

    if ((ret = avresample_open(s->avr)) < 0)
        return ret;

    outlink->time_base = (AVRational){ 1, outlink->sample_rate };
    s->next_pts        = AV_NOPTS_VALUE;

    av_get_channel_layout_string(buf1, sizeof(buf1),
                                 -1, inlink ->channel_layout);
    av_get_channel_layout_string(buf2, sizeof(buf2),
                                 -1, outlink->channel_layout);
    av_log(ctx, AV_LOG_VERBOSE,
122
           "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236
           av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);

    return 0;
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    ResampleContext   *s = ctx->priv;
    int ret = avfilter_request_frame(ctx->inputs[0]);

    /* flush the lavr delay buffer */
    if (ret == AVERROR_EOF && s->avr) {
        AVFilterBufferRef *buf;
        int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
                                        outlink->sample_rate,
                                        ctx->inputs[0]->sample_rate,
                                        AV_ROUND_UP);

        if (!nb_samples)
            return ret;

        buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
        if (!buf)
            return AVERROR(ENOMEM);

        ret = avresample_convert(s->avr, (void**)buf->extended_data,
                                 buf->linesize[0], nb_samples,
                                 NULL, 0, 0);
        if (ret <= 0) {
            avfilter_unref_buffer(buf);
            return (ret == 0) ? AVERROR_EOF : ret;
        }

        buf->pts = s->next_pts;
        ff_filter_samples(outlink, buf);
        return 0;
    }
    return ret;
}

static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
    AVFilterContext  *ctx = inlink->dst;
    ResampleContext    *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];

    if (s->avr) {
        AVFilterBufferRef *buf_out;
        int delay, nb_samples, ret;

        /* maximum possible samples lavr can output */
        delay      = avresample_get_delay(s->avr);
        nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
                                    outlink->sample_rate, inlink->sample_rate,
                                    AV_ROUND_UP);

        buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
        ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
                                     buf_out->linesize[0], nb_samples,
                                     (void**)buf->extended_data, buf->linesize[0],
                                     buf->audio->nb_samples);

        av_assert0(!avresample_available(s->avr));

        if (s->next_pts == AV_NOPTS_VALUE) {
            if (buf->pts == AV_NOPTS_VALUE) {
                av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
                       "assuming 0.\n");
                s->next_pts = 0;
            } else
                s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
                                           outlink->time_base);
        }

        if (ret > 0) {
            buf_out->audio->nb_samples = ret;
            if (buf->pts != AV_NOPTS_VALUE) {
                buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
                                            outlink->time_base) -
                               av_rescale(delay, outlink->sample_rate,
                                          inlink->sample_rate);
            } else
                buf_out->pts = s->next_pts;

            s->next_pts = buf_out->pts + buf_out->audio->nb_samples;

            ff_filter_samples(outlink, buf_out);
        }
        avfilter_unref_buffer(buf);
    } else
        ff_filter_samples(outlink, buf);
}

AVFilter avfilter_af_resample = {
    .name          = "resample",
    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
    .priv_size     = sizeof(ResampleContext),

    .uninit         = uninit,
    .query_formats  = query_formats,

    .inputs    = (const AVFilterPad[]) {{ .name            = "default",
                                          .type            = AVMEDIA_TYPE_AUDIO,
                                          .filter_samples  = filter_samples,
                                          .min_perms       = AV_PERM_READ },
                                        { .name = NULL}},
    .outputs   = (const AVFilterPad[]) {{ .name          = "default",
                                          .type          = AVMEDIA_TYPE_AUDIO,
                                          .config_props  = config_output,
                                          .request_frame = request_frame },
                                        { .name = NULL}},
};