libavcodec/wmavoice.c File Reference

Windows Media Audio Voice compatible decoder. More...

#include <math.h>
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#include "wmavoice_data.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_vectors.h"
#include "acelp_filters.h"
#include "lsp.h"
#include "libavutil/lzo.h"
#include "dct.h"
#include "rdft.h"
#include "sinewin.h"

Go to the source code of this file.

Data Structures

struct  frame_type_desc
 Description of frame types. More...
struct  WMAVoiceContext
 WMA Voice decoding context. More...

Defines

#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
#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
#define VLC_NBITS   6
 number of bits to read per VLC iteration
#define log_range(var, assign)

Enumerations

enum  { ACB_TYPE_NONE = 0, ACB_TYPE_ASYMMETRIC = 1, ACB_TYPE_HAMMING = 2 }
 Adaptive codebook types. More...
enum  { FCB_TYPE_SILENCE = 0, FCB_TYPE_HARDCODED = 1, FCB_TYPE_AW_PULSES = 2, FCB_TYPE_EXC_PULSES = 3 }
 Fixed codebook types. More...

Functions

static av_cold int decode_vbmtree (GetBitContext *gb, int8_t vbm_tree[25])
 Set up the variable bit mode (VBM) tree from container extradata.
static av_cold int wmavoice_decode_init (AVCodecContext *ctx)
 Set up decoder with parameters from demuxer (extradata etc.
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)
 Dequantize LSPs.
static int pRNG (int frame_cntr, int block_num, int block_size)
 Generate a random number from frame_cntr and block_idx, which will lief in the range [0, 1000 - block_size] (so it can be used as an index in a table of size 1000 of which you want to read block_size entries).
static void synth_block_hardcoded (WMAVoiceContext *s, GetBitContext *gb, int block_idx, int size, const struct frame_type_desc *frame_desc, float *excitation)
 Parse hardcoded signal for a single 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)
 Parse FCB/ACB signal for a single block.
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)
 Parse data in a single block.
static int synth_frame (AVCodecContext *ctx, GetBitContext *gb, int frame_idx, float *samples, const double *lsps, const double *prev_lsps, float *excitation, float *synth)
 Synthesize output samples for a single frame.
static void stabilize_lsps (double *lsps, int num)
 Ensure minimum value for first item, maximum value for last value, proper spacing between each value and proper ordering.
static int check_bits_for_superframe (GetBitContext *orig_gb, WMAVoiceContext *s)
 Test if there's enough bits to read 1 superframe.
static int synth_superframe (AVCodecContext *ctx, float *samples, int *data_size)
 Synthesize output samples for a single superframe.
static int parse_packet_header (WMAVoiceContext *s)
 Parse the packet header at the start of each packet (input data to this decoder).
static void copy_bits (PutBitContext *pb, const uint8_t *data, int size, GetBitContext *gb, int nbits)
 Copy (unaligned) bits from gb/data/size to pb.
static int wmavoice_decode_packet (AVCodecContext *ctx, void *data, int *data_size, AVPacket *avpkt)
 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).
static av_cold int wmavoice_decode_end (AVCodecContext *ctx)
static av_cold void wmavoice_flush (AVCodecContext *ctx)
Postfilter functions
Postfilter functions (gain control, wiener denoise filter, DC filter, kalman smoothening, plus surrounding code to wrap it)

static void adaptive_gain_control (float *out, const float *in, const float *speech_synth, int size, float alpha, float *gain_mem)
 Adaptive gain control (as used in postfilter).
static int kalman_smoothen (WMAVoiceContext *s, int pitch, const float *in, float *out, int size)
 Kalman smoothing function.
static float tilt_factor (const float *lpcs, int n_lpcs)
 Get the tilt factor of a formant filter from its transfer function.
static void calc_input_response (WMAVoiceContext *s, float *lpcs, int fcb_type, float *coeffs, int remainder)
 Derive denoise filter coefficients (in real domain) from the LPCs.
static void wiener_denoise (WMAVoiceContext *s, int fcb_type, float *synth_pf, int size, const float *lpcs)
 This function applies a Wiener filter on the (noisy) speech signal as a means to denoise it.
static void postfilter (WMAVoiceContext *s, const float *synth, float *samples, int size, const float *lpcs, float *zero_exc_pf, int fcb_type, int pitch)
 Averaging projection filter, the postfilter used in WMAVoice.
LSP dequantization routines
LSP dequantization routines, for 10/16LSPs and independent/residual coding.

