FFmpeg
dvaudiodec.c
Go to the documentation of this file.
1 /*
2  * Copyright (c) 2012 Laurent Aimar
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/intreadwrite.h"
22 #include "avcodec.h"
23 #include "internal.h"
24 #include "dvaudio.h"
25 
26 typedef struct DVAudioContext {
28  int is_12bit;
29  int is_pal;
30  int16_t shuffle[2000];
32 
34 {
35  DVAudioContext *s = avctx->priv_data;
36  int i;
37 
38  if (avctx->channels != 2) {
39  av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
40  return AVERROR(EINVAL);
41  }
42 
43  if (avctx->codec_tag == 0x0215) {
44  s->block_size = 7200;
45  } else if (avctx->codec_tag == 0x0216) {
46  s->block_size = 8640;
47  } else if (avctx->block_align == 7200 ||
48  avctx->block_align == 8640) {
49  s->block_size = avctx->block_align;
50  } else {
51  return AVERROR(EINVAL);
52  }
53 
54  s->is_pal = s->block_size == 8640;
55  s->is_12bit = avctx->bits_per_coded_sample == 12;
58 
59  for (i = 0; i < FF_ARRAY_ELEMS(s->shuffle); i++) {
60  const unsigned a = s->is_pal ? 18 : 15;
61  const unsigned b = 3 * a;
62 
63  s->shuffle[i] = 80 * ((21 * (i % 3) + 9 * (i / 3) + ((i / a) % 3)) % b) +
64  (2 + s->is_12bit) * (i / b) + 8;
65  }
66 
67  return 0;
68 }
69 
70 static inline uint16_t dv_audio_12to16(uint16_t sample)
71 {
72  uint16_t shift, result;
73 
74  sample = (sample < 0x800) ? sample : sample | 0xf000;
75  shift = (sample & 0xf00) >> 8;
76 
77  if (shift < 0x2 || shift > 0xd) {
78  result = sample;
79  } else if (shift < 0x8) {
80  shift--;
81  result = (sample - (256 * shift)) << shift;
82  } else {
83  shift = 0xe - shift;
84  result = ((sample + ((256 * shift) + 1)) << shift) - 1;
85  }
86 
87  return result;
88 }
89 
90 static int decode_frame(AVCodecContext *avctx, void *data,
91  int *got_frame_ptr, AVPacket *pkt)
92 {
93  DVAudioContext *s = avctx->priv_data;
94  AVFrame *frame = data;
95  const uint8_t *src = pkt->data;
96  int16_t *dst;
97  int ret, i;
98 
99  if (pkt->size < s->block_size)
100  return AVERROR_INVALIDDATA;
101 
102  frame->nb_samples = dv_get_audio_sample_count(pkt->data + 244, s->is_pal);
103  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
104  return ret;
105  dst = (int16_t *)frame->data[0];
106 
107  for (i = 0; i < frame->nb_samples; i++) {
108  const uint8_t *v = &src[s->shuffle[i]];
109 
110  if (s->is_12bit) {
111  *dst++ = dv_audio_12to16((v[0] << 4) | ((v[2] >> 4) & 0x0f));
112  *dst++ = dv_audio_12to16((v[1] << 4) | ((v[2] >> 0) & 0x0f));
113  } else {
114  *dst++ = AV_RB16(&v[0]);
115  *dst++ = AV_RB16(&v[s->is_pal ? 4320 : 3600]);
116  }
117  }
118 
119  *got_frame_ptr = 1;
120 
121  return s->block_size;
122 }
123 
125  .name = "dvaudio",
126  .long_name = NULL_IF_CONFIG_SMALL("Ulead DV Audio"),
127  .type = AVMEDIA_TYPE_AUDIO,
128  .id = AV_CODEC_ID_DVAUDIO,
129  .init = decode_init,
130  .decode = decode_frame,
131  .capabilities = AV_CODEC_CAP_DR1,
132  .priv_data_size = sizeof(DVAudioContext),
133 };
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
static int shift(int a, int b)
Definition: sonic.c:82
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
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:36
static AVPacket pkt
#define AV_CH_LAYOUT_STEREO
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_RB16
Definition: bytestream.h:87
#define src
Definition: vp8dsp.c:254
#define sample
AVCodec.
Definition: avcodec.h:3492
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2265
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
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:2792
#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:3499
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2279
#define b
Definition: input.c:41
static av_cold int decode_init(AVCodecContext *avctx)
Definition: dvaudiodec.c:33
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
#define s(width, name)
Definition: cbs_vp9.c:257
static uint16_t dv_audio_12to16(uint16_t sample)
Definition: dvaudiodec.c:70
#define FF_ARRAY_ELEMS(a)
Libavcodec external API header.
main external API structure.
Definition: avcodec.h:1568
unsigned int codec_tag
fourcc (LSB first, so "ABCD" -> (&#39;D&#39;<<24) + (&#39;C&#39;<<16) + (&#39;B&#39;<<8) + &#39;A&#39;).
Definition: avcodec.h:1593
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1968
static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *pkt)
Definition: dvaudiodec.c:90
static int dv_get_audio_sample_count(const uint8_t *buffer, int dsf)
Definition: dvaudio.h:24
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
void * priv_data
Definition: avcodec.h:1595
int channels
number of audio channels
Definition: avcodec.h:2229
int16_t shuffle[2000]
Definition: dvaudiodec.c:30
AVCodec ff_dvaudio_decoder
Definition: dvaudiodec.c:124
and forward the result(frame or status change) to the corresponding input.If nothing is possible
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
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)