[FFmpeg-cvslog] r19796 - in trunk/libavcodec: Makefile atrac.c atrac.h atrac3.c
banan
subversion
Tue Sep 8 21:25:54 CEST 2009
Author: banan
Date: Tue Sep 8 21:25:54 2009
New Revision: 19796
Log:
Split out common routines needed in the atrac1 decoder from atrac3.c to atrac.c.
Added:
trunk/libavcodec/atrac.c (contents, props changed)
- copied, changed from r19791, trunk/libavcodec/atrac3.c
trunk/libavcodec/atrac.h
Modified:
trunk/libavcodec/Makefile
trunk/libavcodec/atrac3.c
Modified: trunk/libavcodec/Makefile
==============================================================================
--- trunk/libavcodec/Makefile Tue Sep 8 11:11:56 2009 (r19795)
+++ trunk/libavcodec/Makefile Tue Sep 8 21:25:54 2009 (r19796)
@@ -48,7 +48,7 @@ OBJS-$(CONFIG_ASV1_DECODER) +
OBJS-$(CONFIG_ASV1_ENCODER) += asv1.o mpeg12data.o
OBJS-$(CONFIG_ASV2_DECODER) += asv1.o mpeg12data.o
OBJS-$(CONFIG_ASV2_ENCODER) += asv1.o mpeg12data.o
-OBJS-$(CONFIG_ATRAC3_DECODER) += atrac3.o
+OBJS-$(CONFIG_ATRAC3_DECODER) += atrac3.o atrac.o
OBJS-$(CONFIG_AVS_DECODER) += avs.o
OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o
OBJS-$(CONFIG_BFI_DECODER) += bfi.o
Copied and modified: trunk/libavcodec/atrac.c (from r19791, trunk/libavcodec/atrac3.c)
==============================================================================
--- trunk/libavcodec/atrac3.c Mon Sep 7 12:49:51 2009 (r19791, copy source)
+++ trunk/libavcodec/atrac.c Tue Sep 8 21:25:54 2009 (r19796)
@@ -1,5 +1,5 @@
/*
- * Atrac 3 compatible decoder
+ * Atrac common functions
* Copyright (c) 2006-2008 Maxim Poliakovski
* Copyright (c) 2006-2008 Benjamin Larsson
*
@@ -21,15 +21,7 @@
*/
/**
- * @file libavcodec/atrac3.c
- * Atrac 3 compatible decoder.
- * 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 in the containers above.
+ * @file libavcodec/atrac.c
*/
#include <math.h>
@@ -37,98 +29,42 @@
#include <stdio.h>
#include "avcodec.h"
-#include "get_bits.h"
#include "dsputil.h"
-#include "bytestream.h"
-
-#include "atrac3data.h"
-#define JOINT_STEREO 0x12
-#define STEREO 0x2
-
-
-/* These structures are needed to store the parsed gain control data. */
-typedef struct {
- int num_gain_data;
- int levcode[8];
- int loccode[8];
-} gain_info;
-
-typedef struct {
- gain_info gBlock[4];
-} gain_block;
-
-typedef struct {
- int pos;
- int numCoefs;
- float coef[8];
-} tonal_component;
-
-typedef struct {
- int bandsCoded;
- int numComponents;
- tonal_component components[64];
- float prevFrame[1024];
- int gcBlkSwitch;
- gain_block gainBlock[2];
-
- DECLARE_ALIGNED_16(float, spectrum[1024]);
- DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);
+float sf_table[64];
+float qmf_window[48];
- float delayBuf1[46]; ///<qmf delay buffers
- float delayBuf2[46];
- float delayBuf3[46];
-} channel_unit;
+static const float qmf_48tap_half[24] = {
+ -0.00001461907, -0.00009205479,-0.000056157569,0.00030117269,
+ 0.0002422519, -0.00085293897,-0.0005205574, 0.0020340169,
+ 0.00078333891, -0.0042153862, -0.00075614988, 0.0078402944,
+ -0.000061169922,-0.01344162, 0.0024626821, 0.021736089,
+ -0.007801671, -0.034090221, 0.01880949, 0.054326009,
+ -0.043596379, -0.099384367, 0.13207909, 0.46424159
+};
-typedef struct {
- GetBitContext gb;
- //@{
- /** stream data */
- int channels;
- int codingMode;
- int bit_rate;
- int sample_rate;
- int samples_per_channel;
- int samples_per_frame;
+/**
+ * Generate common tables
+ */
- int bits_per_frame;
- int bytes_per_frame;
- int pBs;
- channel_unit* pUnits;
- //@}
- //@{
- /** joint-stereo related variables */
- int matrix_coeff_index_prev[4];
- int matrix_coeff_index_now[4];
- int matrix_coeff_index_next[4];
- int weighting_delay[6];
- //@}
- //@{
- /** data buffers */
- float outSamples[2048];
- uint8_t* decoded_bytes_buffer;
- float tempBuf[1070];
- //@}
- //@{
- /** extradata */
- int atrac3version;
- int delay;
- int scrambled_stream;
- int frame_factor;
