FFmpeg  4.3
ra144dec.c
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1 /*
2  * Real Audio 1.0 (14.4K)
3  *
4  * Copyright (c) 2008 Vitor Sessak
5  * Copyright (c) 2003 Nick Kurshev
6  * Based on public domain decoder at http://www.honeypot.net/audio
7  *
8  * This file is part of FFmpeg.
9  *
10  * FFmpeg is free software; you can redistribute it and/or
11  * modify it under the terms of the GNU Lesser General Public
12  * License as published by the Free Software Foundation; either
13  * version 2.1 of the License, or (at your option) any later version.
14  *
15  * FFmpeg is distributed in the hope that it will be useful,
16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18  * Lesser General Public License for more details.
19  *
20  * You should have received a copy of the GNU Lesser General Public
21  * License along with FFmpeg; if not, write to the Free Software
22  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23  */
24 
26 #include "avcodec.h"
27 #include "get_bits.h"
28 #include "internal.h"
29 #include "ra144.h"
30 
31 
33 {
34  RA144Context *ractx = avctx->priv_data;
35 
36  ractx->avctx = avctx;
37  ff_audiodsp_init(&ractx->adsp);
38 
39  ractx->lpc_coef[0] = ractx->lpc_tables[0];
40  ractx->lpc_coef[1] = ractx->lpc_tables[1];
41 
42  avctx->channels = 1;
45 
46  return 0;
47 }
48 
49 static void do_output_subblock(RA144Context *ractx, const int16_t *lpc_coefs,
50  int gval, GetBitContext *gb)
51 {
52  int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none
53  int gain = get_bits(gb, 8);
54  int cb1_idx = get_bits(gb, 7);
55  int cb2_idx = get_bits(gb, 7);
56 
57  ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, gval,
58  gain);
59 }
60 
61 /** Uncompress one block (20 bytes -> 160*2 bytes). */
62 static int ra144_decode_frame(AVCodecContext * avctx, void *data,
63  int *got_frame_ptr, AVPacket *avpkt)
64 {
65  AVFrame *frame = data;
66  const uint8_t *buf = avpkt->data;
67  int buf_size = avpkt->size;
68  static const uint8_t sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
69  unsigned int refl_rms[NBLOCKS]; // RMS of the reflection coefficients
70  int16_t block_coefs[NBLOCKS][LPC_ORDER]; // LPC coefficients of each sub-block
71  unsigned int lpc_refl[LPC_ORDER]; // LPC reflection coefficients of the frame
72  int i, j;
73  int ret;
74  int16_t *samples;
75  unsigned int energy;
76 
77  RA144Context *ractx = avctx->priv_data;
78  GetBitContext gb;
79 
80  if (buf_size < FRAME_SIZE) {
81  av_log(avctx, AV_LOG_ERROR,
82  "Frame too small (%d bytes). Truncated file?\n", buf_size);
83  *got_frame_ptr = 0;
84  return AVERROR_INVALIDDATA;
85  }
86 
87  /* get output buffer */
88  frame->nb_samples = NBLOCKS * BLOCKSIZE;
89  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
90  return ret;
91  samples = (int16_t *)frame->data[0];
92 
93  init_get_bits8(&gb, buf, FRAME_SIZE);
94 
95  for (i = 0; i < LPC_ORDER; i++)
96  lpc_refl[i] = ff_lpc_refl_cb[i][get_bits(&gb, sizes[i])];
97 
98  ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
99  ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
100 
101  energy = ff_energy_tab[get_bits(&gb, 5)];
102 
103  refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
104  refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
105  energy <= ractx->old_energy,
106  ff_t_sqrt(energy*ractx->old_energy) >> 12);
107  refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
108  refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
109 
110  ff_int_to_int16(block_coefs[3], ractx->lpc_coef[0]);
111 
112  for (i=0; i < NBLOCKS; i++) {
113  do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb);
114 
115  for (j=0; j < BLOCKSIZE; j++)
116  *samples++ = av_clip_int16(ractx->curr_sblock[j + 10] * (1 << 2));
117  }
118 
119  ractx->old_energy = energy;
120  ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
121 
122  FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
123 
124  *got_frame_ptr = 1;
125 
126  return FRAME_SIZE;
127 }
128 
130  .name = "real_144",
131  .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
132  .type = AVMEDIA_TYPE_AUDIO,
133  .id = AV_CODEC_ID_RA_144,
134  .priv_data_size = sizeof(RA144Context),
137  .capabilities = AV_CODEC_CAP_DR1,
138 };
do_output_subblock
static void do_output_subblock(RA144Context *ractx, const int16_t *lpc_coefs, int gval, GetBitContext *gb)
Definition: ra144dec.c:49
AVCodec
AVCodec.
