| /* |
| * 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 "libavutil/thread.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 |
| } frame_descs[17] = { |
| { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0 }, |
| { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0 }, |
| { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0 }, |
| { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, |
| { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, |
| { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0 }, |
| { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, |
| { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, |
| { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, |
| { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, |
| { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, |
| { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, |
| { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, |
| { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, |
| { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, |
| { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, |
| { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 } |
| }; |
| |
| /** |
| * WMA Voice decoding context. |
| */ |
| typedef struct WMAVoiceContext { |
| /** |
| * @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 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 + AV_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() |
| int nb_superframes; ///< number of superframes in current packet |
| float gain_pred_err[6]; ///< cache for gain prediction |
| float excitation_history[MAX_SIGNAL_HISTORY]; |
| ///< cache of the signal of previous |
| ///< superframes, used as a history for |
| ///< signal generation |
| float synth_history[MAX_LSPS]; ///< see #excitation_history |
| /** |
| * @} |
| * |
| * @name Postfilter values |
| * |
| * Variables used for postfilter implementation, mostly history for |
| * smoothing and so on, and context variables for FFT/iFFT. |
| * @{ |
| */ |
| RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the |
| ///< postfilter (for denoise filter) |
| DCTContext dct, dst; ///< contexts for phase shift (in Hilbert |
| ///< transform, part of postfilter) |
| float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] |
| ///< range |
| float postfilter_agc; ///< gain control memory, used in |
| ///< #adaptive_gain_control() |
| float dcf_mem[2]; ///< DC filter history |
| float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; |
| ///< zero filter output (i.e. excitation) |
| ///< by postfilter |
| float denoise_filter_cache[MAX_FRAMESIZE]; |
| int denoise_filter_cache_size; ///< samples in #denoise_filter_cache |
| DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80]; |
| ///< aligned buffer for LPC tilting |
| DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80]; |
| ///< aligned buffer for denoise coefficients |
| DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; |
| ///< aligned buffer for postfilter speech |
| ///< synthesis |
| /** |
| * @} |
| */ |
| } WMAVoiceContext; |
| |
| /** |
| * Set up the variable bit mode (VBM) tree from container extradata. |
| * @param gb bit I/O context. |
| * The bit context (s->gb) should be loaded with byte 23-46 of the |
| * container extradata (i.e. the ones containing the VBM tree). |
| * @param vbm_tree pointer to array to which the decoded VBM tree will be |
| * written. |
| * @return 0 on success, <0 on error. |
| */ |
| static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) |
| { |
| int cntr[8] = { 0 }, n, res; |
| |
| memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); |
| for (n = 0; n < 17; n++) { |
| res = get_bits(gb, 3); |
| if (cntr[res] > 3) // should be >= 3 + (res == 7)) |
| return -1; |
| vbm_tree[res * 3 + cntr[res]++] = n; |
| } |
| return 0; |
| } |
| |
| static av_cold void wmavoice_init_static_data(void) |
| { |
| static const uint8_t bits[] = { |
| 2, 2, 2, 4, 4, 4, |
| 6, 6, 6, 8, 8, 8, |
| 10, 10, 10, 12, 12, 12, |
| 14, 14, 14, 14 |
| }; |
| static const uint16_t codes[] = { |
| 0x0000, 0x0001, 0x0002, // 00/01/10 |
| 0x000c, 0x000d, 0x000e, // 11+00/01/10 |
| 0x003c, 0x003d, 0x003e, // 1111+00/01/10 |
| 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 |
| 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 |
| 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 |
| 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx |
| }; |
| |
| INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), |
| bits, 1, 1, codes, 2, 2, 132); |
| } |
| |
| 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)); |
| } |
| } |
| |
| /** |
| * Set up decoder with parameters from demuxer (extradata etc.). |
| */ |
| static av_cold int wmavoice_decode_init(AVCodecContext *ctx) |
| { |
| static AVOnce init_static_once = AV_ONCE_INIT; |
| int n, flags, pitch_range, lsp16_flag; |
| WMAVoiceContext *s = ctx->priv_data; |
| |
| ff_thread_once(&init_static_once, wmavoice_init_static_data); |
| |
| /** |
| * Extradata layout: |
| * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), |
| * - byte 19-22: flags field (annoyingly in LE; see below for known |
| * values), |
| * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, |
| * rest is 0). |
| */ |
| if (ctx->extradata_size != 46) { |
| av_log(ctx, AV_LOG_ERROR, |
| "Invalid extradata size %d (should be 46)\n", |
| ctx->extradata_size); |
| return AVERROR_INVALIDDATA; |
| } |
