Mercurial > libavcodec.hg
diff wmavoice.c @ 11123:d59349627f52 libavcodec
WMAVoice decoder.
| author | rbultje |
|---|---|
| date | Fri, 12 Feb 2010 14:22:41 +0000 |
| parents | |
| children | b94e1810ce4c |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/wmavoice.c Fri Feb 12 14:22:41 2010 +0000 @@ -0,0 +1,1568 @@ +/* + * Windows Media Audio Voice decoder. + * Copyright (c) 2009 Ronald S. Bultje + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file libavcodec/wmavoice.c + * @brief Windows Media Audio Voice compatible decoder + * @author Ronald S. Bultje <rsbultje@gmail.com> + */ + +#include <math.h> +#include "avcodec.h" +#include "get_bits.h" +#include "put_bits.h" +#include "wmavoice_data.h" +#include "celp_math.h" +#include "celp_filters.h" +#include "acelp_vectors.h" +#include "acelp_filters.h" +#include "lsp.h" +#include "libavutil/lzo.h" + +#define MAX_BLOCKS 8 ///< maximum number of blocks per frame +#define MAX_LSPS 16 ///< maximum filter order +#define MAX_FRAMES 3 ///< maximum number of frames per superframe +#define MAX_FRAMESIZE 160 ///< maximum number of samples per frame +#define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history +#define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES) + ///< maximum number of samples per superframe +#define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that + ///< was split over two packets +#define VLC_NBITS 6 ///< number of bits to read per VLC iteration + +/** + * Frame type VLC coding. + */ +static VLC frame_type_vlc; + +/** + * Adaptive codebook types. + */ +enum { + ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed) + ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which + ///< we interpolate to get a per-sample pitch. + ///< Signal is generated using an asymmetric sinc + ///< window function + ///< @note see #wmavoice_ipol1_coeffs + ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using + ///< a Hamming sinc window function + ///< @note see #wmavoice_ipol2_coeffs +}; + +/** + * Fixed codebook types. + */ +enum { + FCB_TYPE_SILENCE = 0, ///< comfort noise during silence + ///< generated from a hardcoded (fixed) codebook + ///< with per-frame (low) gain values + FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block + ///< gain values + FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals, + ///< used in particular for low-bitrate streams + FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in + ///< combinations of either single pulses or + ///< pulse pairs +}; + +/** + * Description of frame types. + */ +static const struct frame_type_desc { + uint8_t n_blocks; ///< amount of blocks per frame (each block + ///< (contains 160/#n_blocks samples) + uint8_t log_n_blocks; ///< log2(#n_blocks) + uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*) + uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*) + uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs + ///< (rather than just one single pulse) + ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES + uint16_t frame_size; ///< the amount of bits that make up the block + ///< data (per frame) +} frame_descs[17] = { + { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 }, + { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 }, + { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 }, + { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 }, + { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 }, + { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 }, + { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 }, + { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 }, + { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 }, + { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 }, + { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 }, + { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 }, + { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 }, + { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 }, + { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 }, + { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 }, + { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 } +}; + +/** + * WMA Voice decoding context. + */ +typedef struct { + /** + * @defgroup struct_global Global values + * Global values, specified in the stream header / extradata or used + * all over. + * @{ + */ + GetBitContext gb; ///< packet bitreader. During decoder init, + ///< it contains the extradata from the + ///< demuxer. During decoding, it contains + ///< packet data. + int8_t vbm_tree[25]; ///< converts VLC codes to frame type + + int spillover_bitsize; ///< number of bits used to specify + ///< #spillover_nbits in the packet header + ///< = ceil(log2(ctx->block_align << 3)) + int history_nsamples; ///< number of samples in history for signal + ///< prediction (through ACB) + + int do_apf; ///< whether to apply the averaged + ///< projection filter (APF) + + int lsps; ///< number of LSPs per frame [10 or 16] + int lsp_q_mode; ///< defines quantizer defaults [0, 1] + int lsp_def_mode; ///< defines different sets of LSP defaults + ///< [0, 1] + int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded + ///< per-frame (independent coding) + int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded + ///< per superframe (residual coding) + + int min_pitch_val; ///< base value for pitch parsing code + int max_pitch_val; ///< max value + 1 for pitch parsing + int pitch_nbits; ///< number of bits used to specify the + ///< pitch value in the frame header + int block_pitch_nbits; ///< number of bits used to specify the + ///< first block's pitch value + int block_pitch_range; ///< range of the block pitch + int block_delta_pitch_nbits; ///< number of bits used to specify the + ///< delta pitch between this and the last + ///< block's pitch value, used in all but + ///< first block + int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is + ///< from -this to +this-1) + uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale + ///< conversion + + /** + * @} + * @defgroup struct_packet Packet values + * Packet values, specified in the packet header or related to a packet. + * A packet is considered to be a single unit of data provided to this + * decoder by the demuxer. + * @{ + */ + int spillover_nbits; ///< number of bits of the previous packet's + ///< last superframe preceeding this + ///< packet's first full superframe (useful + ///< for re-synchronization also) + int has_residual_lsps; ///< if set, superframes