diff options
Diffstat (limited to 'ffmpeg/libavcodec/cook.c')
| -rw-r--r-- | ffmpeg/libavcodec/cook.c | 1292 |
1 files changed, 1292 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/cook.c b/ffmpeg/libavcodec/cook.c new file mode 100644 index 0000000..08cd401 --- /dev/null +++ b/ffmpeg/libavcodec/cook.c @@ -0,0 +1,1292 @@ +/* + * 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 "avcodec.h" +#include "get_bits.h" +#include "dsputil.h" +#include "bytestream.h" +#include "fft.h" +#include "internal.h" +#include "sinewin.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 { + int *now; + int *previous; +} cook_gains; + +typedef struct { + 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; + DSPContext dsp; + 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 + + /* generatable 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) +{ + int i; + for (i = -63; i < 64; i++) { + pow2tab[63 + i] = pow(2, i); + rootpow2tab[63 + i] = sqrt(pow(2, i)); + } +} + +/* 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(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_free(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_free(q->mlt_window); + av_free(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; + + while (get_bits1(gb)) { + /* NOTHING */ + } + + n = get_bits_count(gb) - 1; // 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((i - quant_index_table[index] + bias) / 2, 0, 7); + 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((bias - quant_index_table[i]) / 2, 0, 7); + 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->dsp.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; +} + +#ifdef DEBUG +static void dump_cook_context(COOKContext *q) +{ + //int i=0; +#define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b); + av_dlog(q->avctx, "COOKextradata\n"); + av_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); + } + av_dlog(q->avctx, "COOKContext\n"); + PRINT("nb_channels", q->avctx->channels); + PRINT("bit_rate", 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); +} +#endif + +/** + * 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 <= 0) { + 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_dsputil_init(&q->dsp, avctx); + + 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, othervise 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 > MAX_SUBPACKETS) { + avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS); + 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(), + FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */ + q->decoded_bytes_buffer = + av_mallocz(avctx->block_align + + DECODE_BYTES_PAD1(avctx->block_align) + + FF_INPUT_BUFFER_PADDING_SIZE); + if (q->decoded_bytes_buffer == NULL) + 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, othervise 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; + +#ifdef DEBUG + dump_cook_context(q); +#endif + return 0; +} + +AVCodec ff_cook_decoder = { + .name = "cook", + .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 = CODEC_CAP_DR1, + .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"), + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_NONE }, +}; |