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-rw-r--r--Changelog1
-rw-r--r--doc/ffmpeg-doc.texi1
-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/allcodecs.c1
-rw-r--r--libavcodec/atrac3.c1069
-rw-r--r--libavcodec/atrac3data.h133
-rw-r--r--libavcodec/avcodec.h2
-rw-r--r--libavformat/riff.c1
-rw-r--r--libavformat/rm.c9
9 files changed, 1215 insertions, 3 deletions
diff --git a/Changelog b/Changelog
index 2f1b6feb3d..729ca5d449 100644
--- a/Changelog
+++ b/Changelog
@@ -80,6 +80,7 @@ version <next>
- Interplay C93 demuxer and video decoder
- Bethsoft VID demuxer and video decoder
- CRYO APC demuxer
+- Atrac3 decoder
version 0.4.9-pre1:
diff --git a/doc/ffmpeg-doc.texi b/doc/ffmpeg-doc.texi
index a33ab56caf..bae8851d4f 100644
--- a/doc/ffmpeg-doc.texi
+++ b/doc/ffmpeg-doc.texi
@@ -1098,6 +1098,7 @@ following image formats are supported:
@item Musepack @tab @tab X
@tab Only SV7 is supported
@item DT$ Coherent Audio @tab @tab X
+@item ATRAC 3 @tab @tab X
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index bebaf71466..6345f64a49 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -53,6 +53,7 @@ OBJS-$(CONFIG_ASV1_DECODER) += asv1.o
OBJS-$(CONFIG_ASV1_ENCODER) += asv1.o
OBJS-$(CONFIG_ASV2_DECODER) += asv1.o
OBJS-$(CONFIG_ASV2_ENCODER) += asv1.o
+OBJS-$(CONFIG_ATRAC3_DECODER) += atrac3.o
OBJS-$(CONFIG_AVS_DECODER) += avs.o
OBJS-$(CONFIG_BETHSOFTVID_DECODER) += bethsoftvideo.o
OBJS-$(CONFIG_BMP_DECODER) += bmp.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 5464dc7abc..99c046457d 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -167,6 +167,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(ALAC, alac);
REGISTER_ENCDEC (AMR_NB, amr_nb);
REGISTER_ENCDEC (AMR_WB, amr_wb);
+ REGISTER_DECODER(ATRAC3, atrac3);
REGISTER_DECODER(COOK, cook);
REGISTER_DECODER(DSICINAUDIO, dsicinaudio);
REGISTER_DECODER(DTS, dts);
diff --git a/libavcodec/atrac3.c b/libavcodec/atrac3.c
new file mode 100644
index 0000000000..4d676c1494
--- /dev/null
+++ b/libavcodec/atrac3.c
@@ -0,0 +1,1069 @@
+/*
+ * Atrac 3 compatible decoder
+ * Copyright (c) 2006-2007 Maxim Poliakovski
+ * Copyright (c) 2006-2007 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 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.
+ *
+ * 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.
+ */
+
+#include <math.h>
+#include <stddef.h>
+#include <stdio.h>
+
+#include "avcodec.h"
+#include "bitstream.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 delayBuf1[46]; ///<qmf delay buffers
+ float delayBuf2[46];
+ float delayBuf3[46];
+} channel_unit;
+
+typedef struct {
+ GetBitContext gb;
+ //@{
+ /** stream data */
+ int channels;
+ int codingMode;
+ int bit_rate;
+ int sample_rate;
+ int samples_per_channel;
+ int samples_per_frame;
+
+ 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];
+ DECLARE_ALIGNED_16(float,mdct_tmp[512]);
+ //@}
+ //@{
+ /** extradata */
+ int atrac3version;
+ int delay;
+ int scrambled_stream;
+ int frame_factor;
+ //@}
+} 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.
