| 1 | /* |
| 2 | * Copyright (c) 2012 Andrey Utkin |
| 3 | * Copyright (c) 2012 Stefano Sabatini |
| 4 | * |
| 5 | * This file is part of FFmpeg. |
| 6 | * |
| 7 | * FFmpeg is free software; you can redistribute it and/or |
| 8 | * modify it under the terms of the GNU Lesser General Public |
| 9 | * License as published by the Free Software Foundation; either |
| 10 | * version 2.1 of the License, or (at your option) any later version. |
| 11 | * |
| 12 | * FFmpeg is distributed in the hope that it will be useful, |
| 13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 15 | * Lesser General Public License for more details. |
| 16 | * |
| 17 | * You should have received a copy of the GNU Lesser General Public |
| 18 | * License along with FFmpeg; if not, write to the Free Software |
| 19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| 20 | */ |
| 21 | |
| 22 | /** |
| 23 | * @file |
| 24 | * Filter that changes number of samples on single output operation |
| 25 | */ |
| 26 | |
| 27 | #include "libavutil/audio_fifo.h" |
| 28 | #include "libavutil/avassert.h" |
| 29 | #include "libavutil/channel_layout.h" |
| 30 | #include "libavutil/opt.h" |
| 31 | #include "avfilter.h" |
| 32 | #include "audio.h" |
| 33 | #include "internal.h" |
| 34 | #include "formats.h" |
| 35 | |
| 36 | typedef struct { |
| 37 | const AVClass *class; |
| 38 | int nb_out_samples; ///< how many samples to output |
| 39 | AVAudioFifo *fifo; ///< samples are queued here |
| 40 | int64_t next_out_pts; |
| 41 | int pad; |
| 42 | } ASNSContext; |
| 43 | |
| 44 | #define OFFSET(x) offsetof(ASNSContext, x) |
| 45 | #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| 46 | |
| 47 | static const AVOption asetnsamples_options[] = { |
| 48 | { "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS }, |
| 49 | { "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS }, |
| 50 | { "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS }, |
| 51 | { "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS }, |
| 52 | { NULL } |
| 53 | }; |
| 54 | |
| 55 | AVFILTER_DEFINE_CLASS(asetnsamples); |
| 56 | |
| 57 | static av_cold int init(AVFilterContext *ctx) |
| 58 | { |
| 59 | ASNSContext *asns = ctx->priv; |
| 60 | |
| 61 | asns->next_out_pts = AV_NOPTS_VALUE; |
| 62 | av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad); |
| 63 | |
| 64 | return 0; |
| 65 | } |
| 66 | |
| 67 | static av_cold void uninit(AVFilterContext *ctx) |
| 68 | { |
| 69 | ASNSContext *asns = ctx->priv; |
| 70 | av_audio_fifo_free(asns->fifo); |
| 71 | } |
| 72 | |
| 73 | static int config_props_output(AVFilterLink *outlink) |
| 74 | { |
| 75 | ASNSContext *asns = outlink->src->priv; |
| 76 | |
| 77 | asns->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, asns->nb_out_samples); |
| 78 | if (!asns->fifo) |
| 79 | return AVERROR(ENOMEM); |
| 80 | outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP; |
| 81 | |
| 82 | return 0; |
| 83 | } |
| 84 | |
| 85 | static int push_samples(AVFilterLink *outlink) |
| 86 | { |
| 87 | ASNSContext *asns = outlink->src->priv; |
| 88 | AVFrame *outsamples = NULL; |
| 89 | int ret, nb_out_samples, nb_pad_samples; |
| 90 | |
| 91 | if (asns->pad) { |
| 92 | nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0; |
| 93 | nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo)); |
| 94 | } else { |
| 95 | nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo)); |
| 96 | nb_pad_samples = 0; |