| if (ctx->block_align <= 0 || ctx->block_align > (1<<22)) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid block alignment %d.\n", ctx->block_align); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| flags = AV_RL32(ctx->extradata + 18); |
| s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); |
| s->do_apf = flags & 0x1; |
| if (s->do_apf) { |
| ff_rdft_init(&s->rdft, 7, DFT_R2C); |
| ff_rdft_init(&s->irdft, 7, IDFT_C2R); |
| ff_dct_init(&s->dct, 6, DCT_I); |
| ff_dct_init(&s->dst, 6, DST_I); |
| |
| ff_sine_window_init(s->cos, 256); |
| memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); |
| for (n = 0; n < 255; n++) { |
| s->sin[n] = -s->sin[510 - n]; |
| s->cos[510 - n] = s->cos[n]; |
| } |
| } |
| s->denoise_strength = (flags >> 2) & 0xF; |
| if (s->denoise_strength >= 12) { |
| av_log(ctx, AV_LOG_ERROR, |
| "Invalid denoise filter strength %d (max=11)\n", |
| s->denoise_strength); |
| return AVERROR_INVALIDDATA; |
| } |
| s->denoise_tilt_corr = !!(flags & 0x40); |
| s->dc_level = (flags >> 7) & 0xF; |
| s->lsp_q_mode = !!(flags & 0x2000); |
| s->lsp_def_mode = !!(flags & 0x4000); |
| lsp16_flag = flags & 0x1000; |
| if (lsp16_flag) { |
| s->lsps = 16; |
| } else { |
| s->lsps = 10; |
| } |
| for (n = 0; n < s->lsps; n++) |
| s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); |
| |
| init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); |
| if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (ctx->sample_rate >= INT_MAX / (256 * 37)) |
| return AVERROR_INVALIDDATA; |
| |
| s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; |
| s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; |
| pitch_range = s->max_pitch_val - s->min_pitch_val; |
| if (pitch_range <= 0) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| s->pitch_nbits = av_ceil_log2(pitch_range); |
| s->last_pitch_val = 40; |
| s->last_acb_type = ACB_TYPE_NONE; |
| s->history_nsamples = s->max_pitch_val + 8; |
| |
| if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { |
| int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, |
| max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; |
| |
| av_log(ctx, AV_LOG_ERROR, |
| "Unsupported samplerate %d (min=%d, max=%d)\n", |
| ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz |
| |
| return AVERROR(ENOSYS); |
| } |
| |
| s->block_conv_table[0] = s->min_pitch_val; |
| s->block_conv_table[1] = (pitch_range * 25) >> 6; |
| s->block_conv_table[2] = (pitch_range * 44) >> 6; |
| s->block_conv_table[3] = s->max_pitch_val - 1; |
| s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; |
| if (s->block_delta_pitch_hrange <= 0) { |
| av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); |
| s->block_pitch_range = s->block_conv_table[2] + |
| s->block_conv_table[3] + 1 + |
| 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); |
| s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); |
| |
| ctx->channels = 1; |
| ctx->channel_layout = AV_CH_LAYOUT_MONO; |
| ctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
| |
| return 0; |
| } |
| |
| /** |
| * @name Postfilter functions |
| * Postfilter functions (gain control, wiener denoise filter, DC filter, |
| * kalman smoothening, plus surrounding code to wrap it) |
| * @{ |
| */ |
| /** |
| * Adaptive gain control (as used in postfilter). |
| * |
| * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except |
| * that the energy here is calculated using sum(abs(...)), whereas the |
| * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). |
| * |
| * @param out output buffer for filtered samples |
| * @param in input buffer containing the samples as they are after the |
| * postfilter steps so far |
| * @param speech_synth input buffer containing speech synth before postfilter |
| * @param size input buffer size |
| * @param alpha exponential filter factor |
| * @param gain_mem pointer to filter memory (single float) |
| */ |
| static void adaptive_gain_control(float *out, const float *in, |
| const float *speech_synth, |
| int size, float alpha, float *gain_mem) |
| { |
| int i; |
| float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; |
| float mem = *gain_mem; |
| |
| for (i = 0; i < size; i++) { |
| speech_energy += fabsf(speech_synth[i]); |
| postfilter_energy += fabsf(in[i]); |
| } |
| gain_scale_factor = postfilter_energy == 0.0 ? 0.0 : |
| (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 = lrint((max - lpcs[n]) * irange - 1); |
| idx = FFMAX(0, idx); |
| pwr = wmavoice_denoise_power_table[s->denoise_strength][idx]; |
| lpcs[n] = angle_mul * pwr; |
| |
| /* 70.57 =~ 1/log10(1.0331663) */ |
| idx = av_clipf((pwr * gain_mul - 0.0295) * 70.570526123, 0, INT_MAX / 2); |
| |
| if (idx > 127) { // fall back if index falls outside table range |
| coeffs[n] = wmavoice_energy_table[127] * |
| powf(1.0331663, idx - 127); |
| } else |
| coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; |
| } |
| |
| /* calculate the Hilbert transform of the gains, which we do (since this |
| * is a sine input) by doing a phase shift (in theory, H(sin())=cos()). |
| * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the |
| * "moment" of the LPCs in this filter. */ |
| s->dct.dct_calc(&s->dct, lpcs); |
| s->dst.dct_calc(&s->dst, lpcs); |
| |
| /* Split out the coefficient indexes into phase/magnitude pairs */ |
| idx = 255 + av_clip(lpcs[64], -255, 255); |
| coeffs[0] = coeffs[0] * s->cos[idx]; |
| idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); |
| last_coeff = coeffs[64] * s->cos[idx]; |
| for (n = 63;; n--) { |
| idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); |
| coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; |
| coeffs[n * 2] = coeffs[n] * s->cos[idx]; |
| |
