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rockbox/lib/rbcodec/codecs/libwmapro/wmaprodec.c
Michael Giacomelli 6ffbee954b wmapro: fix the level and accuracy of 16 bit streams 
A WMA Professional stream of 16 bits per sample played about 48 dB
too quiet and with a noise floor near -70 dBFS.

ffmpeg scales the transform's output by the stream's sample size.
That was dropped when the decoder was converted to fixed point
(d884af2b99, 16284ae8ae), and the output is passed to the DSP as if
every stream had 24 bits. A stream of fewer bits has a lower
quantization step to match, so it came out low by the difference,
and it used the bottom of the integer quantization table, where the
factors have only a few significant bits.

Decode a 16 or 20 bit stream at the level of a 24 bit stream: use
that stream's quantization step, and scale each band's factor by
the ratio that is left. 24 bit streams are not affected.

Checked with perfsim (Sansa e200v1 build) against ffmpeg's decode
of a 16 bit, 192 kbps stereo file, the only such file to hand:

            level      SNR vs ffmpeg   noise
  before    -48 dB     37 dB           -70 dBFS
  after     correct    85 dB           -117 dBFS

A 24 bit file's output is byte-identical before and after. The
20 bit case follows the same rule but is untested.

Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
2026-10-05 20:13:48 -04:00

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/*
* Wmapro compatible decoder
* Copyright (c) 2007 Baptiste Coudurier, Benjamin Larsson, Ulion
* Copyright (c) 2008 - 2009 Sascha Sommer, Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file libavcodec/wmaprodec.c
* @brief wmapro decoder implementation
* Wmapro is an MDCT based codec comparable to wma standard or AAC.
* The decoding therefore consists of the following steps:
* - bitstream decoding
* - reconstruction of per-channel data
* - rescaling and inverse quantization
* - IMDCT
* - windowing and overlapp-add
*
* The compressed wmapro bitstream is split into individual packets.
* Every such packet contains one or more wma frames.
* The compressed frames may have a variable length and frames may
* cross packet boundaries.
* Common to all wmapro frames is the number of samples that are stored in
* a frame.
* The number of samples and a few other decode flags are stored
* as extradata that has to be passed to the decoder.
*
* The wmapro frames themselves are again split into a variable number of
* subframes. Every subframe contains the data for 2^N time domain samples
* where N varies between 7 and 12.
*
* Example wmapro bitstream (in samples):
*
* || packet 0 || packet 1 || packet 2 packets
* ---------------------------------------------------
* || frame 0 || frame 1 || frame 2 || frames
* ---------------------------------------------------
* || | | || | | | || || subframes of channel 0
* ---------------------------------------------------
* || | | || | | | || || subframes of channel 1
* ---------------------------------------------------
*
* The frame layouts for the individual channels of a wma frame does not need
* to be the same.
*
* However, if the offsets and lengths of several subframes of a frame are the
* same, the subframes of the channels can be grouped.
* Every group may then use special coding techniques like M/S stereo coding
* to improve the compression ratio. These channel transformations do not
* need to be applied to a whole subframe. Instead, they can also work on
* individual scale factor bands (see below).
* The coefficients that carry the audio signal in the frequency domain
* are transmitted as huffman-coded vectors with 4, 2 and 1 elements.
* In addition to that, the encoder can switch to a runlevel coding scheme
* by transmitting subframe_length / 128 zero coefficients.
*
* Before the audio signal can be converted to the time domain, the
* coefficients have to be rescaled and inverse quantized.
* A subframe is therefore split into several scale factor bands that get
* scaled individually.
* Scale factors are submitted for every frame but they might be shared
* between the subframes of a channel. Scale factors are initially DPCM-coded.
* Once scale factors are shared, the differences are transmitted as runlevel
* codes.
* Every subframe length and offset combination in the frame layout shares a
* common quantization factor that can be adjusted for every channel by a
* modifier.
* After the inverse quantization, the coefficients get processed by an IMDCT.
* The resulting values are then windowed with a sine window and the first half
* of the values are added to the second half of the output from the previous
* subframe in order to reconstruct the output samples.
*/
#include "ffmpeg_get_bits.h"
#include "ffmpeg_put_bits.h"
#include "wmaprodata.h"
#include "wma.h"
#include "wmaprodec.h"
//#include "wmapro_mdct.h"
#include "mdct_tables.h"
#include "quant.h"
#include "wmapro_math.h"
#include "codecs.h"
#include "codeclib.h"
#include "../libasf/asf.h"
/* Uncomment the following line to enable some debug output */
//#define WMAPRO_DUMP_CTX_EN
#undef DEBUGF
#ifdef WMAPRO_DUMP_CTX_EN
# define DEBUGF printf
#else
# define DEBUGF(...)
#endif
/* Some defines to make it compile */
#define AVERROR_INVALIDDATA -1
#define AVERROR_PATCHWELCOME -2
#define av_log_ask_for_sample(...)
/* Taken from avcodec.h */
#define FF_INPUT_BUFFER_PADDING_SIZE 8
/* Taken from libavutil/mem.h */
#define DECLARE_ALIGNED(n,t,v) t __attribute__ ((aligned (n))) v
/* Taken from libavutil/common.h */
#define FFMIN(a,b) ((a) > (b) ? (b) : (a))
#define FFMAX(a,b) ((a) > (b) ? (a) : (b))
/* Define some multiple used constants */
#define SQRT2_FRACT16 0x00016A0A /* 0x00016A0A = (sqrt(2)*(1<<16)) */
#define COS_PI4_FRACT16 0x0000B505 /* 0x0000B505 = (cos(pi/4)<<16) */
#define ONE_FRACT16 0x00010000 /* 0x00010000 = (1<<16) */
/* Enable multichannel for large-memory targets only */
#if (MEMORYSIZE > 2)
#define WMAPRO_MAX_CHANNELS 8 ///< max number of handled channels
#else
#define WMAPRO_MAX_CHANNELS 2 ///< max number of handled channels
#endif
/* Current decoder limitations */
#define MAX_SUBFRAMES 32 ///< max number of subframes per channel
#define MAX_BANDS 29 ///< max number of scale factor bands
#define MAX_FRAMESIZE 32768 ///< maximum compressed frame size
#define WMAPRO_BLOCK_MIN_BITS 6 ///< log2 of min block size
#define WMAPRO_BLOCK_MAX_BITS 12 ///< log2 of max block size
#define WMAPRO_BLOCK_MAX_SIZE (1 << WMAPRO_BLOCK_MAX_BITS) ///< maximum block size
#define WMAPRO_BLOCK_SIZES (WMAPRO_BLOCK_MAX_BITS - BLOCK_MIN_BITS + 1) ///< possible block sizes
#define WMAPRO_OUT_BUF_SIZE (WMAPRO_BLOCK_MAX_SIZE + WMAPRO_BLOCK_MAX_SIZE / 2)
#define VLCBITS 9
#define SCALEVLCBITS 8
#define VEC4MAXDEPTH ((HUFF_VEC4_MAXBITS+VLCBITS-1)/VLCBITS)
#define VEC2MAXDEPTH ((HUFF_VEC2_MAXBITS+VLCBITS-1)/VLCBITS)
