FFmpeg
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audio.c
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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"
24 #include "libavutil/common.h"
25 #include "libavcodec/avcodec.h"
26 
27 #include "audio.h"
28 #include "avfilter.h"
29 #include "internal.h"
30 
31 int avfilter_ref_get_channels(AVFilterBufferRef *ref)
32 {
33  return ref->audio ? ref->audio->channels : 0;
34 }
35 
37 {
38  return ff_get_audio_buffer(link->dst->outputs[0], nb_samples);
39 }
40 
42 {
44  int channels = link->channels;
45  int ret;
46 
48 
49  if (!frame)
50  return NULL;
51 
52  frame->nb_samples = nb_samples;
53  frame->format = link->format;
54  av_frame_set_channels(frame, link->channels);
55  frame->channel_layout = link->channel_layout;
56  frame->sample_rate = link->sample_rate;
57  ret = av_frame_get_buffer(frame, 0);
58  if (ret < 0) {
59  av_frame_free(&frame);
60  return NULL;
61  }
62 
63  av_samples_set_silence(frame->extended_data, 0, nb_samples, channels,
64  link->format);
65 
66 
67  return frame;
68 }
69 
70 AVFrame *ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
71 {
72  AVFrame *ret = NULL;
73 
74  if (link->dstpad->get_audio_buffer)
75  ret = link->dstpad->get_audio_buffer(link, nb_samples);
76 
77  if (!ret)
78  ret = ff_default_get_audio_buffer(link, nb_samples);
79 
80  return ret;
81 }
82 
83 #if FF_API_AVFILTERBUFFER
84 AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays_channels(uint8_t **data,
85  int linesize,int perms,
86  int nb_samples,
87  enum AVSampleFormat sample_fmt,
88  int channels,
89  uint64_t channel_layout)
90 {
91  int planes;
92  AVFilterBuffer *samples = av_mallocz(sizeof(*samples));
93  AVFilterBufferRef *samplesref = av_mallocz(sizeof(*samplesref));
94 
95  if (!samples || !samplesref)
96  goto fail;
97 
98  av_assert0(channels);
99  av_assert0(channel_layout == 0 ||
100  channels == av_get_channel_layout_nb_channels(channel_layout));
101 
102  samplesref->buf = samples;
103  samplesref->buf->free = ff_avfilter_default_free_buffer;
104  if (!(samplesref->audio = av_mallocz(sizeof(*samplesref->audio))))
105  goto fail;
106 
107  samplesref->audio->nb_samples = nb_samples;
108  samplesref->audio->channel_layout = channel_layout;
109  samplesref->audio->channels = channels;
110 
111  planes = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
112 
113  /* make sure the buffer gets read permission or it's useless for output */
114  samplesref->perms = perms | AV_PERM_READ;
115 
116  samples->refcount = 1;
117  samplesref->type = AVMEDIA_TYPE_AUDIO;
118  samplesref->format = sample_fmt;
119 
120  memcpy(samples->data, data,
121  FFMIN(FF_ARRAY_ELEMS(samples->data), planes)*sizeof(samples->data[0]));
122  memcpy(samplesref->data, samples->data, sizeof(samples->data));
123 
124  samples->linesize[0] = samplesref->linesize[0] = linesize;
125 
126  if (planes > FF_ARRAY_ELEMS(samples->data)) {
127  samples-> extended_data = av_mallocz(sizeof(*samples->extended_data) *
128  planes);
129  samplesref->extended_data = av_mallocz(sizeof(*samplesref->extended_data) *
130  planes);
131 
132  if (!samples->extended_data || !samplesref->extended_data)
133  goto fail;
134 
135  memcpy(samples-> extended_data, data, sizeof(*data)*planes);
136  memcpy(samplesref->extended_data, data, sizeof(*data)*planes);
137  } else {
138  samples->extended_data = samples->data;
139  samplesref->extended_data = samplesref->data;
140  }
141 
142  samplesref->pts = AV_NOPTS_VALUE;
143 
144  return samplesref;
145 
146 fail:
147  if (samples && samples->extended_data != samples->data)
148  av_freep(&samples->extended_data);
149  if (samplesref) {
150  av_freep(&samplesref->audio);
151  if (samplesref->extended_data != samplesref->data)
152  av_freep(&samplesref->extended_data);
153  }
154  av_freep(&samplesref);
155  av_freep(&samples);
156  return NULL;
157 }
158 
159 AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
160  int linesize,int perms,
161  int nb_samples,
162  enum AVSampleFormat sample_fmt,
163  uint64_t channel_layout)
164 {
165  int channels = av_get_channel_layout_nb_channels(channel_layout);
166  return avfilter_get_audio_buffer_ref_from_arrays_channels(data, linesize, perms,
167  nb_samples, sample_fmt,
168  channels, channel_layout);
169 }
170 #endif