FFmpeg
bmvaudio.c
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1 /*
2  * Discworld II BMV audio decoder
3  * Copyright (c) 2011 Konstantin Shishkov
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 
23 #include "libavutil/common.h"
24 
25 #include "avcodec.h"
26 #include "internal.h"
27 
28 static const int bmv_aud_mults[16] = {
29  16512, 8256, 4128, 2064, 1032, 516, 258, 192, 129, 88, 64, 56, 48, 40, 36, 32
30 };
31 
33 {
34  avctx->channels = 2;
37 
38  return 0;
39 }
40 
41 static int bmv_aud_decode_frame(AVCodecContext *avctx, void *data,
42  int *got_frame_ptr, AVPacket *avpkt)
43 {
44  AVFrame *frame = data;
45  const uint8_t *buf = avpkt->data;
46  int buf_size = avpkt->size;
47  int blocks = 0, total_blocks, i;
48  int ret;
49  int16_t *output_samples;
50  int scale[2];
51 
52  total_blocks = *buf++;
53  if (buf_size < total_blocks * 65 + 1) {
54  av_log(avctx, AV_LOG_ERROR, "expected %d bytes, got %d\n",
55  total_blocks * 65 + 1, buf_size);
56  return AVERROR_INVALIDDATA;
57  }
58 
59  /* get output buffer */
60  frame->nb_samples = total_blocks * 32;
61  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
62  return ret;
63  output_samples = (int16_t *)frame->data[0];
64 
65  for (blocks = 0; blocks < total_blocks; blocks++) {
66  uint8_t code = *buf++;
67  code = (code >> 1) | (code << 7);
68  scale[0] = bmv_aud_mults[code & 0xF];
69  scale[1] = bmv_aud_mults[code >> 4];
70  for (i = 0; i < 32; i++) {
71  *output_samples++ = av_clip_int16((scale[0] * (int8_t)*buf++) >> 5);
72  *output_samples++ = av_clip_int16((scale[1] * (int8_t)*buf++) >> 5);
73  }
74  }
75 
76  *got_frame_ptr = 1;
77 
78  return buf_size;
79 }
80 
82  .name = "bmv_audio",
83  .long_name = NULL_IF_CONFIG_SMALL("Discworld II BMV audio"),
84  .type = AVMEDIA_TYPE_AUDIO,
86  .init = bmv_aud_decode_init,
87  .decode = bmv_aud_decode_frame,
88  .capabilities = AV_CODEC_CAP_DR1,
89 };
#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
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
int size
Definition: avcodec.h:1481
AVCodec ff_bmv_audio_decoder
Definition: bmvaudio.c:81
#define AV_CH_LAYOUT_STEREO
AVCodec.
Definition: avcodec.h:3484
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2236
uint8_t
#define av_cold
Definition: attributes.h:82
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
audio channel layout utility functions
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
static const int bmv_aud_mults[16]
Definition: bmvaudio.c:28
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
void * buf
Definition: avisynth_c.h:766
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
common internal api header.
common internal and external API header
signed 16 bits
Definition: samplefmt.h:61
static int bmv_aud_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: bmvaudio.c:41
int channels
number of audio channels
Definition: avcodec.h:2229
static av_cold int bmv_aud_decode_init(AVCodecContext *avctx)
Definition: bmvaudio.c:32
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)