| if (!--n) break; |
| |
| idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); |
| coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; |
| coeffs[n * 2] = coeffs[n] * s->cos[idx]; |
| } |
| coeffs[1] = last_coeff; |
| |
| /* move into real domain */ |
| s->irdft.rdft_calc(&s->irdft, coeffs); |
| |
| /* tilt correction and normalize scale */ |
| memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); |
| if (s->denoise_tilt_corr) { |
| float tilt_mem = 0; |
| |
| coeffs[remainder - 1] = 0; |
| ff_tilt_compensation(&tilt_mem, |
| -1.8 * tilt_factor(coeffs, remainder - 1), |
| coeffs, remainder); |
| } |
| sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs, |
| remainder)); |
| for (n = 0; n < remainder; n++) |
| coeffs[n] *= sq; |
| } |
| |
| /** |
| * This function applies a Wiener filter on the (noisy) speech signal as |
| * a means to denoise it. |
| * |
| * - take RDFT of LPCs to get the power spectrum of the noise + speech; |
| * - using this power spectrum, calculate (for each frequency) the Wiener |
| * filter gain, which depends on the frequency power and desired level |
| * of noise subtraction (when set too high, this leads to artifacts) |
| * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse |
| * of 4-8kHz); |
| * - by doing a phase shift, calculate the Hilbert transform of this array |
| * of per-frequency filter-gains to get the filtering coefficients; |
| * - smoothen/normalize/de-tilt these filter coefficients as desired; |
| * - take RDFT of noisy sound, apply the coefficients and take its IRDFT |
| * to get the denoised speech signal; |
| * - the leftover (i.e. output of the IRDFT on denoised speech data beyond |
| * the frame boundary) are saved and applied to subsequent frames by an |
| * overlap-add method (otherwise you get clicking-artifacts). |
| * |
| * @param s WMA Voice decoding context |
| * @param fcb_type Frame (codebook) type |
| * @param synth_pf input: the noisy speech signal, output: denoised speech |
| * data; should be 16-byte aligned (for ASM purposes) |
| * @param size size of the speech data |
| * @param lpcs LPCs used to synthesize this frame's speech data |
| */ |
| static void wiener_denoise(WMAVoiceContext *s, int fcb_type, |
| float *synth_pf, int size, |
| const float *lpcs) |
| { |
| int remainder, lim, n; |
| |
| if (fcb_type != FCB_TYPE_SILENCE) { |
| float *tilted_lpcs = s->tilted_lpcs_pf, |
| *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; |
| |
| tilted_lpcs[0] = 1.0; |
| memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); |
| memset(&tilted_lpcs[s->lsps + 1], 0, |
| sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); |
| ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), |
| tilted_lpcs, s->lsps + 2); |
| |
| /* The IRDFT output (127 samples for 7-bit filter) beyond the frame |
| * size is applied to the next frame. All input beyond this is zero, |
| * and thus all output beyond this will go towards zero, hence we can |
| * limit to min(size-1, 127-size) as a performance consideration. */ |
| remainder = FFMIN(127 - size, size - 1); |
| calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); |
| |
| /* apply coefficients (in frequency spectrum domain), i.e. complex |
| * number multiplication */ |
| memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); |
| s->rdft.rdft_calc(&s->rdft, synth_pf); |
| s->rdft.rdft_calc(&s->rdft, coeffs); |
| synth_pf[0] *= coeffs[0]; |
| synth_pf[1] *= coeffs[1]; |
| for (n = 1; n < 64; n++) { |
| float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; |
| synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; |
| synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; |
| } |
| s->irdft.rdft_calc(&s->irdft, synth_pf); |
| } |
| |
| /* merge filter output with the history of previous runs */ |
| if (s->denoise_filter_cache_size) { |
| lim = FFMIN(s->denoise_filter_cache_size, size); |
| for (n = 0; n < lim; n++) |
| synth_pf[n] += s->denoise_filter_cache[n]; |
| s->denoise_filter_cache_size -= lim; |
| memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], |
| sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); |
| } |
| |
| /* move remainder of filter output into a cache for future runs */ |
| if (fcb_type != FCB_TYPE_SILENCE) { |
| lim = FFMIN(remainder, s->denoise_filter_cache_size); |
| for (n = 0; n < lim; n++) |
| s->denoise_filter_cache[n] += synth_pf[size + n]; |
| if (lim < remainder) { |
| memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], |
| sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); |
| s->denoise_filter_cache_size = remainder; |
| } |
| } |
| } |
| |
| /** |
| * Averaging projection filter, the postfilter used in WMAVoice. |
| * |
| * This uses the following steps: |
| * - A zero-synthesis filter (generate excitation from synth signal) |
| * - Kalman smoothing on excitation, based on pitch |
| * - Re-synthesized smoothened output |
| * - Iterative Wiener denoise filter |
| * - Adaptive gain filter |
| * - DC filter |
| * |
| * @param s WMAVoice decoding context |
| * @param synth Speech synthesis output (before postfilter) |
| * @param samples Output buffer for filtered samples |
| * @param size Buffer size of synth & samples |
| * @param lpcs Generated LPCs used for speech synthesis |
| * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned) |
| * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) |
| * @param pitch Pitch of the input signal |
| */ |
| static void postfilter(WMAVoiceContext *s, const float *synth, |
| float *samples, int size, |
| const float *lpcs, float *zero_exc_pf, |