- //@}
-} ATRAC3Context;
+void atrac_generate_tables(void)
+{
+ int i;
+ float s;
-static DECLARE_ALIGNED_16(float,mdct_window[512]);
-static float qmf_window[48];
-static VLC spectral_coeff_tab[7];
-static float SFTable[64];
-static float gain_tab1[16];
-static float gain_tab2[31];
-static MDCTContext mdct_ctx;
-static DSPContext dsp;
+ /* Generate scale factors */
+ if (!sf_table[63])
+ for (i=0 ; i<64 ; i++)
+ sf_table[i] = pow(2.0, (i - 15) / 3.0);
+ /* Generate the QMF window. */
+ if (!qmf_window[47])
+ for (i=0 ; i<24; i++) {
+ s = qmf_48tap_half[i] * 2.0;
+ qmf_window[i] = qmf_window[47 - i] = s;
+ }
+}
-/* quadrature mirror synthesis filter */
/**
* Quadrature mirror synthesis filter.
@@ -142,7 +78,7 @@ static DSPContext dsp;
*/
-static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
+void atrac_iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
{
int i, j;
float *p1, *p3;
@@ -181,909 +117,3 @@ static void iqmf (float *inlo, float *in
memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
}
-/**
- * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
- * caused by the reverse spectra of the QMF.
- *
- * @param pInput float input
- * @param pOutput float output
- * @param odd_band 1 if the band is an odd band
- */
-
-static void IMLT(float *pInput, float *pOutput, int odd_band)
-{
- int i;
-
- if (odd_band) {
- /**
- * Reverse the odd bands before IMDCT, this is an effect of the QMF transform
- * or it gives better compression to do it this way.
- * FIXME: It should be possible to handle this in ff_imdct_calc
- * for that to happen a modification of the prerotation step of
- * all SIMD code and C code is needed.
- * Or fix the functions before so they generate a pre reversed spectrum.
- */
-
- for (i=0; i<128; i++)
- FFSWAP(float, pInput[i], pInput[255-i]);
- }
-
- ff_imdct_calc(&mdct_ctx,pOutput,pInput);
-
- /* Perform windowing on the output. */
- dsp.vector_fmul(pOutput,mdct_window,512);
-
-}
-
-
-/**
- * Atrac 3 indata descrambling, only used for data coming from the rm container
- *
- * @param in pointer to 8 bit array of indata
- * @param bits amount of bits
- * @param out pointer to 8 bit array of outdata
- */
-
-static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
- int i, off;
- uint32_t c;
- const uint32_t* buf;
- uint32_t* obuf = (uint32_t*) out;
-
- off = (intptr_t)inbuffer & 3;
- buf = (const uint32_t*) (inbuffer - off);
- c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
- bytes += 3 + off;
- for (i = 0; i < bytes/4; i++)
- obuf[i] = c ^ buf[i];
-
- if (off)
- av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
-
- return off;
-}
-
-
-static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
- float enc_window[256];
- float s;
- int i;
-
- /* Generate the mdct window, for details see
- * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
- for (i=0 ; i<256; i++)
- enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
-
- if (!mdct_window[0])
- for (i=0 ; i<256; i++) {
- mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
- mdct_window[511-i] = mdct_window[i];
- }
-
- /* Generate the QMF window. */
- for (i=0 ; i<24; i++) {
- s = qmf_48tap_half[i] * 2.0;
- qmf_window[i] = s;
- qmf_window[47 - i] = s;
- }
-
- /* Initialize the MDCT transform. */
- ff_mdct_init(&mdct_ctx, 9, 1, 1.0);
-}
-
-/**
- * Atrac3 uninit, free all allocated memory
- */
-
-static av_cold int atrac3_decode_close(AVCodecContext *avctx)
-{
- ATRAC3Context *q = avctx->priv_data;
-
- av_free(q->pUnits);
- av_free(q->decoded_bytes_buffer);
-
- return 0;
-}
-
-/**
-/ * Mantissa decoding
- *
- * @param gb the GetBit context
- * @param selector what table is the output values coded with
- * @param codingFlag constant length coding or variable length coding
- * @param mantissas mantissa output table
- * @param numCodes amount of values to get
- */
-
-static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