+ *
+ * @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)
+{
+ 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]);
+ }
+
+ mdct_ctx.fft.imdct_calc(&mdct_ctx,pOutput,pInput,mdct_tmp);
+
+ /* 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(uint8_t* inbuffer, uint8_t* out, int bytes){
+ int i, off;
+ uint32_t c;
+ uint32_t* buf;
+ uint32_t* obuf = (uint32_t*) out;
+
+ off = (int)((long)inbuffer & 3);
+ buf = (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 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);
+}
+
+/**
+ * Atrac3 uninit, free all allocated memory
+ */
+
+static 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) */
+ //FIXME we don't have any samples coded in CLC mode
+ 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_bits(gb, 1); // 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 numComponents tonal components to report back
+ * @param pComponent tone component
+ * @param numBands amount of coded bands
+ */
+
+static int decodeTonalComponents (GetBitContext *gb, int *numComponents, tonal_component *pComponent, int numBands)
+{
+ int i,j,k,cnt;
+ int component_count, 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;
+
+ component_count = 0;
+ *numComponents = 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[k].coef;
+ for (cnt = 0; cnt < coded_values; cnt++)
+ pCoef[cnt] = mantissa[cnt] * scalefactor;
+
+ component_count++;
+ }
+ }
+ }
+
+ *numComponents = component_count;
+
+ return 0;
+}
+
+/**
+ * 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
+ *
+ * @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)
+{
+ int cnt, i;
+ float *pIn, *pOut;
+
+ for (cnt = 0; cnt < numComponents; cnt++){
+ pIn = pComponent[cnt].coef;
+ pOut = &(pSpectrum[pComponent[cnt].pos]);
+
+ for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
+ pOut[i] += pIn[i];
+ }
+}
+
+
+#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, 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;
+
+ result = decodeTonalComponents (gb, &pSnd->numComponents, pSnd->components, pSnd->bandsCoded);
+ if (result) return result;
+
+ numSubbands = decodeSpectrum (gb, pSnd->spectrum);
+
+ /* Merge the decoded spectrum and tonal components. */
+ addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
+
+
+ /* Convert number of subbands into number of MLT/QMF bands */
+ numBands = (subbandTab[numSubbands] - 1) >> 8;
+
+
+ /* 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);
+ } 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, uint8_t* databuf)
+{
+ int result, i;
+ float *p1, *p2, *p3, *p4;
+ uint8_t *ptr1, *ptr2;
+
+ 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. */
+ 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);
+ }
+
+ /* Skip the sync codes (0xF8). */
+ ptr1 = databuf;
+ 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_bits(&q->gb,1);
+ 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,
+ uint8_t *buf, int buf_size) {
+ ATRAC3Context *q = avctx->priv_data;
+ int result = 0, i;
+ 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(round(q->outSamples[i]), -32768, 32767);
+ *data_size = 1024 * sizeof(int16_t);
+ } else {
+ /* stereo */
+ for (i = 0; i < 1024; i++) {
+ samples[i*2] = av_clip(round(q->outSamples[i]), -32768, 32767);