| int fcb_type, int pitch) |
| { |
| float synth_filter_in_buf[MAX_FRAMESIZE / 2], |
| *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], |
| *synth_filter_in = zero_exc_pf; |
| |
| av_assert0(size <= MAX_FRAMESIZE / 2); |
| |
| /* generate excitation from input signal */ |
| ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); |
| |
| if (fcb_type >= FCB_TYPE_AW_PULSES && |
| !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) |
| synth_filter_in = synth_filter_in_buf; |
| |
| /* re-synthesize speech after smoothening, and keep history */ |
| ff_celp_lp_synthesis_filterf(synth_pf, lpcs, |
| synth_filter_in, size, s->lsps); |
| memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], |
| sizeof(synth_pf[0]) * s->lsps); |
| |
| wiener_denoise(s, fcb_type, synth_pf, size, lpcs); |
| |
| adaptive_gain_control(samples, synth_pf, synth, size, 0.99, |
| &s->postfilter_agc); |
| |
| if (s->dc_level > 8) { |
| /* remove ultra-low frequency DC noise / highpass filter; |
| * coefficients are identical to those used in SIPR decoding, |
| * and very closely resemble those used in AMR-NB decoding. */ |
| ff_acelp_apply_order_2_transfer_function(samples, samples, |
| (const float[2]) { -1.99997, 1.0 }, |
| (const float[2]) { -1.9330735188, 0.93589198496 }, |
| 0.93980580475, s->dcf_mem, size); |
| } |
| } |
| /** |
| * @} |
| */ |
| |
| /** |
| * Dequantize LSPs |
| * @param lsps output pointer to the array that will hold the LSPs |
| * @param num number of LSPs to be dequantized |
| * @param values quantized values, contains n_stages values |
| * @param sizes range (i.e. max value) of each quantized value |
| * @param n_stages number of dequantization runs |
| * @param table dequantization table to be used |
| * @param mul_q LSF multiplier |
| * @param base_q base (lowest) LSF values |
| */ |
| static void dequant_lsps(double *lsps, int num, |
| const uint16_t *values, |
| const uint16_t *sizes, |
| int n_stages, const uint8_t *table, |
| const double *mul_q, |
| const double *base_q) |
| { |
| int n, m; |
| |
| memset(lsps, 0, num * sizeof(*lsps)); |
| for (n = 0; n < n_stages; n++) { |
| const uint8_t *t_off = &table[values[n] * num]; |
| double base = base_q[n], mul = mul_q[n]; |
| |
| for (m = 0; m < num; m++) |
| lsps[m] += base + mul * t_off[m]; |
| |
| table += sizes[n] * num; |
| } |
| } |
| |
| /** |
| * @name LSP dequantization routines |
| * LSP dequantization routines, for 10/16LSPs and independent/residual coding. |
| * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; |
| * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. |
| * @{ |
| */ |
| /** |
| * Parse 10 independently-coded LSPs. |
| */ |
| static void dequant_lsp10i(GetBitContext *gb, double *lsps) |
| { |
| static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; |
| static const double mul_lsf[4] = { |
| 5.2187144800e-3, 1.4626986422e-3, |
| 9.6179549166e-4, 1.1325736225e-3 |
| }; |
| static const double base_lsf[4] = { |
| M_PI * -2.15522e-1, M_PI * -6.1646e-2, |
| M_PI * -3.3486e-2, M_PI * -5.7408e-2 |
| }; |
| uint16_t v[4]; |
| |
| v[0] = get_bits(gb, 8); |
| v[1] = get_bits(gb, 6); |
| v[2] = get_bits(gb, 5); |
| v[3] = get_bits(gb, 5); |
| |
| dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, |
| mul_lsf, base_lsf); |
| } |
| |
| /** |
| * Parse 10 independently-coded LSPs, and then derive the tables to |
| * generate LSPs for the other frames from them (residual coding). |
| */ |
| static void dequant_lsp10r(GetBitContext *gb, |
| double *i_lsps, const double *old, |
| double *a1, double *a2, int q_mode) |
| { |
| static const uint16_t vec_sizes[3] = { 128, 64, 64 }; |
| static const double mul_lsf[3] = { |
| 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 |
| }; |
| static const double base_lsf[3] = { |
| M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 |
| }; |
| const float (*ipol_tab)[2][10] = q_mode ? |
| wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; |
| uint16_t interpol, v[3]; |
| int n; |
| |
| dequant_lsp10i(gb, i_lsps); |
| |
| interpol = get_bits(gb, 5); |
| v[0] = get_bits(gb, 7); |
| v[1] = get_bits(gb, 6); |
| v[2] = get_bits(gb, 6); |
| |
| for (n = 0; n < 10; n++) { |
| double delta = old[n] - i_lsps[n]; |
| a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; |
| a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; |
| } |
| |
| dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, |
| mul_lsf, base_lsf); |
| } |
| |
| /** |
| * Parse 16 independently-coded LSPs. |
| */ |
| static void dequant_lsp16i(GetBitContext *gb, double *lsps) |
| { |
| static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; |
| static const double mul_lsf[5] = { |
| 3.3439586280e-3, 6.9908173703e-4, |
| 3.3216608306e-3, 1.0334960326e-3, |
| 3.1899104283e-3 |
| }; |
| static const double base_lsf[5] = { |
| M_PI * -1.27576e-1, M_PI * -2.4292e-2, |
| M_PI * -1.28094e-1, M_PI * -3.2128e-2, |
| M_PI * -1.29816e-1 |
| }; |
| uint16_t v[5]; |
| |
| v[0] = get_bits(gb, 8); |
| v[1] = get_bits(gb, 6); |
| v[2] = get_bits(gb, 7); |
| v[3] = get_bits(gb, 6); |
| v[4] = get_bits(gb, 7); |
| |
| dequant_lsps( lsps, 5, v, vec_sizes, 2, |
| wmavoice_dq_lsp16i1, mul_lsf, base_lsf); |
| dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, |
| wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); |
| dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, |
| wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); |
| } |
| |
| /** |
| * Parse 16 independently-coded LSPs, and then derive the tables to |
| * generate LSPs for the other frames from them (residual coding). |
| */ |