#define VEC1MAXDEPTH ((HUFF_VEC1_MAXBITS+VLCBITS-1)/VLCBITS)
#define SCALEMAXDEPTH ((HUFF_SCALE_MAXBITS+SCALEVLCBITS-1)/SCALEVLCBITS)
#define SCALERLMAXDEPTH ((HUFF_SCALE_RL_MAXBITS+VLCBITS-1)/VLCBITS)
static VLC sf_vlc; ///< scale factor DPCM vlc
static VLC sf_rl_vlc; ///< scale factor run length vlc
static VLC vec4_vlc; ///< 4 coefficients per symbol
static VLC vec2_vlc; ///< 2 coefficients per symbol
static VLC vec1_vlc; ///< 1 coefficient per symbol
static VLC coef_vlc[2]; ///< coefficient run length vlc codes
//static float sin64[33]; ///< sinus table for decorrelation
/* Global defined arrays to allow IRAM usage for some models. */
static int32_t g_tmp[WMAPRO_BLOCK_MAX_SIZE] IBSS_ATTR_WMAPRO_LARGE_IRAM;
static int32_t g_out_ch0[WMAPRO_OUT_BUF_SIZE] IBSS_ATTR;
static int32_t g_out_ch1[WMAPRO_OUT_BUF_SIZE] IBSS_ATTR_WMAPRO_LARGE_IRAM;
#if (WMAPRO_MAX_CHANNELS > 2)
static int32_t g_out_multichannel[WMAPRO_MAX_CHANNELS-2][WMAPRO_OUT_BUF_SIZE];
#endif
/**
* @brief frame specific decoder context for a single channel
*/
typedef struct {
int16_t prev_block_len; ///< length of the previous block
uint8_t transmit_coefs;
uint8_t num_subframes;
uint16_t subframe_len[MAX_SUBFRAMES]; ///< subframe length in samples
uint16_t subframe_offset[MAX_SUBFRAMES]; ///< subframe positions in the current frame
uint8_t cur_subframe; ///< current subframe number
uint16_t decoded_samples; ///< number of already processed samples
uint8_t grouped; ///< channel is part of a group
int quant_step; ///< quantization step for the current subframe
int8_t reuse_sf; ///< share scale factors between subframes
int8_t scale_factor_step; ///< scaling step for the current subframe
int max_scale_factor; ///< maximum scale factor for the current subframe
int saved_scale_factors[2][MAX_BANDS]; ///< resampled and (previously) transmitted scale factor values
int8_t scale_factor_idx; ///< index for the transmitted scale factor values (used for resampling)
int* scale_factors; ///< pointer to the scale factor values used for decoding
uint8_t table_idx; ///< index in sf_offsets for the scale factor reference block
int32_t* coeffs; ///< pointer to the subframe decode buffer
uint16_t num_vec_coeffs; ///< number of vector coded coefficients
int32_t* out; ///< output buffer
} WMAProChannelCtx;
/**
* @brief channel group for channel transformations
*/
typedef struct {
uint8_t num_channels; ///< number of channels in the group
int8_t transform; ///< transform on / off
int8_t transform_band[MAX_BANDS]; ///< controls if the transform is enabled for a certain band
//float decorrelation_matrix[WMAPRO_MAX_CHANNELS*WMAPRO_MAX_CHANNELS];
int32_t* channel_data[WMAPRO_MAX_CHANNELS]; ///< transformation coefficients
int32_t fixdecorrelation_matrix[WMAPRO_MAX_CHANNELS*WMAPRO_MAX_CHANNELS];
} WMAProChannelGrp;
/**
* @brief main decoder context
*/
typedef struct WMAProDecodeCtx {
/* generic decoder variables */
uint8_t frame_data[MAX_FRAMESIZE +
FF_INPUT_BUFFER_PADDING_SIZE];///< compressed frame data
PutBitContext pb; ///< context for filling the frame_data buffer
int32_t* tmp; ///< IMDCT input buffer
/* frame size dependent frame information (set during initialization) */
uint32_t decode_flags; ///< used compression features
uint8_t len_prefix; ///< frame is prefixed with its length
uint8_t dynamic_range_compression; ///< frame contains DRC data
uint8_t bits_per_sample; ///< integer audio sample size for the unscaled IMDCT output (used to scale to [-1.0, 1.0])
uint8_t quant_step_bias; ///< added to the quantization step of a stream of less than 24 bits
int32_t quant_scale; ///< s1.30 factor for its quantization factors, 0 if none
uint16_t samples_per_frame; ///< number of samples to output
uint16_t log2_frame_size;
int8_t num_channels; ///< number of channels in the stream (same as AVCodecContext.num_channels)
int8_t lfe_channel; ///< lfe channel index
uint8_t max_num_subframes;
uint8_t subframe_len_bits; ///< number of bits used for the subframe length
uint8_t max_subframe_len_bit; ///< flag indicating that the subframe is of maximum size when the first subframe length bit is 1
uint16_t min_samples_per_subframe;
int8_t num_sfb[WMAPRO_BLOCK_SIZES]; ///< scale factor bands per block size
int16_t sfb_offsets[WMAPRO_BLOCK_SIZES][MAX_BANDS]; ///< scale factor band offsets (multiples of 4)
int8_t sf_offsets[WMAPRO_BLOCK_SIZES][WMAPRO_BLOCK_SIZES][MAX_BANDS]; ///< scale factor resample matrix
int16_t subwoofer_cutoffs[WMAPRO_BLOCK_SIZES]; ///< subwoofer cutoff values
/* packet decode state */
GetBitContext pgb; ///< bitstream reader context for the packet
uint8_t packet_offset; ///< frame offset in the packet
uint8_t packet_sequence_number; ///< current packet number
int num_saved_bits; ///< saved number of bits
int frame_offset; ///< frame offset in the bit reservoir
int subframe_offset; ///< subframe offset in the bit reservoir
uint8_t packet_loss; ///< set in case of bitstream error
uint8_t packet_done; ///< set when a packet is fully decoded
/* frame decode state */
uint32_t frame_num; ///< current frame number
GetBitContext gb; ///< bitstream reader context
int buf_bit_size; ///< buffer size in bits
int32_t samples;
int32_t* samples_end; ///< maximum samplebuffer pointer
uint8_t drc_gain; ///< gain for the DRC tool
int8_t skip_frame; ///< skip output step
int8_t parsed_all_subframes; ///< all subframes decoded?
/* subframe/block decode state */
int16_t subframe_len; ///< current subframe length
int8_t channels_for_cur_subframe; ///< number of channels that contain the subframe
int8_t channel_indexes_for_cur_subframe[WMAPRO_MAX_CHANNELS];
int8_t num_bands; ///< number of scale factor bands
int8_t transmit_num_vec_coeffs; ///< number of vector coded coefficients is part of the bitstream
int16_t* cur_sfb_offsets; ///< sfb offsets for the current block
uint8_t table_idx; ///< index for the num_sfb, sfb_offsets, sf_offsets and subwoofer_cutoffs tables
int8_t esc_len; ///< length of escaped coefficients
uint8_t num_chgroups; ///< number of channel groups
WMAProChannelGrp chgroup[WMAPRO_MAX_CHANNELS]; ///< channel group information
WMAProChannelCtx channel[WMAPRO_MAX_CHANNELS]; ///< per channel data
} WMAProDecodeCtx;
/* static decode context, to avoid malloc */
static WMAProDecodeCtx globWMAProDecCtx;
/**
*@brief helper function to print the most important members of the context
*@param s context
*/
#ifdef WMAPRO_DUMP_CTX_EN
static void dump_context(WMAProDecodeCtx *s)
{
#define PRINT(a, b) printf(" %s = %d\n", a, b);
#define PRINT_HEX(a, b) printf(" %s = %x\n", a, b);
PRINT("ed sample bit depth", s->bits_per_sample);
PRINT_HEX("ed decode flags", s->decode_flags);
PRINT("samples per frame", s->samples_per_frame);
PRINT("log2 frame size", s->log2_frame_size);
PRINT("max num subframes", s->max_num_subframes);
PRINT("len prefix", s->len_prefix);
PRINT("num channels", s->num_channels);
}
#endif
/**
*@brief Initialize the decoder.
