FFmpeg: libavcodec/ra144dec.c Source File
Go to the documentation of this file. 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
31
32
34 {
36
39
42
46
47 return 0;
48 }
49
52 {
53 int cba_idx =
get_bits(gb, 7);
// index of the adaptive CB, 0 if none
57
59 gain);
60 }
61
62 /** Uncompress one block (20 bytes -> 160*2 bytes). */
65 {
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
75 unsigned int energy;
76
79
82 "Frame too small (%d bytes). Truncated file?\n", buf_size);
83 *got_frame_ptr = 0;
85 }
86
87 /* get output buffer */
92
94
97
100
102
104 refl_rms[1] =
ff_interp(ractx, block_coefs[1], 2,
105 energy <= ractx->old_energy,
107 refl_rms[2] =
ff_interp(ractx, block_coefs[2], 3, 0, energy);
109
111
114
117 }
118
121
123
124 *got_frame_ptr = 1;
125
127 }
128
130 .
p.
name =
"real_144",
138 };
static void do_output_subblock(RA144Context *ractx, const int16_t *lpc_coefs, int gval, GetBitContext *gb)
#define NBLOCKS
number of subblocks within a block
const int16_t ff_energy_tab[32]
This structure describes decoded (raw) audio or video data.
void ff_eval_coefs(int *coefs, const int *refl)
Evaluate the LPC filter coefficients from the reflection coefficients.
av_cold void ff_audiodsp_init(AudioDSPContext *c)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static int ra144_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Uncompress one block (20 bytes -> 160*2 bytes).
AVCodec p
The public AVCodec.
AVChannelLayout ch_layout
Audio channel layout.
unsigned int * lpc_coef[2]
LPC coefficients: lpc_coef[0] is the coefficients of the current frame and lpc_coef[1] of the previou...
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
const int16_t *const ff_lpc_refl_cb[10]
#define FF_CODEC_DECODE_CB(func)
#define CODEC_LONG_NAME(str)
const FFCodec ff_ra_144_decoder
unsigned int ff_rescale_rms(unsigned int rms, unsigned int energy)
static const int sizes[][2]
unsigned int lpc_tables[2][10]
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
int16_t curr_sblock[50]
The current subblock padded by the last 10 values of the previous one.
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
int(* init)(AVBSFContext *ctx)
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
An AVChannelLayout holds information about the channel layout of audio data.
enum AVSampleFormat sample_fmt
audio sample format
unsigned int ff_rms(const int *data)
void ff_int_to_int16(int16_t *out, const int *inp)
#define i(width, name, range_min, range_max)
unsigned int old_energy
previous frame energy
@ AV_SAMPLE_FMT_S16
signed 16 bits
const char * name
Name of the codec implementation.
void ff_subblock_synthesis(RA144Context *ractx, const int16_t *lpc_coefs, int cba_idx, int cb1_idx, int cb2_idx, int gval, int gain)
#define FFSWAP(type, a, b)
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
main external API structure.
int ff_t_sqrt(unsigned int x)
Evaluate sqrt(x << 24).
void av_channel_layout_uninit(AVChannelLayout *channel_layout)
Free any allocated data in the channel layout and reset the channel count to 0.
int ff_interp(RA144Context *ractx, int16_t *out, int a, int copyold, int energy)
Filter the word "frame" indicates either a video frame or a group of audio samples
#define AV_CHANNEL_LAYOUT_MONO
This structure stores compressed data.
static av_cold int ra144_decode_init(AVCodecContext *avctx)
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
unsigned int lpc_refl_rms[2]
#define BLOCKSIZE
subblock size in 16-bit words
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