Definition: codec.h:190
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
NBLOCKS
#define NBLOCKS
number of subblocks within a block
Definition: ra144.h:30
RA144Context::avctx
AVCodecContext * avctx
Definition: ra144.h:38
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
FFSWAP
#define FFSWAP(type, a, b)
Definition: common.h:99
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
ff_energy_tab
const int16_t ff_energy_tab[32]
Definition: ra144.c:1440
FRAME_SIZE
#define FRAME_SIZE
AV_CODEC_ID_RA_144
@ AV_CODEC_ID_RA_144
Definition: codec_id.h:396
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
ff_eval_coefs
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
Definition: ra144.c:1593
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:355
data
const char data[16]
Definition: mxf.c:91
ff_audiodsp_init
av_cold void ff_audiodsp_init(AudioDSPContext *c)
Definition: audiodsp.c:106
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
ff_ra_144_decoder
AVCodec ff_ra_144_decoder
Definition: ra144dec.c:129
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
GetBitContext
Definition: get_bits.h:61
RA144Context::lpc_coef
unsigned int * lpc_coef[2]
LPC coefficients: lpc_coef[0] is the coefficients of the current frame and lpc_coef[1] of the previou...
Definition: ra144.h:50
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold
#define av_cold
Definition: attributes.h:90
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
ff_lpc_refl_cb
const int16_t *const ff_lpc_refl_cb[10]
Definition: ra144.c:1502
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
get_bits.h
LPC_ORDER
#define LPC_ORDER
Definition: g723_1.h:40
ff_rescale_rms
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
Definition: ra144.c:1678
sizes
static const int sizes[][2]
Definition: img2dec.c:53
RA144Context
Definition: ra144.h:37
ra144_decode_frame
static int ra144_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Uncompress one block (20 bytes -> 160*2 bytes).
Definition: ra144dec.c:62
RA144Context::adsp
AudioDSPContext adsp
Definition: ra144.h:39
RA144Context::lpc_tables
unsigned int lpc_tables[2][10]
Definition: ra144.h:46
RA144Context::curr_sblock
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
Definition: ra144.h:57
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1854
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:50
AVPacket::size
int size
Definition: packet.h:356
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:186
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
ff_rms
unsigned int ff_rms(const int *data)
Definition: ra144.c:1636
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1187
ff_int_to_int16
void ff_int_to_int16(int16_t *out, const int *inp)
Definition: ra144.c:1613
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
RA144Context::old_energy
unsigned int old_energy
previous frame energy
Definition: ra144.h:44
uint8_t
uint8_t
Definition: audio_convert.c:194
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:197
ff_subblock_synthesis
void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
Definition: ra144.c:1694
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
AVCodecContext
main external API structure.
Definition: avcodec.h:526
channel_layout.h
ff_t_sqrt
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
Definition: ra144.c:1625
ff_interp
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
Definition: ra144.c:1657
ra144.h
AVPacket
This structure stores compressed data.
Definition: packet.h:332
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:553
ra144_decode_init
static av_cold int ra144_decode_init(AVCodecContext *avctx)
Definition: ra144dec.c:32
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
RA144Context::lpc_refl_rms
unsigned int lpc_refl_rms[2]
Definition: ra144.h:52
BLOCKSIZE
#define BLOCKSIZE
subblock size in 16-bit words
Definition: ra144.h:31