*@param avctx codec context
*@return 0 on success, -1 otherwise
*/
int decode_init(asf_waveformatex_t *wfx)
{
memset(&globWMAProDecCtx, 0, sizeof(WMAProDecodeCtx));
WMAProDecodeCtx *s = &globWMAProDecCtx;
uint8_t *edata_ptr = wfx->data;
unsigned int channel_mask;
int i, bits;
int log2_max_num_subframes;
int num_possible_block_sizes;
/* Use globally defined array. Allows IRAM usage for models with large IRAM. */
s->tmp = g_tmp;
/* Use globally defined arrays. Allows IRAM usage for up to 2 channels. */
s->channel[0].out = g_out_ch0;
s->channel[1].out = g_out_ch1;
#if (WMAPRO_MAX_CHANNELS > 2)
for (i=2; i<WMAPRO_MAX_CHANNELS; ++i)
s->channel[i].out = g_out_multichannel[i-2];
#endif
#if defined(CPU_COLDFIRE)
coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
#endif
init_put_bits(&s->pb, s->frame_data, MAX_FRAMESIZE);
if (wfx->datalen >= 18) {
s->decode_flags = AV_RL16(edata_ptr+14);
channel_mask = AV_RL32(edata_ptr+2);
s->bits_per_sample = AV_RL16(edata_ptr);
/* A stream of fewer bits has a lower quantization step. Decode it at
* the level of a 24 bit stream instead: use that stream's quantization
* step, and scale the factors by what is left,
* 2^(24-bits) / 10^(bias/20). */
if (s->bits_per_sample == 16) {
s->quant_step_bias = (90 * 24 >> 4) - (90 * 16 >> 4);
s->quant_scale = 1545752065; /* 256 / 10^(45/20) */
} else if (s->bits_per_sample == 20) {
s->quant_step_bias = (90 * 24 >> 4) - (90 * 20 >> 4);
s->quant_scale = 1216241597; /* 16 / 10^(23/20) */
}
/** dump the extradata */
for (i = 0; i < wfx->datalen; i++)
DEBUGF("[%x] ", wfx->data[i]);
DEBUGF("\n");
} else {
DEBUGF("Unknown extradata size\n");
return AVERROR_INVALIDDATA;
}
/** generic init */
s->log2_frame_size = av_log2(wfx->blockalign) + 4;
/** frame info */
s->skip_frame = 1; /** skip first frame */
s->packet_loss = 1;
s->len_prefix = (s->decode_flags & 0x40);
if (!s->len_prefix) {
DEBUGF("no length prefix\n");
return AVERROR_INVALIDDATA;
}
/** get frame len */
bits = ff_wma_get_frame_len_bits(wfx->rate, 3, s->decode_flags);
if (bits > WMAPRO_BLOCK_MAX_BITS) {
DEBUGF("14-bits block sizes\n");
return AVERROR_INVALIDDATA;
}
s->samples_per_frame = 1 << bits;
/** init previous block len */
for (i = 0; i < wfx->channels; i++)
s->channel[i].prev_block_len = s->samples_per_frame;
/** subframe info */
log2_max_num_subframes = ((s->decode_flags & 0x38) >> 3);
s->max_num_subframes = 1 << log2_max_num_subframes;
if (s->max_num_subframes == 16 || s->max_num_subframes == 4)
s->max_subframe_len_bit = 1;
s->subframe_len_bits = av_log2(log2_max_num_subframes) + 1;
num_possible_block_sizes = log2_max_num_subframes + 1;
s->min_samples_per_subframe = s->samples_per_frame / s->max_num_subframes;
s->dynamic_range_compression = (s->decode_flags & 0x80);
if (s->max_num_subframes > MAX_SUBFRAMES) {
DEBUGF("invalid number of subframes %i\n",
s->max_num_subframes);
return AVERROR_INVALIDDATA;
}
if (s->min_samples_per_subframe < (1<<WMAPRO_BLOCK_MIN_BITS)) {
av_log(avctx, AV_LOG_ERROR, "min_samples_per_subframe of %d too small\n",
s->min_samples_per_subframe);
return AVERROR_INVALIDDATA;
}
s->num_channels = wfx->channels;
/** extract lfe channel position */
s->lfe_channel = -1;
if (channel_mask & 8) {
unsigned int mask;
for (mask = 1; mask < 16; mask <<= 1) {
if (channel_mask & mask)
++s->lfe_channel;
}
}
if (s->num_channels < 0) {
DEBUGF("invalid number of channels %d\n", s->num_channels);
return AVERROR_INVALIDDATA;
} else if (s->num_channels > WMAPRO_MAX_CHANNELS) {
DEBUGF("unsupported number of channels\n");
return AVERROR_PATCHWELCOME;
}
INIT_VLC_STATIC(&sf_vlc, SCALEVLCBITS, HUFF_SCALE_SIZE,
scale_huffbits, 1, 1,
scale_huffcodes, 2, 2, 616, IBSS_ATTR_WMAPRO_VLC_TABLES);
INIT_VLC_STATIC(&sf_rl_vlc, VLCBITS, HUFF_SCALE_RL_SIZE,
scale_rl_huffbits, 1, 1,
scale_rl_huffcodes, 4, 4, 1406, IBSS_ATTR_WMAPRO_VLC_TABLES);
INIT_VLC_STATIC(&coef_vlc[0], VLCBITS, HUFF_COEF0_SIZE,
coef0_huffbits, 1, 1,
coef0_huffcodes, 4, 4, 2108, IBSS_ATTR_WMAPRO_VLC_TABLES);
INIT_VLC_STATIC(&coef_vlc[1], VLCBITS, HUFF_COEF1_SIZE,
coef1_huffbits, 1, 1,
coef1_huffcodes, 4, 4, 3912, IBSS_ATTR_WMAPRO_VLC_TABLES);
INIT_VLC_STATIC(&vec4_vlc, VLCBITS, HUFF_VEC4_SIZE,
vec4_huffbits, 1, 1,
vec4_huffcodes, 2, 2, 604, IBSS_ATTR_WMAPRO_VLC_TABLES);
INIT_VLC_STATIC(&vec2_vlc, VLCBITS, HUFF_VEC2_SIZE,
vec2_huffbits, 1, 1,
vec2_huffcodes, 2, 2, 562, IBSS_ATTR_WMAPRO_VLC_TABLES);
INIT_VLC_STATIC(&vec1_vlc, VLCBITS, HUFF_VEC1_SIZE,
vec1_huffbits, 1, 1,
vec1_huffcodes, 2, 2, 562, IBSS_ATTR_WMAPRO_VLC_TABLES);
/** calculate number of scale factor bands and their offsets
for every possible block size */
for (i = 0; i < num_possible_block_sizes; i++) {
int subframe_len = s->samples_per_frame >> i;
int x;
int band = 1;
s->sfb_offsets[i][0] = 0;
for (x = 0; x < MAX_BANDS-1 && s->sfb_offsets[i][band - 1] < subframe_len; x++) {
int offset = (subframe_len * 2 * critical_freq[x])
/ wfx->rate + 2;
offset &= ~3;
if (offset > s->sfb_offsets[i][band - 1])
s->sfb_offsets[i][band++] = offset;
if (offset >= subframe_len)
break;
}
s->sfb_offsets[i][band - 1] = subframe_len;
s->num_sfb[i] = band - 1;
if (s->num_sfb[i] <= 0) {
av_log(avctx, AV_LOG_ERROR, "num_sfb invalid\n");
return AVERROR_INVALIDDATA;
}
}
/** Scale factors can be shared between blocks of different size
as every block has a different scale factor band layout.
The matrix sf_offsets is needed to find the correct scale factor.
*/
for (i = 0; i < num_possible_block_sizes; i++) {
int b;
for (b = 0; b < s->num_sfb[i]; b++) {
int x;
int offset = ((s->sfb_offsets[i][b]
+ s->sfb_offsets[i][b + 1] - 1) << i) >> 1;
for (x = 0; x < num_possible_block_sizes; x++) {
int v = 0;
while (s->sfb_offsets[x][v + 1] << x < offset)
++v;
s->sf_offsets[i][x][b] = v;
}
}
}
/** calculate subwoofer cutoff values */
for (i = 0; i < num_possible_block_sizes; i++) {
int block_size = s->samples_per_frame >> i;
int cutoff = (440*block_size + 3 * (wfx->rate >> 1) - 1)
/ wfx->rate;
s->subwoofer_cutoffs[i] = av_clip(cutoff, 4, block_size);
}
#if 0
/** calculate sine values for the decorrelation matrix */
for (i = 0; i < 33; i++)
sin64[i] = sin(i*M_PI / 64.0);
#endif
#ifdef WMAPRO_DUMP_CTX_EN
dump_context(s);
#endif
return 0;
}
/**
*@brief Decode the subframe length.