+ samples[i*2+1] = av_clip(round(q->outSamples[1024+i]), -32768, 32767);
+ }
+ *data_size = 2048 * sizeof(int16_t);
+ }
+
+ return avctx->block_align;
+}
+
+
+/**
+ * Atrac3 initialization
+ *
+ * @param avctx pointer to the AVCodecContext
+ */
+
+static int atrac3_decode_init(AVCodecContext *avctx)
+{
+ int i;
+ uint8_t *edata_ptr = avctx->extradata;
+ ATRAC3Context *q = avctx->priv_data;
+
+ /* 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 -1;
+
+
+ /* Initialize the VLC tables. */
+ for (i=0 ; i<7 ; 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_STATIC);
+ }
+
+ 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);
+
+ return 0;
+}
+
+
+AVCodec atrac3_decoder =
+{
+ .name = "atrac 3",
+ .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,
+};
diff --git a/libavcodec/atrac3data.h b/libavcodec/atrac3data.h
new file mode 100644
index 0000000000..0017d60c06
--- /dev/null
+++ b/libavcodec/atrac3data.h
@@ -0,0 +1,133 @@
+/*
+ * Atrac 3 compatible decoder data
+ * Copyright (c) 2006-2007 Maxim Poliakovski
+ * Copyright (c) 2006-2007 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 atrac3data.h
+ * Atrac 3 AKA RealAudio 8 compatible decoder data
+ */
+
+/* VLC tables */
+
+static const uint8_t huffcode1[9] = {
+ 0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
+};
+
+static const uint8_t huffbits1[9] = {
+ 1,3,3,4,4,5,5,5,5,
+};
+
+static const uint8_t huffcode2[5] = {
+ 0x0,0x4,0x5,0x6,0x7,
+};
+
+static const uint8_t huffbits2[5] = {
+ 1,3,3,3,3,
+};
+
+static const uint8_t huffcode3[7] = {
+0x0,0x4,0x5,0xC,0xD,0xE,0xF,
+};
+
+static const uint8_t huffbits3[7] = {
+ 1,3,3,4,4,4,4,
+};
+
+static const uint8_t huffcode4[9] = {
+ 0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
+};
+
+static const uint8_t huffbits4[9] = {
+ 1,3,3,4,4,5,5,5,5,
+};
+
+static const uint8_t huffcode5[15] = {
+ 0x0,0x2,0x3,0x8,0x9,0xA,0xB,0x1C,0x1D,0x3C,0x3D,0x3E,0x3F,0xC,0xD,
+};
+
+static const uint8_t huffbits5[15] = {
+ 2,3,3,4,4,4,4,5,5,6,6,6,6,4,4
+};
+
+static const uint8_t huffcode6[31] = {
+ 0x0,0x2,0x3,0x4,0x5,0x6,0x7,0x14,0x15,0x16,0x17,0x18,0x19,0x34,0x35,
+ 0x36,0x37,0x38,0x39,0x3A,0x3B,0x78,0x79,0x7A,0x7B,0x7C,0x7D,0x7E,0x7F,0x8,0x9,
+};
+
+static const uint8_t huffbits6[31] = {
+ 3,4,4,4,4,4,4,5,5,5,5,5,5,6,6,6,6,6,6,6,6,7,7,7,7,7,7,7,7,4,4
+};
+
+static const uint8_t huffcode7[63] = {
+ 0x0,0x8,0x9,0xA,0xB,0xC,0xD,0xE,0xF,0x10,0x11,0x24,0x25,0x26,0x27,0x28,
+ 0x29,0x2A,0x2B,0x2C,0x2D,0x2E,0x2F,0x30,0x31,0x32,0x33,0x68,0x69,0x6A,0x6B,0x6C,
+ 0x6D,0x6E,0x6F,0x70,0x71,0x72,0x73,0x74,0x75,0xEC,0xED,0xEE,0xEF,0xF0,0xF1,0xF2,
+ 0xF3,0xF4,0xF5,0xF6,0xF7,0xF8,0xF9,0xFA,0xFB,0xFC,0xFD,0xFE,0xFF,0x2,0x3,
+};
+
+static const uint8_t huffbits7[63] = {
+ 3,5,5,5,5,5,5,5,5,5,5,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,4,4
+};
+
+static const uint8_t huff_tab_sizes[7] = {
+ 9, 5, 7, 9, 15, 31, 63,
+};
+
+static const uint8_t* huff_codes[7] = {
+ huffcode1,huffcode2,huffcode3,huffcode4,huffcode5,huffcode6,huffcode7,
+};
+
+static const uint8_t* huff_bits[7] = {
+ huffbits1,huffbits2,huffbits3,huffbits4,huffbits5,huffbits6,huffbits7,
+};
+
+/* selector tables */
+
+static const uint8_t CLCLengthTab[8] = {0, 4, 3, 3, 4, 4, 5, 6};
