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;
45 int full_perms = AV_PERM_READ | AV_PERM_WRITE | AV_PERM_PRESERVE |
46 AV_PERM_REUSE | AV_PERM_REUSE2 | AV_PERM_ALIGN;
48 av_assert1(!(perms & ~(full_perms | AV_PERM_NEG_LINESIZES)));
50 if (!(data = av_mallocz(sizeof(*data) * planes)))
53 if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
56 samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, full_perms,
57 nb_samples, link->format,
58 link->channel_layout);
62 samplesref->audio->sample_rate = link->sample_rate;
73 AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
76 AVFilterBufferRef *ret = NULL;
78 if (link->dstpad->get_audio_buffer)
79 ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
82 ret = ff_default_get_audio_buffer(link, perms, nb_samples);
85 ret->type = AVMEDIA_TYPE_AUDIO;
90 AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
91 int linesize,int perms,
93 enum AVSampleFormat sample_fmt,
94 uint64_t channel_layout)
97 AVFilterBuffer *samples = av_mallocz(sizeof(*samples));
98 AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
100 if (!samples || !samplesref)
103 samplesref->buf = samples;
104 samplesref->buf->free = ff_avfilter_default_free_buffer;
105 if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
108 samplesref->audio->nb_samples = nb_samples;
109 samplesref->audio->channel_layout = channel_layout;
111 planes = av_sample_fmt_is_planar(sample_fmt) ?
112 av_get_channel_layout_nb_channels(channel_layout) : 1;
114 /* make sure the buffer gets read permission or it's useless for output */
115 samplesref->perms = perms | AV_PERM_READ;
117 samples->refcount = 1;
118 samplesref->type = AVMEDIA_TYPE_AUDIO;
119 samplesref->format = sample_fmt;
121 memcpy(samples->data, data,
122 FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
123 memcpy(samplesref->data, samples->data, sizeof(samples->data));
125 samples->linesize[0] = samplesref->linesize[0] = linesize;
127 if (planes > FF_ARRAY_ELEMS(samples->data)) {
128 samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) *
130 samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
133 if (!samples->extended_data || !samplesref->extended_data)
136 memcpy(samples-> extended_data, data, sizeof(*data)*planes);
137 memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
139 samples->extended_data = samples->data;
140 samplesref->extended_data = samplesref->data;
143 samplesref->pts = AV_NOPTS_VALUE;
148 if (samples && samples->extended_data != samples->data)
149 av_freep(&samples->extended_data);
151 av_freep(&samplesref->audio);
152 if (samplesref->extended_data != samplesref->data)
153 av_freep(&samplesref->extended_data);
155 av_freep(&samplesref);
160 static int default_filter_samples(AVFilterLink *link,
161 AVFilterBufferRef *samplesref)
163 return ff_filter_samples(link->dst->outputs[0], samplesref);
166 int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
168 int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
169 AVFilterPad *src = link->srcpad;
170 AVFilterPad *dst = link->dstpad;
172 AVFilterBufferRef *buf_out;
175 FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
178 avfilter_unref_buffer(samplesref);
182 if (!(filter_samples = dst->filter_samples))
183 filter_samples = default_filter_samples;
185 av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
186 samplesref->perms &= ~ src->rej_perms;
188 /* prepare to copy the samples if the buffer has insufficient permissions */
189 if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
190 dst->rej_perms & samplesref->perms) {
191 av_log(link->dst, AV_LOG_DEBUG,
192 "Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
193 samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
195 buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
196 samplesref->audio->nb_samples);
198 avfilter_unref_buffer(samplesref);
199 return AVERROR(ENOMEM);
201 buf_out->pts = samplesref->pts;
202 buf_out->audio->sample_rate = samplesref->audio->sample_rate;
204 /* Copy actual data into new samples buffer */
205 av_samples_copy(buf_out->extended_data, samplesref->extended_data,
206 0, 0, samplesref->audio->nb_samples,
207 av_get_channel_layout_nb_channels(link->channel_layout),
210 avfilter_unref_buffer(samplesref);
212 buf_out = samplesref;
214 link->cur_buf = buf_out;
216 ret = filter_samples(link, buf_out);
217 ff_update_link_current_pts(link, pts);
221 int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
223 int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
224 AVFilterBufferRef *pbuf = link->partial_buf;
225 int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
228 av_assert1(samplesref->format == link->format);
229 av_assert1(samplesref->audio->channel_layout == link->channel_layout);
230 av_assert1(samplesref->audio->sample_rate == link->sample_rate);
232 if (!link->min_samples ||
234 insamples >= link->min_samples && insamples <= link->max_samples)) {
235 return ff_filter_samples_framed(link, samplesref);
237 /* Handle framing (min_samples, max_samples) */
240 AVRational samples_tb = { 1, link->sample_rate };
241 int perms = link->dstpad->min_perms | AV_PERM_WRITE;
242 pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size);
244 av_log(link->dst, AV_LOG_WARNING,
245 "Samples dropped due to memory allocation failure.\n");
248 avfilter_copy_buffer_ref_props(pbuf, samplesref);
249 pbuf->pts = samplesref->pts +
250 av_rescale_q(inpos, samples_tb, link->time_base);
251 pbuf->audio->nb_samples = 0;
253 nb_samples = FFMIN(insamples,
254 link->partial_buf_size - pbuf->audio->nb_samples);
255 av_samples_copy(pbuf->extended_data, samplesref->extended_data,
256 pbuf->audio->nb_samples, inpos,
257 nb_samples, nb_channels, link->format);
259 insamples -= nb_samples;
260 pbuf->audio->nb_samples += nb_samples;
261 if (pbuf->audio->nb_samples >= link->min_samples) {
262 ret = ff_filter_samples_framed(link, pbuf);
266 avfilter_unref_buffer(samplesref);
267 link->partial_buf = pbuf;