Mercurial > libavcodec.hg
view wmavoice.c @ 11926:43e39953f4f9 libavcodec
Improve some uses of ff_cropTbl with constant offset
| author | mru |
|---|---|
| date | Tue, 22 Jun 2010 23:12:48 +0000 |
| parents | 5c1363c233b8 |
| children | 18c23a632001 |
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/* * 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 * @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" #include "avfft.h" #include "fft.h" #define MAX_BLOCKS 8 ///< maximum number of blocks per frame #define MAX_LSPS 16 ///< maximum filter order #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple ///< of 16 for ASM input buffer alignment #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) /* postfilter specific values */ int do_apf; ///< whether to apply the averaged ///< projection filter (APF) int denoise_strength; ///< strength of denoising in Wiener filter ///< [0-11] int denoise_tilt_corr; ///< Whether to apply tilt correction to the ///< Wiener filter coefficients (postfilter) int dc_level; ///< Predicted amount of DC noise, based ///< on which a DC removal filter is used 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 /** * @} * @defgroup post_filter Postfilter values * Varibales used for postfilter implementation, mostly history for * smoothing and so on, and context variables for FFT/iFFT. * @{ */ RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the ///< postfilter (for denoise filter) DCTContext dct, dst; ///< contexts for phase shift (in Hilbert ///< transform, part of postfilter) float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] ///< range float postfilter_agc; ///< gain control memory, used in ///< #adaptive_gain_control() float dcf_mem[2]; ///< DC filter history float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; ///< zero filter output (i.e. excitation) ///< by postfilter float denoise_filter_cache[MAX_FRAMESIZE]; int denoise_filter_cache_size; ///< samples in #denoise_filter_cache DECLARE_ALIGNED(16, float, tilted_lpcs_pf)[0x80]; ///< aligned buffer for LPC tilting DECLARE_ALIGNED(16, float, denoise_coeffs_pf)[0x80]; ///< aligned buffer for denoise coefficients DECLARE_ALIGNED(16, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; ///< aligned buffer for postfilter speech ///< synthesis /** * @} */ } 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; if (s->do_apf) { ff_rdft_init(&s->rdft, 7, DFT_R2C); ff_rdft_init(&s->irdft, 7, IDFT_C2R); ff_dct_init(&s->dct, 6, DCT_I); ff_dct_init(&s->dst, 6, DST_I); ff_sine_window_init(s->cos, 256); memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); for (n = 0; n < 255; n++) { s->sin[n] = -s->sin[510 - n]; s->cos[510 - n] = s->cos[n]; } } s->denoise_strength = (flags >> 2) & 0xF; if (s->denoise_strength >= 12) { av_log(ctx, AV_LOG_ERROR, "Invalid denoise filter strength %d (max=11)\n", s->denoise_strength); return -1; } s->denoise_tilt_corr = !!(flags & 0x40); s->dc_level = (flags >> 7) & 0xF; 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; } /** * @defgroup postfilter Postfilter functions * Postfilter functions (gain control, wiener denoise filter, DC filter, * kalman smoothening, plus surrounding code to wrap it) * @{ */ /** * Adaptive gain control (as used in postfilter). * * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except * that the energy here is calculated using sum(abs(...)), whereas the * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). * * @param out output buffer for filtered samples * @param in input buffer containing the samples as they are after the * postfilter steps so far * @param speech_synth input buffer containing speech synth before postfilter * @param size input buffer size * @param alpha exponential filter factor * @param gain_mem pointer to filter memory (single float) */ static void adaptive_gain_control(float *out, const float *in, const float *speech_synth, int size, float alpha, float *gain_mem) { int i; float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; float mem = *gain_mem; for (i = 0; i < size; i++) { speech_energy += fabsf(speech_synth[i]); postfilter_energy += fabsf(in[i]); } gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy; for (i = 0; i < size; i++) { mem = alpha * mem + gain_scale_factor; out[i] = in[i] * mem; } *gain_mem = mem; } /** * Kalman smoothing function. * * This function looks back pitch +/- 3 samples back into history to find * the best fitting curve (that one giving the optimal gain of the two * signals, i.e. the highest dot product between the two), and then * uses that signal history to smoothen the output of the speech synthesis * filter. * * @param s WMA Voice decoding context * @param pitch pitch of the speech signal * @param in input speech signal * @param out output pointer for smoothened signal * @param size input/output buffer size * * @returns -1 if no smoothening took place, e.g. because no optimal * fit could be found, or 0 on success. */ static int kalman_smoothen(WMAVoiceContext *s, int pitch, const float *in, float *out, int size) { int n; float optimal_gain = 0, dot; const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], *best_hist_ptr; /* find best fitting point in history */ do { dot = ff_dot_productf(in, ptr, size); if (dot > optimal_gain) { optimal_gain = dot; best_hist_ptr = ptr; } } while (--ptr >= end); if (optimal_gain <= 0) return -1; dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size); if (dot <= 0) // would be 1.0 return -1; if (optimal_gain <= dot) { dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000 } else dot = 