*@param s context
*@param offset sample offset in the frame
*@return decoded subframe length on success, < 0 in case of an error
*/
static int decode_subframe_length(WMAProDecodeCtx *s, int offset)
{
int frame_len_shift = 0;
int subframe_len;
/** no need to read from the bitstream when only one length is possible */
if (offset == s->samples_per_frame - s->min_samples_per_subframe)
return s->min_samples_per_subframe;
/** 1 bit indicates if the subframe is of maximum length */
if (s->max_subframe_len_bit) {
if (get_bits1(&s->gb))
frame_len_shift = 1 + get_bits(&s->gb, s->subframe_len_bits-1);
} else
frame_len_shift = get_bits(&s->gb, s->subframe_len_bits);
subframe_len = s->samples_per_frame >> frame_len_shift;
/** sanity check the length */
if (subframe_len < s->min_samples_per_subframe ||
subframe_len > s->samples_per_frame) {
DEBUGF("broken frame: subframe_len %i\n",
subframe_len);
return AVERROR_INVALIDDATA;
}
return subframe_len;
}
/**
*@brief Decode how the data in the frame is split into subframes.
* Every WMA frame contains the encoded data for a fixed number of
* samples per channel. The data for every channel might be split
* into several subframes. This function will reconstruct the list of
* subframes for every channel.
*
* If the subframes are not evenly split, the algorithm estimates the
* channels with the lowest number of total samples.
* Afterwards, for each of these channels a bit is read from the
* bitstream that indicates if the channel contains a subframe with the
* next subframe size that is going to be read from the bitstream or not.
* If a channel contains such a subframe, the subframe size gets added to
* the channel's subframe list.
* The algorithm repeats these steps until the frame is properly divided
* between the individual channels.
*
*@param s context
*@return 0 on success, < 0 in case of an error
*/
static int decode_tilehdr(WMAProDecodeCtx *s)
{
uint16_t num_samples[WMAPRO_MAX_CHANNELS]; /** sum of samples for all currently known subframes of a channel */
uint8_t contains_subframe[WMAPRO_MAX_CHANNELS]; /** flag indicating if a channel contains the current subframe */
int channels_for_cur_subframe = s->num_channels; /** number of channels that contain the current subframe */
int fixed_channel_layout = 0; /** flag indicating that all channels use the same subframe offsets and sizes */
int min_channel_len = 0; /** smallest sum of samples (channels with this length will be processed first) */
int c;
/* Should never consume more than 3073 bits (256 iterations for the
* while loop when always the minimum amount of 128 samples is substracted
* from missing samples in the 8 channel case).
* 1 + BLOCK_MAX_SIZE * MAX_CHANNELS / BLOCK_MIN_SIZE * (MAX_CHANNELS + 4)
*/
/** reset tiling information */
for (c = 0; c < s->num_channels; c++)
s->channel[c].num_subframes = 0;
memset(num_samples, 0, sizeof(num_samples));
if (s->max_num_subframes == 1 || get_bits1(&s->gb))
fixed_channel_layout = 1;
/** loop until the frame data is split between the subframes */
do {
int subframe_len;
/** check which channels contain the subframe */
for (c = 0; c < s->num_channels; c++) {
if (num_samples[c] == min_channel_len) {
if (fixed_channel_layout || channels_for_cur_subframe == 1 ||
(min_channel_len == s->samples_per_frame - s->min_samples_per_subframe))
contains_subframe[c] = 1;
else
contains_subframe[c] = get_bits1(&s->gb);
} else
contains_subframe[c] = 0;
}
/** get subframe length, subframe_len == 0 is not allowed */
if ((subframe_len = decode_subframe_length(s, min_channel_len)) <= 0)
return AVERROR_INVALIDDATA;
/** add subframes to the individual channels and find new min_channel_len */
min_channel_len += subframe_len;
for (c = 0; c < s->num_channels; c++) {
WMAProChannelCtx* chan = &s->channel[c];
if (contains_subframe[c]) {
if (chan->num_subframes >= MAX_SUBFRAMES) {
DEBUGF("broken frame: num subframes > 31\n");
return AVERROR_INVALIDDATA;
}
chan->subframe_len[chan->num_subframes] = subframe_len;
num_samples[c] += subframe_len;
++chan->num_subframes;
if (num_samples[c] > s->samples_per_frame) {
DEBUGF("broken frame: "
"channel len > samples_per_frame\n");
return AVERROR_INVALIDDATA;
}
} else if (num_samples[c] <= min_channel_len) {
if (num_samples[c] < min_channel_len) {
channels_for_cur_subframe = 0;
min_channel_len = num_samples[c];
}
++channels_for_cur_subframe;
}
}
} while (min_channel_len < s->samples_per_frame);
for (c = 0; c < s->num_channels; c++) {
int i;
int offset = 0;
for (i = 0; i < s->channel[c].num_subframes; i++) {
DEBUGF("frame[%i] channel[%i] subframe[%i]"
" len %i\n", s->frame_num, c, i,
s->channel[c].subframe_len[i]);
s->channel[c].subframe_offset[i] = offset;
offset += s->channel[c].subframe_len[i];
}
}
return 0;
}
#if 0
/**
*@brief Calculate a decorrelation matrix from the bitstream parameters.
*@param s codec context
*@param chgroup channel group for which the matrix needs to be calculated
*/
static void decode_decorrelation_matrix(WMAProDecodeCtx *s,
WMAProChannelGrp *chgroup)
{
int i;
int offset = 0;
int8_t rotation_offset[WMAPRO_MAX_CHANNELS * WMAPRO_MAX_CHANNELS];
memset(chgroup->decorrelation_matrix, 0, s->num_channels *
s->num_channels * sizeof(*chgroup->decorrelation_matrix));
for (i = 0; i < chgroup->num_channels * (chgroup->num_channels - 1) >> 1; i++)
rotation_offset[i] = get_bits(&s->gb, 6);
for (i = 0; i < chgroup->num_channels; i++) {
chgroup->decorrelation_matrix[chgroup->num_channels * i + i] =
get_bits1(&s->gb) ? 1.0 : -1.0;
if(chgroup->decorrelation_matrix[chgroup->num_channels * i + i] > 0)
chgroup->fixdecorrelation_matrix[chgroup->num_channels * i + i] = ONE_FRACT16;
else
chgroup->fixdecorrelation_matrix[chgroup->num_channels * i + i] = -ONE_FRACT16;
}
for (i = 1; i < chgroup->num_channels; i++) {
int x;
for (x = 0; x < i; x++) {
int y;
for (y = 0; y < i + 1; y++) {
float v1 = chgroup->decorrelation_matrix[x * chgroup->num_channels + y];
float v2 = chgroup->decorrelation_matrix[i * chgroup->num_channels + y];
int32_t f1 = chgroup->fixdecorrelation_matrix[x * chgroup->num_channels + y];
int32_t f2 = chgroup->fixdecorrelation_matrix[i * chgroup->num_channels + y];
int n = rotation_offset[offset + x];
float sinv;
float cosv;
int32_t fixsinv;
int32_t fixcosv;
if (n < 32) {
sinv = sin64[n];
cosv = sin64[32 - n];
fixsinv = fixed_sin64[n];
fixcosv = fixed_sin64[32-n];
} else {
sinv = sin64[64 - n];