contain one set of + ///< LSPs that cover all frames, encoded as + ///< independent and residual LSPs; if not + ///< set, each frame contains its own, fully + ///< independent, LSPs + int skip_bits_next; ///< number of bits to skip at the next call + ///< to #wmavoice_decode_packet() (since + ///< they're part of the previous superframe) + + uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE]; + ///< cache for superframe data split over + ///< multiple packets + int sframe_cache_size; ///< set to >0 if we have data from an + ///< (incomplete) superframe from a previous + ///< packet that spilled over in the current + ///< packet; specifies the amount of bits in + ///< #sframe_cache + PutBitContext pb; ///< bitstream writer for #sframe_cache + + /** + * @} + * @defgroup struct_frame Frame and superframe values + * Superframe and frame data - these can change from frame to frame, + * although some of them do in that case serve as a cache / history for + * the next frame or superframe. + * @{ + */ + double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous + ///< superframe + int last_pitch_val; ///< pitch value of the previous frame + int last_acb_type; ///< frame type [0-2] of the previous frame + int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val) + ///< << 16) / #MAX_FRAMESIZE + float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE + + int aw_idx_is_ext; ///< whether the AW index was encoded in + ///< 8 bits (instead of 6) + int aw_pulse_range; ///< the range over which #aw_pulse_set1() + ///< can apply the pulse, relative to the + ///< value in aw_first_pulse_off. The exact + ///< position of the first AW-pulse is within + ///< [pulse_off, pulse_off + this], and + ///< depends on bitstream values; [16 or 24] + int aw_n_pulses[2]; ///< number of AW-pulses in each block; note + ///< that this number can be negative (in + ///< which case it basically means "zero") + int aw_first_pulse_off[2]; ///< index of first sample to which to + ///< apply AW-pulses, or -0xff if unset + int aw_next_pulse_off_cache; ///< the position (relative to start of the + ///< second block) at which pulses should + ///< start to be positioned, serves as a + ///< cache for pitch-adaptive window pulses + ///< between blocks + + int frame_cntr; ///< current frame index [0 - 0xFFFE]; is + ///< only used for comfort noise in #pRNG() + float gain_pred_err[6]; ///< cache for gain prediction + float excitation_history[MAX_SIGNAL_HISTORY]; + ///< cache of the signal of previous + ///< superframes, used as a history for + ///< signal generation + float synth_history[MAX_LSPS]; ///< see #excitation_history + /** + * @} + */ +} WMAVoiceContext; + +/** + * Sets up the variable bit mode (VBM) tree from container extradata. + * @param gb bit I/O context. + * The bit context (s->gb) should be loaded with byte 23-46 of the + * container extradata (i.e. the ones containing the VBM tree). + * @param vbm_tree pointer to array to which the decoded VBM tree will be + * written. + * @return 0 on success, <0 on error. + */ +static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) +{ + static const uint8_t bits[] = { + 2, 2, 2, 4, 4, 4, + 6, 6, 6, 8, 8, 8, + 10, 10, 10, 12, 12, 12, + 14, 14, 14, 14 + }; + static const uint16_t codes[] = { + 0x0000, 0x0001, 0x0002, // 00/01/10 + 0x000c, 0x000d, 0x000e, // 11+00/01/10 + 0x003c, 0x003d, 0x003e, // 1111+00/01/10 + 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 + 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 + 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 + 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx + }; + int cntr[8], n, res; + + memset(vbm_tree, 0xff, sizeof(vbm_tree)); + memset(cntr, 0, sizeof(cntr)); + for (n = 0; n < 17; n++) { + res = get_bits(gb, 3); + if (cntr[res] > 3) // should be >= 3 + (res == 7)) + return -1; + vbm_tree[res * 3 + cntr[res]++] = n; + } + INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), + bits, 1, 1, codes, 2, 2, 132); + return 0; +} + +/** + * Set up decoder with parameters from demuxer (extradata etc.). + */ +static av_cold int wmavoice_decode_init(AVCodecContext *ctx) +{ + int n, flags, pitch_range, lsp16_flag; + WMAVoiceContext *s = ctx->priv_data; + + /** + * Extradata layout: + * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), + * - byte 19-22: flags field (annoyingly in LE; see below for known + * values), + * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, + * rest is 0). + */ + if (ctx->extradata_size != 46) { + av_log(ctx, AV_LOG_ERROR, + "Invalid extradata size %d (should be 46)\n", + ctx->extradata_size); + return -1; + } + flags = AV_RL32(ctx->extradata + 18); + s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); + s->do_apf = flags & 0x1; + s->lsp_q_mode = !!(flags & 0x2000); + s->lsp_def_mode = !!(flags & 0x4000); + lsp16_flag = flags & 0x1000; + if (lsp16_flag) { + s->lsps = 16; + s->frame_lsp_bitsize = 34; + s->sframe_lsp_bitsize = 60; + } else { + s->lsps = 10; + s->frame_lsp_bitsize = 24; + s->sframe_lsp_bitsize = 48; + } + for (n = 0; n < s->lsps; n++) + s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); + + init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); + if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { + av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); + return -1; + } + + s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; + s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; + pitch_range = s->max_pitch_val - s->min_pitch_val; + s->pitch_nbits = av_ceil_log2(pitch_range); + s->last_pitch_val = 40; + s->last_acb_type = ACB_TYPE_NONE; + s->history_nsamples = s->max_pitch_val + 8; + + if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { + int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, + max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; + + av_log(ctx, AV_LOG_ERROR, + "Unsupported samplerate %d (min=%d, max=%d)\n", + ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz + + return -1; + } + + s->block_conv_table[0] = s->min_pitch_val; + s->block_conv_table[1] = (pitch_range * 25) >> 6; + s->block_conv_table[2] = (pitch_range * 44) >> 6; + s->block_conv_table[3] = s->max_pitch_val - 1; + s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; + s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); + s->block_pitch_range = s->block_conv_table[2] + + s->block_conv_table[3] + 1 + + 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); + s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); + + ctx->sample_fmt = SAMPLE_FMT_FLT; + + return 0; +} + +/** + * Dequantize LSPs + * @param lsps output pointer to the array that will hold the LSPs + * @param num number of LSPs to be dequantized + * @param values quantized values, contains n_stages values + * @param sizes range (i.e. max value) of each quantized value + * @param n_stages number of dequantization runs + * @param table dequantization table to be used + * @param mul_q LSF multiplier + * @param base_q base (lowest) LSF values + */ +static void dequant_lsps(double *lsps, int num, + const uint16_t *values, + const uint16_t *sizes, + int n_stages, const uint8_t *table, + const double *mul_q, + const double *base_q) +{ + int n, m; + + memset(lsps, 0, num * sizeof(*lsps)); + for (n = 0; n < n_stages; n++) { + const uint8_t *t_off = &table[values[n] * num]; + double base = base_q[n], mul = mul_q[n]; + + for (m = 0; m < num; m++) + lsps[m] += base + mul * t_off[m]; + + table += sizes[n] * num; + } +} + +/** + * @defgroup lsp_dequant LSP dequantization routines + * LSP dequantization routines, for 10/16LSPs and independent/residual coding. + * @note we assume enough bits are available, caller should check. + * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; + * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. + * @{ + */ +/** + * Parse 10 independently-coded LSPs. + */ +static void dequant_lsp10i(GetBitContext *gb, double *lsps) +{ + static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; + static const double mul_lsf[4] = { + 5.2187144800e-3, 1.4626986422e-3, + 9.6179549166e-4, 1.1325736225e-3 + }; + static const double base_lsf[4] = { + M_PI * -2.15522e-1, M_PI * -6.1646e-2, + M_PI * -3.3486e-2, M_PI * -5.7408e-2 + }; + uint16_t v[4]; + + v[0] = get_bits(gb, 8); + v[1] = get_bits(gb, 6); + v[2] = get_bits(gb, 5); + v[3] = get_bits(gb, 5); + + dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, + mul_lsf, base_lsf); +} + +/** + * Parse 10 independently-coded LSPs, and then derive the tables to + * generate LSPs for the other frames from them (residual coding). + */ +static void dequant_lsp10r(GetBitContext *gb, + double *i_lsps, const double *old, + double *a1, double *a2, int q_mode) +{ + static const uint16_t vec_sizes[3] = { 128, 64, 64 }; + static const double mul_lsf[3] = { + 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 + }; + static const double base_lsf[3] = { + M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 + }; + const float (*ipol_tab)[2][10] = q_mode ? + wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; + uint16_t interpol, v[3]; + int n; + + dequant_lsp10i(gb, i_lsps); + + interpol = get_bits(gb, 5); + v[0] = get_bits(gb, 7); + v[1] = get_bits(gb, 6); + v[2] = get_bits(gb, 6); + + for (n = 0; n < 10; n++) { + double delta = old[n] - i_lsps[n]; + a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; + a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; + } + + dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, + mul_lsf, base_lsf); +} + +/** + * Parse 16 independently-coded LSPs. + */ +static void dequant_lsp16i(GetBitContext *gb, double *lsps) +{ + static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; + static const double mul_lsf[5] = { + 3.3439586280e-3, 6.9908173703e-4, + 3.3216608306e-3, 1.0334960326e-3, + 3.1899104283e-3 + }; + static const double base_lsf[5] = { + M_PI * -1.27576e-1, M_PI * -2.4292e-2, + M_PI * -1.28094e-1, M_PI * -3.2128e-2, + M_PI * -1.29816e-1 + }; + uint16_t v[5]; + + v[0] = get_bits(gb, 8); + v[1] = get_bits(gb, 6); + v[2] = get_bits(gb, 7); + v[3] = get_bits(gb, 6); + v[4] = get_bits(gb, 7); + + dequant_lsps( lsps, 5, v, vec_sizes, 2, + wmavoice_dq_lsp16i1, mul_lsf, base_lsf); + dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, + wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); + dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, + wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); +} + +/** + * Parse 16 independently-coded LSPs, and then derive the tables to + * generate LSPs for the other frames from them (residual coding). + */ +static void dequant_lsp16r(GetBitContext *gb, + double *i_lsps, const double *old, + double *a1, double *a2, int q_mode) +{ + static const uint16_t vec_sizes[3] = { 128, 128, 128 }; + static const double mul_lsf[3] = { + 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 + }; + static const double base_lsf[3] = { + M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 + }; + const float (*ipol_tab)[2][16] = q_mode ? + wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; + uint16_t interpol, v[3]; + int n; + + dequant_lsp16i(gb, i_lsps); + + interpol = get_bits(gb, 5); + v[0] = get_bits(gb, 7); + v[1] = get_bits(gb, 7); + v[2] = get_bits(gb, 7); + + for (n = 0; n < 16; n++) { + double delta = old[n] - i_lsps[n]; + a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; + a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; + } + + dequant_lsps( a2, 10, v, vec_sizes, 1, + wmavoice_dq_lsp16r1, mul_lsf, base_lsf); + dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, + wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); + dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, + wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); +} + +/** + * @} + * @defgroup aw Pitch-adaptive window coding functions + * The next few functions are for pitch-adaptive window coding. + * @{ + */ +/** + * Parse the offset of the first pitch-adaptive window pulses, and + * the distribution of pulses between the two blocks in this frame. + * @param s WMA Voice decoding context private data + * @param gb bit I/O context + * @param pitch pitch for each block in this frame + */ +static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, + const int *pitch) +{ + static const int16_t start_offset[94] = { + -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, + 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, + 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, + 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, + 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, + 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, + 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, + 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 + }; + int bits, offset; + + /* position of pulse */ + s->aw_idx_is_ext = 0; + if ((bits = get_bits(gb, 6)) >= 54) { + s->aw_idx_is_ext = 1; + bits += (bits - 54) * 3 + get_bits(gb, 2); + } + + /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count + * the