virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-12 19:18:44 +02:00
Code Issues Releases Activity
31c714bf0b
FFmpeg/libavcodec/wmavoice.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +02:00

2014 lines
79 KiB
C
Raw Blame History

/*
* 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_internal.h"
#include "libavutil/thread.h"
#include "avcodec.h"
#include "codec_internal.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, ret;
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) {
if ((ret = ff_rdft_init(&s->rdft, 7, DFT_R2C)) < 0 ||
(ret = ff_rdft_init(&s->irdft, 7, IDFT_C2R)) < 0 ||
(ret = ff_dct_init (&s->dct, 6, DCT_I)) < 0 ||
(ret = ff_dct_init (&s->dst, 6, DST_I)) < 0)
return ret;
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);
av_channel_layout_uninit(&ctx->ch_layout);
ctx->ch_layout = (AVChannelLayout)AV_CHANNEL_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 > put_bits_left(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, AVFrame *frame,
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, frame, 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, frame, 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;
}
const FFCodec ff_wmavoice_decoder = {
.p.name = "wmavoice",
.p.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_WMAVOICE,
.priv_data_size = sizeof(WMAVoiceContext),
.init = wmavoice_decode_init,
.close = wmavoice_decode_end,
FF_CODEC_DECODE_CB(wmavoice_decode_packet),
.p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
.flush = wmavoice_flush,
};
Reference in New Issue View Git Blame Copy Permalink