cosv = -sin64[n - 32];
fixsinv = fixed_sin64[64-n];
fixcosv = -fixed_sin64[n-32];
}
chgroup->decorrelation_matrix[y + x * chgroup->num_channels] =
(v1 * sinv) - (v2 * cosv);
chgroup->decorrelation_matrix[y + i * chgroup->num_channels] =
(v1 * cosv) + (v2 * sinv);
chgroup->fixdecorrelation_matrix[y + x * chgroup->num_channels] =
fixmul31(f1, fixsinv) - fixmul31(f2, fixcosv);
chgroup->fixdecorrelation_matrix[y + i * chgroup->num_channels] =
fixmul31(f1, fixcosv) + fixmul31(f2, fixsinv);
}
}
offset += i;
}
}
#endif
/**
*@brief Decode channel transformation parameters
*@param s codec context
*@return 0 in case of success, < 0 in case of bitstream errors
*/
static int decode_channel_transform(WMAProDecodeCtx* s)
{
int i;
/* should never consume more than 1921 bits for the 8 channel case
* 1 + MAX_CHANNELS * (MAX_CHANNELS + 2 + 3 * MAX_CHANNELS * MAX_CHANNELS
* + MAX_CHANNELS + MAX_BANDS + 1)
*/
/** in the one channel case channel transforms are pointless */
s->num_chgroups = 0;
if (s->num_channels > 1) {
int remaining_channels = s->channels_for_cur_subframe;
if (get_bits1(&s->gb)) {
DEBUGF("unsupported channel transform bit\n");
return AVERROR_INVALIDDATA;
}
for (s->num_chgroups = 0; remaining_channels &&
s->num_chgroups < s->channels_for_cur_subframe; s->num_chgroups++) {
WMAProChannelGrp* chgroup = &s->chgroup[s->num_chgroups];
int32_t** channel_data = chgroup->channel_data;
chgroup->num_channels = 0;
chgroup->transform = 0;
/** decode channel mask */
if (remaining_channels > 2) {
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int channel_idx = s->channel_indexes_for_cur_subframe[i];
if (!s->channel[channel_idx].grouped
&& get_bits1(&s->gb)) {
++chgroup->num_channels;
s->channel[channel_idx].grouped = 1;
*channel_data++ = s->channel[channel_idx].coeffs;
}
}
} else {
chgroup->num_channels = remaining_channels;
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int channel_idx = s->channel_indexes_for_cur_subframe[i];
if (!s->channel[channel_idx].grouped)
*channel_data++ = s->channel[channel_idx].coeffs;
s->channel[channel_idx].grouped = 1;
}
}
/** decode transform type */
if (chgroup->num_channels == 2) {
if (get_bits1(&s->gb)) {
if (get_bits1(&s->gb)) {
DEBUGF("unsupported channel transform type\n");
}
} else {
chgroup->transform = 1;
if (s->num_channels == 2) {
chgroup->fixdecorrelation_matrix[0] = ONE_FRACT16;
chgroup->fixdecorrelation_matrix[1] = -ONE_FRACT16;
chgroup->fixdecorrelation_matrix[2] = ONE_FRACT16;
chgroup->fixdecorrelation_matrix[3] = ONE_FRACT16;
} else {
/** cos(pi/4) */
chgroup->fixdecorrelation_matrix[0] = COS_PI4_FRACT16;
chgroup->fixdecorrelation_matrix[1] = -COS_PI4_FRACT16;
chgroup->fixdecorrelation_matrix[2] = COS_PI4_FRACT16;
chgroup->fixdecorrelation_matrix[3] = COS_PI4_FRACT16;
}
}
} else if (chgroup->num_channels > 2) {
DEBUGF("in wmaprodec.c: Multichannel streams still not supported\n");
return -1;
#if 0
if (get_bits1(&s->gb)) {
chgroup->transform = 1;
if (get_bits1(&s->gb)) {
decode_decorrelation_matrix(s, chgroup);
} else {
/** FIXME: more than 6 coupled channels not supported */
if (chgroup->num_channels > 6) {
av_log_ask_for_sample(s->avctx,
"coupled channels > 6\n");
} else {
memcpy(chgroup->decorrelation_matrix,
default_decorrelation[chgroup->num_channels],
chgroup->num_channels * chgroup->num_channels *
sizeof(*chgroup->decorrelation_matrix));
}
}
}
#endif
}
/** decode transform on / off */
if (chgroup->transform) {
if (!get_bits1(&s->gb)) {
int i;
/** transform can be enabled for individual bands */
for (i = 0; i < s->num_bands; i++) {
chgroup->transform_band[i] = get_bits1(&s->gb);
}
} else {
memset(chgroup->transform_band, 1, s->num_bands);
}
}
remaining_channels -= chgroup->num_channels;
}
}
return 0;
}
/**
*@brief Extract the coefficients from the bitstream.
*@param s codec context
*@param c current channel number
*@return 0 on success, < 0 in case of bitstream errors
*/
static int decode_coeffs(WMAProDecodeCtx *s, int c)
{
int vlctable;
VLC* vlc;
WMAProChannelCtx* ci = &s->channel[c];
int rl_mode = 0;
int cur_coeff = 0;
int num_zeros = 0;
const uint16_t* run;
const int32_t* level;
DEBUGF("decode coefficients for channel %i\n", c);
vlctable = get_bits1(&s->gb);
vlc = &coef_vlc[vlctable];
if (vlctable) {
run = coef1_run;
level = coef1_level;
} else {
run = coef0_run;
level = coef0_level;
}
/** decode vector coefficients (consumes up to 167 bits per iteration for
4 vector coded large values) */
while ((s->transmit_num_vec_coeffs || !rl_mode) &&
(cur_coeff + 3 < ci->num_vec_coeffs)) {
int32_t vals[4];
int i;
unsigned int idx;
idx = get_vlc2(&s->gb, vec4_vlc.table, VLCBITS, VEC4MAXDEPTH);
if (idx == HUFF_VEC4_SIZE - 1) {
for (i = 0; i < 4; i += 2) {
idx = get_vlc2(&s->gb, vec2_vlc.table, VLCBITS, VEC2MAXDEPTH);
if (idx == HUFF_VEC2_SIZE - 1) {
int v0, v1;
v0 = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH);
if (v0 == HUFF_VEC1_SIZE - 1)
v0 += ff_wma_get_large_val(&s->gb);
v1 = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH);
if (v1 == HUFF_VEC1_SIZE - 1)
v1 += ff_wma_get_large_val(&s->gb);
vals[i ] = v0;
vals[i+1] = v1;
} else {
vals[i ] = symbol_to_vec2[idx] >> 4;
vals[i+1] = symbol_to_vec2[idx] & 0xF;
}
}
} else {
vals[0] = (symbol_to_vec4[idx] >> 12);
vals[1] = (symbol_to_vec4[idx] >> 8) & 0xF;
vals[2] = (symbol_to_vec4[idx] >> 4) & 0xF;
vals[3] = (symbol_to_vec4[idx] ) & 0xF;
}
/* Rockbox: To be able to use rockbox' optimized mdct we need to
* pre-shift the values by >>(nbits-3). */
const int nbits = av_log2(s->subframe_len)+1;
const int shift = WMAPRO_FRACT-(nbits-3);
/** decode sign */
for (i = 0; i < 4; i++) {
if (vals[i]) {
int sign = get_bits1(&s->gb) - 1;
/* Rockbox: To be able to use rockbox' optimized mdct we need
* invert the sign. */
ci->coeffs[cur_coeff] = (sign == -1)? vals[i]<<shift : -vals[i]<<shift;
num_zeros = 0;
} else {
ci->coeffs[cur_coeff] = 0;
/** switch to run level mode when subframe_len / 128 zeros
were found in a row */
rl_mode |= (++num_zeros > s->subframe_len >> 8);
}
++cur_coeff;
}
}
/** decode run level coded coefficients */
if (cur_coeff < s->subframe_len) {
memset(&ci->coeffs[cur_coeff], 0,
sizeof(*ci->coeffs) * (s->subframe_len - cur_coeff));
if (ff_wma_run_level_decode(&s->gb, vlc,
level, run, 1, ci->coeffs,
cur_coeff, s->subframe_len,
s->subframe_len, s->esc_len, 0))
return AVERROR_INVALIDDATA;
}
return 0;
}
/**
*@brief Extract scale factors from the bitstream.
