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Diffstat (limited to 'ffmpeg/libavcodec/wmavoice.c')
| -rw-r--r-- | ffmpeg/libavcodec/wmavoice.c | 2075 |
1 files changed, 0 insertions, 2075 deletions
diff --git a/ffmpeg/libavcodec/wmavoice.c b/ffmpeg/libavcodec/wmavoice.c deleted file mode 100644 index c2737ab..0000000 --- a/ffmpeg/libavcodec/wmavoice.c +++ /dev/null @@ -1,2075 +0,0 @@ -/* - * 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 "libavutil/channel_layout.h" -#include "libavutil/float_dsp.h" -#include "libavutil/mem.h" -#include "avcodec.h" -#include "internal.h" -#include "get_bits.h" -#include "put_bits.h" -#include "wmavoice_data.h" -#include "celp_filters.h" -#include "acelp_vectors.h" -#include "acelp_filters.h" -#include "lsp.h" -#include "dct.h" -#include "rdft.h" -#include "sinewin.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 { - /** - * @name 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 - - /** - * @} - * - * @name 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 preceding 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 - - /** - * @} - * - * @name 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 - /** - * @} - * - * @name Postfilter values - * - * Variables 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(32, float, tilted_lpcs_pf)[0x80]; - ///< aligned buffer for LPC tilting - DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80]; - ///< aligned buffer for denoise coefficients - DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; - ///< aligned buffer for postfilter speech - ///< synthesis - /** - * @} - */ -} WMAVoiceContext; - -/** - * Set 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]) -{ - int cntr[8] = { 0 }, n, res; - - memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); - 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; - } - return 0; -} - -static av_cold void wmavoice_init_static_data(AVCodec *codec) -{ - 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 - }; - - INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), - bits, 1, 1, codes, 2, 2, 132); -} - -/** - * 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 AVERROR_INVALIDDATA; - } - 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 AVERROR_INVALIDDATA; - } - 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 AVERROR_INVALIDDATA; - } - - 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; - if (pitch_range <= 0) { - av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n"); - return AVERROR_INVALIDDATA; - } - 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 AVERROR(ENOSYS); - } - - 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; - if (s->block_delta_pitch_hrange <= 0) { - av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n"); - return AVERROR_INVALIDDATA; - } - 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->channels = 1; - ctx->channel_layout = AV_CH_LAYOUT_MONO; - ctx->sample_fmt = AV_SAMPLE_FMT_FLT; - - return 0; -} - -/** - * @name 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 = NULL; - - /* find best fitting point in history */ - do { - dot = avpriv_scalarproduct_float_c(in, ptr, size); - if (dot > optimal_gain) { - optimal_gain = dot; - best_hist_ptr = ptr; - } - } while (--ptr >= end); - - if (optimal_gain <= 0) - return -1; - dot = avpriv_scalarproduct_float_c(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 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs); - rh1 = lpcs[0] + avpriv_scalarproduct_float_c(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) */ - s->rdft.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 pwr; - - idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1); - pwr = wmavoice_denoise_power_table[s->denoise_strength][idx]; - lpcs[n] = angle_mul * pwr; - - /* 70.57 =~ 1/log10(1.0331663) */ - idx = (pwr * gain_mul - 0.0295) * 70.570526123; - if (idx > 127) { // fall back 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 sine 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. */ - s->dct.dct_calc(&s->dct, lpcs); - s->dst.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 */ - s->irdft.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 / avpriv_scalarproduct_float_c(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 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)); - s->rdft.rdft_calc(&s->rdft, synth_pf); - s->rdft.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]; - } - s->irdft.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 zero_exc_pf destination for zero synthesis filter (16-byte aligned) - * @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; - - av_assert0(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; - } -} - -/** - * @name 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]); -} - -/** - * @} - * @name 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 - * @return -1 on error, 0 otherwise - */ -static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, - int block_idx, AMRFixed *fcb) -{ - uint16_t use_mask_mem[9]; // only 5 are used, rest is padding - uint16_t *use_mask = use_mask_mem + 2; - /* 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[-2], 0, 2 * sizeof(use_mask[0])); - 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++ &= 0xFFFFu << 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 -1; - 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; - return 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; - - av_assert0(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; - - av_assert0(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); - if (aw_pulse_set2(s, gb, block_idx, &fcb)) { - /* Conceal the block with silence and return. - * Skip the correct amount of bits to read the next - * block from the correct offset. */ - int r_idx = pRNG(s->frame_cntr, block_idx, size); - - for (n = 0; n < size; n++) - excitation[n] = - wmavoice_std_codebook[r_idx + n] * s->silence_gain; - skip_bits(gb, 7 + 1); - return; - } - } 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(avpriv_scalarproduct_float_c(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((uint8_t *) 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, av_uninit(cur_pitch_val); - int pitch[MAX_BLOCKS], av_uninit(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; - - if (bd_idx < 0) { - av_log(ctx, AV_LOG_ERROR, - "Invalid frame type VLC code, skipping\n"); - return AVERROR_INVALIDDATA; - } - - block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; - - /* 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 < 0 on error, 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)); - av_assert1(get_bits_left(gb) == get_bits_left(orig_gb)); - - /* superframe header */ - if (get_bits_left(gb) < 14) - return 1; - if (!get_bits1(gb)) - return AVERROR(ENOSYS); // 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 AVERROR_INVALIDDATA; // 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 - * @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, AVFrame *frame, - int *got_frame_ptr) -{ - 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]; - float *samples; - - 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) { - *got_frame_ptr = 0; - return 1; - } else if (res < 0) - return res; - - /* 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)) { - avpriv_request_sample(ctx, "WMAPro-in-WMAVoice"); - return AVERROR_PATCHWELCOME; - } - - /* (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 AVERROR_INVALIDDATA; - } - } - /* 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); - } - - /* get output buffer */ - frame->nb_samples = 480; - if ((res = ff_get_buffer(ctx, frame, 0)) < 0) - return res; - frame->nb_samples = n_samples; - samples = (float *)frame->data[0]; - - /* Parse frames, optionally preceded 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]))) { - *got_frame_ptr = 0; - 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)); - } - - *got_frame_ptr = 1; - - /* 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 #avpriv_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; - if (nbits > pb->size_in_bits - put_bits_count(pb)) - 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)); - avpriv_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 *got_frame_ptr, AVPacket *avpkt) -{ - WMAVoiceContext *s = ctx->priv_data; - GetBitContext *gb = &s->gb; - int size, res, pos; - - /* 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) { - *got_frame_ptr = 0; - 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, got_frame_ptr)) == 0 && - *got_frame_ptr) { - 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, got_frame_ptr)) < 0) { - return res; - } else if (*got_frame_ptr) { - 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); - av_assert1(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 ff_wmavoice_decoder = { - .name = "wmavoice", - .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_WMAVOICE, - .priv_data_size = sizeof(WMAVoiceContext), - .init = wmavoice_decode_init, - .init_static_data = wmavoice_init_static_data, - .close = wmavoice_decode_end, - .decode = wmavoice_decode_packet, - .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, - .flush = wmavoice_flush, -}; |