0.625; /* actual smoothing */ for (n = 0; n < size; n++) out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]); return 0; } /** * Get the tilt factor of a formant filter from its transfer function * @see #tilt_factor() in amrnbdec.c, which does essentially the same, * but somehow (??) it does a speech synthesis filter in the * middle, which is missing here * * @param lpcs LPC coefficients * @param n_lpcs Size of LPC buffer * @returns the tilt factor */ static float tilt_factor(const float *lpcs, int n_lpcs) { float rh0, rh1; rh0 = 1.0 + ff_dot_productf(lpcs, lpcs, n_lpcs); rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1); return rh1 / rh0; } /** * Derive denoise filter coefficients (in real domain) from the LPCs. */ static void calc_input_response(WMAVoiceContext *s, float *lpcs, int fcb_type, float *coeffs, int remainder) { float last_coeff, min = 15.0, max = -15.0; float irange, angle_mul, gain_mul, range, sq; int n, idx; /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ ff_rdft_calc(&s->rdft, lpcs); #define log_range(var, assign) do { \ float tmp = log10f(assign); var = tmp; \ max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ } while (0) log_range(last_coeff, lpcs[1] * lpcs[1]); for (n = 1; n < 64; n++) log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); log_range(lpcs[0], lpcs[0] * lpcs[0]); #undef log_range range = max - min; lpcs[64] = last_coeff; /* Now, use this spectrum to pick out these frequencies with higher * (relative) power/energy (which we then take to be "not noise"), * and set up a table (still in lpc[]) of (relative) gains per frequency. * These frequencies will be maintained, while others ("noise") will be * decreased in the filter output. */ irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63] gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : (5.0 / 14.7)); angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); for (n = 0; n <= 64; n++) { float pow; idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1); pow = wmavoice_denoise_power_table[s->denoise_strength][idx]; lpcs[n] = angle_mul * pow; /* 70.57 =~ 1/log10(1.0331663) */ idx = (pow * gain_mul - 0.0295) * 70.570526123; if (idx > 127) { // fallback if index falls outside table range coeffs[n] = wmavoice_energy_table[127] * powf(1.0331663, idx - 127); } else coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; } /* calculate the Hilbert transform of the gains, which we do (since this * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()). * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the * "moment" of the LPCs in this filter. */ ff_dct_calc(&s->dct, lpcs); ff_dct_calc(&s->dst, lpcs); /* Split out the coefficient indexes into phase/magnitude pairs */ idx = 255 + av_clip(lpcs[64], -255, 255); coeffs[0] = coeffs[0] * s->cos[idx]; idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); last_coeff = coeffs[64] * s->cos[idx]; for (n = 63;; n--) { idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; coeffs[n * 2] = coeffs[n] * s->cos[idx]; if (!--n) break; idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; coeffs[n * 2] = coeffs[n] * s->cos[idx]; } coeffs[1] = last_coeff; /* move into real domain */ ff_rdft_calc(&s->irdft, coeffs); /* tilt correction and normalize scale */ memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); if (s->denoise_tilt_corr) { float tilt_mem = 0; coeffs[remainder - 1] = 0; ff_tilt_compensation(&tilt_mem, -1.8 * tilt_factor(coeffs, remainder - 1), coeffs, remainder); } sq = (1.0 / 64.0) * sqrtf(1 / ff_dot_productf(coeffs, coeffs, remainder)); for (n = 0; n < remainder; n++) coeffs[n] *= sq; } /** * This function applies a Wiener filter on the (noisy) speech signal as * a means to denoise it. * * - take RDFT of LPCs to get the power spectrum of the noise + speech; * - using this power spectrum, calculate (for each frequency) the Wiener * filter gain, which depends on the frequency power and desired level * of noise subtraction (when set too high, this leads to artifacts) * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse * of 4-8kHz); * - by doing a phase shift, calculate the Hilbert transform of this array * of per-frequency filter-gains to get the filtering coefficients; * - smoothen/normalize/de-tilt these filter coefficients as desired; * - take RDFT of noisy sound, apply the coefficients and take its IRDFT * to get the denoised speech signal; * - the leftover (i.e. output of the IRDFT on denoised speech data beyond * the frame boundary) are saved and applied to subsequent frames by an * overlap-add method (otherwise you get clicking-artifacts). * * @param s WMA Voice decoding context * @param s fcb_type Frame (codebook) type * @param synth_pf input: the noisy speech signal, output: denoised speech * data; should be 16-byte aligned (for ASM purposes) * @param size size of the speech data * @param lpcs LPCs used to synthesize this frame's speech data */ static void wiener_denoise(WMAVoiceContext *s, int fcb_type, float *synth_pf, int size, const float *lpcs) { int remainder, lim, n; if (fcb_type != FCB_TYPE_SILENCE) { float *tilted_lpcs = s->tilted_lpcs_pf, *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; tilted_lpcs[0] = 1.0; memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); memset(&tilted_lpcs[s->lsps + 1], 0, sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), tilted_lpcs, s->lsps + 2); /* The IRDFT output (127 samples for 7-bit filter) beyond the frame * size is applied to the next frame. All input beyond this is zero, * and thus all output beyond this will go towards