*@param s codec context
*@return 0 on success, < 0 in case of bitstream errors
*/
static int decode_scale_factors(WMAProDecodeCtx* s)
{
int i;
/** should never consume more than 5344 bits
* MAX_CHANNELS * (1 + MAX_BANDS * 23)
*/
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
int* sf;
int* sf_end;
s->channel[c].scale_factors = s->channel[c].saved_scale_factors[!s->channel[c].scale_factor_idx];
sf_end = s->channel[c].scale_factors + s->num_bands;
/** resample scale factors for the new block size
* as the scale factors might need to be resampled several times
* before some new values are transmitted, a backup of the last
* transmitted scale factors is kept in saved_scale_factors
*/
if (s->channel[c].reuse_sf) {
const int8_t* sf_offsets = s->sf_offsets[s->table_idx][s->channel[c].table_idx];
int b;
for (b = 0; b < s->num_bands; b++)
s->channel[c].scale_factors[b] =
s->channel[c].saved_scale_factors[s->channel[c].scale_factor_idx][*sf_offsets++];
}
if (!s->channel[c].cur_subframe || get_bits1(&s->gb)) {
if (!s->channel[c].reuse_sf) {
int val;
/** decode DPCM coded scale factors */
s->channel[c].scale_factor_step = get_bits(&s->gb, 2) + 1;
val = 45 / s->channel[c].scale_factor_step;
for (sf = s->channel[c].scale_factors; sf < sf_end; sf++) {
val += get_vlc2(&s->gb, sf_vlc.table, SCALEVLCBITS, SCALEMAXDEPTH) - 60;
*sf = val;
}
} else {
int i;
/** run level decode differences to the resampled factors */
for (i = 0; i < s->num_bands; i++) {
int idx;
int skip;
int val;
int sign;
idx = get_vlc2(&s->gb, sf_rl_vlc.table, VLCBITS, SCALERLMAXDEPTH);
if (!idx) {
uint32_t code = get_bits(&s->gb, 14);
val = code >> 6;
sign = (code & 1) - 1;
skip = (code & 0x3f) >> 1;
} else if (idx == 1) {
break;
} else {
skip = scale_rl_run[idx];
val = scale_rl_level[idx];
sign = get_bits1(&s->gb)-1;
}
i += skip;
if (i >= s->num_bands) {
DEBUGF("invalid scale factor coding\n");
return AVERROR_INVALIDDATA;
}
s->channel[c].scale_factors[i] += (val ^ sign) - sign;
}
}
/** swap buffers */
s->channel[c].scale_factor_idx = !s->channel[c].scale_factor_idx;
s->channel[c].table_idx = s->table_idx;
s->channel[c].reuse_sf = 1;
}
/** calculate new scale factor maximum */
s->channel[c].max_scale_factor = s->channel[c].scale_factors[0];
for (sf = s->channel[c].scale_factors + 1; sf < sf_end; sf++) {
s->channel[c].max_scale_factor =
FFMAX(s->channel[c].max_scale_factor, *sf);
}
}
return 0;
}
/**
*@brief Apply a 2x2 decorrelation matrix to one band of two channels.
* Two channels are by far the common case; the results are the
* same as the general loop's in inverse_channel_transform().
*@param ch0 first channel's coefficients
*@param ch1 second channel's coefficients
*@param mat the matrix, as 16.16 fixed point
*@param len number of coefficients
*/
static inline void decorrelate_stereo(int32_t *ch0, int32_t *ch1,
const int32_t *mat, int len)
{
const int32_t m0 = mat[0], m1 = mat[1], m2 = mat[2], m3 = mat[3];
if (m0 == ONE_FRACT16 && m1 == -ONE_FRACT16 &&
m2 == ONE_FRACT16 && m3 == ONE_FRACT16) {
/* The matrix of a stereo stream. Multiplying by +-1.0 is exact,
* so add and subtract; unsigned, as the multiply's result wraps. */
for (; len > 0; len--) {
uint32_t a = *ch0, b = *ch1;
*ch0++ = a - b;
*ch1++ = a + b;
}
} else {
for (; len > 0; len--) {
int32_t a = *ch0, b = *ch1;
*ch0++ = fixmul16(m0, a) + fixmul16(m1, b);
*ch1++ = fixmul16(m2, a) + fixmul16(m3, b);
}
}
}
/**
*@brief Reconstruct the individual channel data.
*@param s codec context
*/
static void inverse_channel_transform(WMAProDecodeCtx *s)
{
int i;
for (i = 0; i < s->num_chgroups; i++) {
if (s->chgroup[i].transform) {
const int num_channels = s->chgroup[i].num_channels;
int32_t data[WMAPRO_MAX_CHANNELS];
int32_t** ch_data = s->chgroup[i].channel_data;
int32_t** ch_end = ch_data + num_channels;
const int8_t* tb = s->chgroup[i].transform_band;
int16_t* sfb;
/** multichannel decorrelation */
for (sfb = s->cur_sfb_offsets;
sfb < s->cur_sfb_offsets + s->num_bands; sfb++) {
int y;
if (*tb++ == 1) {
if (num_channels == 2) {
decorrelate_stereo(ch_data[0] + sfb[0],
ch_data[1] + sfb[0],
s->chgroup[i].fixdecorrelation_matrix,
FFMIN(sfb[1], s->subframe_len) - sfb[0]);
} else {
/** multiply values with the decorrelation_matrix */
for (y = sfb[0];
y < FFMIN(sfb[1], s->subframe_len); y++) {
const int32_t* mat =
s->chgroup[i].fixdecorrelation_matrix;
const int32_t* data_end = data + num_channels;
int32_t* data_ptr = data;
int32_t** ch;
for (ch = ch_data; ch < ch_end; ch++)
*data_ptr++ = (*ch)[y];
for (ch = ch_data; ch < ch_end; ch++) {
int32_t sum = 0;
data_ptr = data;
while (data_ptr < data_end)
sum += fixmul16(*mat++, *data_ptr++);
(*ch)[y] = sum;
}
}
}
} else if (s->num_channels == 2) {
/* Scale with sqrt(2) */
int len = FFMIN(sfb[1], s->subframe_len) - sfb[0];
vector_fixmul_scalar(ch_data[0] + sfb[0],
ch_data[0] + sfb[0],
SQRT2_FRACT16, len);
vector_fixmul_scalar(ch_data[1] + sfb[0],
ch_data[1] + sfb[0],
SQRT2_FRACT16, len);
}
}
}
}
}
/**
*@brief Apply sine window and reconstruct the output buffer.
*@param s codec context
*/
static void wmapro_window(WMAProDecodeCtx *s)
{
int i;
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
const int32_t* window;
int winlen = s->channel[c].prev_block_len;
int32_t *xstart= s->channel[c].coeffs - (winlen >> 1);
if (s->subframe_len < winlen) {
xstart += (winlen - s->subframe_len) >> 1;
winlen = s->subframe_len;
}
window = sine_windows[av_log2(winlen) - BLOCK_MIN_BITS];
winlen >>= 1;
vector_fixmul_window(xstart, xstart, xstart + winlen,
window, winlen);
s->channel[c].prev_block_len = s->subframe_len;
}
}
/**
*@brief Decode a single subframe (block).
