FFmpeg
fastaudio.c
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1 /*
2  * MOFLEX Fast Audio decoder
3  * Copyright (c) 2020 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 "bytestream.h"
26 #include "internal.h"
27 #include "mathops.h"
28 
29 typedef struct ChannelItems {
30  float f[8];
31  float last;
32 } ChannelItems;
33 
34 typedef struct FastAudioContext {
35  float table[8][64];
36 
39 
41 {
42  FastAudioContext *s = avctx->priv_data;
43 
45 
46  for (int i = 0; i < 8; i++)
47  s->table[0][i] = (i - 159.5f) / 160.f;
48  for (int i = 0; i < 11; i++)
49  s->table[0][i + 8] = (i - 37.5f) / 40.f;
50  for (int i = 0; i < 27; i++)
51  s->table[0][i + 8 + 11] = (i - 13.f) / 20.f;
52  for (int i = 0; i < 11; i++)
53  s->table[0][i + 8 + 11 + 27] = (i + 27.5f) / 40.f;
54  for (int i = 0; i < 7; i++)
55  s->table[0][i + 8 + 11 + 27 + 11] = (i + 152.5f) / 160.f;
56 
57  memcpy(s->table[1], s->table[0], sizeof(s->table[0]));
58 
59  for (int i = 0; i < 7; i++)
60  s->table[2][i] = (i - 33.5f) / 40.f;
61  for (int i = 0; i < 25; i++)
62  s->table[2][i + 7] = (i - 13.f) / 20.f;
63 
64  for (int i = 0; i < 32; i++)
65  s->table[3][i] = -s->table[2][31 - i];
66 
67  for (int i = 0; i < 16; i++)
68  s->table[4][i] = i * 0.22f / 3.f - 0.6f;
69 
70  for (int i = 0; i < 16; i++)
71  s->table[5][i] = i * 0.20f / 3.f - 0.3f;
72 
73  for (int i = 0; i < 8; i++)
74  s->table[6][i] = i * 0.36f / 3.f - 0.4f;
75 
76  for (int i = 0; i < 8; i++)
77  s->table[7][i] = i * 0.34f / 3.f - 0.2f;
78 
79  s->ch = av_calloc(avctx->channels, sizeof(*s->ch));
80  if (!s->ch)
81  return AVERROR(ENOMEM);
82 
83  return 0;
84 }
85 
86 static int read_bits(int bits, int *ppos, unsigned *src)
87 {
88  int r, pos;
89 
90  pos = *ppos;
91  pos += bits;
92  r = src[(pos - 1) / 32] >> (32 - pos % 32);
93  *ppos = pos;
94 
95  return r & ((1 << bits) - 1);
96 }
97 
98 static const uint8_t bits[8] = { 6, 6, 5, 5, 4, 0, 3, 3, };
99 
100 static void set_sample(int i, int j, int v, float *result, int *pads, float value)
101 {
102  result[i * 64 + pads[i] + j * 3] = value * (2 * v - 7);
103 }
104 
105 static int fastaudio_decode(AVCodecContext *avctx, void *data,
106  int *got_frame, AVPacket *pkt)
107 {
108  FastAudioContext *s = avctx->priv_data;
109  GetByteContext gb;
110  AVFrame *frame = data;
111  int subframes;
112  int ret;
113 
114  subframes = pkt->size / (40 * avctx->channels);
115  frame->nb_samples = subframes * 256;
116  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
117  return ret;
118 
119  bytestream2_init(&gb, pkt->data, pkt->size);
120 
121  for (int subframe = 0; subframe < subframes; subframe++) {
122  for (int channel = 0; channel < avctx->channels; channel++) {
123  ChannelItems *ch = &s->ch[channel];
124  float result[256] = { 0 };
125  unsigned src[10];
126  int inds[4], pads[4];
127  float m[8];
128  int pos = 0;
129 
130  for (int i = 0; i < 10; i++)
131  src[i] = bytestream2_get_le32(&gb);
132 
133  for (int i = 0; i < 8; i++)
134  m[7 - i] = s->table[i][read_bits(bits[i], &pos, src)];
135 
136  for (int i = 0; i < 4; i++)
137  inds[3 - i] = read_bits(6, &pos, src);
138 
139  for (int i = 0; i < 4; i++)
140  pads[3 - i] = read_bits(2, &pos, src);
141 
142  for (int i = 0, index5 = 0; i < 4; i++) {
143  float value = av_int2float((inds[i] + 1) << 20) * powf(2.f, 116.f);
144 
145  for (int j = 0, tmp = 0; j < 21; j++) {
146  set_sample(i, j, j == 20 ? tmp / 2 : read_bits(3, &pos, src), result, pads, value);
147  if (j % 10 == 9)
148  tmp = 4 * tmp + read_bits(2, &pos, src);
149  if (j == 20)
150  index5 = FFMIN(2 * index5 + tmp % 2, 63);
151  }
152 
153  m[2] = s->table[5][index5];
154  }
155 
156  for (int i = 0; i < 256; i++) {
157  float x = result[i];
158 