Note:
we assume enough bits are available, caller should check. lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.


static void dequant_lsp10i (GetBitContext *gb, double *lsps)
 Parse 10 independently-coded LSPs.
static void dequant_lsp10r (GetBitContext *gb, double *i_lsps, const double *old, double *a1, double *a2, int q_mode)
 Parse 10 independently-coded LSPs, and then derive the tables to generate LSPs for the other frames from them (residual coding).
static void dequant_lsp16i (GetBitContext *gb, double *lsps)
 Parse 16 independently-coded LSPs.
static void dequant_lsp16r (GetBitContext *gb, double *i_lsps, const double *old, double *a1, double *a2, int q_mode)
 Parse 16 independently-coded LSPs, and then derive the tables to generate LSPs for the other frames from them (residual coding).
Pitch-adaptive window coding functions
The next few functions are for pitch-adaptive window coding.

static void aw_parse_coords (WMAVoiceContext *s, GetBitContext *gb, const int *pitch)
 Parse the offset of the first pitch-adaptive window pulses, and the distribution of pulses between the two blocks in this frame.
static void aw_pulse_set2 (WMAVoiceContext *s, GetBitContext *gb, int block_idx, AMRFixed *fcb)
 Apply second set of pitch-adaptive window pulses.
static void aw_pulse_set1 (WMAVoiceContext *s, GetBitContext *gb, int block_idx, AMRFixed *fcb)
 Apply first set of pitch-adaptive window pulses.

Variables

static VLC frame_type_vlc
 Frame type VLC coding.
static struct frame_type_desc frame_descs [17]
 Description of frame types.
AVCodec ff_wmavoice_decoder


Detailed Description

Windows Media Audio Voice compatible decoder.

Author:
Ronald S. Bultje <rsbultje@gmail.com>

Definition in file wmavoice.c.


Define Documentation

#define log_range ( var,
assign   ) 

Value:

do { \
        float tmp = log10f(assign);  var = tmp; \
        max       = FFMAX(max, tmp); min = FFMIN(min, tmp); \
    } while (0)

Referenced by calc_input_response().

#define MAX_BLOCKS   8

maximum number of blocks per frame

Definition at line 43 of file wmavoice.c.

Referenced by synth_frame().

#define MAX_FRAMES   3

maximum number of frames per superframe

Definition at line 47 of file wmavoice.c.

Referenced by check_bits_for_superframe(), and synth_superframe().

#define MAX_FRAMESIZE   160

maximum number of samples per frame

Definition at line 48 of file wmavoice.c.

#define MAX_LSPS   16

maximum filter order

Definition at line 44 of file wmavoice.c.

Referenced by synth_block(), synth_frame(), synth_superframe(), and wmavoice_flush().

#define MAX_LSPS_ALIGN16   16

same as MAX_LSPS; needs to be multiple

of 16 for ASM input buffer alignment

Definition at line 45 of file wmavoice.c.

Referenced by postfilter(), and wmavoice_flush().

#define MAX_SFRAMESIZE   (MAX_FRAMESIZE * MAX_FRAMES)

maximum number of samples per superframe

Definition at line 50 of file wmavoice.c.

Referenced by synth_superframe().

#define MAX_SIGNAL_HISTORY   416

maximum excitation signal history

Definition at line 49 of file wmavoice.c.

Referenced by synth_superframe(), wmavoice_decode_init(), and wmavoice_flush().

#define SFRAME_CACHE_MAXSIZE   256

maximum cache size for frame data that

was split over two packets

Definition at line 52 of file wmavoice.c.

Referenced by wmavoice_decode_packet().

#define VLC_NBITS   6

number of bits to read per VLC iteration

Definition at line 54 of file wmavoice.c.

Referenced by decode_vbmtree().