| static void dequant_lsp16r(GetBitContext *gb, |
| double *i_lsps, const double *old, |
| double *a1, double *a2, int q_mode) |
| { |
| static const uint16_t vec_sizes[3] = { 128, 128, 128 }; |
| static const double mul_lsf[3] = { |
| 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 |
| }; |
| static const double base_lsf[3] = { |
| M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 |
| }; |
| const float (*ipol_tab)[2][16] = q_mode ? |
| wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; |
| uint16_t interpol, v[3]; |
| int n; |
| |
| dequant_lsp16i(gb, i_lsps); |
| |
| interpol = get_bits(gb, 5); |
| v[0] = get_bits(gb, 7); |
| v[1] = get_bits(gb, 7); |
| v[2] = get_bits(gb, 7); |
| |
| for (n = 0; n < 16; n++) { |
| double delta = old[n] - i_lsps[n]; |
| a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; |
| a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; |
| } |
| |
| dequant_lsps( a2, 10, v, vec_sizes, 1, |
| wmavoice_dq_lsp16r1, mul_lsf, base_lsf); |
| dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, |
| wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); |
| dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, |
| wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); |
| } |
| |
| /** |
| * @} |
| * @name Pitch-adaptive window coding functions |
| * The next few functions are for pitch-adaptive window coding. |
| * @{ |
| */ |
| /** |
| * Parse the offset of the first pitch-adaptive window pulses, and |
| * the distribution of pulses between the two blocks in this frame. |
| * @param s WMA Voice decoding context private data |
| * @param gb bit I/O context |
| * @param pitch pitch for each block in this frame |
| */ |
| static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, |
| const int *pitch) |
| { |
| static const int16_t start_offset[94] = { |
| -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, |
| 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, |
| 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, |
| 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, |
| 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, |
| 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, |
| 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, |
| 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 |
| }; |
| int bits, offset; |
| |
| /* position of pulse */ |
| s->aw_idx_is_ext = 0; |
| if ((bits = get_bits(gb, 6)) >= 54) { |
| s->aw_idx_is_ext = 1; |
| bits += (bits - 54) * 3 + get_bits(gb, 2); |
| } |
| |
| /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count |
| * the distribution of the pulses in each block contained in this frame. */ |
| s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; |
| for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; |
| s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; |
| s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; |
| offset += s->aw_n_pulses[0] * pitch[0]; |
| s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; |
| s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; |
| |
| /* if continuing from a position before the block, reset position to |
| * start of block (when corrected for the range over which it can be |
| * spread in aw_pulse_set1()). */ |
| if (start_offset[bits] < MAX_FRAMESIZE / 2) { |
| while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) |
| s->aw_first_pulse_off[1] -= pitch[1]; |
| if (start_offset[bits] < 0) |
| while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) |
| s->aw_first_pulse_off[0] -= pitch[0]; |
| } |
| } |
| |
| /** |
| * Apply second set of pitch-adaptive window pulses. |
| * @param s WMA Voice decoding context private data |
| * @param gb bit I/O context |
| * @param block_idx block index in frame [0, 1] |
| * @param fcb structure containing fixed codebook vector info |
| * @return -1 on error, 0 otherwise |
| */ |
| static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, |
| int block_idx, AMRFixed *fcb) |
| { |
| uint16_t use_mask_mem[9]; // only 5 are used, rest is padding |
| uint16_t *use_mask = use_mask_mem + 2; |
| /* in this function, idx is the index in the 80-bit (+ padding) use_mask |
| * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits |
| * of idx are the position of the bit within a particular item in the |
| * array (0 being the most significant bit, and 15 being the least |
| * significant bit), and the remainder (>> 4) is the index in the |
| * use_mask[]-array. This is faster and uses less memory than using a |
| * 80-byte/80-int array. */ |
| int pulse_off = s->aw_first_pulse_off[block_idx], |
| pulse_start, n, idx, range, aidx, start_off = 0; |
| |
| /* set offset of first pulse to within this block */ |
| if (s->aw_n_pulses[block_idx] > 0) |
| while (pulse_off + s->aw_pulse_range < 1) |
| pulse_off += fcb->pitch_lag; |
| |
| /* find range per pulse */ |
| if (s->aw_n_pulses[0] > 0) { |
| if (block_idx == 0) { |
| range = 32; |
| } else /* block_idx = 1 */ { |
| range = 8; |
| if (s->aw_n_pulses[block_idx] > 0) |
| pulse_off = s->aw_next_pulse_off_cache; |
| } |
| } else |
| range = 16; |
| pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; |
| |
| /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, |
| * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus |
| * we exclude that range from being pulsed again in this function. */ |
| memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0])); |
| memset( use_mask, -1, 5 * sizeof(use_mask[0])); |
| memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); |
| if (s->aw_n_pulses[block_idx] > 0) |