zero, hence we can * limit to min(size-1, 127-size) as a performance consideration. */ remainder = FFMIN(127 - size, size - 1); calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); /* apply coefficients (in frequency spectrum domain), i.e. complex * number multiplication */ memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); ff_rdft_calc(&s->rdft, synth_pf); ff_rdft_calc(&s->rdft, coeffs); synth_pf[0] *= coeffs[0]; synth_pf[1] *= coeffs[1]; for (n = 1; n < 64; n++) { float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; } ff_rdft_calc(&s->irdft, synth_pf); } /* merge filter output with the history of previous runs */ if (s->denoise_filter_cache_size) { lim = FFMIN(s->denoise_filter_cache_size, size); for (n = 0; n < lim; n++) synth_pf[n] += s->denoise_filter_cache[n]; s->denoise_filter_cache_size -= lim; memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); } /* move remainder of filter output into a cache for future runs */ if (fcb_type != FCB_TYPE_SILENCE) { lim = FFMIN(remainder, s->denoise_filter_cache_size); for (n = 0; n < lim; n++) s->denoise_filter_cache[n] += synth_pf[size + n]; if (lim < remainder) { memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); s->denoise_filter_cache_size = remainder; } } } /** * Averaging projection filter, the postfilter used in WMAVoice. * * This uses the following steps: * - A zero-synthesis filter (generate excitation from synth signal) * - Kalman smoothing on excitation, based on pitch * - Re-synthesized smoothened output * - Iterative Wiener denoise filter * - Adaptive gain filter * - DC filter * * @param s WMAVoice decoding context * @param synth Speech synthesis output (before postfilter) * @param samples Output buffer for filtered samples * @param size Buffer size of synth & samples * @param lpcs Generated LPCs used for speech synthesis * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) * @param pitch Pitch of the input signal */ static void postfilter(WMAVoiceContext *s, const float *synth, float *samples, int size, const float *lpcs, float *zero_exc_pf, int fcb_type, int pitch) { float synth_filter_in_buf[MAX_FRAMESIZE / 2], *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], *synth_filter_in = zero_exc_pf; assert(size <= MAX_FRAMESIZE / 2); /* generate excitation from input signal */ ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); if (fcb_type >= FCB_TYPE_AW_PULSES && !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) synth_filter_in = synth_filter_in_buf; /* re-synthesize speech after smoothening, and keep history */ ff_celp_lp_synthesis_filterf(synth_pf, lpcs, synth_filter_in, size, s->lsps); memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], sizeof(synth_pf[0]) * s->lsps); wiener_denoise(s, fcb_type, synth_pf, size, lpcs); adaptive_gain_control(samples, synth_pf, synth, size, 0.99, &s->postfilter_agc); if (s->dc_level > 8) { /* remove ultra-low frequency DC noise / highpass filter; * coefficients are identical to those used in SIPR decoding, * and very closely resemble those used in AMR-NB decoding. */ ff_acelp_apply_order_2_transfer_function(samples, samples, (const float[2]) { -1.99997, 1.0 }, (const float[2]) { -1.9330735188, 0.93589198496 }, 0.93980580475, s->dcf_mem, size); } } /** * @} */ /** * 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 * @return 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 frame_idx Frame number within superframe [0-2] * @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, int frame_idx, 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) { double i_lsps[MAX_LSPS]; float lpcs[MAX_LSPS]; for (n = 0; n < s->lsps; n++) // LSF -> LSP i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); postfilter(s, synth, samples, 80, lpcs, &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], frame_descs[bd_idx].fcb_type, pitch[0]); for (n = 0; n < s->lsps; n++) // LSF -> LSP i_lsps[n] = cos(lsps[n]); ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); postfilter(s, &synth[80], &samples[80], 80, lpcs, &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], frame_descs[bd_idx].fcb_type, pitch[0]); } else memcpy(samples, synth, 160 * sizeof(synth[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 * @return -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, n, &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)); if (s->do_apf) memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], s->history_nsamples * sizeof(*s->zero_exc_pf)); 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 * @return 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 int wmavoice_decode_end(AVCodecContext *ctx) { WMAVoiceContext *s = ctx->priv_data; if (s->do_apf) { ff_rdft_end(&s->rdft); ff_rdft_end(&s->irdft); ff_dct_end(&s->dct); ff_dct_end(&s->dst); } return 0; } static av_cold void wmavoice_flush(AVCodecContext *ctx) { WMAVoiceContext *s = ctx->priv_data; int n; s->postfilter_agc = 0; 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)); if (s->do_apf) { memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, sizeof(*s->synth_filter_out_buf) * s->lsps); memset(s->dcf_mem, 0, sizeof(*s->dcf_mem) * 2); memset(s->zero_exc_pf, 0, sizeof(*s->zero_exc_pf) * s->history_nsamples); memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); } } AVCodec wmavoice_decoder = { "wmavoice", AVMEDIA_TYPE_AUDIO, CODEC_ID_WMAVOICE, sizeof(WMAVoiceContext), wmavoice_decode_init, NULL, wmavoice_decode_end, wmavoice_decode_packet, CODEC_CAP_SUBFRAMES, .flush = wmavoice_flush, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), };