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/channel_layout.h"
24 #include "libavutil/common.h"
30 int avfilter_ref_get_channels(AVFilterBufferRef *ref)
32 return ref->audio ? ref->audio->channels : 0;
35 AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
38 return ff_get_audio_buffer(link->dst->outputs[0], perms, nb_samples);
41 AVFilterBufferRef *ff_default_get_audio_buffer(AVFilterLink *link, int perms,
44 AVFilterBufferRef *samplesref = NULL;
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;
50 int full_perms = AV_PERM_READ | AV_PERM_WRITE | AV_PERM_PRESERVE |
51 AV_PERM_REUSE | AV_PERM_REUSE2 | AV_PERM_ALIGN;
53 av_assert1(!(perms & ~(full_perms | AV_PERM_NEG_LINESIZES)));
55 if (!(data = av_mallocz(sizeof(*data) * planes)))
58 if (av_samples_alloc(data, &linesize, nb_channels, nb_samples, link->format, 0) < 0)
61 samplesref = avfilter_get_audio_buffer_ref_from_arrays(data, linesize, full_perms,
62 nb_samples, link->format,
63 link->channel_layout);
67 samplesref->audio->sample_rate = link->sample_rate;
78 AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
81 AVFilterBufferRef *ret = NULL;
83 if (link->dstpad->get_audio_buffer)
84 ret = link->dstpad->get_audio_buffer(link, perms, nb_samples);
87 ret = ff_default_get_audio_buffer(link, perms, nb_samples);
90 ret->type = AVMEDIA_TYPE_AUDIO;
95 AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
96 int linesize,int perms,
98 enum AVSampleFormat sample_fmt,
99 uint64_t channel_layout)
101 int channels = av_get_channel_layout_nb_channels(channel_layout);
103 AVFilterBuffer *samples = av_mallocz(sizeof(*samples));
104 AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
106 if (!samples || !samplesref)
109 samplesref->buf = samples;
110 samplesref->buf->free = ff_avfilter_default_free_buffer;
111 if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
114 samplesref->audio->nb_samples = nb_samples;
115 samplesref->audio->channel_layout = channel_layout;
116 samplesref->audio->channels = channels;
118 planes = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
120 /* make sure the buffer gets read permission or it's useless for output */
121 samplesref->perms = perms | AV_PERM_READ;
123 samples->refcount = 1;
124 samplesref->type = AVMEDIA_TYPE_AUDIO;
125 samplesref->format = sample_fmt;
127 memcpy(samples->data, data,
128 FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
129 memcpy(samplesref->data, samples->data, sizeof(samples->data));
131 samples->linesize[0] = samplesref->linesize[0] = linesize;
133 if (planes > FF_ARRAY_ELEMS(samples->data)) {
134 samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) *
136 samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
139 if (!samples->extended_data || !samplesref->extended_data)
142 memcpy(samples-> extended_data, data, sizeof(*data)*planes);
143 memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
145 samples->extended_data = samples->data;
146 samplesref->extended_data = samplesref->data;
149 samplesref->pts = AV_NOPTS_VALUE;
154 if (samples && samples->extended_data != samples->data)
155 av_freep(&samples->extended_data);
157 av_freep(&samplesref->audio);
158 if (samplesref->extended_data != samplesref->data)
159 av_freep(&samplesref->extended_data);
161 av_freep(&samplesref);
166 static int default_filter_frame(AVFilterLink *link, AVFilterBufferRef *frame)
168 return ff_filter_frame(link->dst->outputs[0], frame);
171 int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
173 int (*filter_frame)(AVFilterLink *, AVFilterBufferRef *);
174 AVFilterPad *src = link->srcpad;
175 AVFilterPad *dst = link->dstpad;
177 AVFilterBufferRef *buf_out;
180 FF_TPRINTF_START(NULL, filter_frame); ff_tlog_link(NULL, link, 1);
183 avfilter_unref_buffer(samplesref);
187 if (!(filter_frame = dst->filter_frame))
188 filter_frame = default_filter_frame;
190 av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
191 samplesref->perms &= ~ src->rej_perms;
193 /* prepare to copy the samples if the buffer has insufficient permissions */
194 if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
195 dst->rej_perms & samplesref->perms) {
196 av_log(link->dst, AV_LOG_DEBUG,
197 "Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
198 samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
200 buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
201 samplesref->audio->nb_samples);
203 avfilter_unref_buffer(samplesref);
204 return AVERROR(ENOMEM);
206 buf_out->pts = samplesref->pts;
207 buf_out->audio->sample_rate = samplesref->audio->sample_rate;
209 /* Copy actual data into new samples buffer */
210 av_samples_copy(buf_out->extended_data, samplesref->extended_data,
211 0, 0, samplesref->audio->nb_samples,
212 av_get_channel_layout_nb_channels(link->channel_layout),
215 avfilter_unref_buffer(samplesref);
217 buf_out = samplesref;
219 link->cur_buf = buf_out;
221 ret = filter_frame(link, buf_out);
222 ff_update_link_current_pts(link, pts);
226 int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
228 int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
229 AVFilterBufferRef *pbuf = link->partial_buf;
230 int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
233 av_assert1(samplesref->format == link->format);
234 av_assert1(samplesref->audio->channels == link->channels);
235 av_assert1(samplesref->audio->channel_layout == link->channel_layout);
236 av_assert1(samplesref->audio->sample_rate == link->sample_rate);
238 if (!link->min_samples ||
240 insamples >= link->min_samples && insamples <= link->max_samples)) {
241 return ff_filter_samples_framed(link, samplesref);
243 /* Handle framing (min_samples, max_samples) */
246 AVRational samples_tb = { 1, link->sample_rate };
247 int perms = link->dstpad->min_perms | AV_PERM_WRITE;
248 pbuf = ff_get_audio_buffer(link, perms, link->partial_buf_size);
250 av_log(link->dst, AV_LOG_WARNING,
251 "Samples dropped due to memory allocation failure.\n");
254 avfilter_copy_buffer_ref_props(pbuf, samplesref);
255 pbuf->pts = samplesref->pts +
256 av_rescale_q(inpos, samples_tb, link->time_base);
257 pbuf->audio->nb_samples = 0;
259 nb_samples = FFMIN(insamples,
260 link->partial_buf_size - pbuf->audio->nb_samples);
261 av_samples_copy(pbuf->extended_data, samplesref->extended_data,
262 pbuf->audio->nb_samples, inpos,
263 nb_samples, nb_channels, link->format);
265 insamples -= nb_samples;
266 pbuf->audio->nb_samples += nb_samples;
267 if (pbuf->audio->nb_samples >= link->min_samples) {
268 ret = ff_filter_samples_framed(link, pbuf);
272 avfilter_unref_buffer(samplesref);
273 link->partial_buf = pbuf;