-{
- int numBits, cnt, code, huffSymb;
-
- if (selector == 1)
- numCodes /= 2;
-
- if (codingFlag != 0) {
- /* constant length coding (CLC) */
- numBits = CLCLengthTab[selector];
-
- if (selector > 1) {
- for (cnt = 0; cnt < numCodes; cnt++) {
- if (numBits)
- code = get_sbits(gb, numBits);
- else
- code = 0;
- mantissas[cnt] = code;
- }
- } else {
- for (cnt = 0; cnt < numCodes; cnt++) {
- if (numBits)
- code = get_bits(gb, numBits); //numBits is always 4 in this case
- else
- code = 0;
- mantissas[cnt*2] = seTab_0[code >> 2];
- mantissas[cnt*2+1] = seTab_0[code & 3];
- }
- }
- } else {
- /* variable length coding (VLC) */
- if (selector != 1) {
- for (cnt = 0; cnt < numCodes; cnt++) {
- huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
- huffSymb += 1;
- code = huffSymb >> 1;
- if (huffSymb & 1)
- code = -code;
- mantissas[cnt] = code;
- }
- } else {
- for (cnt = 0; cnt < numCodes; cnt++) {
- huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
- mantissas[cnt*2] = decTable1[huffSymb*2];
- mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
- }
- }
- }
-}
-
-/**
- * Restore the quantized band spectrum coefficients
- *
- * @param gb the GetBit context
- * @param pOut decoded band spectrum
- * @return outSubbands subband counter, fix for broken specification/files
- */
-
-static int decodeSpectrum (GetBitContext *gb, float *pOut)
-{
- int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
- int subband_vlc_index[32], SF_idxs[32];
- int mantissas[128];
- float SF;
-
- numSubbands = get_bits(gb, 5); // number of coded subbands
- codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
-
- /* Get the VLC selector table for the subbands, 0 means not coded. */
- for (cnt = 0; cnt <= numSubbands; cnt++)
- subband_vlc_index[cnt] = get_bits(gb, 3);
-
- /* Read the scale factor indexes from the stream. */
- for (cnt = 0; cnt <= numSubbands; cnt++) {
- if (subband_vlc_index[cnt] != 0)
- SF_idxs[cnt] = get_bits(gb, 6);
- }
-
- for (cnt = 0; cnt <= numSubbands; cnt++) {
- first = subbandTab[cnt];
- last = subbandTab[cnt+1];
-
- subbWidth = last - first;
-
- if (subband_vlc_index[cnt] != 0) {
- /* Decode spectral coefficients for this subband. */
- /* TODO: This can be done faster is several blocks share the
- * same VLC selector (subband_vlc_index) */
- readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
-
- /* Decode the scale factor for this subband. */
- SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
-
- /* Inverse quantize the coefficients. */
- for (pIn=mantissas ; first<last; first++, pIn++)
- pOut[first] = *pIn * SF;
- } else {
- /* This subband was not coded, so zero the entire subband. */
- memset(pOut+first, 0, subbWidth*sizeof(float));
- }
- }
-
- /* Clear the subbands that were not coded. */
- first = subbandTab[cnt];
- memset(pOut+first, 0, (1024 - first) * sizeof(float));
- return numSubbands;
-}
-
-/**
- * Restore the quantized tonal components
- *
- * @param gb the GetBit context
- * @param pComponent tone component
- * @param numBands amount of coded bands
- */
-
-static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
-{
- int i,j,k,cnt;
- int components, coding_mode_selector, coding_mode, coded_values_per_component;
- int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
- int band_flags[4], mantissa[8];
- float *pCoef;
- float scalefactor;
- int component_count = 0;
-
- components = get_bits(gb,5);
-
- /* no tonal components */
- if (components == 0)
- return 0;
-
- coding_mode_selector = get_bits(gb,2);
- if (coding_mode_selector == 2)
- return -1;
-
- coding_mode = coding_mode_selector & 1;
-
- for (i = 0; i < components; i++) {
- for (cnt = 0; cnt <= numBands; cnt++)
- band_flags[cnt] = get_bits1(gb);
-
- coded_values_per_component = get_bits(gb,3);
-
- quant_step_index = get_bits(gb,3);
- if (quant_step_index <= 1)
- return -1;
-