159  for (int j = 0; j < 8; j++) {
160  x -= m[j] * ch->f[j];
161  ch->f[j] += m[j] * x;
162  }
163 
164  memmove(&ch->f[0], &ch->f[1], sizeof(float) * 7);
165  ch->f[7] = x;
166  ch->last = x + ch->last * 0.86f;
167  result[i] = ch->last * 2.f;
168  }
169 
170  memcpy(frame->extended_data[channel] + 1024 * subframe, result, 256 * sizeof(float));
171  }
172  }
173 
174  *got_frame = 1;
175 
176  return pkt->size;
177 }
178 
180 {
181  FastAudioContext *s = avctx->priv_data;
182 
183  av_freep(&s->ch);
184 
185  return 0;
186 }
187 
189  .name = "fastaudio",
190  .long_name = NULL_IF_CONFIG_SMALL("MobiClip FastAudio"),
191  .type = AVMEDIA_TYPE_AUDIO,
192  .id = AV_CODEC_ID_FASTAUDIO,
193  .priv_data_size = sizeof(FastAudioContext),
194  .init = fastaudio_init,
196  .close = fastaudio_close,
197  .capabilities = AV_CODEC_CAP_DR1,
198  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
200 };
float, planar
Definition: samplefmt.h:69
float last
Definition: fastaudio.c:31
static void set_sample(int i, int j, int v, float *result, int *pads, float value)
Definition: fastaudio.c:100
This structure describes decoded (raw) audio or video data.
Definition: frame.h:308
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static av_always_inline float av_int2float(uint32_t i)
Reinterpret a 32-bit integer as a float.
Definition: intfloat.h:40
int size
Definition: packet.h:364
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:137
static AVPacket pkt
static av_cold int fastaudio_close(AVCodecContext *avctx)
Definition: fastaudio.c:179
static av_cold int fastaudio_init(AVCodecContext *avctx)
Definition: fastaudio.c:40
AVCodec.
Definition: codec.h:190
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:245
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
uint8_t
#define av_cold
Definition: attributes.h:88
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:108
uint8_t * data
Definition: packet.h:363
static const uint16_t table[]
Definition: prosumer.c:206
#define src
Definition: vp8dsp.c:254
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:153
const char * r
Definition: vf_curves.c:114
unsigned int pos
Definition: spdifenc.c:410
const char * name
Name of the codec implementation.
Definition: codec.h:197
#define powf(x, y)
Definition: libm.h:50
AVCodec ff_fastaudio_decoder
Definition: fastaudio.c:188
#define FFMIN(a, b)
Definition: common.h:96
float f[8]
Definition: fastaudio.c:30
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 int fastaudio_decode(AVCodecContext *avctx, void *data, int *got_frame, AVPacket *pkt)
Definition: fastaudio.c:105
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
main external API structure.
Definition: avcodec.h:526
static const uint8_t bits[8]
Definition: fastaudio.c:98
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1879
float table[8][64]
Definition: fastaudio.c:35
common internal api header.
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
void * priv_data
Definition: avcodec.h:553
ChannelItems * ch
Definition: fastaudio.c:37
int channels
number of audio channels
Definition: avcodec.h:1187
and forward the result(frame or status change) to the corresponding input.If nothing is possible
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:884
static int read_bits(int bits, int *ppos, unsigned *src)
Definition: fastaudio.c:86
#define av_freep(p)
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
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:355
This structure stores compressed data.
Definition: packet.h:340
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:374
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
for(j=16;j >0;--j)
int i
Definition: input.c:407
static uint8_t tmp[11]
Definition: aes_ctr.c:26