distribution of the pulses in each block contained in this frame. */ + s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; + for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; + s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; + s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; + offset += s->aw_n_pulses[0] * pitch[0]; + s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; + s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; + + /* if continuing from a position before the block, reset position to + * start of block (when corrected for the range over which it can be + * spread in aw_pulse_set1()). */ + if (start_offset[bits] < MAX_FRAMESIZE / 2) { + while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) + s->aw_first_pulse_off[1] -= pitch[1]; + if (start_offset[bits] < 0) + while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) + s->aw_first_pulse_off[0] -= pitch[0]; + } +} + +/** + * Apply second set of pitch-adaptive window pulses. + * @param s WMA Voice decoding context private data + * @param gb bit I/O context + * @param block_idx block index in frame [0, 1] + * @param fcb structure containing fixed codebook vector info + */ +static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, + int block_idx, AMRFixed *fcb) +{ + uint16_t use_mask[7]; // only 5 are used, rest is padding + /* in this function, idx is the index in the 80-bit (+ padding) use_mask + * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits + * of idx are the position of the bit within a particular item in the + * array (0 being the most significant bit, and 15 being the least + * significant bit), and the remainder (>> 4) is the index in the + * use_mask[]-array. This is faster and uses less memory than using a + * 80-byte/80-int array. */ + int pulse_off = s->aw_first_pulse_off[block_idx], + pulse_start, n, idx, range, aidx, start_off = 0; + + /* set offset of first pulse to within this block */ + if (s->aw_n_pulses[block_idx] > 0) + while (pulse_off + s->aw_pulse_range < 1) + pulse_off += fcb->pitch_lag; + + /* find range per pulse */ + if (s->aw_n_pulses[0] > 0) { + if (block_idx == 0) { + range = 32; + } else /* block_idx = 1 */ { + range = 8; + if (s->aw_n_pulses[block_idx] > 0) + pulse_off = s->aw_next_pulse_off_cache; + } + } else + range = 16; + pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; + + /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, + * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus + * we exclude that range from being pulsed again in this function. */ + memset( use_mask, -1, 5 * sizeof(use_mask[0])); + memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); + if (s->aw_n_pulses[block_idx] > 0) + for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { + int excl_range = s->aw_pulse_range; // always 16 or 24 + uint16_t *use_mask_ptr = &use_mask[idx >> 4]; + int first_sh = 16 - (idx & 15); + *use_mask_ptr++ &= 0xFFFF << first_sh; + excl_range -= first_sh; + if (excl_range >= 16) { + *use_mask_ptr++ = 0; + *use_mask_ptr &= 0xFFFF >> (excl_range - 16); + } else + *use_mask_ptr &= 0xFFFF >> excl_range; + } + + /* find the 'aidx'th offset that is not excluded */ + aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); + for (n = 0; n <= aidx; pulse_start++) { + for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; + if (idx >= MAX_FRAMESIZE / 2) { // find from zero + if (use_mask[0]) idx = 0x0F; + else if (use_mask[1]) idx = 0x1F; + else if (use_mask[2]) idx = 0x2F; + else if (use_mask[3]) idx = 0x3F; + else if (use_mask[4]) idx = 0x4F; + else return; + idx -= av_log2_16bit(use_mask[idx >> 4]); + } + if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { + use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); + n++; + start_off = idx; + } + } + + fcb->x[fcb->n] = start_off; + fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; + fcb->n++; + + /* set offset for next block, relative to start of that block */ + n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; + s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; +} + +/** + * Apply first set of pitch-adaptive window pulses. + * @param s WMA Voice decoding context private data + * @param gb bit I/O context + * @param block_idx block index in frame [0, 1] + * @param fcb storage location for fixed codebook pulse info + */ +static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, + int block_idx, AMRFixed *fcb) +{ + int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); + float v; + + if (s->aw_n_pulses[block_idx] > 0) { + int n, v_mask, i_mask, sh, n_pulses; + + if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each + n_pulses = 3; + v_mask = 8; + i_mask = 7; + sh = 4; + } else { // 4 pulses, 1:sign + 2:index each + n_pulses = 4; + v_mask = 4; + i_mask = 3; + sh = 3; + } + + for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { + fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; + fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + + s->aw_first_pulse_off[block_idx]; + while (fcb->x[fcb->n] < 0) + fcb->x[fcb->n] += fcb->pitch_lag; + if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) + fcb->n++; + } + } else { + int num2 = (val & 0x1FF) >> 1, delta, idx; + + if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } + else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } + else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } + else { delta = 7; idx = num2 + 1 - 3 * 75; } + v = (val & 0x200) ? -1.0 : 1.0; + + fcb->no_repeat_mask |= 3 << fcb->n; + fcb->x[fcb->n] = idx - delta; + fcb->y[fcb->n] = v; + fcb->x[fcb->n + 1] = idx; + fcb->y[fcb->n + 1] = (val & 1) ? -v : v; + fcb->n += 2; + } +} + +/** + * @} + * + * Generate a random number from frame_cntr and block_idx, which will lief + * in the range [0, 1000 - block_size] (so it can be used as an index in a + * table of size 1000 of which you want to read block_size entries). + * + * @param frame_cntr current frame number + * @param block_num current block index + * @param block_size amount of entries we want to read from a table + * that has 1000 entries + * @returns a (non-)random number in the [0, 1000 - block_size] range. + */ +static int pRNG(int frame_cntr, int block_num, int block_size) +{ + /* array to simplify the calculation of z: + * y = (x % 9) * 5 + 6; + * z = (49995 * x) / y; + * Since y only has 9 values, we can remove the division by using a + * LUT and using FASTDIV-style divisions. For each of the 9 values + * of y, we can rewrite z as: + * z = x * (49995 / y) + x * ((49995 % y) / y) + * In this table, each col represents one possible value of y, the + * first number is 49995 / y, and the second is the FASTDIV variant + * of 49995 % y / y. */ + static const unsigned int div_tbl[9][2] = { + { 8332, 3 * 715827883U }, // y = 6 + { 4545, 0 * 390451573U }, // y = 11 + { 3124, 11 * 268435456U }, // y = 16 + { 2380, 15 * 204522253U }, // y = 21 + { 1922, 23 * 165191050U }, // y = 26 + { 1612, 23 * 138547333U }, // y = 31 + { 1388, 27 * 119304648U }, // y = 36 + { 1219, 16 * 104755300U }, // y = 41 + { 1086, 39 * 93368855U } // y = 46 + }; + unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; + if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, + // so this is effectively a modulo (%) + y = x - 9 * MULH(477218589, x); // x % 9 + z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); + // z = x * 49995 / (y * 5 + 6) + return z % (1000 - block_size); +} + +/** + * Parse hardcoded signal for a single block. + * @note see #synth_block(). + */ +static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, + int block_idx, int size, + const struct frame_type_desc *frame_desc, + float *excitation) +{ + float gain; + int n, r_idx; + + assert(size <= MAX_FRAMESIZE); + + /* Set the offset from which we start reading wmavoice_std_codebook */ + if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { + r_idx = pRNG(s->frame_cntr, block_idx, size); + gain = s->silence_gain; + } else /* FCB_TYPE_HARDCODED */ { + r_idx = get_bits(gb, 8); + gain = wmavoice_gain_universal[get_bits(gb, 6)]; + } + + /* Clear gain prediction parameters */ + memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); + + /* Apply gain to hardcoded codebook and use that as excitation signal */ + for (n = 0; n < size; n++) + excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; +} + +/** + * Parse FCB/ACB signal for a single block. + * @note see #synth_block(). + */ +static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, + int block_idx, int size, + int block_pitch_sh2, + const struct frame_type_desc *frame_desc, + float *excitation) +{ + static const float gain_coeff[6] = { + 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 + }; + float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; + int n, idx, gain_weight; + AMRFixed fcb; + + assert(size <= MAX_FRAMESIZE / 2); + memset(pulses, 0, sizeof(*pulses) * size); + + fcb.pitch_lag = block_pitch_sh2 >> 2; + fcb.pitch_fac = 1.0; + fcb.no_repeat_mask = 0; + fcb.n = 0; + + /* For the other frame types, this is where we apply the innovation + * (fixed) codebook pulses of the speech signal. */ + if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { + aw_pulse_set1(s, gb, block_idx, &fcb); + aw_pulse_set2(s, gb, block_idx, &fcb); + } else /* FCB_TYPE_EXC_PULSES */ { + int offset_nbits = 5 - frame_desc->log_n_blocks; + + fcb.no_repeat_mask = -1; + /* similar to ff_decode_10_pulses_35bits(), but with single pulses + * (instead of double) for a subset of pulses */ + for (n = 0; n < 5; n++) { + float sign; + int pos1, pos2; + + sign = get_bits1(gb) ? 1.0 : -1.0; + pos1 = get_bits(gb, offset_nbits); + fcb.x[fcb.n] = n + 5 * pos1; + fcb.y[fcb.n++] = sign; + if (n < frame_desc->dbl_pulses) { + pos2 = get_bits(gb, offset_nbits); + fcb.x[fcb.n] = n + 5 * pos2; + fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; + } + } + } + ff_set_fixed_vector(pulses, &fcb, 1.0, size); + + /* Calculate gain for adaptive & fixed codebook signal. + * see ff_amr_set_fixed_gain(). */ + idx = get_bits(gb, 7); + fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) - + 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); + acb_gain = wmavoice_gain_codebook_acb[idx]; + pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], + -2.9957322736 /* log(0.05) */, + 1.6094379124 /* log(5.0) */); + + gain_weight = 8 >> frame_desc->log_n_blocks; + memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, + sizeof(*s->gain_pred_err) * (6 - gain_weight)); + for (n = 0; n < gain_weight; n++) + s->gain_pred_err[n] = pred_err; + + /* Calculation of adaptive codebook */ + if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { + int len; + for (n = 0; n < size; n += len) { + int next_idx_sh16; + int abs_idx = block_idx * size + n; + int pitch_sh16 = (s->last_pitch_val << 16) + + s->pitch_diff_sh16 * abs_idx; + int pitch = (pitch_sh16 + 0x6FFF) >> 16; + int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; + idx = idx_sh16 >> 16; + if (s->pitch_diff_sh16) { + if (s->pitch_diff_sh16 > 0) { + next_idx_sh16 = (idx_sh16) &~ 0xFFFF; + } else + next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; + len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, + 1, size - n); + } else + len = size; + + ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], + wmavoice_ipol1_coeffs, 17, + idx, 9, len); + } + } else /* ACB_TYPE_HAMMING */ { + int block_pitch = block_pitch_sh2 >> 2; + idx = block_pitch_sh2 & 3; + if (idx) { + ff_acelp_interpolatef(excitation, &excitation[-block_pitch], + wmavoice_ipol2_coeffs, 4, + idx, 8, size); + } else + av_memcpy_backptr(excitation, sizeof(float) * block_pitch, + sizeof(float) * size); + } + + /* Interpolate ACB/FCB and use as excitation signal */ + ff_weighted_vector_sumf(excitation, excitation, pulses, + acb_gain, fcb_gain, size); +} + +/** + * Parse data in a single block. + * @note we assume enough bits are available, caller should check. + * + * @param s WMA Voice decoding context private data + * @param gb bit I/O context + * @param block_idx index of the to-be-read block + * @param size amount of samples to be read in this block + * @param block_pitch_sh2 pitch for this block << 2 + * @param lsps LSPs for (the end of) this frame + * @param prev_lsps LSPs for the last frame + * @param frame_desc frame type descriptor + * @param excitation target memory for the ACB+FCB interpolated signal + * @param synth target memory for the speech synthesis filter output + * @return 0 on success, <0 on error. + */ +static void synth_block(WMAVoiceContext *s, GetBitContext *gb, + int block_idx, int size, + int block_pitch_sh2, + const double *lsps, const double *prev_lsps, + const struct frame_type_desc *frame_desc, + float *excitation, float *synth) +{ + double i_lsps[MAX_LSPS]; + float lpcs[MAX_LSPS]; + float fac; + int n; + + if (frame_desc->acb_type == ACB_TYPE_NONE) + synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); + else + synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, + frame_desc, excitation); + + /* convert interpolated LSPs