- if (coding_mode_selector == 3)
- coding_mode = get_bits1(gb);
-
- for (j = 0; j < (numBands + 1) * 4; j++) {
- if (band_flags[j >> 2] == 0)
- continue;
-
- coded_components = get_bits(gb,3);
-
- for (k=0; k<coded_components; k++) {
- sfIndx = get_bits(gb,6);
- pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
- max_coded_values = 1024 - pComponent[component_count].pos;
- coded_values = coded_values_per_component + 1;
- coded_values = FFMIN(max_coded_values,coded_values);
-
- scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];
-
- readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
-
- pComponent[component_count].numCoefs = coded_values;
-
- /* inverse quant */
- pCoef = pComponent[component_count].coef;
- for (cnt = 0; cnt < coded_values; cnt++)
- pCoef[cnt] = mantissa[cnt] * scalefactor;
-
- component_count++;
- }
- }
- }
-
- return component_count;
-}
-
-/**
- * Decode gain parameters for the coded bands
- *
- * @param gb the GetBit context
- * @param pGb the gainblock for the current band
- * @param numBands amount of coded bands
- */
-
-static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
-{
- int i, cf, numData;
- int *pLevel, *pLoc;
-
- gain_info *pGain = pGb->gBlock;
-
- for (i=0 ; i<=numBands; i++)
- {
- numData = get_bits(gb,3);
- pGain[i].num_gain_data = numData;
- pLevel = pGain[i].levcode;
- pLoc = pGain[i].loccode;
-
- for (cf = 0; cf < numData; cf++){
- pLevel[cf]= get_bits(gb,4);
- pLoc [cf]= get_bits(gb,5);
- if(cf && pLoc[cf] <= pLoc[cf-1])
- return -1;
- }
- }
-
- /* Clear the unused blocks. */
- for (; i<4 ; i++)
- pGain[i].num_gain_data = 0;
-
- return 0;
-}
-
-/**
- * Apply gain parameters and perform the MDCT overlapping part
- *
- * @param pIn input float buffer
- * @param pPrev previous float buffer to perform overlap against
- * @param pOut output float buffer
- * @param pGain1 current band gain info
- * @param pGain2 next band gain info
- */
-
-static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
-{
- /* gain compensation function */
- float gain1, gain2, gain_inc;
- int cnt, numdata, nsample, startLoc, endLoc;
-
-
- if (pGain2->num_gain_data == 0)
- gain1 = 1.0;
- else
- gain1 = gain_tab1[pGain2->levcode[0]];
-
- if (pGain1->num_gain_data == 0) {
- for (cnt = 0; cnt < 256; cnt++)
- pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
- } else {
- numdata = pGain1->num_gain_data;
- pGain1->loccode[numdata] = 32;
- pGain1->levcode[numdata] = 4;
-
- nsample = 0; // current sample = 0
-
- for (cnt = 0; cnt < numdata; cnt++) {
- startLoc = pGain1->loccode[cnt] * 8;
- endLoc = startLoc + 8;
-
- gain2 = gain_tab1[pGain1->levcode[cnt]];
- gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
-
- /* interpolate */
- for (; nsample < startLoc; nsample++)
- pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
-
- /* interpolation is done over eight samples */
- for (; nsample < endLoc; nsample++) {
- pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
- gain2 *= gain_inc;
- }
- }
-
- for (; nsample < 256; nsample++)
- pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
- }
-
- /* Delay for the overlapping part. */
- memcpy(pPrev, &pIn[256], 256*sizeof(float));
-}
-
-/**
- * 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 int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
-{
- 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;
-}
-
-
-#define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
-
-static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
-{
- int i, band, nsample, s1, s2;
- float c1, c2;
- float mc1_l, mc1_r, mc2_l, mc2_r;
-
- for (i=0,band = 0; band < 4*256; band+=256,i++) {
- s1 = pPrevCode[i];
- s2 = pCurrCode[i];
- nsample = 0;
-
- if (s1 != s2) {
- /* Selector value changed, interpolation needed. */
- mc1_l = matrixCoeffs[s1*2];
- mc1_r = matrixCoeffs[s1*2+1];
- mc2_l = matrixCoeffs[s2*2];
- mc2_r = matrixCoeffs[s2*2+1];
