lavfi: add avfilter_get_audio_buffer_ref_from_arrays_channels.
[ffmpeg.git] / libavfilter / audio.c
1 /*
2  * Copyright (c) Stefano Sabatini | stefasab at gmail.com
3  * Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu
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 #include "libavutil/avassert.h"
23 #include "libavutil/channel_layout.h"
24 #include "libavutil/common.h"
25
26 #include "audio.h"
27 #include "avfilter.h"
28 #include "internal.h"
29
30 int avfilter_ref_get_channels(AVFilterBufferRef *ref)
31 {
32     return ref->audio ? ref->audio->channels : 0;
33 }
34
35 AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
36                                             int nb_samples)
37 {
38     return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
39 }
40
41 AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
42                                                int nb_samples)
43 {
44     AVFilterBufferRef *samplesref = NULL;
45     uint8_t **data;
46     int planar      = av_sample_fmt_is_planar(link->format);
47     int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
48     int planes      = planar ? nb_channels : 1;
49     int linesize;
50     int full_perms = AV_PERM_READ | AV_PERM_WRITE | AV_PERM_PRESERVE |
51                      AV_PERM_REUSE | AV_PERM_REUSE2 | AV_PERM_ALIGN;
52
53     av_assert1(!(perms & ~(full_perms | AV_PERM_NEG_LINESIZES)));
54
55     if (!(data = av_mallocz(sizeof(*data) * planes)))
56         goto fail;
57
58     if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
59         goto fail;
60
61     samplesref = avfilter_get_audio_buffer_ref_from_arrays_channels(
62         data, linesize, full_perms, nb_samples, link->format,
63         link->channels, link->channel_layout);
64     if (!samplesref)
65         goto fail;
66
67     samplesref->audio->sample_rate = link->sample_rate;
68
69     av_freep(&data);
70
71 fail:
72     if (data)
73         av_freep(&data[0]);
74     av_freep(&data);
75     return samplesref;
76 }
77
78 AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
79                                        int nb_samples)
80 {
81     AVFilterBufferRef *ret = NULL;
82
83     if (link->dstpad->get_audio_buffer)
84         ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
85
86     if (!ret)
87         ret = ff_default_get_audio_buffer(link, perms, nb_samples);
88
89     if (ret)
90         ret->type = AVMEDIA_TYPE_AUDIO;
91
92     return ret;
93 }
94
95 AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_channels(uint8_t **data,
96                                                                       int linesize,
97                                                                       int perms,
98                                                                       int nb_samples,
99                                                                       enum AVSampleFormat sample_fmt,
100                                                                       int channels,
101                                                                       uint64_t channel_layout)
102 {
103     int planes;
104     AVFilterBuffer    *samples    = av_mallocz(sizeof(*samples));
105     AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
106
107     if (!samples || !samplesref)
108         goto fail;
109
110     av_assert0(channels);
111     av_assert0(channel_layout == 0 ||
112                channels == av_get_channel_layout_nb_channels(channel_layout));
113
114     samplesref->buf         = samples;
115     samplesref->buf->free   = ff_avfilter_default_free_buffer;
116     if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
117         goto fail;
118
119     samplesref->audio->nb_samples     = nb_samples;
120     samplesref->audio->channel_layout = channel_layout;
121     samplesref->audio->channels       = channels;
122
123     planes = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
124
125     /* make sure the buffer gets read permission or it's useless for output */
126     samplesref->perms = perms | AV_PERM_READ;
127
128     samples->refcount  = 1;
129     samplesref->type   = AVMEDIA_TYPE_AUDIO;
130     samplesref->format = sample_fmt;
131
132     memcpy(samples->data, data,
133            FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
134     memcpy(samplesref->data, samples->data, sizeof(samples->data));
135
136     samples->linesize[0] = samplesref->linesize[0] = linesize;
137
138     if (planes > FF_ARRAY_ELEMS(samples->data)) {
139         samples->   extended_data = av_mallocz(sizeof(*samples->extended_data) *
140                                                planes);
141         samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
142                                                planes);
143
144         if (!samples->extended_data || !samplesref->extended_data)
145             goto fail;
146
147         memcpy(samples->   extended_data, data, sizeof(*data)*planes);
148         memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
149     } else {
150         samples->extended_data    = samples->data;
151         samplesref->extended_data = samplesref->data;
152     }
153
154     samplesref->pts = AV_NOPTS_VALUE;
155
156     return samplesref;
157
158 fail:
159     if (samples && samples->extended_data != samples->data)
160         av_freep(&samples->extended_data);
161     if (samplesref) {
162         av_freep(&samplesref->audio);
163         if (samplesref->extended_data != samplesref->data)
164             av_freep(&samplesref->extended_data);
