2 * Copyright (c) Stefano Sabatini | stefasab at gmail.com
3 * Copyright (c) S.N. Hemanth Meenakshisundaram | smeenaks at ucsd.edu
5 * This file is part of FFmpeg.
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.
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.
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
22 #include "libavutil/avassert.h"
23 #include "libavutil/audioconvert.h"
24 #include "libavutil/common.h"
30 AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
33 return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
36 AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
39 AVFilterBufferRef *samplesref = NULL;
41 int planar = av_sample_fmt_is_planar(link->format);
42 int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
43 int planes = planar ? nb_channels : 1;
46 if (!(data = av_mallocz(sizeof(*data) * planes)))
49 if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
52 samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, perms,
53 nb_samples, link->format,
54 link->channel_layout);
67 AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
70 AVFilterBufferRef *ret = NULL;
72 if (link->dstpad->get_audio_buffer)
73 ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
76 ret = ff_default_get_audio_buffer(link, perms, nb_samples);
79 ret->type = AVMEDIA_TYPE_AUDIO;
84 AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
85 int linesize,int perms,
87 enum AVSampleFormat sample_fmt,
88 uint64_t channel_layout)
91 AVFilterBuffer *samples = av_mallocz(sizeof(*samples));
92 AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
94 if (!samples || !samplesref)
97 samplesref->buf = samples;
98 samplesref->buf->free = ff_avfilter_default_free_buffer;
99 if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
102 samplesref->audio->nb_samples = nb_samples;
103 samplesref->audio->channel_layout = channel_layout;
105 planes = av_sample_fmt_is_planar(sample_fmt) ?
106 av_get_channel_layout_nb_channels(channel_layout) : 1;
108 /* make sure the buffer gets read permission or it's useless for output */
109 samplesref->perms = perms | AV_PERM_READ;
111 samples->refcount = 1;
112 samplesref->type = AVMEDIA_TYPE_AUDIO;
113 samplesref->format = sample_fmt;
115 memcpy(samples->data, data,
116 FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
117 memcpy(samplesref->data, samples->data, sizeof(samples->data));
119 samples->linesize[0] = samplesref->linesize[0] = linesize;
121 if (planes > FF_ARRAY_ELEMS(samples->data)) {
122 samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) *
124 samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
127 if (!samples->extended_data || !samplesref->extended_data)
130 memcpy(samples-> extended_data, data, sizeof(*data)*planes);
131 memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
133 samples->extended_data = samples->data;
134 samplesref->extended_data = samplesref->data;
137 samplesref->pts = AV_NOPTS_VALUE;
142 if (samples && samples->extended_data != samples->data)
143 av_freep(&samples->extended_data);
145 av_freep(&samplesref->audio);
146 if (samplesref->extended_data != samplesref->data)
147 av_freep(&samplesref->extended_data);
149 av_freep(&samplesref);
154 static int default_filter_samples(AVFilterLink *link,
155 AVFilterBufferRef *samplesref)
157 return ff_filter_samples(link->dst->outputs[0], samplesref);
160 int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
162 int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
163 AVFilterPad *src = link->srcpad;
164 AVFilterPad *dst = link->dstpad;
166 AVFilterBufferRef *buf_out;
169 FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
171 if (!(filter_samples = dst->filter_samples))
172 filter_samples = default_filter_samples;
174 av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
175 samplesref->perms &= ~ src->rej_perms;
177 /* prepare to copy the samples if the buffer has insufficient permissions */
178 if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
179 dst->rej_perms & samplesref->perms) {
180 av_log(link->dst, AV_LOG_DEBUG,
181 "Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
182 samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
184 buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
185 samplesref->audio->nb_samples);
187 avfilter_unref_buffer(samplesref);
188 return AVERROR(ENOMEM);
190 buf_out->pts = samplesref->pts;
191 buf_out->audio->sample_rate = samplesref->audio->sample_rate;
193 /* Copy actual data into new samples buffer */
194 av_samples_copy(buf_out->extended_data, samplesref->extended_data,
195 0, 0, samplesref->audio->nb_samples,
196 av_get_channel_layout_nb_channels(link->channel_layout),
199 avfilter_unref_buffer(samplesref);
201 buf_out = samplesref;
203 link->cur_buf = buf_out;
205 ret = filter_samples(link, buf_out);
206 ff_update_link_current_pts(link, pts);
210 int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
212 int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
213 AVFilterBufferRef *pbuf = link->partial_buf;
214 int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
217 if (!link->min_samples ||
219 insamples >= link->min_samples && insamples <= link->max_samples)) {
220 return ff_filter_samples_framed(link, samplesref);
222 /* Handle framing (min_samples, max_samples) */
225 AVRational samples_tb = { 1, link->sample_rate };
226 int perms = link->dstpad->min_perms | AV_PERM_WRITE;
227 pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size);
229 av_log(link->dst, AV_LOG_WARNING,
230 "Samples dropped due to memory allocation failure.\n");
233 avfilter_copy_buffer_ref_props(pbuf, samplesref);
234 pbuf->pts = samplesref->pts +
235 av_rescale_q(inpos, samples_tb, link->time_base);
236 pbuf->audio->nb_samples = 0;
238 nb_samples = FFMIN(insamples,
239 link->partial_buf_size - pbuf->audio->nb_samples);
240 av_samples_copy(pbuf->extended_data, samplesref->extended_data,
241 pbuf->audio->nb_samples, inpos,
242 nb_samples, nb_channels, link->format);
244 insamples -= nb_samples;
245 pbuf->audio->nb_samples += nb_samples;
246 if (pbuf->audio->nb_samples >= link->min_samples) {
247 ret = ff_filter_samples_framed(link, pbuf);
251 avfilter_unref_buffer(samplesref);
252 link->partial_buf = pbuf;