-
- /* Interpolation is done over the first eight samples. */
- for(; nsample < 8; nsample++) {
- c1 = su1[band+nsample];
- c2 = su2[band+nsample];
- c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
- su1[band+nsample] = c2;
- su2[band+nsample] = c1 * 2.0 - c2;
- }
- }
-
- /* Apply the matrix without interpolation. */
- switch (s2) {
- case 0: /* M/S decoding */
- for (; nsample < 256; nsample++) {
- c1 = su1[band+nsample];
- c2 = su2[band+nsample];
- su1[band+nsample] = c2 * 2.0;
- su2[band+nsample] = (c1 - c2) * 2.0;
- }
- break;
-
- case 1:
- for (; nsample < 256; nsample++) {
- c1 = su1[band+nsample];
- c2 = su2[band+nsample];
- su1[band+nsample] = (c1 + c2) * 2.0;
- su2[band+nsample] = c2 * -2.0;
- }
- break;
- case 2:
- case 3:
- for (; nsample < 256; nsample++) {
- c1 = su1[band+nsample];
- c2 = su2[band+nsample];
- su1[band+nsample] = c1 + c2;
- su2[band+nsample] = c1 - c2;
- }
- break;
- default:
- assert(0);
- }
- }
-}
-
-static void getChannelWeights (int indx, int flag, float ch[2]){
-
- if (indx == 7) {
- ch[0] = 1.0;
- ch[1] = 1.0;
- } else {
- ch[0] = (float)(indx & 7) / 7.0;
- ch[1] = sqrt(2 - ch[0]*ch[0]);
- if(flag)
- FFSWAP(float, ch[0], ch[1]);
- }
-}
-
-static void channelWeighting (float *su1, float *su2, int *p3)
-{
- int band, nsample;
- /* w[x][y] y=0 is left y=1 is right */
- float w[2][2];
-
- if (p3[1] != 7 || p3[3] != 7){
- getChannelWeights(p3[1], p3[0], w[0]);
- getChannelWeights(p3[3], p3[2], w[1]);
-
- for(band = 1; band < 4; band++) {
- /* scale the channels by the weights */
- for(nsample = 0; nsample < 8; nsample++) {
- su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
- su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
- }
-
- for(; nsample < 256; nsample++) {
- su1[band*256+nsample] *= w[1][0];
- su2[band*256+nsample] *= w[1][1];
- }
- }
- }
-}
-
-
-/**
- * Decode a Sound Unit
- *
- * @param gb the GetBit context
- * @param pSnd the channel unit to be used
- * @param pOut the decoded samples before IQMF in float representation
- * @param channelNum channel number
- * @param codingMode the coding mode (JOINT_STEREO or regular stereo/mono)
- */
-
-
-static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
-{
- int band, result=0, numSubbands, lastTonal, numBands;
-
- if (codingMode == JOINT_STEREO && channelNum == 1) {
- if (get_bits(gb,2) != 3) {
- av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
- return -1;
- }
- } else {
- if (get_bits(gb,6) != 0x28) {
- av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
- return -1;
- }
- }
-
- /* number of coded QMF bands */
- pSnd->bandsCoded = get_bits(gb,2);
-
- result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
- if (result) return result;
-
- pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
- if (pSnd->numComponents == -1) return -1;
-
- numSubbands = decodeSpectrum (gb, pSnd->spectrum);
-
- /* Merge the decoded spectrum and tonal components. */
- lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
-
-
- /* 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);
- } else
- memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
-
- /* gain compensation and overlapping */
- gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
- &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
- &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
- }
-
- /* Swap the gain control buffers for the next frame. */
- pSnd->gcBlkSwitch ^= 1;
-
- return 0;
-}
-
-/**
- * Frame handling
- *
- * @param q Atrac3 private context
- * @param databuf the input data
- */
-
-static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
-{
- int result, i;
- float *p1, *p2, *p3, *p4;
- uint8_t *ptr1;
-
- if (q->codingMode == JOINT_STEREO) {
-
- /* channel coupling mode */
- /* decode Sound Unit 1 */
- init_get_bits(&q->gb,databuf,q->bits_per_frame);
-
- result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
