/*
* Atrac 3 compatible decoder
- * Copyright (c) 2006-2007 Maxim Poliakovski
- * Copyright (c) 2006-2007 Benjamin Larsson
+ * Copyright (c) 2006-2008 Maxim Poliakovski
+ * Copyright (c) 2006-2008 Benjamin Larsson
*
* This file is part of FFmpeg.
*
*/
/**
- * @file atrac3.c
+ * @file libavcodec/atrac3.c
* Atrac 3 compatible decoder.
- * This decoder handles RealNetworks, RealAudio atrc data.
- * Atrac 3 is identified by the codec name atrc in RealMedia files.
+ * This decoder handles Sony's ATRAC3 data.
+ *
+ * Container formats used to store atrac 3 data:
+ * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
*
* To use this decoder, a calling application must supply the extradata
- * bytes provided from the RealMedia container: 10 bytes or 14 bytes
- * from the WAV container.
+ * bytes provided in the containers above.
*/
#include <math.h>
#include <stdio.h>
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "dsputil.h"
#include "bytestream.h"
float outSamples[2048];
uint8_t* decoded_bytes_buffer;
float tempBuf[1070];
- DECLARE_ALIGNED_16(float,mdct_tmp[512]);
//@}
//@{
/** extradata */
* @param pInput float input
* @param pOutput float output
* @param odd_band 1 if the band is an odd band
- * @param mdct_tmp aligned temporary buffer for the mdct
*/
-static void IMLT(float *pInput, float *pOutput, int odd_band, float* mdct_tmp)
+static void IMLT(float *pInput, float *pOutput, int odd_band)
{
int i;
FFSWAP(float, pInput[i], pInput[255-i]);
}
- mdct_ctx.fft.imdct_calc(&mdct_ctx,pOutput,pInput,mdct_tmp);
+ ff_imdct_calc(&mdct_ctx,pOutput,pInput);
/* Perform windowing on the output. */
dsp.vector_fmul(pOutput,mdct_window,512);
const uint32_t* buf;
uint32_t* obuf = (uint32_t*) out;
- off = (int)((long)inbuffer & 3);
+ off = (intptr_t)inbuffer & 3;
buf = (const uint32_t*) (inbuffer - off);
c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
bytes += 3 + off;
}
-static void init_atrac3_transforms(ATRAC3Context *q) {
+static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
float enc_window[256];
float s;
int i;
* Atrac3 uninit, free all allocated memory
*/
-static int atrac3_decode_close(AVCodecContext *avctx)
+static av_cold int atrac3_decode_close(AVCodecContext *avctx)
{
ATRAC3Context *q = avctx->priv_data;
if (codingFlag != 0) {
/* constant length coding (CLC) */
- //FIXME we don't have any samples coded in CLC mode
numBits = CLCLengthTab[selector];
if (selector > 1) {
pComponent[component_count].numCoefs = coded_values;
/* inverse quant */
- pCoef = pComponent[k].coef;
+ pCoef = pComponent[component_count].coef;
for (cnt = 0; cnt < coded_values; cnt++)
pCoef[cnt] = mantissa[cnt] * scalefactor;
/**
* Combine the tonal band spectrum and regular band spectrum
+ * Return position of the last tonal coefficient
*
* @param pSpectrum output spectrum buffer
* @param numComponents amount of tonal components
* @param pComponent tonal components for this band
*/
-static void addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
+static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
{
- int cnt, i;
+ int cnt, i, lastPos = -1;
float *pIn, *pOut;
for (cnt = 0; cnt < numComponents; cnt++){
+ lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
pIn = pComponent[cnt].coef;
pOut = &(pSpectrum[pComponent[cnt].pos]);
for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
pOut[i] += pIn[i];
}
+
+ return lastPos;
}
static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
{
- int band, result=0, numSubbands, numBands;
+ int band, result=0, numSubbands, lastTonal, numBands;
if (codingMode == JOINT_STEREO && channelNum == 1) {
if (get_bits(gb,2) != 3) {
numSubbands = decodeSpectrum (gb, pSnd->spectrum);
/* Merge the decoded spectrum and tonal components. */
- addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
+ lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
- /* Convert number of subbands into number of MLT/QMF bands */
+ /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
numBands = (subbandTab[numSubbands] - 1) >> 8;
+ if (lastTonal >= 0)
+ numBands = FFMAX((lastTonal + 256) >> 8, numBands);
/* Reconstruct time domain samples. */
for (band=0; band<4; band++) {
/* Perform the IMDCT step without overlapping. */
if (band <= numBands) {
- IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1,q->mdct_tmp);
+ IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
} else
memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
* @param databuf the input data
*/
-static int decodeFrame(ATRAC3Context *q, uint8_t* databuf)
+static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
{
int result, i;
float *p1, *p2, *p3, *p4;
- uint8_t *ptr1, *ptr2;
+ uint8_t *ptr1;
if (q->codingMode == JOINT_STEREO) {
/* Framedata of the su2 in the joint-stereo mode is encoded in
* reverse byte order so we need to swap it first. */
- ptr1 = databuf;
- ptr2 = databuf+q->bytes_per_frame-1;
- for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
- FFSWAP(uint8_t,*ptr1,*ptr2);
+ if (databuf == q->decoded_bytes_buffer) {
+ uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
+ ptr1 = q->decoded_bytes_buffer;
+ for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
+ FFSWAP(uint8_t,*ptr1,*ptr2);
+ }
+ } else {
+ const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
+ for (i = 0; i < q->bytes_per_frame; i++)
+ q->decoded_bytes_buffer[i] = *ptr2--;
}
/* Skip the sync codes (0xF8). */
- ptr1 = databuf;
+ ptr1 = q->decoded_bytes_buffer;
for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
if (i >= q->bytes_per_frame)
return -1;
static int atrac3_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
- const uint8_t *buf, int buf_size) {
+ AVPacket *avpkt) {
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
ATRAC3Context *q = avctx->priv_data;
int result = 0, i;
- uint8_t* databuf;
+ const uint8_t* databuf;
int16_t* samples = data;
if (buf_size < avctx->block_align)
* @param avctx pointer to the AVCodecContext
*/
-static int atrac3_decode_init(AVCodecContext *avctx)
+static av_cold int atrac3_decode_init(AVCodecContext *avctx)
{
int i;
const uint8_t *edata_ptr = avctx->extradata;
return AVERROR(ENOMEM);
}
+ avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
.init = atrac3_decode_init,
.close = atrac3_decode_close,
.decode = atrac3_decode_frame,
- .long_name = "Atrac 3 (Adaptive TRansform Acoustic Coding 3)",
+ .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
};
projects / frescor / ffmpeg.git / blobdiff