+static const int8_t seTab_0[4] = {0, 1, -2, -1};
+static const int8_t decTable1[18] = {0,0, 0,1, 0,-1, 1,0, -1,0, 1,1, 1,-1, -1,1, -1,-1};
+
+
+/* tables for the scalefactor decoding */
+
+static const float iMaxQuant[8] = {
+ 0.0, 1.0/1.5, 1.0/2.5, 1.0/3.5, 1.0/4.5, 1.0/7.5, 1.0/15.5, 1.0/31.5
+};
+
+static const uint16_t subbandTab[33] = {
+ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224,
+ 256, 288, 320, 352, 384, 416, 448, 480, 512, 576, 640, 704, 768, 896, 1024
+};
+
+/* transform data */
+
+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
+};
+
+/* joint stereo related tables */
+static const float matrixCoeffs[8] = {0.0, 2.0, 2.0, 2.0, 0.0, 0.0, 1.0, 1.0};
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 596f8efebc..8c754537ba 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -253,6 +253,7 @@ enum CodecID {
CODEC_ID_MUSEPACK7,
CODEC_ID_MLP,
CODEC_ID_GSM_MS, /* as found in WAV */
+ CODEC_ID_ATRAC3,
/* subtitle codecs */
CODEC_ID_DVD_SUBTITLE= 0x17000,
@@ -2252,6 +2253,7 @@ extern AVCodec amr_nb_decoder;
extern AVCodec amr_wb_decoder;
extern AVCodec asv1_decoder;
extern AVCodec asv2_decoder;
+extern AVCodec atrac3_decoder;
extern AVCodec avs_decoder;
extern AVCodec bethsoftvid_decoder;
extern AVCodec bmp_decoder;
diff --git a/libavformat/riff.c b/libavformat/riff.c
index 4a5553fa41..0f5e975692 100644
--- a/libavformat/riff.c
+++ b/libavformat/riff.c
@@ -204,6 +204,7 @@ const AVCodecTag codec_wav_tags[] = {
{ CODEC_ID_FLAC, 0xF1AC },
{ CODEC_ID_IMC, 0x401 },
{ CODEC_ID_GSM_MS, 0x31 },
+ { CODEC_ID_ATRAC3, 0x270 },
/* FIXME: All of the IDs below are not 16 bit and thus illegal. */
// for NuppelVideo (nuv.c)
diff --git a/libavformat/rm.c b/libavformat/rm.c
index d2e41acbb6..96752f225f 100644
--- a/libavformat/rm.c
+++ b/libavformat/rm.c
@@ -565,7 +565,7 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVStream *st,
}
rm->audiobuf = av_malloc(rm->audio_framesize * sub_packet_h);
- } else if (!strcmp(buf, "cook")) {
+ } else if ((!strcmp(buf, "cook")) || (!strcmp(buf, "atrc"))) {
int codecdata_length, i;
get_be16(pb); get_byte(pb);
if (((version >> 16) & 0xff) == 5)
@@ -576,7 +576,8 @@ static int rm_read_audio_stream_info(AVFormatContext *s, AVStream *st,
return -1;
}
- st->codec->codec_id = CODEC_ID_COOK;
+ if (!strcmp(buf, "cook")) st->codec->codec_id = CODEC_ID_COOK;
+ else st->codec->codec_id = CODEC_ID_ATRAC3;
st->codec->extradata_size= codecdata_length;
st->codec->extradata= av_mallocz(st->codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
for(i = 0; i < codecdata_length; i++)
@@ -957,7 +958,8 @@ resync:
} else if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
if ((st->codec->codec_id == CODEC_ID_RA_288) ||
- (st->codec->codec_id == CODEC_ID_COOK)) {
+ (st->codec->codec_id == CODEC_ID_COOK) ||
+ (st->codec->codec_id == CODEC_ID_ATRAC3)) {
int x;
int sps = rm->sub_packet_size;
int cfs = rm->coded_framesize;
@@ -975,6 +977,7 @@ resync:
for (x = 0; x < h/2; x++)
get_buffer(pb, rm->audiobuf+x*2*w+y*cfs, cfs);
break;
+ case CODEC_ID_ATRAC3:
case CODEC_ID_COOK:
for (x = 0; x < w/sps; x++)
get_buffer(pb, rm->audiobuf+sps*(h*x+((h+1)/2)*(y&1)+(y>>1)), sps);
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-08 15:31:37 +0000