- if (result != 0)
- return (result);
-
- /* Framedata of the su2 in the joint-stereo mode is encoded in
- * reverse byte order so we need to swap it first. */
- 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 = q->decoded_bytes_buffer;
- for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
- if (i >= q->bytes_per_frame)
- return -1;
- }
-
-
- /* set the bitstream reader at the start of the second Sound Unit*/
- init_get_bits(&q->gb,ptr1,q->bits_per_frame);
-
- /* Fill the Weighting coeffs delay buffer */
- memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
- q->weighting_delay[4] = get_bits1(&q->gb);
- q->weighting_delay[5] = get_bits(&q->gb,3);
-
- for (i = 0; i < 4; i++) {
- q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
- q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
- q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
- }
-
- /* Decode Sound Unit 2. */
- result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
- if (result != 0)
- return (result);
-
- /* Reconstruct the channel coefficients. */
- reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
-
- channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
-
- } else {
- /* normal stereo mode or mono */
- /* Decode the channel sound units. */
- for (i=0 ; i<q->channels ; i++) {
-
- /* Set the bitstream reader at the start of a channel sound unit. */
- init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
-
- result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
- if (result != 0)
- return (result);
- }
- }
-
- /* Apply the iQMF synthesis filter. */
- p1= q->outSamples;
- for (i=0 ; i<q->channels ; i++) {
- p2= p1+256;
- p3= p2+256;
- p4= p3+256;
- iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
- iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
- iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
- p1 +=1024;
- }
-
- return 0;
-}
-
-
-/**
- * Atrac frame decoding
- *
- * @param avctx pointer to the AVCodecContext
- */
-
-static int atrac3_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt) {
- const uint8_t *buf = avpkt->data;
- int buf_size = avpkt->size;
- ATRAC3Context *q = avctx->priv_data;
- int result = 0, i;
- const uint8_t* databuf;
- int16_t* samples = data;
-
- if (buf_size < avctx->block_align)
- return buf_size;
-
- /* Check if we need to descramble and what buffer to pass on. */
- if (q->scrambled_stream) {
- decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
- databuf = q->decoded_bytes_buffer;
- } else {
- databuf = buf;
- }
-
- result = decodeFrame(q, databuf);
-
- if (result != 0) {
- av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
- return -1;
- }
-
- if (q->channels == 1) {
- /* mono */
- for (i = 0; i<1024; i++)
- samples[i] = av_clip_int16(round(q->outSamples[i]));
- *data_size = 1024 * sizeof(int16_t);
- } else {
- /* stereo */
- for (i = 0; i < 1024; i++) {
- samples[i*2] = av_clip_int16(round(q->outSamples[i]));
- samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
- }
- *data_size = 2048 * sizeof(int16_t);
- }
-
- return avctx->block_align;
-}
-
-
-/**
- * Atrac3 initialization
- *
- * @param avctx pointer to the AVCodecContext
- */
-
-static av_cold int atrac3_decode_init(AVCodecContext *avctx)
-{
- int i;
- const uint8_t *edata_ptr = avctx->extradata;
- ATRAC3Context *q = avctx->priv_data;
- static VLC_TYPE atrac3_vlc_table[4096][2];
- static int vlcs_initialized = 0;
-
- /* Take data from the AVCodecContext (RM container). */
- q->sample_rate = avctx->sample_rate;
- q->channels = avctx->channels;
- q->bit_rate = avctx->bit_rate;
- q->bits_per_frame = avctx->block_align * 8;
- q->bytes_per_frame = avctx->block_align;
-
- /* Take care of the codec-specific extradata. */
- if (avctx->extradata_size == 14) {
- /* Parse the extradata, WAV format */
- av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1