*@param s codec context
*@return 0 on success, < 0 when decoding failed
*/
static int decode_subframe(WMAProDecodeCtx *s)
{
int offset = s->samples_per_frame;
int subframe_len = s->samples_per_frame;
int i;
int total_samples = s->samples_per_frame * s->num_channels;
int transmit_coeffs = 0;
int cur_subwoofer_cutoff;
s->subframe_offset = get_bits_count(&s->gb);
/** reset channel context and find the next block offset and size
== the next block of the channel with the smallest number of
decoded samples
*/
for (i = 0; i < s->num_channels; i++) {
s->channel[i].grouped = 0;
if (offset > s->channel[i].decoded_samples) {
offset = s->channel[i].decoded_samples;
subframe_len =
s->channel[i].subframe_len[s->channel[i].cur_subframe];
}
}
DEBUGF("processing subframe with offset %i len %i\n", offset, subframe_len);
/** get a list of all channels that contain the estimated block */
s->channels_for_cur_subframe = 0;
for (i = 0; i < s->num_channels; i++) {
const int cur_subframe = s->channel[i].cur_subframe;
/** substract already processed samples */
total_samples -= s->channel[i].decoded_samples;
/** and count if there are multiple subframes that match our profile */
if (offset == s->channel[i].decoded_samples &&
subframe_len == s->channel[i].subframe_len[cur_subframe]) {
total_samples -= s->channel[i].subframe_len[cur_subframe];
s->channel[i].decoded_samples +=
s->channel[i].subframe_len[cur_subframe];
s->channel_indexes_for_cur_subframe[s->channels_for_cur_subframe] = i;
++s->channels_for_cur_subframe;
}
}
/** check if the frame will be complete after processing the
estimated block */
if (!total_samples)
s->parsed_all_subframes = 1;
DEBUGF("subframe is part of %i channels\n", s->channels_for_cur_subframe);
/** calculate number of scale factor bands and their offsets */
s->table_idx = av_log2(s->samples_per_frame/subframe_len);
s->num_bands = s->num_sfb[s->table_idx];
s->cur_sfb_offsets = s->sfb_offsets[s->table_idx];
cur_subwoofer_cutoff = s->subwoofer_cutoffs[s->table_idx];
/** configure the decoder for the current subframe */
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
s->channel[c].coeffs = &s->channel[c].out[(s->samples_per_frame >> 1)
+ offset];
}
s->subframe_len = subframe_len;
s->esc_len = av_log2(s->subframe_len - 1) + 1;
/** skip extended header if any */
if (get_bits1(&s->gb)) {
int num_fill_bits;
if (!(num_fill_bits = get_bits(&s->gb, 2))) {
int len = get_bits(&s->gb, 4);
num_fill_bits = (len ? get_bits(&s->gb, len) : 0) + 1;
}
if (num_fill_bits >= 0) {
if (get_bits_count(&s->gb) + num_fill_bits > s->num_saved_bits) {
DEBUGF("invalid number of fill bits\n");
return AVERROR_INVALIDDATA;
}
skip_bits_long(&s->gb, num_fill_bits);
}
}
/** no idea for what the following bit is used */
if (get_bits1(&s->gb)) {
DEBUGF("reserved bit set\n");
return AVERROR_INVALIDDATA;
}
if (decode_channel_transform(s) < 0)
return AVERROR_INVALIDDATA;
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
if ((s->channel[c].transmit_coefs = get_bits1(&s->gb)))
transmit_coeffs = 1;
}
if (transmit_coeffs) {
int step;
int quant_step = (90 * s->bits_per_sample >> 4) + s->quant_step_bias;
/** decode number of vector coded coefficients */
if ((s->transmit_num_vec_coeffs = get_bits1(&s->gb))) {
int num_bits = av_log2((s->subframe_len + 3)/4) + 1;
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
int num_vec_coeffs = get_bits(&s->gb, num_bits) << 2;
if (num_vec_coeffs > s->subframe_len) {
av_log(s->avctx, AV_LOG_ERROR, "num_vec_coeffs %d is too large\n", num_vec_coeffs);
return AVERROR_INVALIDDATA;
}
s->channel[c].num_vec_coeffs = num_vec_coeffs;
}
} else {
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
s->channel[c].num_vec_coeffs = s->subframe_len;
}
}
/** decode quantization step */
step = get_sbits(&s->gb, 6);
quant_step += step;
if (step == -32 || step == 31) {
const int sign = (step == 31) - 1;
int quant = 0;
while (get_bits_count(&s->gb) + 5 < s->num_saved_bits &&
(step = get_bits(&s->gb, 5)) == 31) {
quant += 31;
}
quant_step += ((quant + step) ^ sign) - sign;
}
if (quant_step < 0) {
DEBUGF("negative quant step\n");
}
/** decode quantization step modifiers for every channel */
if (s->channels_for_cur_subframe == 1) {
s->channel[s->channel_indexes_for_cur_subframe[0]].quant_step = quant_step;
} else {
int modifier_len = get_bits(&s->gb, 3);
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
s->channel[c].quant_step = quant_step;
if (get_bits1(&s->gb)) {
if (modifier_len) {
s->channel[c].quant_step += get_bits(&s->gb, modifier_len) + 1;
} else
++s->channel[c].quant_step;
}
}
}
/** decode scale factors */
if (decode_scale_factors(s) < 0)
return AVERROR_INVALIDDATA;
}
DEBUGF("BITSTREAM: subframe header length was %i\n",
get_bits_count(&s->gb) - s->subframe_offset);
/** parse coefficients */
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
if (s->channel[c].transmit_coefs &&
get_bits_count(&s->gb) < s->num_saved_bits) {
decode_coeffs(s, c);
} else {
memset(s->channel[c].coeffs, 0,
sizeof(*s->channel[c].coeffs) * subframe_len);
}
}
DEBUGF("BITSTREAM: subframe length was %i\n",
get_bits_count(&s->gb) - s->subframe_offset);
if (transmit_coeffs) {
int nbits = av_log2(subframe_len)+1;
/** reconstruct the per channel data */
inverse_channel_transform(s);
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
const int* sf = s->channel[c].scale_factors;
int b;
if (c == s->lfe_channel)
memset(&s->tmp[cur_subwoofer_cutoff], 0, sizeof(*s->tmp) *
(subframe_len - cur_subwoofer_cutoff));
/** inverse quantization and rescaling */
for (b = 0; b < s->num_bands; b++) {
const int end = FFMIN(s->cur_sfb_offsets[b+1], s->subframe_len);
const int exp = s->channel[c].quant_step -
(s->channel[c].max_scale_factor - *sf++) *
s->channel[c].scale_factor_step;
if(exp < EXP_MIN || exp > EXP_MAX) {
DEBUGF("in wmaprodec.c : unhandled value for exp (%d), please report sample.\n", exp);
return -1;
}
int32_t quant = QUANT(exp);
int start = s->cur_sfb_offsets[b];
if (s->quant_scale)
quant = (int64_t)quant * s->quant_scale >> 30;
vector_fixmul_scalar(s->tmp+start,
s->channel[c].coeffs + start,
quant, end-start);
}
/** apply imdct (ff_imdct_half == DCTIV with reverse) */
ff_imdct_half(nbits,s->channel[c].coeffs, s->tmp);
}
}
/** window and overlapp-add */
wmapro_window(s);
/** handled one subframe */
for (i = 0; i < s->channels_for_cur_subframe; i++) {
int c = s->channel_indexes_for_cur_subframe[i];
if (s->channel[c].cur_subframe >= s->channel[c].num_subframes) {
DEBUGF("broken subframe\n");
return AVERROR_INVALIDDATA;
}
++s->channel[c].cur_subframe;
}
return 0;
}
/**
*@brief Decode one WMA frame.
*@param s codec context
*@return 0 if the trailer bit indicates that this is the last frame,
* 1 if there are additional frames
*/
static int decode_frame(WMAProDecodeCtx *s)
{
GetBitContext* gb = &s->gb;
int more_frames = 0;
int len = 0;
int i;
#if 0
/** check for potential output buffer overflow */
/* Rockbox : No need to check that anymore since we work directly on the
buffers in the WMAProDecCtx */
if (s->num_channels * s->samples_per_frame > s->samples_end - s->samples) {
/** return an error if no frame could be decoded at all */
DEBUGF("not enough space for the output samples\n");
s->packet_loss = 1;
return 0;
}
#endif
/** get frame length */
if (s->len_prefix)
len = get_bits(gb, s->log2_frame_size);
DEBUGF("decoding frame with length %x\n", len);
/** decode tile information */
if (decode_tilehdr(s)) {
s->packet_loss = 1;
return 0;
}
/** read postproc transform */
if (s->num_channels > 1 && get_bits1(gb)) {
DEBUGF("Unsupported postproc transform found\n");
s->packet_loss = 1;
return 0;
}
/** read drc info */
if (s->dynamic_range_compression) {
s->drc_gain = get_bits(gb, 8);
DEBUGF("drc_gain %i\n", s->drc_gain);
}
/** no idea what these are for, might be the number of samples
that need to be skipped at the beginning or end of a stream */
if (get_bits1(gb)) {
/** usually true for the first frame */
if (get_bits1(gb)) {
get_bits(gb, av_log2(s->samples_per_frame * 2));
DEBUGF("start skip\n");
}
/** sometimes true for the last frame */
if (get_bits1(gb)) {
get_bits(gb, av_log2(s->samples_per_frame * 2));
DEBUGF("end skip\n");
}
}
DEBUGF("BITSTREAM: frame header length was %i\n",
get_bits_count(gb) - s->frame_offset);
/** reset subframe states */
s->parsed_all_subframes = 0;
for (i = 0; i < s->num_channels; i++) {
s->channel[i].decoded_samples = 0;
s->channel[i].cur_subframe = 0;
s->channel[i].reuse_sf = 0;
}
/** decode all subframes */
while (!s->parsed_all_subframes) {
if (decode_subframe(s) < 0) {
s->packet_loss = 1;
return 0;
}
}
if (s->skip_frame) {
s->skip_frame = 0;
} else
s->samples += s->num_channels * s->samples_per_frame;
if (len != (get_bits_count(gb) - s->frame_offset) + 2) {
/** FIXME: not sure if this is always an error */
DEBUGF("frame[%i] would have to skip %i bits\n",
(int)s->frame_num, len - (get_bits_count(gb) - s->frame_offset) - 1);
s->packet_loss = 1;
return 0;
}
/** skip the rest of the frame data */
skip_bits_long(gb, len - (get_bits_count(gb) - s->frame_offset) - 1);
/** decode trailer bit */
more_frames = get_bits1(gb);
++s->frame_num;
return more_frames;
}
/**
*@brief Calculate remaining input buffer length.