Enumeration Type Documentation

anonymous enum

Adaptive codebook types.

Enumerator:
ACB_TYPE_NONE  no adaptive codebook (only hardcoded fixed)
ACB_TYPE_ASYMMETRIC  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  Per-block pitch with signal generation using a Hamming sinc window function.

Note:
see wmavoice_ipol2_coeffs

Definition at line 64 of file wmavoice.c.

anonymous enum

Fixed codebook types.

Enumerator:
FCB_TYPE_SILENCE  comfort noise during silence generated from a hardcoded (fixed) codebook with per-frame (low) gain values
FCB_TYPE_HARDCODED  hardcoded (fixed) codebook with per-block gain values
FCB_TYPE_AW_PULSES  Pitch-adaptive window (AW) pulse signals, used in particular for low-bitrate streams.
FCB_TYPE_EXC_PULSES  Innovation (fixed) codebook pulse sets in combinations of either single pulses or pulse pairs.

Definition at line 79 of file wmavoice.c.


Function Documentation

static void adaptive_gain_control ( float *  out,
const float *  in,
const float *  speech_synth,
int  size,
float  alpha,
float *  gain_mem 
) [static]

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(...)).

Parameters:
out output buffer for filtered samples
in input buffer containing the samples as they are after the postfilter steps so far
speech_synth input buffer containing speech synth before postfilter
size input buffer size
alpha exponential filter factor
gain_mem pointer to filter memory (single float)

Definition at line 465 of file wmavoice.c.

Referenced by postfilter().

static void aw_parse_coords ( WMAVoiceContext s,
GetBitContext gb,
const int *  pitch 
) [static]

Parse the offset of the first pitch-adaptive window pulses, and the distribution of pulses between the two blocks in this frame.

Parameters:
s WMA Voice decoding context private data
gb bit I/O context
pitch pitch for each block in this frame

Definition at line 993 of file wmavoice.c.

Referenced by synth_frame().

static void aw_pulse_set1 ( WMAVoiceContext s,
GetBitContext gb,
int  block_idx,
AMRFixed fcb 
) [static]

Apply first set of pitch-adaptive window pulses.

Parameters:
s WMA Voice decoding context private data
gb bit I/O context
block_idx block index in frame [0, 1]
fcb storage location for fixed codebook pulse info

Definition at line 1133 of file wmavoice.c.

Referenced by synth_block_fcb_acb().

static void aw_pulse_set2 ( WMAVoiceContext s,
GetBitContext gb,
int  block_idx,
AMRFixed fcb 
) [static]

Apply second set of pitch-adaptive window pulses.

Parameters:
s WMA Voice decoding context private data
gb bit I/O context
block_idx block index in frame [0, 1]
fcb structure containing fixed codebook vector info

Definition at line 1044 of file wmavoice.c.

Referenced by synth_block_fcb_acb().

static void calc_input_response ( WMAVoiceContext s,
float *  lpcs,
int  fcb_type,
float *  coeffs,
int  remainder 
) [static]

Derive denoise filter coefficients (in real domain) from the LPCs.

Definition at line 564 of file wmavoice.c.

Referenced by wiener_denoise().

static int check_bits_for_superframe ( GetBitContext orig_gb,
WMAVoiceContext s 
) [static]

Test if there's enough bits to read 1 superframe.

Parameters:
orig_gb bit I/O context used for reading. This function does not modify the state of the bitreader; it only uses it to copy the current stream position
s WMA Voice decoding context private data
Returns:
-1 if unsupported, 1 on not enough bits or 0 if OK.

Definition at line 1640 of file wmavoice.c.

Referenced by synth_superframe().

static void copy_bits ( PutBitContext pb,
const uint8_t *  data,
int  size,
GetBitContext gb,
int  nbits 
) [static]

Copy (unaligned) bits from gb/data/size to pb.

Parameters:
pb target buffer to copy bits into
data source buffer to copy bits from
size size of the source data, in bytes
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
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.

Definition at line 1882 of file wmavoice.c.

static av_cold int decode_vbmtree ( GetBitContext gb,
int8_t  vbm_tree[25] 
) [static]

Set up the variable bit mode (VBM) tree from container extradata.