- q->samples_per_channel = bytestream_get_le32(&edata_ptr);
- q->codingMode = bytestream_get_le16(&edata_ptr);
- av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
- q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1
- av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0
-
- /* setup */
- q->samples_per_frame = 1024 * q->channels;
- q->atrac3version = 4;
- q->delay = 0x88E;
- if (q->codingMode)
- q->codingMode = JOINT_STEREO;
- else
- q->codingMode = STEREO;
-
- q->scrambled_stream = 0;
-
- if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) {
- } else {
- av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor);
- return -1;
- }
-
- } else if (avctx->extradata_size == 10) {
- /* Parse the extradata, RM format. */
- q->atrac3version = bytestream_get_be32(&edata_ptr);
- q->samples_per_frame = bytestream_get_be16(&edata_ptr);
- q->delay = bytestream_get_be16(&edata_ptr);
- q->codingMode = bytestream_get_be16(&edata_ptr);
-
- q->samples_per_channel = q->samples_per_frame / q->channels;
- q->scrambled_stream = 1;
-
- } else {
- av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
- }
- /* Check the extradata. */
-
- if (q->atrac3version != 4) {
- av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
- return -1;
- }
-
- if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
- av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
- return -1;
- }
-
- if (q->delay != 0x88E) {
- av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
- return -1;
- }
-
- if (q->codingMode == STEREO) {
- av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
- } else if (q->codingMode == JOINT_STEREO) {
- av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
- } else {
- av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
- return -1;
- }
-
- if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
- av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
- return -1;
- }
-
-
- if(avctx->block_align >= UINT_MAX/2)
- return -1;
-
- /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
- * this is for the bitstream reader. */
- if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE))) == NULL)
- return AVERROR(ENOMEM);
-
-
- /* Initialize the VLC tables. */
- if (!vlcs_initialized) {
- for (i=0 ; i<7 ; i++) {
- spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
- spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i];
- init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
- huff_bits[i], 1, 1,
- huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
- }
- vlcs_initialized = 1;
- }
-
- init_atrac3_transforms(q);
-
- /* Generate the scale factors. */
- for (i=0 ; i<64 ; i++)
- SFTable[i] = pow(2.0, (i - 15) / 3.0);
-
- /* Generate gain tables. */
- for (i=0 ; i<16 ; i++)
- gain_tab1[i] = powf (2.0, (4 - i));
-
- for (i=-15 ; i<16 ; i++)
- gain_tab2[i+15] = powf (2.0, i * -0.125);
-
- /* init the joint-stereo decoding data */
- q->weighting_delay[0] = 0;
- q->weighting_delay[1] = 7;
- q->weighting_delay[2] = 0;
- q->weighting_delay[3] = 7;
- q->weighting_delay[4] = 0;
- q->weighting_delay[5] = 7;
-
- for (i=0; i<4; i++) {
- q->matrix_coeff_index_prev[i] = 3;
- q->matrix_coeff_index_now[i] = 3;
- q->matrix_coeff_index_next[i] = 3;
- }
-
- dsputil_init(&dsp, avctx);
-
- q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
- if (!q->pUnits) {
- av_free(q->decoded_bytes_buffer);
- return AVERROR(ENOMEM);
- }
-
- avctx->sample_fmt = SAMPLE_FMT_S16;
- return 0;
-}
-
-
-AVCodec atrac3_decoder =
-{
- .name = "atrac3",
- .type = CODEC_TYPE_AUDIO,
- .id = CODEC_ID_ATRAC3,
- .priv_data_size = sizeof(ATRAC3Context),
- .init = atrac3_decode_init,
- .close = atrac3_decode_close,
- .decode = atrac3_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
-};
Added: trunk/libavcodec/atrac.h
==============================================================================
--- /dev/null 00:00:00 1970 (empty, because file is newly added)
