FFmpeg
pafaudio.c
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1 /*
2  * Packed Animation File audio decoder
3  * Copyright (c) 2012 Paul B Mahol
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/intreadwrite.h"
23 
24 #include "avcodec.h"
25 #include "internal.h"
26 #include "mathops.h"
27 #include "paf.h"
28 
30 {
31  if (avctx->channels != 2) {
32  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
33  return AVERROR_INVALIDDATA;
34  }
35 
38 
39  return 0;
40 }
41 
42 static int paf_audio_decode(AVCodecContext *avctx, void *data,
43  int *got_frame, AVPacket *pkt)
44 {
45  AVFrame *frame = data;
46  int16_t *output_samples;
47  const uint8_t *src = pkt->data;
48  int frames, ret, i, j;
49  int16_t cb[256];
50 
51  frames = pkt->size / PAF_SOUND_FRAME_SIZE;
52  if (frames < 1)
53  return AVERROR_INVALIDDATA;
54 
56  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
57  return ret;
58 
59  output_samples = (int16_t *)frame->data[0];
60  // codebook of 256 16-bit samples and 8-bit indices to it
61  for (j = 0; j < frames; j++) {
62  for (i = 0; i < 256; i++)
63  cb[i] = sign_extend(AV_RL16(src + i * 2), 16);
64  src += 256 * 2;
65  // always 2 channels
66  for (i = 0; i < PAF_SOUND_SAMPLES * 2; i++)
67  *output_samples++ = cb[*src++];
68  }
69  *got_frame = 1;
70 
71  return pkt->size;
72 }
73 
75  .name = "paf_audio",
76  .long_name = NULL_IF_CONFIG_SMALL("Amazing Studio Packed Animation File Audio"),
77  .type = AVMEDIA_TYPE_AUDIO,
79  .init = paf_audio_init,
80  .decode = paf_audio_decode,
81  .capabilities = AV_CODEC_CAP_DR1,
82 };
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
#define PAF_SOUND_FRAME_SIZE
Definition: paf.h:26
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
int size
Definition: avcodec.h:1481
static AVPacket pkt
#define AV_CH_LAYOUT_STEREO
#define src
Definition: vp8dsp.c:254
AVCodec.
Definition: avcodec.h:3484
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
Definition: bytestream.h:87
if it could not because there are no more frames
static double cb(void *priv, double x, double y)
Definition: vf_geq.c:112
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
#define av_cold
Definition: attributes.h:82
static int paf_audio_decode(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *pkt)
Definition: pafaudio.c:42
uint8_t * data
Definition: avcodec.h:1480
#define av_log(a,...)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * name
Name of the codec implementation.
Definition: avcodec.h:3491
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2279
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Libavcodec external API header.
main external API structure.
Definition: avcodec.h:1568
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1963
#define PAF_SOUND_SAMPLES
Definition: paf.h:25
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:130
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
AVCodec ff_paf_audio_decoder
Definition: pafaudio.c:74
static av_cold int paf_audio_init(AVCodecContext *avctx)
Definition: pafaudio.c:29
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
int channels
number of audio channels
Definition: avcodec.h:2229
This structure stores compressed data.
Definition: avcodec.h:1457
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:984
for(j=16;j >0;--j)