| for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { |
| int excl_range = s->aw_pulse_range; // always 16 or 24 |
| uint16_t *use_mask_ptr = &use_mask[idx >> 4]; |
| int first_sh = 16 - (idx & 15); |
| *use_mask_ptr++ &= 0xFFFFu << first_sh; |
| excl_range -= first_sh; |
| if (excl_range >= 16) { |
| *use_mask_ptr++ = 0; |
| *use_mask_ptr &= 0xFFFF >> (excl_range - 16); |
| } else |
| *use_mask_ptr &= 0xFFFF >> excl_range; |
| } |
| |
| /* find the 'aidx'th offset that is not excluded */ |
| aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); |
| for (n = 0; n <= aidx; pulse_start++) { |
| for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; |
| if (idx >= MAX_FRAMESIZE / 2) { // find from zero |
| if (use_mask[0]) idx = 0x0F; |
| else if (use_mask[1]) idx = 0x1F; |
| else if (use_mask[2]) idx = 0x2F; |
| else if (use_mask[3]) idx = 0x3F; |
| else if (use_mask[4]) idx = 0x4F; |
| else return -1; |
| idx -= av_log2_16bit(use_mask[idx >> 4]); |
| } |
| if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { |
| use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); |
| n++; |
| start_off = idx; |
| } |
| } |
| |
| fcb->x[fcb->n] = start_off; |
| fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; |
| fcb->n++; |
| |
| /* set offset for next block, relative to start of that block */ |
| n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; |
| s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; |
| return 0; |
| } |
| |
| /** |
| * Apply first set of pitch-adaptive window pulses. |
| * @param s WMA Voice decoding context private data |
| * @param gb bit I/O context |
| * @param block_idx block index in frame [0, 1] |
| * @param fcb storage location for fixed codebook pulse info |
| */ |
| static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, |
| int block_idx, AMRFixed *fcb) |
| { |
| int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); |
| float v; |
| |
| if (s->aw_n_pulses[block_idx] > 0) { |
| int n, v_mask, i_mask, sh, n_pulses; |
| |
| if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each |
| n_pulses = 3; |
| v_mask = 8; |
| i_mask = 7; |
| sh = 4; |
| } else { // 4 pulses, 1:sign + 2:index each |
| n_pulses = 4; |
| v_mask = 4; |
| i_mask = 3; |
| sh = 3; |
| } |
| |
| for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { |
| fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; |
| fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + |
| s->aw_first_pulse_off[block_idx]; |
| while (fcb->x[fcb->n] < 0) |
| fcb->x[fcb->n] += fcb->pitch_lag; |
| if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) |
| fcb->n++; |
| } |
| } else { |
| int num2 = (val & 0x1FF) >> 1, delta, idx; |
| |
| if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } |
| else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } |
| else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } |
| else { delta = 7; idx = num2 + 1 - 3 * 75; } |
| v = (val & 0x200) ? -1.0 : 1.0; |
| |
| fcb->no_repeat_mask |= 3 << fcb->n; |
| fcb->x[fcb->n] = idx - delta; |
| fcb->y[fcb->n] = v; |
| fcb->x[fcb->n + 1] = idx; |
| fcb->y[fcb->n + 1] = (val & 1) ? -v : v; |
| fcb->n += 2; |
| } |
| } |
| |
| /** |
| * @} |
| * |
| * Generate a random number from frame_cntr and block_idx, which will live |
| * in the range [0, 1000 - block_size] (so it can be used as an index in a |
| * table of size 1000 of which you want to read block_size entries). |
| * |
| * @param frame_cntr current frame number |
| * @param block_num current block index |
| * @param block_size amount of entries we want to read from a table |
| * that has 1000 entries |
| * @return a (non-)random number in the [0, 1000 - block_size] range. |
| */ |
| static int pRNG(int frame_cntr, int block_num, int block_size) |
| { |
| /* array to simplify the calculation of z: |
| * y = (x % 9) * 5 + 6; |
| * z = (49995 * x) / y; |
| * Since y only has 9 values, we can remove the division by using a |
| * LUT and using FASTDIV-style divisions. For each of the 9 values |
| * of y, we can rewrite z as: |
| * z = x * (49995 / y) + x * ((49995 % y) / y) |
| * In this table, each col represents one possible value of y, the |
| * first number is 49995 / y, and the second is the FASTDIV variant |
| * of 49995 % y / y. */ |
| static const unsigned int div_tbl[9][2] = { |
| { 8332, 3 * 715827883U }, // y = 6 |
| { 4545, 0 * 390451573U }, // y = 11 |
| { 3124, 11 * 268435456U }, // y = 16 |
| { 2380, 15 * 204522253U }, // y = 21 |
| { 1922, 23 * 165191050U }, // y = 26 |
| { 1612, 23 * 138547333U }, // y = 31 |
| { 1388, 27 * 119304648U }, // y = 36 |
| { 1219, 16 * 104755300U }, // y = 41 |
| { 1086, 39 * 93368855U } // y = 46 |
| }; |
| unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; |
| if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, |
| // so this is effectively a modulo (%) |
| y = x - 9 * MULH(477218589, x); // x % 9 |
| z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); |
| // z = x * 49995 / (y * 5 + 6) |
| return z % (1000 - block_size); |
| } |
| |
| /** |
| * Parse hardcoded signal for a single block. |
| * @note see #synth_block(). |
| */ |
| static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, |
| int block_idx, int size, |
| const struct frame_type_desc *frame_desc, |
| float *excitation) |
| { |
| float gain; |
| int n, r_idx; |
| |
| av_assert0(size <= MAX_FRAMESIZE); |
| |
| /* Set the offset from which we start reading wmavoice_std_codebook */ |
| if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { |
| r_idx = pRNG(s->frame_cntr, block_idx, size); |
| gain = s->silence_gain; |
