FFmpeg
fastaudio.c
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1 /*
2  * MOFLEX Fast Audio decoder
3  * Copyright (c) 2015-2016 Florian Nouwt
4  * Copyright (c) 2017 Adib Surani
5  * Copyright (c) 2020 Paul B Mahol
6  *
7  * This file is part of FFmpeg.
8  *
9  * FFmpeg is free software; you can redistribute it and/or
10  * modify it under the terms of the GNU Lesser General Public
11  * License as published by the Free Software Foundation; either
12  * version 2.1 of the License, or (at your option) any later version.
13  *
14  * FFmpeg is distributed in the hope that it will be useful,
15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17  * Lesser General Public License for more details.
18  *
19  * You should have received a copy of the GNU Lesser General Public
20  * License along with FFmpeg; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22  */
23 
24 #include "libavutil/intreadwrite.h"
25 
26 #include "avcodec.h"
27 #include "bytestream.h"
28 #include "internal.h"
29 #include "mathops.h"
30 
31 typedef struct ChannelItems {
32  float f[8];
33  float last;
34 } ChannelItems;
35 
36 typedef struct FastAudioContext {
37  float table[8][64];
38 
41 
43 {
44  FastAudioContext *s = avctx->priv_data;
45 
47 
48  for (int i = 0; i < 8; i++)
49  s->table[0][i] = (i - 159.5f) / 160.f;
50  for (int i = 0; i < 11; i++)
51  s->table[0][i + 8] = (i - 37.5f) / 40.f;
52  for (int i = 0; i < 27; i++)
53  s->table[0][i + 8 + 11] = (i - 13.f) / 20.f;
54  for (int i = 0; i < 11; i++)
55  s->table[0][i + 8 + 11 + 27] = (i + 27.5f) / 40.f;
56  for (int i = 0; i < 7; i++)
57  s->table[0][i + 8 + 11 + 27 + 11] = (i + 152.5f) / 160.f;
58 
59  memcpy(s->table[1], s->table[0], sizeof(s->table[0]));
60 
61  for (int i = 0; i < 7; i++)
62  s->table[2][i] = (i - 33.5f) / 40.f;
63  for (int i = 0; i < 25; i++)
64  s->table[2][i + 7] = (i - 13.f) / 20.f;
65 
66  for (int i = 0; i < 32; i++)
67  s->table[3][i] = -s->table[2][31 - i];
68 
69  for (int i = 0; i < 16; i++)
70  s->table[4][i] = i * 0.22f / 3.f - 0.6f;
71 
72  for (int i = 0; i < 16; i++)
73  s->table[5][i] = i * 0.20f / 3.f - 0.3f;
74 
75  for (int i = 0; i < 8; i++)
76  s->table[6][i] = i * 0.36f / 3.f - 0.4f;
77 
78  for (int i = 0; i < 8; i++)
79  s->table[7][i] = i * 0.34f / 3.f - 0.2f;
80 
81  s->ch = av_calloc(avctx->channels, sizeof(*s->ch));
82  if (!s->ch)
83  return AVERROR(ENOMEM);
84 
85  return 0;
86 }
87 
88 static int read_bits(int bits, int *ppos, unsigned *src)
89 {
90  int r, pos;
91 
92  pos = *ppos;
93  pos += bits;
94  r = src[(pos - 1) / 32] >> ((-pos) & 31);
95  *ppos = pos;
96 
97  return r & ((1 << bits) - 1);
98 }
99 
100 static const uint8_t bits[8] = { 6, 6, 5, 5, 4, 0, 3, 3, };
101 
102 static void set_sample(int i, int j, int v, float *result, int *pads, float value)
103 {
104  result[i * 64 + pads[i] + j * 3] = value * (2 * v - 7);
105 }
106 
107 static int fastaudio_decode(AVCodecContext *avctx, void *data,
108  int *got_frame, AVPacket *pkt)
109 {
110  FastAudioContext *s = avctx->priv_data;
111  GetByteContext gb;
112  AVFrame *frame = data;
113  int subframes;
114  int ret;
115 
116  subframes = pkt->size / (40 * avctx->channels);
117  frame->nb_samples = subframes * 256;
118  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
119  return ret;
120 
121  bytestream2_init(&gb, pkt->data, pkt->size);
122 
123  for (int subframe = 0; subframe < subframes; subframe++) {
124  for (int channel = 0; channel < avctx->channels; channel++) {
125  ChannelItems *ch = &s->ch[channel];
126  float result[256] = { 0 };
127  unsigned src[10];
128  int inds[4], pads[4];
129  float m[8];
130  int pos = 0;
131 
132  for (int i = 0; i < 10; i++)
133  src[i] = bytestream2_get_le32(&gb);
134 
135  for (int i = 0; i < 8; i++)
136  m[7 - i] = s->table[i][read_bits(bits[i], &pos, src)];
137 
138  for (int i = 0; i < 4; i++)
139  inds[3 - i] = read_bits(6, &pos, src);
140 
141  for (int i = 0; i < 4; i++)
142  pads[3 - i] = read_bits(2, &pos, src);
143 
144  for (int i = 0, index5 = 0; i < 4; i++) {
145  float value = av_int2float((inds[i] + 1) << 20) * powf(2.f, 116.f);
146 
147  for (int j = 0, tmp = 0; j < 21; j++) {
148  set_sample(i, j, j == 20 ? tmp / 2 : read_bits(3, &pos, src), result, pads, value);
149  if (j % 10 == 9)
150  tmp = 4 * tmp + read_bits(2, &pos, src);
151  if (j == 20)
152  index5 = FFMIN(2 * index5 + tmp % 2, 63);
153  }
154 
155  m[2] = s->table[5][index5];
156  }
157 
158  for (int i = 0; i < 256; i++) {
