47 #define AT1_MAX_BFU 52
48 #define AT1_SU_SIZE 212
49 #define AT1_SU_SAMPLES 512
50 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2
51 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8
52 #define AT1_MAX_CHANNELS 2
54 #define AT1_QMF_BANDS 3
55 #define IDX_LOW_BAND 0
56 #define IDX_MID_BAND 1
57 #define IDX_HIGH_BAND 2
98 int transf_size = 1 << nbits;
102 for (
i = 0;
i < transf_size / 2;
i++)
103 FFSWAP(
float, spec[
i], spec[transf_size - 1 -
i]);
111 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size;
112 unsigned int start_pos, ref_pos = 0,
pos = 0;
119 log2_block_count =
su->log2_block_count[band_num];
123 num_blocks = 1 << log2_block_count;
125 if (num_blocks == 1) {
128 block_size = band_samples >> log2_block_count;
133 if (nbits != 5 && nbits != 7 && nbits != 8)
141 prev_buf = &
su->spectrum[1][ref_pos + band_samples - 16];
142 for (j=0; j < num_blocks; j++) {
147 &
su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16);
149 prev_buf = &
su->spectrum[0][ref_pos+start_pos + 16];
150 start_pos += block_size;
155 memcpy(q->
bands[band_num] + 32, &
su->spectrum[0][ref_pos + 16], 240 *
sizeof(
float));
157 ref_pos += band_samples;
161 FFSWAP(
float*,
su->spectrum[0],
su->spectrum[1]);
172 int log2_block_count_tmp,
i;
174 for (
i = 0;
i < 2;
i++) {
176 log2_block_count_tmp =
get_bits(gb, 2);
177 if (log2_block_count_tmp & 1)
179 log2_block_cnt[
i] = 2 - log2_block_count_tmp;
183 log2_block_count_tmp =
get_bits(gb, 2);
184 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3)
196 int bits_used, band_num, bfu_num,
i;
206 bits_used =
su->num_bfus * 10 + 32 +
211 for (
i = 0;
i <
su->num_bfus;
i++)
215 for (
i = 0;
i <
su->num_bfus;
i++)
220 idwls[
i] = idsfs[
i] = 0;
228 int word_len = !!idwls[bfu_num] + idwls[bfu_num];
230 bits_used += word_len * num_specs;
240 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1);
242 for (
i = 0;
i < num_specs;
i++) {
246 spec[
pos+
i] =
get_sbits(gb, word_len) * scale_factor * max_quant;
249 memset(&spec[
pos], 0, num_specs *
sizeof(
float));
261 float iqmf_temp[512 + 46];
267 memcpy(
su->last_qmf_delay, &
su->last_qmf_delay[256],
sizeof(
float) * 39);
268 memcpy(&
su->last_qmf_delay[39], q->
bands[2],
sizeof(
float) * 256);
276 int *got_frame_ptr,
AVPacket *avpkt)
280 int buf_size = avpkt->
size;
286 if (buf_size < 212 * avctx->
channels) {
296 for (ch = 0; ch < avctx->
channels; ch++) {
390 .priv_data_size =
sizeof(
AT1Ctx),
static enum AVSampleFormat sample_fmts[]
static float win(SuperEqualizerContext *s, float n, int N)
static av_cold int atrac1_decode_end(AVCodecContext *avctx)
static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec)
static const uint16_t samples_per_band[3]
size of the transform in samples in the long mode for each QMF band
static int atrac1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx *su, float *pOut)
static int at1_imdct_block(AT1SUCtx *su, AT1Ctx *q)
static int at1_parse_bsm(GetBitContext *gb, int log2_block_cnt[AT1_QMF_BANDS])
Parse the block size mode byte.
static int at1_unpack_dequant(GetBitContext *gb, AT1SUCtx *su, float spec[AT1_SU_SAMPLES])
static const uint8_t mdct_long_nbits[3]
#define AT1_MAX_BFU
max number of block floating units in a sound unit
AVCodec ff_atrac1_decoder
static av_cold int atrac1_decode_init(AVCodecContext *avctx)
#define AT1_SU_SAMPLES
number of samples in a sound unit
ATRAC1 compatible decoder data.
static const uint16_t bfu_start_short[52]
start position of each BFU in the MDCT spectrum for the short mode
static const uint8_t bfu_amount_tab2[4]
static const uint8_t bfu_amount_tab1[8]
static const uint16_t bfu_start_long[52]
start position of each BFU in the MDCT spectrum for the long mode
static const uint8_t specs_per_bfu[52]
number of spectral lines in each BFU block floating unit = group of spectral frequencies having the s...
static const uint8_t bfu_amount_tab3[8]
static const uint8_t bfu_bands_t[4]
number of BFUs in each QMF band
float ff_atrac_sf_table[64]
av_cold void ff_atrac_generate_tables(void)
Generate common tables.
void ff_atrac_iqmf(float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
Quadrature mirror synthesis filter.
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
#define FFSWAP(type, a, b)
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
bitstream reader API header.
static int get_sbits(GetBitContext *s, int n)
static void skip_bits(GetBitContext *s, int n)
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
AVSampleFormat
Audio sample formats.
@ AV_SAMPLE_FMT_FLTP
float, planar
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
typedef void(RENAME(mix_any_func_type))
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
The atrac1 context, holds all needed parameters for decoding.
AT1SUCtx SUs[AT1_MAX_CHANNELS]
channel sound unit
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
float spec[AT1_SU_SAMPLES]
the mdct spectrum buffer
Sound unit struct, one unit is used per channel.
float fst_qmf_delay[46]
delay line for the 1st stacked QMF filter
float spec1[AT1_SU_SAMPLES]
mdct buffer
int num_bfus
number of Block Floating Units
float snd_qmf_delay[46]
delay line for the 2nd stacked QMF filter
float spec2[AT1_SU_SAMPLES]
mdct buffer
int log2_block_count[AT1_QMF_BANDS]
log2 number of blocks in a band
float last_qmf_delay[256+39]
delay line for the last stacked QMF filter
main external API structure.
enum AVSampleFormat sample_fmt
audio sample format
int flags
AV_CODEC_FLAG_*.
int channels
number of audio channels
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
const char * name
Name of the codec implementation.
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
This structure describes decoded (raw) audio or video data.
int nb_samples
number of audio samples (per channel) described by this frame
uint8_t ** extended_data
pointers to the data planes/channels.
This structure stores compressed data.
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)