Parameters:
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).
vbm_tree pointer to array to which the decoded VBM tree will be written.
Returns:
0 on success, <0 on error.

Definition at line 301 of file wmavoice.c.

Referenced by wmavoice_decode_init().

static void dequant_lsp10i ( GetBitContext gb,
double *  lsps 
) [static]

Parse 10 independently-coded LSPs.

Definition at line 848 of file wmavoice.c.

Referenced by dequant_lsp10r(), and synth_superframe().

static void dequant_lsp10r ( GetBitContext gb,
double *  i_lsps,
const double *  old,
double *  a1,
double *  a2,
int  q_mode 
) [static]

Parse 10 independently-coded LSPs, and then derive the tables to generate LSPs for the other frames from them (residual coding).

Definition at line 874 of file wmavoice.c.

Referenced by synth_superframe().

static void dequant_lsp16i ( GetBitContext gb,
double *  lsps 
) [static]

Parse 16 independently-coded LSPs.

Definition at line 910 of file wmavoice.c.

Referenced by dequant_lsp16r(), and synth_superframe().

static void dequant_lsp16r ( GetBitContext gb,
double *  i_lsps,
const double *  old,
double *  a1,
double *  a2,
int  q_mode 
) [static]

Parse 16 independently-coded LSPs, and then derive the tables to generate LSPs for the other frames from them (residual coding).

Definition at line 943 of file wmavoice.c.

Referenced by synth_superframe().

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 
) [static]

Dequantize LSPs.

Parameters:
lsps output pointer to the array that will hold the LSPs
num number of LSPs to be dequantized
values quantized values, contains n_stages values
sizes range (i.e. max value) of each quantized value
n_stages number of dequantization runs
table dequantization table to be used
mul_q LSF multiplier
base_q base (lowest) LSF values

Definition at line 816 of file wmavoice.c.

Referenced by dequant_lsp10i(), dequant_lsp10r(), dequant_lsp16i(), and dequant_lsp16r().

static int kalman_smoothen ( WMAVoiceContext s,
int  pitch,
const float *  in,
float *  out,
int  size 
) [static]

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.

Parameters:
s WMA Voice decoding context
pitch pitch of the speech signal
in input speech signal
out output pointer for smoothened signal
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.

Definition at line 505 of file wmavoice.c.

Referenced by postfilter().

static int parse_packet_header ( WMAVoiceContext s  )  [static]

Parse the packet header at the start of each packet (input data to this decoder).

Parameters:
s WMA Voice decoding context private data
Returns:
1 if not enough bits were available, or 0 on success.

Definition at line 1847 of file wmavoice.c.

Referenced by gxf_header(), gxf_packet(), gxf_resync_media(), and wmavoice_decode_packet().

static void postfilter ( WMAVoiceContext s,
const float *  synth,
float *  samples,
int  size,
const float *  lpcs,
float *  zero_exc_pf,
int  fcb_type,
int  pitch 
) [static]

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

Parameters:
s WMAVoice decoding context
synth Speech synthesis output (before postfilter)
samples Output buffer for filtered samples
size Buffer size of synth & samples
lpcs Generated LPCs used for speech synthesis
zero_exc_pf destination for zero synthesis filter (16-byte aligned)
fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
pitch Pitch of the input signal

Definition at line 762 of file wmavoice.c.

static int pRNG ( int  frame_cntr,
int  block_num,
int  block_size 
) [static]

Generate a random number from frame_cntr and block_idx, which will lief in the range [0, 1000 - block_size] (so it can be used as an index in a table of size 1000 of which you want to read block_size entries).

Parameters:
frame_cntr current frame number
block_num current block index
block_size amount of entries we want to read from a table that has 1000 entries
Returns:
a (non-)random number in the [0, 1000 - block_size] range.

Definition at line 1194 of file wmavoice.c.

Referenced by synth_block_hardcoded().

static void stabilize_lsps ( double *  lsps,
int  num 
) [static]

Ensure minimum value for first item, maximum value for last value, proper spacing between each value and proper ordering.

Parameters:
lsps array of LSPs
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.

