/* * COOK compatible decoder * Copyright (c) 2003 Sascha Sommer * Copyright (c) 2005 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 * Cook compatible decoder. Bastardization of the G.722.1 standard. * This decoder handles RealNetworks, RealAudio G2 data. * Cook is identified by the codec name cook in RM files. * * To use this decoder, a calling application must supply the extradata * bytes provided from the RM container; 8+ bytes for mono streams and * 16+ for stereo streams (maybe more). * * Codec technicalities (all this assume a buffer length of 1024): * Cook works with several different techniques to achieve its compression. * In the timedomain the buffer is divided into 8 pieces and quantized. If * two neighboring pieces have different quantization index a smooth * quantization curve is used to get a smooth overlap between the different * pieces. * To get to the transformdomain Cook uses a modulated lapped transform. * The transform domain has 50 subbands with 20 elements each. This * means only a maximum of 50*20=1000 coefficients are used out of the 1024 * available. */ #include "libavutil/channel_layout.h" #include "libavutil/lfg.h" #include "audiodsp.h" #include "avcodec.h" #include "get_bits.h" #include "bytestream.h" #include "fft.h" #include "internal.h" #include "sinewin.h" #include "unary.h" #include "cookdata.h" /* the different Cook versions */ #define MONO 0x1000001 #define STEREO 0x1000002 #define JOINT_STEREO 0x1000003 #define MC_COOK 0x2000000 // multichannel Cook, not supported #define SUBBAND_SIZE 20 #define MAX_SUBPACKETS 5 typedef struct cook_gains { int *now; int *previous; } cook_gains; typedef struct COOKSubpacket { int ch_idx; int size; int num_channels; int cookversion; int subbands; int js_subband_start; int js_vlc_bits; int samples_per_channel; int log2_numvector_size; unsigned int channel_mask; VLC channel_coupling; int joint_stereo; int bits_per_subpacket; int bits_per_subpdiv; int total_subbands; int numvector_size; // 1 << log2_numvector_size; float mono_previous_buffer1[1024]; float mono_previous_buffer2[1024]; cook_gains gains1; cook_gains gains2; int gain_1[9]; int gain_2[9]; int gain_3[9]; int gain_4[9]; } COOKSubpacket; typedef struct cook { /* * The following 5 functions provide the lowlevel arithmetic on * the internal audio buffers. */ void (*scalar_dequant)(struct cook *q, int index, int quant_index, int *subband_coef_index, int *subband_coef_sign, float *mlt_p); void (*decouple)(struct cook *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2); void (*imlt_window)(struct cook *q, float *buffer1, cook_gains *gains_ptr, float *previous_buffer); void (*interpolate)(struct cook *q, float *buffer, int gain_index, int gain_index_next); void (*saturate_output)(struct cook *q, float *out); AVCodecContext* avctx; AudioDSPContext adsp; GetBitContext gb; /* stream data */ int num_vectors; int samples_per_channel; /* states */ AVLFG random_state; int discarded_packets; /* transform data */ FFTContext mdct_ctx; float* mlt_window; /* VLC data */ VLC envelope_quant_index[13]; VLC sqvh[7]; // scalar quantization /* generate tables and related variables */ int gain_size_factor; float gain_table[23]; /* data buffers */ uint8_t* decoded_bytes_buffer; DECLARE_ALIGNED(32, float, mono_mdct_output)[2048]; float decode_buffer_1[1024]; float decode_buffer_2[1024]; float decode_buffer_0[1060]; /* static allocation for joint decode */ const float *cplscales[5]; int num_subpackets; COOKSubpacket subpacket[MAX_SUBPACKETS]; } COOKContext; static float pow2tab[127]; static float rootpow2tab[127]; /*************** init functions ***************/ /* table generator */ static av_cold void init_pow2table(void) { /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */ int i; static const float exp2_tab[2] = {1, M_SQRT2}; float exp2_val = powf(2, -63); float root_val = powf(2, -32); for (i = -63; i < 64; i++) { if (!