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