| } else /* FCB_TYPE_HARDCODED */ { |
| r_idx = get_bits(gb, 8); |
| gain = wmavoice_gain_universal[get_bits(gb, 6)]; |
| } |
| |
| /* Clear gain prediction parameters */ |
| memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); |
| |
| /* Apply gain to hardcoded codebook and use that as excitation signal */ |
| for (n = 0; n < size; n++) |
| excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; |
| } |
| |
| /** |
| * Parse FCB/ACB signal for a single block. |
| * @note see #synth_block(). |
| */ |
| static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
| int block_idx, int size, |
| int block_pitch_sh2, |
| const struct frame_type_desc *frame_desc, |
| float *excitation) |
| { |
| static const float gain_coeff[6] = { |
| 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 |
| }; |
| float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; |
| int n, idx, gain_weight; |
| AMRFixed fcb; |
| |
| av_assert0(size <= MAX_FRAMESIZE / 2); |
| memset(pulses, 0, sizeof(*pulses) * size); |
| |
| fcb.pitch_lag = block_pitch_sh2 >> 2; |
| fcb.pitch_fac = 1.0; |
| fcb.no_repeat_mask = 0; |
| fcb.n = 0; |
| |
| /* For the other frame types, this is where we apply the innovation |
| * (fixed) codebook pulses of the speech signal. */ |
| if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
| aw_pulse_set1(s, gb, block_idx, &fcb); |
| if (aw_pulse_set2(s, gb, block_idx, &fcb)) { |
| /* Conceal the block with silence and return. |
| * Skip the correct amount of bits to read the next |
| * block from the correct offset. */ |
| int r_idx = pRNG(s->frame_cntr, block_idx, size); |
| |
| for (n = 0; n < size; n++) |
| excitation[n] = |
| wmavoice_std_codebook[r_idx + n] * s->silence_gain; |
| skip_bits(gb, 7 + 1); |
| return; |
| } |
| } else /* FCB_TYPE_EXC_PULSES */ { |
| int offset_nbits = 5 - frame_desc->log_n_blocks; |
| |
| fcb.no_repeat_mask = -1; |
| /* similar to ff_decode_10_pulses_35bits(), but with single pulses |
| * (instead of double) for a subset of pulses */ |
| for (n = 0; n < 5; n++) { |
| float sign; |
| int pos1, pos2; |
| |
| sign = get_bits1(gb) ? 1.0 : -1.0; |
| pos1 = get_bits(gb, offset_nbits); |
| fcb.x[fcb.n] = n + 5 * pos1; |
| fcb.y[fcb.n++] = sign; |
| if (n < frame_desc->dbl_pulses) { |
| pos2 = get_bits(gb, offset_nbits); |
| fcb.x[fcb.n] = n + 5 * pos2; |
| fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; |
| } |
| } |
| } |
| ff_set_fixed_vector(pulses, &fcb, 1.0, size); |
| |
| /* Calculate gain for adaptive & fixed codebook signal. |
| * see ff_amr_set_fixed_gain(). */ |
| idx = get_bits(gb, 7); |
| fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, |
| gain_coeff, 6) - |
| 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); |
| acb_gain = wmavoice_gain_codebook_acb[idx]; |
| pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], |
| -2.9957322736 /* log(0.05) */, |
| 1.6094379124 /* log(5.0) */); |
| |
| gain_weight = 8 >> frame_desc->log_n_blocks; |
| memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, |
| sizeof(*s->gain_pred_err) * (6 - gain_weight)); |
| for (n = 0; n < gain_weight; n++) |
| s->gain_pred_err[n] = pred_err; |
| |
| /* Calculation of adaptive codebook */ |
| if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { |
| int len; |
| for (n = 0; n < size; n += len) { |
| int next_idx_sh16; |
| int abs_idx = block_idx * size + n; |
| int pitch_sh16 = (s->last_pitch_val << 16) + |
| s->pitch_diff_sh16 * abs_idx; |
| int pitch = (pitch_sh16 + 0x6FFF) >> 16; |
| int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; |
| idx = idx_sh16 >> 16; |
| if (s->pitch_diff_sh16) { |
| if (s->pitch_diff_sh16 > 0) { |
| next_idx_sh16 = (idx_sh16) &~ 0xFFFF; |
| } else |
| next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; |
| len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, |
| 1, size - n); |
| } else |
| len = size; |
| |
| ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], |
| wmavoice_ipol1_coeffs, 17, |
| idx, 9, len); |
| } |
| } else /* ACB_TYPE_HAMMING */ { |
| int block_pitch = block_pitch_sh2 >> 2; |
| idx = block_pitch_sh2 & 3; |
| if (idx) { |
| ff_acelp_interpolatef(excitation, &excitation[-block_pitch], |
| wmavoice_ipol2_coeffs, 4, |
| idx, 8, size); |
| } else |
| av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, |
| sizeof(float) * size); |
| } |
| |
| /* Interpolate ACB/FCB and use as excitation signal */ |
| ff_weighted_vector_sumf(excitation, excitation, pulses, |
| acb_gain, fcb_gain, size); |
| } |
| |
| /** |
| * Parse data in a single block. |
| * |
| * @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. |
| * |
| * @param ctx WMA Voice decoder context |
| * @param gb bit I/O context (s->gb or one for cross-packet superframes) |
| * @param frame_idx Frame number within superframe [0-2] |
| * @param samples pointer to output sample buffer, has space for at least 160 |
| * samples |
| * @param lsps LSP array |
| * @param prev_lsps array of previous frame's LSPs |
| * @param excitation target buffer for excitation signal |
| * @param synth target buffer for synthesized speech data |
| * @return 0 on success, <0 on error. |
| */ |
| static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
| float *samples, |
| const double *lsps, const double *prev_lsps, |
| float *excitation, float *synth) |
| { |
| WMAVoiceContext *s = ctx->priv_data; |
| int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val); |
| int pitch[MAX_BLOCKS], av_uninit(last_block_pitch); |
| |
| /* Parse frame type ("frame header"), see frame_descs */ |
| int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples; |