*@param s codec context
*@param gb bitstream reader context
*@return remaining size in bits
*/
static int remaining_bits(WMAProDecodeCtx *s, GetBitContext *gb)
{
return s->buf_bit_size - get_bits_count(gb);
}
/**
*@brief Fill the bit reservoir with a (partial) frame.
*@param s codec context
*@param gb bitstream reader context
*@param len length of the partial frame
*@param append decides wether to reset the buffer or not
*/
static void save_bits(WMAProDecodeCtx *s, GetBitContext* gb, int len,
int append)
{
int buflen;
/** when the frame data does not need to be concatenated, the input buffer
is resetted and additional bits from the previous frame are copyed
and skipped later so that a fast byte copy is possible */
if (!append) {
s->frame_offset = get_bits_count(gb) & 7;
s->num_saved_bits = s->frame_offset;
init_put_bits(&s->pb, s->frame_data, MAX_FRAMESIZE);
}
buflen = (put_bits_count(&s->pb) + len + 8) >> 3;
if (len <= 0 || buflen > MAX_FRAMESIZE) {
DEBUGF("input buffer too small\n");
s->packet_loss = 1;
return;
}
#if 0
if (len > put_bits_left(&s->pb)) {
av_log(s->avctx, AV_LOG_ERROR,
"Cannot append %d bits, only %d bits available.\n",
len, put_bits_left(&s->pb));
s->packet_loss = 1;
return;
}
#endif
s->num_saved_bits += len;
if (!append) {
ff_copy_bits(&s->pb, gb->buffer + (get_bits_count(gb) >> 3),
s->num_saved_bits);
} else {
int align = 8 - (get_bits_count(gb) & 7);
align = FFMIN(align, len);
put_bits(&s->pb, align, get_bits(gb, align));
len -= align;
ff_copy_bits(&s->pb, gb->buffer + (get_bits_count(gb) >> 3), len);
}
skip_bits_long(gb, len);
{
PutBitContext tmp = s->pb;
flush_put_bits(&tmp);
}
init_get_bits(&s->gb, s->frame_data, s->num_saved_bits);
skip_bits(&s->gb, s->frame_offset);
}
/**
*@brief Decode a single WMA packet.
*@param avctx codec context
*@param data the output buffer
*@param data_size number of bytes that were written to the output buffer
*@param avpkt input packet
*@return number of bytes that were read from the input buffer
*/
int decode_packet(asf_waveformatex_t *wfx, int32_t *dec[2], int *data_size,
void* pktdata, int size)
{
WMAProDecodeCtx *s = &globWMAProDecCtx;
GetBitContext* gb = &s->pgb;
const uint8_t* buf = pktdata;
int buf_size = size;
int num_bits_prev_frame;
int packet_sequence_number;\
int i;
/** reuse second half of the IMDCT output for the next frame */
/* NOTE : Relies on the WMAProDecCtx being static */
for(i = 0; i < s->num_channels; i++)
memcpy(&s->channel[i].out[0],
&s->channel[i].out[s->samples_per_frame],
s->samples_per_frame * sizeof(*s->channel[i].out) >> 1);
s->samples = 0;
*data_size = 0;
if (s->packet_done || s->packet_loss) {
s->packet_done = 0;
s->buf_bit_size = buf_size << 3;
/** sanity check for the buffer length */
if (buf_size < wfx->blockalign) {
av_log(wfx, AV_LOG_ERROR, "Input packet too small (%d < %d)\n",
buf_size, wfx->block_align);
return AVERROR_INVALIDDATA;
}
buf_size = wfx->blockalign;
/** parse packet header */
init_get_bits(gb, buf, s->buf_bit_size);
packet_sequence_number = get_bits(gb, 4);
skip_bits(gb, 2);
/** get number of bits that need to be added to the previous frame */
num_bits_prev_frame = get_bits(gb, s->log2_frame_size);
DEBUGF("packet[%d]: nbpf %x\n", s->frame_num,
num_bits_prev_frame);
/** check for packet loss */
if (!s->packet_loss &&
((s->packet_sequence_number + 1) & 0xF) != packet_sequence_number) {
s->packet_loss = 1;
DEBUGF("Packet loss detected! seq %x vs %x\n",
s->packet_sequence_number, packet_sequence_number);
}
s->packet_sequence_number = packet_sequence_number;
if (num_bits_prev_frame > 0) {
int remaining_packet_bits = s->buf_bit_size - get_bits_count(gb);
if (num_bits_prev_frame >= remaining_packet_bits) {
num_bits_prev_frame = remaining_packet_bits;
s->packet_done = 1;
}
/** append the previous frame data to the remaining data from the
previous packet to create a full frame */
save_bits(s, gb, num_bits_prev_frame, 1);
DEBUGF("accumulated %x bits of frame data\n",
s->num_saved_bits - s->frame_offset);
/** decode the cross packet frame if it is valid */
if (!s->packet_loss)
decode_frame(s);
} else if (s->num_saved_bits - s->frame_offset) {
DEBUGF("ignoring %x previously saved bits\n",
s->num_saved_bits - s->frame_offset);
}
if (s->packet_loss) {
/** reset number of saved bits so that the decoder
does not start to decode incomplete frames in the
s->len_prefix == 0 case */
s->num_saved_bits = 0;
s->packet_loss = 0;
}
} else {
int frame_size;
s->buf_bit_size = size << 3;
init_get_bits(gb, pktdata, s->buf_bit_size);
skip_bits(gb, s->packet_offset);
if (remaining_bits(s, gb) > s->log2_frame_size &&
(frame_size = show_bits(gb, s->log2_frame_size)) &&
frame_size <= remaining_bits(s, gb)) {
save_bits(s, gb, frame_size, 0);
if (!s->packet_loss)
s->packet_done = !decode_frame(s);
} else
s->packet_done = 1;
}
if (remaining_bits(s, gb) < 0) {
av_log(wctx, AV_LOG_ERROR, "Overread %d\n", -remaining_bits(s, gb));
s->packet_loss = 1;
}
if (s->packet_done && !s->packet_loss &&
remaining_bits(s, gb) > 0) {
/** save the rest of the data so that it can be decoded
with the next packet */
save_bits(s, gb, remaining_bits(s, gb), 0);
}
dec[0] = s->channel[0].out;
dec[1] = s->channel[1].out;
*data_size = s->samples;
s->packet_offset = get_bits_count(gb) & 7;
s->frame_num++;
return (s->packet_loss) ? AVERROR_INVALIDDATA : get_bits_count(gb) >> 3;
}
#if 0
/**
*@brief wmapro decoder
*/
AVCodec wmapro_decoder = {
"wmapro",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_WMAPRO,
sizeof(WMAProDecodeCtx),
decode_init,
NULL,
decode_end,
decode_packet,
.capabilities = CODEC_CAP_SUBFRAMES,
.flush= flush,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"),
};
#endif
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