| 97 | } |
| 98 | |
| 99 | if (!nb_out_samples) |
| 100 | return 0; |
| 101 | |
| 102 | outsamples = ff_get_audio_buffer(outlink, nb_out_samples); |
| 103 | if (!outsamples) |
| 104 | return AVERROR(ENOMEM); |
| 105 | |
| 106 | av_audio_fifo_read(asns->fifo, |
| 107 | (void **)outsamples->extended_data, nb_out_samples); |
| 108 | |
| 109 | if (nb_pad_samples) |
| 110 | av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples, |
| 111 | nb_pad_samples, outlink->channels, |
| 112 | outlink->format); |
| 113 | outsamples->nb_samples = nb_out_samples; |
| 114 | outsamples->channel_layout = outlink->channel_layout; |
| 115 | outsamples->sample_rate = outlink->sample_rate; |
| 116 | outsamples->pts = asns->next_out_pts; |
| 117 | |
| 118 | if (asns->next_out_pts != AV_NOPTS_VALUE) |
| 119 | asns->next_out_pts += av_rescale_q(nb_out_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); |
| 120 | |
| 121 | ret = ff_filter_frame(outlink, outsamples); |
| 122 | if (ret < 0) |
| 123 | return ret; |
| 124 | return nb_out_samples; |
| 125 | } |
| 126 | |
| 127 | static int filter_frame(AVFilterLink *inlink, AVFrame *insamples) |
| 128 | { |
| 129 | AVFilterContext *ctx = inlink->dst; |
| 130 | ASNSContext *asns = ctx->priv; |
| 131 | AVFilterLink *outlink = ctx->outputs[0]; |
| 132 | int ret; |
| 133 | int nb_samples = insamples->nb_samples; |
| 134 | |
| 135 | if (av_audio_fifo_space(asns->fifo) < nb_samples) { |
| 136 | av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples); |
| 137 | ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples); |
| 138 | if (ret < 0) { |
| 139 | av_log(ctx, AV_LOG_ERROR, |
| 140 | "Stretching audio fifo failed, discarded %d samples\n", nb_samples); |
| 141 | return -1; |
| 142 | } |
| 143 | } |
| 144 | av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples); |
| 145 | if (asns->next_out_pts == AV_NOPTS_VALUE) |
| 146 | asns->next_out_pts = insamples->pts; |
| 147 | av_frame_free(&insamples); |
| 148 | |
| 149 | while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples) |
| 150 | push_samples(outlink); |
| 151 | return 0; |
| 152 | } |
| 153 | |
| 154 | static int request_frame(AVFilterLink *outlink) |
| 155 | { |
| 156 | AVFilterLink *inlink = outlink->src->inputs[0]; |
| 157 | int ret; |
| 158 | |
| 159 | ret = ff_request_frame(inlink); |
| 160 | if (ret == AVERROR_EOF) { |
| 161 | ret = push_samples(outlink); |
| 162 | return ret < 0 ? ret : ret > 0 ? 0 : AVERROR_EOF; |
| 163 | } |
| 164 | |
| 165 | return ret; |
| 166 | } |
| 167 | |
| 168 | static const AVFilterPad asetnsamples_inputs[] = { |
| 169 | { |
| 170 | .name = "default", |
| 171 | .type = AVMEDIA_TYPE_AUDIO, |
| 172 | .filter_frame = filter_frame, |
| 173 | }, |
| 174 | { NULL } |
| 175 | }; |
| 176 | |
| 177 | static const AVFilterPad asetnsamples_outputs[] = { |
| 178 | { |
| 179 | .name = "default", |
| 180 | .type = AVMEDIA_TYPE_AUDIO, |
| 181 | .request_frame = request_frame, |
| 182 | .config_props = config_props_output, |
| 183 | }, |
| 184 | { NULL } |
| 185 | }; |
| 186 | |
| 187 | AVFilter ff_af_asetnsamples = { |
| 188 | .name = "asetnsamples", |
| 189 | .description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."), |
| 190 | .priv_size = sizeof(ASNSContext), |
| 191 | .priv_class = &asetnsamples_class, |
| 192 | .init = init, |
| 193 | .uninit = uninit, |
| 194 | .inputs = asetnsamples_inputs, |
| 195 | .outputs = asetnsamples_outputs, |
| 196 | }; |