| |
| if (bd_idx < 0) { |
| av_log(ctx, AV_LOG_ERROR, |
| "Invalid frame type VLC code, skipping\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; |
| |
| /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ |
| if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { |
| /* Pitch is provided per frame, which is interpreted as the pitch of |
| * the last sample of the last block of this frame. We can interpolate |
| * the pitch of other blocks (and even pitch-per-sample) by gradually |
| * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ |
| n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; |
| log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; |
| cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); |
| cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); |
| if (s->last_acb_type == ACB_TYPE_NONE || |
| 20 * abs(cur_pitch_val - s->last_pitch_val) > |
| (cur_pitch_val + s->last_pitch_val)) |
| s->last_pitch_val = cur_pitch_val; |
| |
| /* pitch per block */ |
| for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { |
| int fac = n * 2 + 1; |
| |
| pitch[n] = (MUL16(fac, cur_pitch_val) + |
| MUL16((n_blocks_x2 - fac), s->last_pitch_val) + |
| frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; |
| } |
| |
| /* "pitch-diff-per-sample" for calculation of pitch per sample */ |
| s->pitch_diff_sh16 = |
| (cur_pitch_val - s->last_pitch_val) * (1 << 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-logarithmic 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; |
| } |
| } |
| } |
| |
| /** |
| * 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 = MAX_SFRAMESIZE; |
| 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; |
| } |
| |
| /* 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)) > MAX_SFRAMESIZE) { |
| av_log(ctx, AV_LOG_ERROR, |
| "Superframe encodes > %d samples (%d), not allowed\n", |
| MAX_SFRAMESIZE, n_samples); |
| return AVERROR_INVALIDDATA; |
| } |
| } |
| |
| /* Parse LSPs, if global for the superframe (can also be per-frame). */ |
| if (s->has_residual_lsps) { |
| double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; |
| |
| for (n = 0; n < s->lsps; n++) |
| prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; |
| |
| if (s->lsps == 10) { |
| dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
| } else /* s->lsps == 16 */ |
| dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
| |
| for (n = 0; n < s->lsps; n++) { |
| lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); |
| lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); |
| lsps[2][n] += mean_lsf[n]; |
| } |
| for (n = 0; n < 3; n++) |
| stabilize_lsps(lsps[n], s->lsps); |
| } |
| |
| /* synth_superframe can run multiple times per packet |
| * free potential previous frame */ |
| av_frame_unref(frame); |
| |
| /* get output buffer */ |
| frame->nb_samples = MAX_SFRAMESIZE; |
| 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)); |
| } |
| |
| if (get_bits_left(gb) < 0) { |
| wmavoice_flush(ctx); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| *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 <0 on error, nb_superframes on success. |
| */ |
| static int parse_packet_header(WMAVoiceContext *s) |
| { |
| GetBitContext *gb = &s->gb; |
| unsigned int res, n_superframes = 0; |
| |
| skip_bits(gb, 4); // packet sequence number |
| s->has_residual_lsps = get_bits1(gb); |
| do { |
| if (get_bits_left(gb) < 6 + s->spillover_bitsize) |
| return AVERROR_INVALIDDATA; |
| |
| res = get_bits(gb, 6); // number of superframes per packet |
| // (minus first one if there is spillover) |
| n_superframes += res; |
| } while (res == 0x3F); |
| s->spillover_nbits = get_bits(gb, s->spillover_bitsize); |
| |
| return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA; |
| } |
| |
| /** |
| * 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; |
| 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)); |
| 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 *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); |
| init_get_bits8(&s->gb, avpkt->data, size); |
| |
| /* 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 (!size) { |
| s->spillover_nbits = 0; |
| s->nb_superframes = 0; |
| } else { |
| if ((res = parse_packet_header(s)) < 0) |
| return res; |
| s->nb_superframes = 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->sframe_cache_size > 0) { |
| int cnt = get_bits_count(gb); |
| if (cnt + s->spillover_nbits > avpkt->size * 8) { |
| s->spillover_nbits = avpkt->size * 8 - cnt; |
| } |
| 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; |
| res = cnt >> 3; |
| return res; |
| } else |
| skip_bits_long (gb, s->spillover_nbits - cnt + |
| get_bits_count(gb)); // resync |
| } else if (s->spillover_nbits) { |
| 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 (s->nb_superframes-- == 0) { |
| *got_frame_ptr = 0; |
| return size; |
| } else if (s->nb_superframes > 0) { |
| 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; |
| res = cnt >> 3; |
| return res; |
| } |
| } else if ((s->sframe_cache_size = pos) > 0) { |
| /* ... 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; |
| } |
| |
| AVCodec ff_wmavoice_decoder = { |
| .name = "wmavoice", |
| .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_WMAVOICE, |
| .priv_data_size = sizeof(WMAVoiceContext), |
| .init = wmavoice_decode_init, |
| .close = wmavoice_decode_end, |
| .decode = wmavoice_decode_packet, |
| .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, |
| .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, |
| .flush = wmavoice_flush, |
| }; |