Definition at line 1602 of file wmavoice.c.

Referenced by synth_superframe().

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 
) [static]

Parse data in a single block.

Note:
we assume enough bits are available, caller should check.
Parameters:
s WMA Voice decoding context private data
gb bit I/O context
block_idx index of the to-be-read block
size amount of samples to be read in this block
block_pitch_sh2 pitch for this block << 2
lsps LSPs for (the end of) this frame
prev_lsps LSPs for the last frame
frame_desc frame type descriptor
excitation target memory for the ACB+FCB interpolated signal
synth target memory for the speech synthesis filter output
Returns:
0 on success, <0 on error.

Definition at line 1384 of file wmavoice.c.

Referenced by synth_frame().

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]

Parse FCB/ACB signal for a single block.

Note:
see synth_block().

Definition at line 1261 of file wmavoice.c.

Referenced by 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 
) [static]

Parse hardcoded signal for a single block.

Note:
see synth_block().

Definition at line 1230 of file wmavoice.c.

Referenced by synth_block().

static int synth_frame ( AVCodecContext ctx,
GetBitContext gb,
int  frame_idx,
float *  samples,
const double *  lsps,
const double *  prev_lsps,
float *  excitation,
float *  synth 
) [static]

Synthesize output samples for a single frame.

Note:
we assume enough bits are available, caller should check.
Parameters:
ctx WMA Voice decoder context
gb bit I/O context (s->gb or one for cross-packet superframes)
frame_idx Frame number within superframe [0-2]
samples pointer to output sample buffer, has space for at least 160 samples
lsps LSP array
prev_lsps array of previous frame's LSPs
excitation target buffer for excitation signal
synth target buffer for synthesized speech data
Returns:
0 on success, <0 on error.

Definition at line 1427 of file wmavoice.c.

Referenced by synth_superframe().

static int synth_superframe ( AVCodecContext ctx,
float *  samples,
int *  data_size 
) [static]

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.

Parameters:
ctx WMA Voice decoder context
samples pointer to output buffer for voice samples
data_size pointer containing the size of samples on input, and the amount of samples filled on output
Returns:
0 on success, <0 on error or 1 if there was not enough data to fully parse the superframe

Definition at line 1728 of file wmavoice.c.

Referenced by wmavoice_decode_packet().

static float tilt_factor ( const float *  lpcs,
int  n_lpcs 
) [static]

Get the tilt factor of a formant filter from its transfer function.

See also:
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
Parameters:
lpcs LPC coefficients
n_lpcs Size of LPC buffer
Returns:
the tilt factor

Definition at line 551 of file wmavoice.c.

static void wiener_denoise ( WMAVoiceContext s,
int  fcb_type,
float *  synth_pf,
int  size,
const float *  lpcs 
) [static]

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).

Parameters:
s WMA Voice decoding context
fcb_type Frame (codebook) type
synth_pf input: the noisy speech signal, output: denoised speech data; should be 16-byte aligned (for ASM purposes)
size size of the speech data
lpcs LPCs used to synthesize this frame's speech data

Definition at line 680 of file wmavoice.c.

Referenced by postfilter().

static av_cold int wmavoice_decode_end ( AVCodecContext ctx  )  [static]

Definition at line 1992 of file wmavoice.c.

static av_cold int wmavoice_decode_init ( AVCodecContext ctx  )  [static]

Set up decoder with parameters from demuxer (extradata etc.

).

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).

Definition at line 336 of file wmavoice.c.

static int wmavoice_decode_packet ( AVCodecContext ctx,
void *  data,
int *  data_size,
AVPacket avpkt 
) [static]

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().

Definition at line 1911 of file wmavoice.c.

static av_cold void wmavoice_flush ( AVCodecContext ctx  )  [static]

Definition at line 2006 of file wmavoice.c.


Variable Documentation

Initial value:

Definition at line 2034 of file wmavoice.c.

struct frame_type_desc frame_descs[17] [static]

Description of frame types.

Referenced by check_bits_for_superframe(), and synth_frame().

VLC frame_type_vlc [static]

Frame type VLC coding.

Definition at line 59 of file wmavoice.c.


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