(i & 1)) root_val *= 2; pow2tab[63 + i] = exp2_val; rootpow2tab[63 + i] = root_val * exp2_tab[i & 1]; exp2_val *= 2; } } /* table generator */ static av_cold void init_gain_table(COOKContext *q) { int i; q->gain_size_factor = q->samples_per_channel / 8; for (i = 0; i < 23; i++) q->gain_table[i] = pow(pow2tab[i + 52], (1.0 / (double) q->gain_size_factor)); } static av_cold int init_cook_vlc_tables(COOKContext *q) { int i, result; result = 0; for (i = 0; i < 13; i++) { result |= init_vlc(&q->envelope_quant_index[i], 9, 24, envelope_quant_index_huffbits[i], 1, 1, envelope_quant_index_huffcodes[i], 2, 2, 0); } av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n"); for (i = 0; i < 7; i++) { result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i], cvh_huffbits[i], 1, 1, cvh_huffcodes[i], 2, 2, 0); } for (i = 0; i < q->num_subpackets; i++) { if (q->subpacket[i].joint_stereo == 1) { result |= init_vlc(&q->subpacket[i].channel_coupling, 6, (1 << q->subpacket[i].js_vlc_bits) - 1, ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1, ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0); av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i); } } av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n"); return result; } static av_cold int init_cook_mlt(COOKContext *q) { int j, ret; int mlt_size = q->samples_per_channel; if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0) return AVERROR(ENOMEM); /* Initialize the MLT window: simple sine window. */ ff_sine_window_init(q->mlt_window, mlt_size); for (j = 0; j < mlt_size; j++) q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel); /* Initialize the MDCT. */ if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) { av_freep(&q->mlt_window); return ret; } av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n", av_log2(mlt_size) + 1); return 0; } static av_cold void init_cplscales_table(COOKContext *q) { int i; for (i = 0; i < 5; i++) q->cplscales[i] = cplscales[i]; } /*************** init functions end ***********/ #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4) #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes))) /** * Cook indata decoding, every 32 bits are XORed with 0x37c511f2. * Why? No idea, some checksum/error detection method maybe. * * Out buffer size: extra bytes are needed to cope with * padding/misalignment. * Subpackets passed to the decoder can contain two, consecutive * half-subpackets, of identical but arbitrary size. * 1234 1234 1234 1234 extraA extraB * Case 1: AAAA BBBB 0 0 * Case 2: AAAA ABBB BB-- 3 3 * Case 3: AAAA AABB BBBB 2 2 * Case 4: AAAA AAAB BBBB BB-- 1 5 * * Nice way to waste CPU cycles. * * @param inbuffer pointer to byte array of indata * @param out pointer to byte array of outdata * @param bytes number of bytes */ static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes) { static const uint32_t tab[4] = { AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u), AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u), }; int i, off; uint32_t c; const uint32_t *buf; uint32_t *obuf = (uint32_t *) out; /* FIXME: 64 bit platforms would be able to do 64 bits at a time. * I'm too lazy though, should be something like * for (i = 0; i < bitamount / 64; i++) * (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]); * Buffer alignment needs to be checked. */ off = (intptr_t) inbuffer & 3; buf = (const uint32_t *) (inbuffer - off); c = tab[off]; bytes += 3 + off; for (i = 0; i < bytes / 4; i++) obuf[i] = c ^ buf[i]; return off; } static av_cold int cook_decode_close(AVCodecContext *avctx) { int i; COOKContext *q = avctx->priv_data; av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n"); /* Free allocated memory buffers. */ av_freep(&q->mlt_window); av_freep(&q->decoded_bytes_buffer); /* Free the transform. */ ff_mdct_end(&q->mdct_ctx); /* Free the VLC tables. */ for (i = 0; i < 13; i++) ff_free_vlc(&q->envelope_quant_index[i]); for (i = 0; i < 7; i++) ff_free_vlc(&q->sqvh[i]); for (i = 0; i < q->num_subpackets; i++) ff_free_vlc(&q->subpacket[i].channel_coupling); av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n"); return 0; } /** * Fill the gain array for the timedomain quantization. * * @param gb pointer to the GetBitContext * @param gaininfo array[9] of gain indexes */ static void decode_gain_info(GetBitContext *gb, int *gaininfo) { int i, n; n = get_unary(gb, 0, get_bits_left(gb)); // amount of elements*2 to update i = 0; while (n--) { int index = get_bits(gb, 3); int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1; while (i <= index) gaininfo[i++] = gain; } while (i <= 8) gaininfo[i++] = 0; } /** * Create the quant index table needed for the envelope. * * @param q pointer