to LPCs */ + fac = (block_idx + 0.5) / frame_desc->n_blocks; + for (n = 0; n < s->lsps; n++) // LSF -> LSP + i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); + ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); + + /* Speech synthesis */ + ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); +} + +/** + * Synthesize output samples for a single frame. + * @note we assume enough bits are available, caller should check. + * + * @param ctx WMA Voice decoder context + * @param gb bit I/O context (s->gb or one for cross-packet superframes) + * @param samples pointer to output sample buffer, has space for at least 160 + * samples + * @param lsps LSP array + * @param prev_lsps array of previous frame's LSPs + * @param excitation target buffer for excitation signal + * @param synth target buffer for synthesized speech data + * @return 0 on success, <0 on error. + */ +static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, + float *samples, + const double *lsps, const double *prev_lsps, + float *excitation, float *synth) +{ + WMAVoiceContext *s = ctx->priv_data; + int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val; + int pitch[MAX_BLOCKS], last_block_pitch; + + /* Parse frame type ("frame header"), see frame_descs */ + int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], + block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; + + if (bd_idx < 0) { + av_log(ctx, AV_LOG_ERROR, + "Invalid frame type VLC code, skipping\n"); + return -1; + } + + /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ + if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { + /* Pitch is provided per frame, which is interpreted as the pitch of + * the last sample of the last block of this frame. We can interpolate + * the pitch of other blocks (and even pitch-per-sample) by gradually + * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ + n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; + log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; + cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); + cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); + if (s->last_acb_type == ACB_TYPE_NONE || + 20 * abs(cur_pitch_val - s->last_pitch_val) > + (cur_pitch_val + s->last_pitch_val)) + s->last_pitch_val = cur_pitch_val; + + /* pitch per block */ + for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { + int fac = n * 2 + 1; + + pitch[n] = (MUL16(fac, cur_pitch_val) + + MUL16((n_blocks_x2 - fac), s->last_pitch_val) + + frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; + } + + /* "pitch-diff-per-sample" for calculation of pitch per sample */ + s->pitch_diff_sh16 = + ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE; + } + + /* Global gain (if silence) and pitch-adaptive window coordinates */ + switch (frame_descs[bd_idx].fcb_type) { + case FCB_TYPE_SILENCE: + s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; + break; + case FCB_TYPE_AW_PULSES: + aw_parse_coords(s, gb, pitch); + break; + } + + for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { + int bl_pitch_sh2; + + /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ + switch (frame_descs[bd_idx].acb_type) { + case ACB_TYPE_HAMMING: { + /* Pitch is given per block. Per-block pitches are encoded as an + * absolute value for the first block, and then delta values + * relative to this value) for all subsequent blocks. The scale of + * this pitch value is semi-logaritmic compared to its use in the + * decoder, so we convert it to normal scale also. */ + int block_pitch, + t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, + t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, + t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; + + if (n == 0) { + block_pitch = get_bits(gb, s->block_pitch_nbits); + } else + block_pitch = last_block_pitch - s->block_delta_pitch_hrange + + get_bits(gb, s->block_delta_pitch_nbits); + /* Convert last_ so that any next delta is within _range */ + last_block_pitch = av_clip(block_pitch, + s->block_delta_pitch_hrange, + s->block_pitch_range - + s->block_delta_pitch_hrange); + + /* Convert semi-log-style scale back to normal scale */ + if (block_pitch < t1) { + bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; + } else { + block_pitch -= t1; + if (block_pitch < t2) { + bl_pitch_sh2 = + (s->block_conv_table[1] << 2) + (block_pitch << 1); + } else { + block_pitch -= t2; + if (block_pitch < t3) { + bl_pitch_sh2 = + (s->block_conv_table[2] + block_pitch) << 2; + } else + bl_pitch_sh2 = s->block_conv_table[3] << 2; + } + } + pitch[n] = bl_pitch_sh2 >> 2; + break; + } + + case ACB_TYPE_ASYMMETRIC: { + bl_pitch_sh2 = pitch[n] << 2; + break; + } + + default: // ACB_TYPE_NONE has no pitch + bl_pitch_sh2 = 0; + break; + } + + synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, + lsps, prev_lsps, &frame_descs[bd_idx], + &excitation[n * block_nsamples], + &synth[n * block_nsamples]); + } + + /* Averaging projection filter, if applicable. Else, just copy samples + * from synthesis buffer */ + if (s->do_apf) { + // FIXME this is where APF would take place, currently not implemented + av_log_missing_feature(ctx, "APF", 0); + s->do_apf = 0; + } //else + for (n = 0; n < 160; n++) + samples[n] = av_clipf(synth[n], -1.0, 1.0); + + /* Cache values for next frame */ + s->frame_cntr++; + if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) + s->last_acb_type = frame_descs[bd_idx].acb_type; + switch (frame_descs[bd_idx].acb_type) { + case ACB_TYPE_NONE: + s->last_pitch_val = 0; + break; + case ACB_TYPE_ASYMMETRIC: + s->last_pitch_val = cur_pitch_val; + break; + case ACB_TYPE_HAMMING: + s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; + break; + } + + return 0; +} + +/** + * Ensure minimum value for first item, maximum value for last value, + * proper spacing between each value and proper ordering. + * + * @param lsps array of LSPs + * @param num size of LSP array + * + * @note basically a double version of #ff_acelp_reorder_lsf(), might be + * useful to put in a generic location later on. Parts are also + * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), + * which is in float. + */ +static void stabilize_lsps(double *lsps, int num) +{ + int n, m, l; + + /* set minimum value for first, maximum value for last and minimum + * spacing between LSF values. + * Very similar to ff_set_min_dist_lsf(), but in double. */ + lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); + for (n = 1; n < num; n++) + lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); + lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); + + /* reorder (looks like one-time / non-recursed bubblesort). + * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ + for (n = 1; n < num; n++) { + if (lsps[n] < lsps[n - 1]) { + for (m = 1; m < num; m++) { + double tmp = lsps[m]; + for (l = m - 1; l >= 0; l--) { + if (lsps[l] <= tmp) break; + lsps[l + 1] = lsps[l]; + } + lsps[l + 1] = tmp; + } + break; + } + } +} + +/** + * Test if there's enough bits to read 1 superframe. + * + * @param orig_gb bit I/O context used for reading. This function + * does not modify the state of the bitreader; it + * only uses it to copy the current stream position + * @param s WMA Voice decoding context private data + * @returns -1 if unsupported, 1 on not enough bits or 0 if OK. + */ +static int check_bits_for_superframe(GetBitContext *orig_gb, + WMAVoiceContext *s) +{ + GetBitContext s_gb, *gb = &s_gb; + int n, need_bits, bd_idx; + const struct frame_type_desc *frame_desc; + + /* initialize a copy */ + init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); + skip_bits_long(gb, get_bits_count(orig_gb)); + assert(get_bits_left(gb) == get_bits_left(orig_gb)); + + /* superframe header */ + if (get_bits_left(gb) < 14) + return 1; + if (!get_bits1(gb)) + return -1; // WMAPro-in-WMAVoice superframe + if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe + if (s->has_residual_lsps) { // residual LSPs (for all frames) + if (get_bits_left(gb) < s->sframe_lsp_bitsize) + return 1; + skip_bits_long(gb, s->sframe_lsp_bitsize); + } + + /* frames */ + for (n = 0; n < MAX_FRAMES; n++) { + int aw_idx_is_ext = 0; + + if (!s->has_residual_lsps) { // independent LSPs (per-frame) + if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; + skip_bits_long(gb, s->frame_lsp_bitsize); + } + bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; + if (bd_idx < 0) + return -1; // invalid frame type VLC code + frame_desc = &frame_descs[bd_idx]; + if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { + if (get_bits_left(gb) < s->pitch_nbits) + return 1; + skip_bits_long(gb, s->pitch_nbits); + } + if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { + skip_bits(gb, 8); + } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { + int tmp = get_bits(gb, 6); + if (tmp >= 0x36) { + skip_bits(gb, 2); + aw_idx_is_ext = 1; + } + } + + /* blocks */ + if (frame_desc->acb_type == ACB_TYPE_HAMMING) { + need_bits = s->block_pitch_nbits + + (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; + } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { + need_bits = 2 * !aw_idx_is_ext; + } else + need_bits = 0; + need_bits += frame_desc->frame_size; + if (get_bits_left(gb) < need_bits) + return 1; + skip_bits_long(gb, need_bits); + } + + return 0; +} + +/** + * Synthesize output samples for a single superframe. If we have any data + * cached in s->sframe_cache, that will be used instead of whatever is loaded + * in s->gb. + * + * WMA Voice superframes contain 3 frames, each containing 160 audio samples, + * to give a total of 480 samples per frame. See #synth_frame() for frame + * parsing. In addition to 3 frames, superframes can also contain the LSPs + * (if these are globally specified for all frames (residually); they can + * also be specified individually per-frame. See the s->has_residual_lsps + * option), and can specify the number of samples encoded in this superframe + * (if less than 480), usually used to prevent blanks at track boundaries. + * + * @param ctx WMA Voice decoder context + * @param samples pointer to output buffer for voice samples + * @param data_size pointer containing the size of #samples on input, and the + * amount of #samples filled on output + * @return 0 on success, <0 on error or 1 if there was not enough data to + * fully parse the superframe + */ +static int synth_superframe(AVCodecContext *ctx, + float *samples, int *data_size) +{ + WMAVoiceContext *s = ctx->priv_data; + GetBitContext *gb = &s->gb, s_gb; + int n, res, n_samples = 480; + double lsps[MAX_FRAMES][MAX_LSPS]; + const double *mean_lsf = s->lsps == 16 ? + wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; + float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; + float synth[MAX_LSPS + MAX_SFRAMESIZE]; + + memcpy(synth, s->synth_history, + s->lsps * sizeof(*synth)); + memcpy(excitation, s->excitation_history, + s->history_nsamples * sizeof(*excitation)); + + if (s->sframe_cache_size > 0) { + gb = &s_gb; + init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); + s->sframe_cache_size = 0; + } + + if ((res = check_bits_for_superframe(gb, s)) == 1) return 1; + + /* First bit is speech/music bit, it differentiates between WMAVoice + * speech samples (the actual codec) and WMAVoice music samples, which + * are really WMAPro-in-WMAVoice-superframes. I've never seen those in + * the wild yet. */ + if (!get_bits1(gb)) { + av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1); + return -1; + } + + /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ + if (get_bits1(gb)) { + if ((n_samples = get_bits(gb, 12)) > 480) { + av_log(ctx, AV_LOG_ERROR, + "Superframe encodes >480 samples (%d), not allowed\n", + n_samples); + return -1; + } + } + /* Parse LSPs, if global for the superframe (can also be per-frame). */ + if (s->has_residual_lsps) { + double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; + + for (n = 0; n < s->lsps; n++) + prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; + + if (s->lsps == 10) { + dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); + } else /* s->lsps == 16 */ + dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); + + for (n = 0; n < s->lsps; n++) { + lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); + lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); + lsps[2][n] += mean_lsf[n]; + } + for (n = 0; n < 3; n++) + stabilize_lsps(lsps[n], s->lsps); + } + + /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */ + for (n = 0; n < 3; n++) { + if (!s->has_residual_lsps) { + int m; + + if (s->lsps == 10) { + dequant_lsp10i(gb, lsps[n]); + } else /* s->lsps == 16 */ + dequant_lsp16i(gb, lsps[n]); + + for (m = 0; m < s->lsps; m++) + lsps[n][m] += mean_lsf[m]; + stabilize_lsps(lsps[n], s->lsps); + } + + if ((res = synth_frame(ctx, gb, + &samples[n * MAX_FRAMESIZE], + lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], + &excitation[s->history_nsamples + n * MAX_FRAMESIZE], + &synth[s->lsps + n * MAX_FRAMESIZE]))) + return res; + } + + /* Statistics? FIXME - we don't check for length, a