165     }
166     av_freep(&samplesref);
167     av_freep(&samples);
168     return NULL;
169 }
170
171 AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
172                                                              int linesize,int perms,
173                                                              int nb_samples,
174                                                              enum AVSampleFormat sample_fmt,
175                                                              uint64_t channel_layout)
176 {
177     int channels = av_get_channel_layout_nb_channels(channel_layout);
178     return avfilter_get_audio_buffer_ref_from_arrays_channels(data, linesize, perms,
179                                                               nb_samples, sample_fmt,
180                                                               channels, channel_layout);
181 }
182
183 static int default_filter_frame(AVFilterLink *link, AVFilterBufferRef *frame)
184 {
185     return ff_filter_frame(link->dst->outputs[0], frame);
186 }
187
188 int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
189 {
190     int (*filter_frame)(AVFilterLink *, AVFilterBufferRef *);
191     AVFilterPad *src = link->srcpad;
192     AVFilterPad *dst = link->dstpad;
193     int64_t pts;
194     AVFilterBufferRef *buf_out;
195     int ret;
196
197     FF_TPRINTF_START(NULL, filter_frame); ff_tlog_link(NULL, link, 1);
198
199     if (link->closed) {
200         avfilter_unref_buffer(samplesref);
201         return AVERROR_EOF;
202     }
203
204     if (!(filter_frame = dst->filter_frame))
205         filter_frame = default_filter_frame;
206
207     av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
208     samplesref->perms &= ~ src->rej_perms;
209
210     /* prepare to copy the samples if the buffer has insufficient permissions */
211     if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
212         dst->rej_perms & samplesref->perms) {
213         av_log(link->dst, AV_LOG_DEBUG,
214                "Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
215                samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
216
217         buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
218                                               samplesref->audio->nb_samples);
219         if (!buf_out) {
220             avfilter_unref_buffer(samplesref);
221             return AVERROR(ENOMEM);
222         }
223         buf_out->pts                = samplesref->pts;
224         buf_out->audio->sample_rate = samplesref->audio->sample_rate;
225
226         /* Copy actual data into new samples buffer */
227         av_samples_copy(buf_out->extended_data, samplesref->extended_data,
228                         0, 0, samplesref->audio->nb_samples,
229                         av_get_channel_layout_nb_channels(link->channel_layout),
230                         link->format);
231
232         avfilter_unref_buffer(samplesref);
233     } else
234         buf_out = samplesref;
235
236     link->cur_buf = buf_out;
237     pts = buf_out->pts;
238     ret = filter_frame(link, buf_out);
239     ff_update_link_current_pts(link, pts);
240     return ret;
241 }
242
243 int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
244 {
245     int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
246     AVFilterBufferRef *pbuf = link->partial_buf;
247     int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
248     int ret = 0;
249
250     av_assert1(samplesref->format                == link->format);
251     av_assert1(samplesref->audio->channels       == link->channels);
252     av_assert1(samplesref->audio->channel_layout == link->channel_layout);
253     av_assert1(samplesref->audio->sample_rate    == link->sample_rate);
254
255     if (!link->min_samples ||
256         (!pbuf &&
257          insamples >= link->min_samples && insamples <= link->max_samples)) {
258         return ff_filter_samples_framed(link, samplesref);
259     }
260     /* Handle framing (min_samples, max_samples) */
261     while (insamples) {
262         if (!pbuf) {
263             AVRational samples_tb = { 1, link->sample_rate };
264             int perms = link->dstpad->min_perms | AV_PERM_WRITE;
265             pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size);
266             if (!pbuf) {
267                 av_log(link->dst, AV_LOG_WARNING,
268                        "Samples dropped due to memory allocation failure.\n");
269                 return 0;
270             }
271             avfilter_copy_buffer_ref_props(pbuf, samplesref);
272             pbuf->pts = samplesref->pts +
273                         av_rescale_q(inpos, samples_tb, link->time_base);
274             pbuf->audio->nb_samples = 0;
275         }
276         nb_samples = FFMIN(insamples,
277                            link->partial_buf_size - pbuf->audio->nb_samples);
278         av_samples_copy(pbuf->extended_data, samplesref->extended_data,
279                         pbuf->audio->nb_samples, inpos,
280                         nb_samples, nb_channels, link->format);
281         inpos                   += nb_samples;
282         insamples               -= nb_samples;
283         pbuf->audio->nb_samples += nb_samples;
284         if (pbuf->audio->nb_samples >= link->min_samples) {
285             ret = ff_filter_samples_framed(link, pbuf);
286             pbuf = NULL;
287         }
288     }
289     avfilter_unref_buffer(samplesref);
290     link->partial_buf = pbuf;
291     return ret;
292 }