to the COOKContext * @param quant_index_table pointer to the array */ static int decode_envelope(COOKContext *q, COOKSubpacket *p, int *quant_index_table) { int i, j, vlc_index; quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize for (i = 1; i < p->total_subbands; i++) { vlc_index = i; if (i >= p->js_subband_start * 2) { vlc_index -= p->js_subband_start; } else { vlc_index /= 2; if (vlc_index < 1) vlc_index = 1; } if (vlc_index > 13) vlc_index = 13; // the VLC tables >13 are identical to No. 13 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table, q->envelope_quant_index[vlc_index - 1].bits, 2); quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding if (quant_index_table[i] > 63 || quant_index_table[i] < -63) { av_log(q->avctx, AV_LOG_ERROR, "Invalid quantizer %d at position %d, outside [-63, 63] range\n", quant_index_table[i], i); return AVERROR_INVALIDDATA; } } return 0; } /** * Calculate the category and category_index vector. * * @param q pointer to the COOKContext * @param quant_index_table pointer to the array * @param category pointer to the category array * @param category_index pointer to the category_index array */ static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table, int *category, int *category_index) { int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j; int exp_index2[102] = { 0 }; int exp_index1[102] = { 0 }; int tmp_categorize_array[128 * 2] = { 0 }; int tmp_categorize_array1_idx = p->numvector_size; int tmp_categorize_array2_idx = p->numvector_size; bits_left = p->bits_per_subpacket - get_bits_count(&q->gb); if (bits_left > q->samples_per_channel) bits_left = q->samples_per_channel + ((bits_left - q->samples_per_channel) * 5) / 8; bias = -32; /* Estimate bias. */ for (i = 32; i > 0; i = i / 2) { num_bits = 0; index = 0; for (j = p->total_subbands; j > 0; j--) { exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3); index++; num_bits += expbits_tab[exp_idx]; } if (num_bits >= bits_left - 32) bias += i; } /* Calculate total number of bits. */ num_bits = 0; for (i = 0; i < p->total_subbands; i++) { exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3); num_bits += expbits_tab[exp_idx]; exp_index1[i] = exp_idx; exp_index2[i] = exp_idx; } tmpbias1 = tmpbias2 = num_bits; for (j = 1; j < p->numvector_size; j++) { if (tmpbias1 + tmpbias2 > 2 * bits_left) { /* ---> */ int max = -999999; index = -1; for (i = 0; i < p->total_subbands; i++) { if (exp_index1[i] < 7) { v = (-2 * exp_index1[i]) - quant_index_table[i] + bias; if (v >= max) { max = v; index = i; } } } if (index == -1) break; tmp_categorize_array[tmp_categorize_array1_idx++] = index; tmpbias1 -= expbits_tab[exp_index1[index]] - expbits_tab[exp_index1[index] + 1]; ++exp_index1[index]; } else { /* <--- */ int min = 999999; index = -1; for (i = 0; i < p->total_subbands; i++) { if (exp_index2[i] > 0) { v = (-2 * exp_index2[i]) - quant_index_table[i] + bias; if (v < min) { min = v; index = i; } } } if (index == -1) break; tmp_categorize_array[--tmp_categorize_array2_idx] = index; tmpbias2 -= expbits_tab[exp_index2[index]] - expbits_tab[exp_index2[index] - 1]; --exp_index2[index]; } } for (i = 0; i < p->total_subbands; i++) category[i] = exp_index2[i]; for (i = 0; i < p->numvector_size - 1; i++) category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++]; } /** * Expand the category vector. * * @param q pointer to the COOKContext * @param category pointer to the category array * @param category_index pointer to the category_index array */ static inline void expand_category(COOKContext *q, int *category, int *category_index) { int i; for (i = 0; i < q->num_vectors; i++) { int idx = category_index[i]; if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab)) --category[idx]; } } /** * The real requantization of the mltcoefs * * @param q pointer to the COOKContext * @param index index * @param quant_index quantisation index * @param subband_coef_index array of indexes to quant_centroid_tab * @param subband_coef_sign signs of coefficients * @param mlt_p pointer into the mlt buffer */ static void scalar_dequant_float(COOKContext *q, int index, int quant_index, int *subband_coef_index, int *subband_coef_sign, float *mlt_p) { int i; float f1; for (i = 0; i < SUBBAND_SIZE; i++) { if (subband_coef_index[i]) { f1 = quant_centroid_tab[index][subband_coef_index[i]]; if (subband_coef_sign[i]) f1 = -f1; } else { /* noise coding if subband_coef_index[i] == 0 */ f1 = dither_tab[index]; if (av_lfg_get(&q->random_state) < 0x80000000) f1 = -f1; } mlt_p[i] = f1 * rootpow2tab[quant_index + 63]; } } /** * Unpack the subband_coef_index and subband_coef_sign vectors. * * @param q pointer to the COOKContext * @param category pointer to the category array * @param subband_coef_index array of indexes to quant_centroid_tab * @param subband_coef_sign signs of coefficients */ static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, int *subband_coef_index, int *subband_coef_sign) { int i, j; int vlc, vd, tmp, result; vd = vd_tab[category]; result = 0; for (i = 0; i < vpr_tab[category]; i++) { vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3); if (p->bits_per_subpacket < get_bits_count(&q->gb)) { vlc = 0; result = 1; } for (j = vd - 1; j >= 0; j--) { tmp = (vlc * invradix_tab[category]) / 0x100000; subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1); vlc = tmp; } for (j = 0; j < vd; j++) { if (subband_coef_index[i * vd + j]) { if (get_bits_count(&q->gb) < p->bits_per_subpacket) { subband_coef_sign[i * vd + j] = get_bits1(&q->gb); } else { result = 1; subband_coef_sign[i * vd + j] = 0; } } else { subband_coef_sign[i * vd + j] = 0; } } } return result; } /** * Fill the mlt_buffer with mlt coefficients. * * @param q pointer to the COOKContext * @param category pointer to the category array * @param quant_index_table pointer to the array * @param mlt_buffer pointer to mlt coefficients */ static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category, int *quant_index_table, float *mlt_buffer) { /* A zero in this table means that the subband coefficient is random noise coded. */ int subband_coef_index[SUBBAND_SIZE]; /* A zero in this table means that the subband coefficient is a positive multiplicator. */ int subband_coef_sign[SUBBAND_SIZE]; int band, j; int index = 0; for (band = 0; band < p->total_subbands; band++) { index = category[band]; if (category[band] < 7) { if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) { index = 7; for (j = 0; j < p->total_subbands; j++) category[band + j] = 7; } } if (index >= 7) { memset(subband_coef_index, 0, sizeof(subband_coef_index)); memset(subband_coef_sign, 0, sizeof(subband_coef_sign)); } q->scalar_dequant(q, index, quant_index_table[band], subband_coef_index, subband_coef_sign, &mlt_buffer[band * SUBBAND_SIZE]); } /* FIXME: should this be removed, or moved into loop above? */ if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel) return; } static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer) { int category_index[128] = { 0 }; int category[128] = { 0 }; int quant_index_table[102]; int res, i; if ((res = decode_envelope(q, p, quant_index_table)) < 0) return res; q->num_vectors = get_bits(&q->gb, p->log2_numvector_size); categorize(q, p, quant_index_table, category, category_index); expand_category(q, category, category_index); for (i=0; i<p->total_subbands; i++) { if (category[i] > 7) return AVERROR_INVALIDDATA; } decode_vectors(q, p, category, quant_index_table, mlt_buffer); return 0; } /** * the actual requantization of the timedomain samples * * @param q pointer to the COOKContext * @param buffer pointer to the timedomain buffer * @param gain_index index for the block multiplier * @param gain_index_next index for the next block multiplier */ static void interpolate_float(COOKContext *q, float *buffer, int gain_index, int gain_index_next) { int i; float fc1, fc2; fc1 = pow2tab[gain_index + 63]; if (gain_index == gain_index_next) { // static gain for (i = 0; i < q->gain_size_factor; i++) buffer[i] *= fc1; } else { // smooth gain fc2 = q->gain_table[11 + (gain_index_next - gain_index)]; for (i = 0; i < q->gain_size_factor; i++) { buffer[i] *= fc1; fc1 *= fc2; } } } /** * Apply transform window, overlap buffers. * * @param q pointer to the COOKContext * @param inbuffer pointer to the mltcoefficients * @param gains_ptr current and previous gains * @param previous_buffer pointer to the previous buffer to be used for overlapping */ static void imlt_window_float(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer) { const float fc = pow2tab[gains_ptr->previous[0] + 63]; int