Merge remote-tracking branch 'qatar/master'
[ffmpeg.git] / libavfilter / af_resample.c
1 /*
2  *
3  * This file is part of Libav.
4  *
5  * Libav is free software; you can redistribute it and/or
6  * modify it under the terms of the GNU Lesser General Public
7  * License as published by the Free Software Foundation; either
8  * version 2.1 of the License, or (at your option) any later version.
9  *
10  * Libav is distributed in the hope that it will be useful,
11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
13  * Lesser General Public License for more details.
14  *
15  * You should have received a copy of the GNU Lesser General Public
16  * License along with Libav; if not, write to the Free Software
17  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
18  */
19
20 /**
21  * @file
22  * sample format and channel layout conversion audio filter
23  */
24
25 #include "libavutil/avassert.h"
26 #include "libavutil/avstring.h"
27 #include "libavutil/mathematics.h"
28 #include "libavutil/opt.h"
29
30 #include "libavresample/avresample.h"
31
32 #include "audio.h"
33 #include "avfilter.h"
34 #include "internal.h"
35
36 typedef struct ResampleContext {
37     AVAudioResampleContext *avr;
38
39     int64_t next_pts;
40 } ResampleContext;
41
42 static av_cold void uninit(AVFilterContext *ctx)
43 {
44     ResampleContext *s = ctx->priv;
45
46     if (s->avr) {
47         avresample_close(s->avr);
48         avresample_free(&s->avr);
49     }
50 }
51
52 static int query_formats(AVFilterContext *ctx)
53 {
54     AVFilterLink *inlink  = ctx->inputs[0];
55     AVFilterLink *outlink = ctx->outputs[0];
56
57     AVFilterFormats        *in_formats      = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
58     AVFilterFormats        *out_formats     = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
59
60     avfilter_formats_ref(in_formats,  &inlink->out_formats);
61     avfilter_formats_ref(out_formats, &outlink->in_formats);
62
63     return 0;
64 }
65
66 static int config_output(AVFilterLink *outlink)
67 {
68     AVFilterContext *ctx = outlink->src;
69     AVFilterLink *inlink = ctx->inputs[0];
70     ResampleContext   *s = ctx->priv;
71     char buf1[64], buf2[64];
72     int ret;
73
74     if (s->avr) {
75         avresample_close(s->avr);
76         avresample_free(&s->avr);
77     }
78
79     if (inlink->channel_layout == outlink->channel_layout &&
80         inlink->sample_rate    == outlink->sample_rate    &&
81         inlink->format         == outlink->format)
82         return 0;
83
84     if (!(s->avr = avresample_alloc_context()))
85         return AVERROR(ENOMEM);
86
87     av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
88     av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
89     av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
90     av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
91     av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
92     av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
93
94     /* if both the input and output formats are s16 or u8, use s16 as
95        the internal sample format */
96     if (av_get_bytes_per_sample(inlink->format)  <= 2 &&
97         av_get_bytes_per_sample(outlink->format) <= 2)
98         av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
99
100     if ((ret = avresample_open(s->avr)) < 0)
101         return ret;
102
103     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
104     s->next_pts        = AV_NOPTS_VALUE;
105
106     av_get_channel_layout_string(buf1, sizeof(buf1),
107                                  -1, inlink ->channel_layout);
108     av_get_channel_layout_string(buf2, sizeof(buf2),
109                                  -1, outlink->channel_layout);
110     av_log(ctx, AV_LOG_VERBOSE,
111            "fmt:%s srate: %d cl:%s -> fmt:%s srate: %d cl:%s\n",
112            av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
113            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
114
115     return 0;
116 }
117
118 static int request_frame(AVFilterLink *outlink)
119 {
120     AVFilterContext *ctx = outlink->src;
121     ResampleContext   *s = ctx->priv;
122     int ret = avfilter_request_frame(ctx->inputs[0]);
123
124     /* flush the lavr delay buffer */
125     if (ret == AVERROR_EOF && s->avr) {
126         AVFilterBufferRef *buf;
127         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
128                                         outlink->sample_rate,
129                                         ctx->inputs[0]->sample_rate,
130                                         AV_ROUND_UP);
131
132         if (!nb_samples)
133             return ret;
134
135         buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
136         if (!buf)
137             return AVERROR(ENOMEM);
138
139         ret = avresample_convert(s->avr, (void**)buf->extended_data,
140                                  buf->linesize[0], nb_samples,
141                                  NULL, 0, 0);
142         if (ret <= 0) {
143             avfilter_unref_buffer(buf);
144             return (ret == 0) ? AVERROR_EOF : ret;
145         }
146
147         buf->pts = s->next_pts;
148         ff_filter_samples(outlink, buf);
149         return 0;
150     }
151     return ret;
152 }
153
154 static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
155 {
156     AVFilterContext  *ctx = inlink->dst;
157     ResampleContext    *s = ctx->priv;
158     AVFilterLink *outlink = ctx->outputs[0];
159
160     if (s->avr) {
161         AVFilterBufferRef *buf_out;
162         int delay, nb_samples, ret;
163
164         /* maximum possible samples lavr can output */
165         delay      = avresample_get_delay(s->avr);
166         nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
167                                     outlink->sample_rate, inlink->sample_rate,
168                                     AV_ROUND_UP);
169
170         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
171         ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
172                                      buf_out->linesize[0], nb_samples,
173                                      (void**)buf->extended_data, buf->linesize[0],
174                                      buf->audio->nb_samples);
175
176         av_assert0(!avresample_available(s->avr));
177
178         if (s->next_pts == AV_NOPTS_VALUE) {
179             if (buf->pts == AV_NOPTS_VALUE) {
180                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
181                        "assuming 0.\n");
182                 s->next_pts = 0;
183             } else
184                 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
185                                            outlink->time_base);
186         }
187
188         if (ret > 0) {
189             buf_out->audio->nb_samples = ret;
190             if (buf->pts != AV_NOPTS_VALUE) {
191                 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
192                                             outlink->time_base) -
193                                av_rescale(delay, outlink->sample_rate,
194                                           inlink->sample_rate);
195             } else
196                 buf_out->pts = s->next_pts;
197
198             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
199
200             ff_filter_samples(outlink, buf_out);
201         }
202         avfilter_unref_buffer(buf);
203     } else
204         ff_filter_samples(outlink, buf);
205 }
206
207 AVFilter avfilter_af_resample = {
208     .name          = "resample",
209     .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
210     .priv_size     = sizeof(ResampleContext),
211
212     .uninit         = uninit,
213     .query_formats  = query_formats,
214
215     .inputs    = (const AVFilterPad[]) {{ .name            = "default",
216                                           .type            = AVMEDIA_TYPE_AUDIO,
217                                           .filter_samples  = filter_samples,
218                                           .min_perms       = AV_PERM_READ },
219                                         { .name = NULL}},
220     .outputs   = (const AVFilterPad[]) {{ .name          = "default",
221                                           .type          = AVMEDIA_TYPE_AUDIO,
222                                           .config_props  = config_output,
223                                           .request_frame = request_frame },
224                                         { .name = NULL}},
225 };