slight overrun + * will be caught by internal buffer padding, and anything else + * will be skipped, not read. */ + if (get_bits1(gb)) { + res = get_bits(gb, 4); + skip_bits(gb, 10 * (res + 1)); + } + + /* Specify nr. of output samples */ + *data_size = n_samples * sizeof(float); + + /* Update history */ + memcpy(s->prev_lsps, lsps[2], + s->lsps * sizeof(*s->prev_lsps)); + memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], + s->lsps * sizeof(*synth)); + memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], + s->history_nsamples * sizeof(*excitation)); + + return 0; +} + +/** + * Parse the packet header at the start of each packet (input data to this + * decoder). + * + * @param s WMA Voice decoding context private data + * @returns 1 if not enough bits were available, or 0 on success. + */ +static int parse_packet_header(WMAVoiceContext *s) +{ + GetBitContext *gb = &s->gb; + unsigned int res; + + if (get_bits_left(gb) < 11) + return 1; + skip_bits(gb, 4); // packet sequence number + s->has_residual_lsps = get_bits1(gb); + do { + res = get_bits(gb, 6); // number of superframes per packet + // (minus first one if there is spillover) + if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) + return 1; + } while (res == 0x3F); + s->spillover_nbits = get_bits(gb, s->spillover_bitsize); + + return 0; +} + +/** + * Copy (unaligned) bits from gb/data/size to pb. + * + * @param pb target buffer to copy bits into + * @param data source buffer to copy bits from + * @param size size of the source data, in bytes + * @param gb bit I/O context specifying the current position in the source. + * data. This function might use this to align the bit position to + * a whole-byte boundary before calling #ff_copy_bits() on aligned + * source data + * @param nbits the amount of bits to copy from source to target + * + * @note after calling this function, the current position in the input bit + * I/O context is undefined. + */ +static void copy_bits(PutBitContext *pb, + const uint8_t *data, int size, + GetBitContext *gb, int nbits) +{ + int rmn_bytes, rmn_bits; + + rmn_bits = rmn_bytes = get_bits_left(gb); + if (rmn_bits < nbits) + return; + rmn_bits &= 7; rmn_bytes >>= 3; + if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) + put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); + ff_copy_bits(pb, data + size - rmn_bytes, + FFMIN(nbits - rmn_bits, rmn_bytes << 3)); +} + +/** + * Packet decoding: a packet is anything that the (ASF) demuxer contains, + * and we expect that the demuxer / application provides it to us as such + * (else you'll probably get garbage as output). Every packet has a size of + * ctx->block_align bytes, starts with a packet header (see + * #parse_packet_header()), and then a series of superframes. Superframe + * boundaries may exceed packets, i.e. superframes can split data over + * multiple (two) packets. + * + * For more information about frames, see #synth_superframe(). + */ +static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, + int *data_size, AVPacket *avpkt) +{ + WMAVoiceContext *s = ctx->priv_data; + GetBitContext *gb = &s->gb; + int size, res, pos; + + if (*data_size < 480 * sizeof(float)) { + av_log(ctx, AV_LOG_ERROR, + "Output buffer too small (%d given - %lu needed)\n", + *data_size, 480 * sizeof(float)); + return -1; + } + *data_size = 0; + + /* Packets are sometimes a multiple of ctx->block_align, with a packet + * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer + * feeds us ASF packets, which may concatenate multiple "codec" packets + * in a single "muxer" packet, so we artificially emulate that by + * capping the packet size at ctx->block_align. */ + for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); + if (!size) + return 0; + init_get_bits(&s->gb, avpkt->data, size << 3); + + /* size == ctx->block_align is used to indicate whether we are dealing with + * a new packet or a packet of which we already read the packet header + * previously. */ + if (size == ctx->block_align) { // new packet header + if ((res = parse_packet_header(s)) < 0) + return res; + + /* If the packet header specifies a s->spillover_nbits, then we want + * to push out all data of the previous packet (+ spillover) before + * continuing to parse new superframes in the current packet. */ + if (s->spillover_nbits > 0) { + if (s->sframe_cache_size > 0) { + int cnt = get_bits_count(gb); + copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); + flush_put_bits(&s->pb); + s->sframe_cache_size += s->spillover_nbits; + if ((res = synth_superframe(ctx, data, data_size)) == 0 && + *data_size > 0) { + cnt += s->spillover_nbits; + s->skip_bits_next = cnt & 7; + return cnt >> 3; + } else + skip_bits_long (gb, s->spillover_nbits - cnt + + get_bits_count(gb)); // resync + } else + skip_bits_long(gb, s->spillover_nbits); // resync + } + } else if (s->skip_bits_next) + skip_bits(gb, s->skip_bits_next); + + /* Try parsing superframes in current packet */ + s->sframe_cache_size = 0; + s->skip_bits_next = 0; + pos = get_bits_left(gb); + if ((res = synth_superframe(ctx, data, data_size)) < 0) { + return res; + } else if (*data_size > 0) { + int cnt = get_bits_count(gb); + s->skip_bits_next = cnt & 7; + return cnt >> 3; + } else if ((s->sframe_cache_size = pos) > 0) { + /* rewind bit reader to start of last (incomplete) superframe... */ + init_get_bits(gb, avpkt->data, size << 3); + skip_bits_long(gb, (size << 3) - pos); + assert(get_bits_left(gb) == pos); + + /* ...and cache it for spillover in next packet */ + init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); + copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); + // FIXME bad - just copy bytes as whole and add use the + // skip_bits_next field + } + + return size; +} + +static av_cold void wmavoice_flush(AVCodecContext *ctx) +{ + WMAVoiceContext *s = ctx->priv_data; + int n; + + s->sframe_cache_size = 0; + s->skip_bits_next = 0; + for (n = 0; n < s->lsps; n++) + s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); + memset(s->excitation_history, 0, + sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); + memset(s->synth_history, 0, + sizeof(*s->synth_history) * MAX_LSPS); + memset(s->gain_pred_err, 0, + sizeof(s->gain_pred_err)); +} + +AVCodec wmavoice_decoder = { + "wmavoice", + CODEC_TYPE_AUDIO, + CODEC_ID_WMAVOICE, + sizeof(WMAVoiceContext), + wmavoice_decode_init, + NULL, + NULL, + wmavoice_decode_packet, + CODEC_CAP_SUBFRAMES, + .flush = wmavoice_flush, + .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), +};