i; /* The weird thing here, is that the two halves of the time domain * buffer are swapped. Also, the newest data, that we save away for * next frame, has the wrong sign. Hence the subtraction below. * Almost sounds like a complex conjugate/reverse data/FFT effect. */ /* Apply window and overlap */ for (i = 0; i < q->samples_per_channel; i++) inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] - previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i]; } /** * The modulated lapped transform, this takes transform coefficients * and transforms them into timedomain samples. * Apply transform window, overlap buffers, apply gain profile * and buffer management. * * @param q pointer to the COOKContext * @param inbuffer pointer to the mltcoefficients * @param gains_ptr current and previous gains * @param previous_buffer pointer to the previous buffer to be used for overlapping */ static void imlt_gain(COOKContext *q, float *inbuffer, cook_gains *gains_ptr, float *previous_buffer) { float *buffer0 = q->mono_mdct_output; float *buffer1 = q->mono_mdct_output + q->samples_per_channel; int i; /* Inverse modified discrete cosine transform */ q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer); q->imlt_window(q, buffer1, gains_ptr, previous_buffer); /* Apply gain profile */ for (i = 0; i < 8; i++) if (gains_ptr->now[i] || gains_ptr->now[i + 1]) q->interpolate(q, &buffer1[q->gain_size_factor * i], gains_ptr->now[i], gains_ptr->now[i + 1]); /* Save away the current to be previous block. */ memcpy(previous_buffer, buffer0, q->samples_per_channel * sizeof(*previous_buffer)); } /** * function for getting the jointstereo coupling information * * @param q pointer to the COOKContext * @param decouple_tab decoupling array */ static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab) { int i; int vlc = get_bits1(&q->gb); int start = cplband[p->js_subband_start]; int end = cplband[p->subbands - 1]; int length = end - start + 1; if (start > end) return 0; if (vlc) for (i = 0; i < length; i++) decouple_tab[start + i] = get_vlc2(&q->gb, p->channel_coupling.table, p->channel_coupling.bits, 2); else for (i = 0; i < length; i++) { int v = get_bits(&q->gb, p->js_vlc_bits); if (v == (1<<p->js_vlc_bits)-1) { av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n"); return AVERROR_INVALIDDATA; } decouple_tab[start + i] = v; } return 0; } /** * function decouples a pair of signals from a single signal via multiplication. * * @param q pointer to the COOKContext * @param subband index of the current subband * @param f1 multiplier for channel 1 extraction * @param f2 multiplier for channel 2 extraction * @param decode_buffer input buffer * @param mlt_buffer1 pointer to left channel mlt coefficients * @param mlt_buffer2 pointer to right channel mlt coefficients */ static void decouple_float(COOKContext *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2) { int j, tmp_idx; for (j = 0; j < SUBBAND_SIZE; j++) { tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j; mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx]; mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx]; } } /** * function for decoding joint stereo data * * @param q pointer to the COOKContext * @param mlt_buffer1 pointer to left channel mlt coefficients * @param mlt_buffer2 pointer to right channel mlt coefficients */ static int joint_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer_left, float *mlt_buffer_right) { int i, j, res; int decouple_tab[SUBBAND_SIZE] = { 0 }; float *decode_buffer = q->decode_buffer_0; int idx, cpl_tmp; float f1, f2; const float *cplscale; memset(decode_buffer, 0, sizeof(q->decode_buffer_0)); /* Make sure the buffers are zeroed out. */ memset(mlt_buffer_left, 0, 1024 * sizeof(*mlt_buffer_left)); memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right)); if ((res = decouple_info(q, p, decouple_tab)) < 0) return res; if ((res = mono_decode(q, p, decode_buffer)) < 0) return res; /* The two channels are stored interleaved in decode_buffer. */ for (i = 0; i < p->js_subband_start; i++) { for (j = 0; j < SUBBAND_SIZE; j++) { mlt_buffer_left[i * 20 + j] = decode_buffer[i * 40 + j]; mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j]; } } /* When we reach js_subband_start (the higher frequencies) the coefficients are stored