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authorTim Redfern <tim@eclectronics.org>2013-09-05 17:57:22 +0100
committerTim Redfern <tim@eclectronics.org>2013-09-05 17:57:22 +0100
commit8992cb1d0d07edc33d274f6d7924ecdf6f83d994 (patch)
tree3a2c86846b7eec8137c1507e623fc7018f13d453 /ffmpeg/libavcodec/wmavoice.c
parent741fb4b9e135cfb161a749db88713229038577bb (diff)
making act segmenter
Diffstat (limited to 'ffmpeg/libavcodec/wmavoice.c')
-rw-r--r--ffmpeg/libavcodec/wmavoice.c2055
1 files changed, 2055 insertions, 0 deletions
diff --git a/ffmpeg/libavcodec/wmavoice.c b/ffmpeg/libavcodec/wmavoice.c
new file mode 100644
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--- /dev/null
+++ b/ffmpeg/libavcodec/wmavoice.c
@@ -0,0 +1,2055 @@
+/*
+ * 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])
+{
+ 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] = { 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;
+ }
+ 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;
+ if (pitch_range <= 0) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
+ return -1;
+ }
+ 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;
+ if (s->block_delta_pitch_hrange <= 0) {
+ av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
+ return -1;
+ }
+ 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) { // 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. */
+ 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
+ */
+static void 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;
+ 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;
+
+ 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);
+ 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(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 -1;
+ }
+
+ 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 -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));
+ 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 -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
+ * @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;
+ }
+
+ /* 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 -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);
+ }
+
+ /* 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",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_WMAVOICE,
+ .priv_data_size = sizeof(WMAVoiceContext),
+ .init = wmavoice_decode_init,
+ .close = wmavoice_decode_end,
+ .decode = wmavoice_decode_packet,
+ .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
+ .flush = wmavoice_flush,
+ .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
+};
generated by cgit v1.2.3 (git 2.39.1) at 2025-12-06 19:49:36 +0000