+++ trunk/libavcodec/atrac.h Tue Sep 8 21:25:54 2009 (r19796)
@@ -0,0 +1,38 @@
+/*
+ * Atrac common data
+ * Copyright (c) 2009 Maxim Poliakovski
+ * Copyright (c) 2009 Benjamin Larsson
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file libavcodec/atrac1data.h
+ * Atrac 1 compatible decoder data
+ */
+
+#ifndef AVCODEC_ATRAC_H
+#define AVCODEC_ATRAC_H
+
+
+extern float sf_table[64];
+extern float qmf_window[48];
+
+void atrac_generate_tables(void);
+void atrac_iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp);
+
+#endif /* AVCODEC_ATRAC_H */
Modified: trunk/libavcodec/atrac3.c
==============================================================================
--- trunk/libavcodec/atrac3.c Tue Sep 8 11:11:56 2009 (r19795)
+++ trunk/libavcodec/atrac3.c Tue Sep 8 21:25:54 2009 (r19796)
@@ -41,6 +41,7 @@
#include "dsputil.h"
#include "bytestream.h"
+#include "atrac.h"
#include "atrac3data.h"
#define JOINT_STEREO 0x12
@@ -119,68 +120,13 @@ typedef struct {
} ATRAC3Context;
static DECLARE_ALIGNED_16(float,mdct_window[512]);
-static float qmf_window[48];
static VLC spectral_coeff_tab[7];
-static float SFTable[64];
static float gain_tab1[16];
static float gain_tab2[31];
static MDCTContext mdct_ctx;
static DSPContext dsp;
-/* quadrature mirror synthesis filter */
-
-/**
- * Quadrature mirror synthesis filter.
- *
- * @param inlo lower part of spectrum
- * @param inhi higher part of spectrum
- * @param nIn size of spectrum buffer
- * @param pOut out buffer
- * @param delayBuf delayBuf buffer
- * @param temp temp buffer
- */
-
-
-static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
-{
- int i, j;
- float *p1, *p3;
-
- memcpy(temp, delayBuf, 46*sizeof(float));
-
- p3 = temp + 46;
-
- /* loop1 */
- for(i=0; i<nIn; i+=2){
- p3[2*i+0] = inlo[i ] + inhi[i ];
- p3[2*i+1] = inlo[i ] - inhi[i ];
- p3[2*i+2] = inlo[i+1] + inhi[i+1];
- p3[2*i+3] = inlo[i+1] - inhi[i+1];
- }
-
- /* loop2 */
- p1 = temp;
- for (j = nIn; j != 0; j--) {
- float s1 = 0.0;
- float s2 = 0.0;
-
- for (i = 0; i < 48; i += 2) {
- s1 += p1[i] * qmf_window[i];
- s2 += p1[i+1] * qmf_window[i+1];
- }
-
- pOut[0] = s2;
- pOut[1] = s1;
-
- p1 += 2;
- pOut += 2;
- }
-
- /* Update the delay buffer. */
- memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
-}
-
/**
* Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands
* caused by the reverse spectra of the QMF.
@@ -386,7 +332,7 @@ static int decodeSpectrum (GetBitContext
readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
/* Decode the scale factor for this subband. */
- SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
+ SF = sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
/* Inverse quantize the coefficients. */
for (pIn=mantissas ; first<last; first++, pIn++)
@@ -459,7 +405,7 @@ static int decodeTonalComponents (GetBit
coded_values = coded_values_per_component + 1;
coded_values = FFMIN(max_coded_values,coded_values);
- scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];
+ scalefactor = sf_table[sfIndx] * iMaxQuant[quant_step_index];
readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
@@ -860,9 +806,9 @@ static int decodeFrame(ATRAC3Context *q,
p2= p1+256;
p3= p2+256;
p4= p3+256;
- iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
- iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
- iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
+ atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
+ atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
+ atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
p1 +=1024;
}
@@ -1038,9 +984,7 @@ static av_cold int atrac3_decode_init(AV
init_atrac3_transforms(q);
- /* Generate the scale factors. */
- for (i=0 ; i<64 ; i++)
- SFTable[i] = pow(2.0, (i - 15) / 3.0);
+ atrac_generate_tables();
/* Generate gain tables. */
for (i=0 ; i<16 ; i++)
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