in a coupling scheme. */ idx = (1 << p->js_vlc_bits) - 1; for (i = p->js_subband_start; i < p->subbands; i++) { cpl_tmp = cplband[i]; idx -= decouple_tab[cpl_tmp]; cplscale = q->cplscales[p->js_vlc_bits - 2]; // choose decoupler table f1 = cplscale[decouple_tab[cpl_tmp] + 1]; f2 = cplscale[idx]; q->decouple(q, p, i, f1, f2, decode_buffer, mlt_buffer_left, mlt_buffer_right); idx = (1 << p->js_vlc_bits) - 1; } return 0; } /** * First part of subpacket decoding: * decode raw stream bytes and read gain info. * * @param q pointer to the COOKContext * @param inbuffer pointer to raw stream data * @param gains_ptr array of current/prev gain pointers */ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, cook_gains *gains_ptr) { int offset; offset = decode_bytes(inbuffer, q->decoded_bytes_buffer, p->bits_per_subpacket / 8); init_get_bits(&q->gb, q->decoded_bytes_buffer + offset, p->bits_per_subpacket); decode_gain_info(&q->gb, gains_ptr->now); /* Swap current and previous gains */ FFSWAP(int *, gains_ptr->now, gains_ptr->previous); } /** * Saturate the output signal and interleave. * * @param q pointer to the COOKContext * @param out pointer to the output vector */ static void saturate_output_float(COOKContext *q, float *out) { q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel, -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8)); } /** * Final part of subpacket decoding: * Apply modulated lapped transform, gain compensation, * clip and convert to integer. * * @param q pointer to the COOKContext * @param decode_buffer pointer to the mlt coefficients * @param gains_ptr array of current/prev gain pointers * @param previous_buffer pointer to the previous buffer to be used for overlapping * @param out pointer to the output buffer */ static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer, cook_gains *gains_ptr, float *previous_buffer, float *out) { imlt_gain(q, decode_buffer, gains_ptr, previous_buffer); if (out) q->saturate_output(q, out); } /** * Cook subpacket decoding. This function returns one decoded subpacket, * usually 1024 samples per channel. * * @param q pointer to the COOKContext * @param inbuffer pointer to the inbuffer * @param outbuffer pointer to the outbuffer */ static int decode_subpacket(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, float **outbuffer) { int sub_packet_size = p->size; int res; memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1)); decode_bytes_and_gain(q, p, inbuffer, &p->gains1); if (p->joint_stereo) { if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0) return res; } else { if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0) return res; if (p->num_channels == 2) { decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2); if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0) return res; } } mlt_compensate_output(q, q->decode_buffer_1, &p->gains1, p->mono_previous_buffer1, outbuffer ? outbuffer[p->ch_idx] : NULL); if (p->num_channels == 2) { if (p->joint_stereo) mlt_compensate_output(q, q->decode_buffer_2, &p->gains1, p->mono_previous_buffer2, outbuffer ? outbuffer[p->ch_idx + 1] : NULL); else mlt_compensate_output(q, q->decode_buffer_2, &p->gains2, p->mono_previous_buffer2, outbuffer ? outbuffer[p->ch_idx + 1] : NULL); } return 0; } static int cook_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { AVFrame *frame = data; const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; COOKContext *q = avctx->priv_data; float **samples = NULL; int i, ret; int offset = 0; int chidx = 0; if (buf_size < avctx->block_align) return buf_size; /* get output buffer */ if (q->discarded_packets >= 2) { frame->nb_samples = q->samples_per_channel; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; samples = (float **)frame->extended_data; } /* estimate subpacket sizes */ q->subpacket[0].size = avctx->block_align; for (i = 1; i < q->num_subpackets; i++) { q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i]; q->subpacket[0].size -= q->subpacket[i].size + 1; if (q->subpacket[0].size < 0) { av_log(avctx, AV_LOG_DEBUG, "frame subpacket size total > avctx->block_align!