159  float x = result[i];
160 
161  for (int j = 0; j < 8; j++) {
162  x -= m[j] * ch->f[j];
163  ch->f[j] += m[j] * x;
164  }
165 
166  memmove(&ch->f[0], &ch->f[1], sizeof(float) * 7);
167  ch->f[7] = x;
168  ch->last = x + ch->last * 0.86f;
169  result[i] = ch->last * 2.f;
170  }
171 
172  memcpy(frame->extended_data[channel] + 1024 * subframe, result, 256 * sizeof(float));
173  }
174  }
175 
176  *got_frame = 1;
177 
178  return pkt->size;
179 }
180 
182 {
183  FastAudioContext *s = avctx->priv_data;
184 
185  av_freep(&s->ch);
186 
187  return 0;
188 }
189 
191  .name = "fastaudio",
192  .long_name = NULL_IF_CONFIG_SMALL("MobiClip FastAudio"),
193  .type = AVMEDIA_TYPE_AUDIO,
194  .id = AV_CODEC_ID_FASTAUDIO,
195  .priv_data_size = sizeof(FastAudioContext),
196  .init = fastaudio_init,
198  .close = fastaudio_close,
199  .capabilities = AV_CODEC_CAP_DR1,
200  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
202  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
203 };
AVCodec
AVCodec.
Definition: codec.h:202
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
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: internal.h:42
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
r
const char * r
Definition: vf_curves.c:116
AVERROR
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
GetByteContext
Definition: bytestream.h:33
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:948
fastaudio_close
static av_cold int fastaudio_close(AVCodecContext *avctx)
Definition: fastaudio.c:181
AV_CODEC_ID_FASTAUDIO
@ AV_CODEC_ID_FASTAUDIO
Definition: codec_id.h:517
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:310
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:26
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:373
FastAudioContext::ch
ChannelItems * ch
Definition: fastaudio.c:39
data
const char data[16]
Definition: mxf.c:143
FastAudioContext
Definition: fastaudio.c:36
av_int2float
static av_always_inline float av_int2float(uint32_t i)
Reinterpret a 32-bit integer as a float.
Definition: intfloat.h:40
fastaudio_init
static av_cold int fastaudio_init(AVCodecContext *avctx)
Definition: fastaudio.c:42
pkt
AVPacket * pkt
Definition: movenc.c:59
av_cold
#define av_cold
Definition: attributes.h:90
set_sample
static void set_sample(int i, int j, int v, float *result, int *pads, float value)
Definition: fastaudio.c:102
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
f
#define f(width, name)
Definition: cbs_vp9.c:255
ChannelItems::last
float last
Definition: fastaudio.c:33
result
and forward the result(frame or status change) to the corresponding input. If nothing is possible
fastaudio_decode
static int fastaudio_decode(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *pkt)
Definition: fastaudio.c:107
src
#define src
Definition: vp8dsp.c:255
mathops.h
FastAudioContext::table
float table[8][64]
Definition: fastaudio.c:37
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1651
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:374
ChannelItems::f
float f[8]
Definition: fastaudio.c:32
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
powf
#define powf(x, y)
Definition: libm.h:50
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1000
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
read_bits
static int read_bits(int bits, int *ppos, unsigned *src)
Definition: fastaudio.c:88
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:993
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:271
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
value
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default value
Definition: writing_filters.txt:86
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:209
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:271
avcodec.h
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
pos
unsigned int pos
Definition: spdifenc.c:412
ChannelItems
Definition: fastaudio.c:31
ff_fastaudio_decoder
const AVCodec ff_fastaudio_decoder
Definition: fastaudio.c:190
AVCodecContext
main external API structure.
Definition: avcodec.h:383
AVPacket
This structure stores compressed data.
Definition: packet.h:350
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:410
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
bytestream.h
bytestream2_init
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:137
bits
static const uint8_t bits[8]
Definition: fastaudio.c:100
channel
channel
Definition: ebur128.h:39