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 "codec_internal.h"
27 #include "internal.h"
28 
29 static const int bmv_aud_mults[16] = {
30  16512, 8256, 4128, 2064, 1032, 516, 258, 192, 129, 88, 64, 56, 48, 40, 36, 32
31 };
32 
34 {
38 
39  return 0;
40 }
41 
43  int *got_frame_ptr, AVPacket *avpkt)
44 {
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  .p.name = "bmv_audio",
83  .p.long_name = NULL_IF_CONFIG_SMALL("Discworld II BMV audio"),
84  .p.type = AVMEDIA_TYPE_AUDIO,
85  .p.id = AV_CODEC_ID_BMV_AUDIO,
86  .init = bmv_aud_decode_init,
88  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
89  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
90 };
bmv_aud_decode_init
static av_cold int bmv_aud_decode_init(AVCodecContext *avctx)
Definition: bmvaudio.c:33
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:374
FFCodec
Definition: codec_internal.h:112
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:354
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:116
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:2056
scale
static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1389
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
av_cold
#define av_cold
Definition: attributes.h:90
FF_CODEC_DECODE_CB
#define FF_CODEC_DECODE_CB(func)
Definition: codec_internal.h:254
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_clip_int16
#define av_clip_int16
Definition: common.h:110
av_channel_layout_uninit
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
Definition: channel_layout.c:630
AV_CODEC_CAP_CHANNEL_CONF
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:109
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1403
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:375
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:290
codec_internal.h
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1014
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
code
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
Definition: filter_design.txt:178
common.h
AV_CODEC_ID_BMV_AUDIO
@ AV_CODEC_ID_BMV_AUDIO
Definition: codec_id.h:483
FF_CODEC_CAP_INIT_THREADSAFE
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: codec_internal.h:31
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:58
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:203
avcodec.h
bmv_aud_decode_frame
static int bmv_aud_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: bmvaudio.c:42
ret
ret
Definition: filter_design.txt:187
frame
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
Definition: filter_design.txt:264
bmv_aud_mults
static const int bmv_aud_mults[16]
Definition: bmvaudio.c:29
ff_bmv_audio_decoder
const FFCodec ff_bmv_audio_decoder
Definition: bmvaudio.c:81
AVCodecContext
main external API structure.
Definition: avcodec.h:389
channel_layout.h
AVPacket
This structure stores compressed data.
Definition: packet.h:351
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61