\n"); return AVERROR_INVALIDDATA; } } /* decode supbackets */ for (i = 0; i < q->num_subpackets; i++) { q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >> q->subpacket[i].bits_per_subpdiv; q->subpacket[i].ch_idx = chidx; av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] size %i js %i %i block_align %i\n", i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset, avctx->block_align); if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0) return ret; offset += q->subpacket[i].size; chidx += q->subpacket[i].num_channels; av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n", i, q->subpacket[i].size * 8, get_bits_count(&q->gb)); } /* Discard the first two frames: no valid audio. */ if (q->discarded_packets < 2) { q->discarded_packets++; *got_frame_ptr = 0; return avctx->block_align; } *got_frame_ptr = 1; return avctx->block_align; } static void dump_cook_context(COOKContext *q) { //int i=0; #define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b); ff_dlog(q->avctx, "COOKextradata\n"); ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion); if (q->subpacket[0].cookversion > STEREO) { PRINT("js_subband_start", q->subpacket[0].js_subband_start); PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits); } ff_dlog(q->avctx, "COOKContext\n"); PRINT("nb_channels", q->avctx->channels); PRINT("bit_rate", (int)q->avctx->bit_rate); PRINT("sample_rate", q->avctx->sample_rate); PRINT("samples_per_channel", q->subpacket[0].samples_per_channel); PRINT("subbands", q->subpacket[0].subbands); PRINT("js_subband_start", q->subpacket[0].js_subband_start); PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size); PRINT("numvector_size", q->subpacket[0].numvector_size); PRINT("total_subbands", q->subpacket[0].total_subbands); } /** * Cook initialization * * @param avctx pointer to the AVCodecContext */ static av_cold int cook_decode_init(AVCodecContext *avctx) { COOKContext *q = avctx->priv_data; const uint8_t *edata_ptr = avctx->extradata; const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size; int extradata_size = avctx->extradata_size; int s = 0; unsigned int channel_mask = 0; int samples_per_frame = 0; int ret; q->avctx = avctx; /* Take care of the codec specific extradata. */ if (extradata_size < 8) { av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n"); return AVERROR_INVALIDDATA; } av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size); /* Take data from the AVCodecContext (RM container). */ if (!avctx->channels) { av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); return AVERROR_INVALIDDATA; } /* Initialize RNG. */ av_lfg_init(&q->random_state, 0); ff_audiodsp_init(&q->adsp); while (edata_ptr < edata_ptr_end) { /* 8 for mono, 16 for stereo, ? for multichannel Swap to right endianness so we don't need to care later on. */ if (extradata_size >= 8) { q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr); samples_per_frame = bytestream_get_be16(&edata_ptr); q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr); extradata_size -= 8; } if (extradata_size >= 8) { bytestream_get_be32(&edata_ptr); // Unknown unused q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr); if (q->subpacket[s].js_subband_start >= 51) { av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start); return AVERROR_INVALIDDATA; } q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr); extradata_size -= 8; } /* Initialize extradata related variables. */ q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels; q->subpacket[s].bits_per_subpacket = avctx->block_align * 8; /* Initialize default data states. */ q->subpacket[s].log2_numvector_size = 5; q->subpacket[s].total_subbands = q->subpacket[s].subbands; q->subpacket[s].num_channels = 1; /* Initialize version-dependent variables */ av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s, q->subpacket[s].cookversion); q->subpacket[s].joint_stereo = 0; switch (q->subpacket[s].cookversion) { case MONO: if (avctx->channels != 1) { avpriv_request_sample(avctx, "Container channels != 1"); return AVERROR_PATCHWELCOME; } av_log(avctx, AV_LOG_DEBUG, "MONO\n"); break; case STEREO: if (avctx->channels != 1) { q->subpacket[s].bits_per_subpdiv = 1; q->subpacket[s].num_channels = 2; } av_log(avctx, AV_LOG_DEBUG, "STEREO\n"); break; case JOINT_STEREO: if (avctx->channels != 2) { avpriv_request_sample(avctx, "Container channels != 2"); return AVERROR_PATCHWELCOME; } av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n"); if (avctx->extradata_size >= 16) { q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start; q->subpacket[s].joint_stereo = 1; q->subpacket[s].num_channels = 2; } if (q->subpacket[s].samples_per_channel > 256) { q->subpacket[s].log2_numvector_size = 6; } if (q->subpacket[s].samples_per_channel > 512) { q->subpacket[s].log2_numvector_size = 7; } break; case MC_COOK: av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n"); if (extradata_size >= 4) channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr); if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) { q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start; q->subpacket[s].joint_stereo = 1; q->subpacket[s].num_channels = 2; q->subpacket[s].samples_per_channel = samples_per_frame >> 1; if (q->subpacket[s].samples_per_channel > 256) { q->subpacket[s].log2_numvector_size = 6; } if (q->subpacket[s].samples_per_channel > 512) { q->subpacket[s].log2_numvector_size = 7; } } else q->subpacket[s].samples_per_channel = samples_per_frame; break; default: avpriv_request_sample(avctx, "Cook version %d", q->subpacket[s].cookversion); return AVERROR_PATCHWELCOME; } if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) { av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n"); return AVERROR_INVALIDDATA; } else q->samples_per_channel = q->subpacket[0].samples_per_channel; /* Initialize variable relations */ q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size); /* Try to catch some obviously faulty streams, otherwise it might be exploitable */ if (q->subpacket[s].total_subbands > 53) { avpriv_request_sample(avctx, "total_subbands > 53"); return AVERROR_PATCHWELCOME; } if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) { av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n", q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo); return AVERROR_INVALIDDATA; } if (q->subpacket[s].subbands > 50) { avpriv_request_sample(avctx, "subbands > 50"); return AVERROR_PATCHWELCOME; } if (q->subpacket[s].subbands == 0) { avpriv_request_sample(avctx, "subbands = 0"); return AVERROR_PATCHWELCOME; } q->subpacket[s].gains1.now = q->subpacket[s].gain_1; q->subpacket[s].gains1.previous = q->subpacket[s].gain_2; q->subpacket[s].gains2.now = q->subpacket[s].gain_3; q->subpacket[s].gains2.previous = q->subpacket[s].gain_4; if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) { av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels); return AVERROR_INVALIDDATA; } q->num_subpackets++; s++; if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) { avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align)); return AVERROR_PATCHWELCOME; } } /* Generate tables */ init_pow2table(); init_gain_table(q); init_cplscales_table(q); if ((ret = init_cook_vlc_tables(q))) return ret; if (avctx->block_align >= UINT_MAX / 2) return AVERROR(EINVAL); /* Pad the databuffer with: DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(), AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */ q->decoded_bytes_buffer = av_mallocz(avctx->block_align + DECODE_BYTES_PAD1(avctx->block_align) + AV_INPUT_BUFFER_PADDING_SIZE); if (!q->decoded_bytes_buffer) return AVERROR(ENOMEM); /* Initialize transform. */ if ((ret = init_cook_mlt(q))) return ret; /* Initialize COOK signal arithmetic handling */ if (1) { q->scalar_dequant = scalar_dequant_float; q->decouple = decouple_float; q->imlt_window = imlt_window_float; q->interpolate = interpolate_float; q->saturate_output = saturate_output_float; } /* Try to catch some obviously faulty streams, otherwise it might be exploitable */ if (q->samples_per_channel != 256 && q->samples_per_channel != 512 && q->samples_per_channel != 1024) { avpriv_request_sample(avctx, "samples_per_channel = %d", q->samples_per_channel); return AVERROR_PATCHWELCOME; } avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; if (channel_mask) avctx->channel_layout = channel_mask; else avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO; dump_cook_context(q); return 0; } AVCodec ff_cook_decoder = { .name = "cook", .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_COOK, .priv_data_size = sizeof(COOKContext), .init = cook_decode_init, .close = cook_decode_close, .decode = cook_decode_frame, .capabilities = AV_CODEC_CAP_DR1, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, };