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ffmpeg/libavcodec/wmalosslessdec.c
Michael Niedermayer b429440d85 Merge remote-tracking branch 'shariman/wmall' 
* shariman/wmall:
  call revert_cdlms()
  Fix some loop conditions to prevent overreads
  Initialize pred in lms_predict()

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-11-22 21:32:11 +01:00

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/*
* Wmall compatible decoder
* Copyright (c) 2007 Baptiste Coudurier, Benjamin Larsson, Ulion
* Copyright (c) 2008 - 2011 Sascha Sommer, Benjamin Larsson
* Copyright (c) 2011 Andreas Öman
*
* 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 wmall decoder implementation
* Wmall 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 wmall 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 wmall 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 wmall 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 wmall 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 "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "put_bits.h"
#include "dsputil.h"
#include "wma.h"
/** current decoder limitations */
#define WMALL_MAX_CHANNELS 8 ///< max number of handled channels
#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 WMALL_BLOCK_MIN_BITS 6 ///< log2 of min block size
#define WMALL_BLOCK_MAX_BITS 12 ///< log2 of max block size
#define WMALL_BLOCK_MAX_SIZE (1 << WMALL_BLOCK_MAX_BITS) ///< maximum block size
#define WMALL_BLOCK_SIZES (WMALL_BLOCK_MAX_BITS - WMALL_BLOCK_MIN_BITS + 1) ///< possible block sizes
#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 float sin64[33]; ///< sinus table for decorrelation
/**
* @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
float* coeffs; ///< pointer to the subframe decode buffer
uint16_t num_vec_coeffs; ///< number of vector coded coefficients
DECLARE_ALIGNED(16, float, out)[WMALL_BLOCK_MAX_SIZE + WMALL_BLOCK_MAX_SIZE / 2]; ///< output buffer
int transient_counter; ///< number of transient samples from the beginning of transient zone
} WmallChannelCtx;
/**
* @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[WMALL_MAX_CHANNELS*WMALL_MAX_CHANNELS];
float* channel_data[WMALL_MAX_CHANNELS]; ///< transformation coefficients
} WmallChannelGrp;
/**
* @brief main decoder context
*/
typedef struct WmallDecodeCtx {
/* generic decoder variables */
AVCodecContext* avctx; ///< codec context for av_log
DSPContext dsp; ///< accelerated DSP functions
uint8_t frame_data[MAX_FRAMESIZE +
FF_INPUT_BUFFER_PADDING_SIZE];///< compressed frame data
PutBitContext pb; ///< context for filling the frame_data buffer
FFTContext mdct_ctx[WMALL_BLOCK_SIZES]; ///< MDCT context per block size
DECLARE_ALIGNED(16, float, tmp)[WMALL_BLOCK_MAX_SIZE]; ///< IMDCT output buffer
float* windows[WMALL_BLOCK_SIZES]; ///< windows for the different block sizes
/* 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])
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[WMALL_BLOCK_SIZES]; ///< scale factor bands per block size
int16_t sfb_offsets[WMALL_BLOCK_SIZES][MAX_BANDS]; ///< scale factor band offsets (multiples of 4)
int8_t sf_offsets[WMALL_BLOCK_SIZES][WMALL_BLOCK_SIZES][MAX_BANDS]; ///< scale factor resample matrix
int16_t subwoofer_cutoffs[WMALL_BLOCK_SIZES]; ///< subwoofer cutoff values
/* packet decode state */
GetBitContext pgb; ///< bitstream reader context for the packet
int next_packet_start; ///< start offset of the next wma packet in the demuxer 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 (not used for decoding)
GetBitContext gb; ///< bitstream reader context
int buf_bit_size; ///< buffer size in bits
float* samples; ///< current samplebuffer pointer
float* 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[WMALL_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
WmallChannelGrp chgroup[WMALL_MAX_CHANNELS]; ///< channel group information
WmallChannelCtx channel[WMALL_MAX_CHANNELS]; ///< per channel data
// WMA lossless
uint8_t do_arith_coding;
uint8_t do_ac_filter;
uint8_t do_inter_ch_decorr;
uint8_t do_mclms;
uint8_t do_lpc;
int8_t acfilter_order;
int8_t acfilter_scaling;
int acfilter_coeffs[16];
int8_t mclms_order;
int8_t mclms_scaling;
int16_t mclms_coeffs[128];
int16_t mclms_coeffs_cur[4];
int mclms_prevvalues[64]; // FIXME: should be 32-bit / 16-bit depending on bit-depth
int16_t mclms_updates[64];
int mclms_recent;
int movave_scaling;
int quant_stepsize;
struct {
int order;
int scaling;
int coefsend;
int bitsend;
int16_t coefs[256];
int lms_prevvalues[512]; // FIXME: see above
int16_t lms_updates[512]; // and here too
int recent;
} cdlms[2][9]; /* XXX: Here, 2 is the max. no. of channels allowed,
9 is the maximum no. of filters per channel.
Question is, why 2 if WMALL_MAX_CHANNELS == 8 */
int cdlms_ttl[2];
int bV3RTM;
int is_channel_coded[2]; // XXX: same question as above applies here too (and below)
int update_speed[2];
int transient[2];
int transient_pos[2];
int seekable_tile;
int ave_sum[2];
int channel_residues[2][2048];
int lpc_coefs[2][40];
int lpc_order;
int lpc_scaling;
int lpc_intbits;
int channel_coeffs[2][2048];
} WmallDecodeCtx;
#undef dprintf
#define dprintf(pctx, ...) av_log(pctx, AV_LOG_DEBUG, __VA_ARGS__)
static int num_logged_tiles = 0;
/**
*@brief helper function to print the most important members of the context
*@param s context
*/
static void av_cold dump_context(WmallDecodeCtx *s)
{
#define PRINT(a, b) av_log(s->avctx, AV_LOG_DEBUG, " %s = %d\n", a, b);
#define PRINT_HEX(a, b) av_log(s->avctx, AV_LOG_DEBUG, " %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);
}
static int dump_int_buffer(int *buffer, int length, int delimiter)
{
int i;
for (i=0 ; i<length ; i++) {
if (!(i%delimiter))
av_log(0, 0, "\n[%d] ", i);
av_log(0, 0, "%d, ", buffer[i]);
}
av_log(0, 0, "\n");
}
/**
*@brief Uninitialize the decoder and free all resources.
*@param avctx codec context
*@return 0 on success, < 0 otherwise
*/
static av_cold int decode_end(AVCodecContext *avctx)
{
WmallDecodeCtx *s = avctx->priv_data;
int i;
for (i = 0; i < WMALL_BLOCK_SIZES; i++)
ff_mdct_end(&s->mdct_ctx[i]);
return 0;
}
/**
*@brief Initialize the decoder.
*@param avctx codec context
*@return 0 on success, -1 otherwise
*/
static av_cold int decode_init(AVCodecContext *avctx)
{
WmallDecodeCtx *s = avctx->priv_data;
uint8_t *edata_ptr = avctx->extradata;
unsigned int channel_mask;
int i;
int log2_max_num_subframes;
int num_possible_block_sizes;
s->avctx = avctx;
dsputil_init(&s->dsp, avctx);
init_put_bits(&s->pb, s->frame_data, MAX_FRAMESIZE);
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (avctx->extradata_size >= 18) {
s->decode_flags = AV_RL16(edata_ptr+14);
channel_mask = AV_RL32(edata_ptr+2);
s->bits_per_sample = AV_RL16(edata_ptr);
/** dump the extradata */
for (i = 0; i < avctx->extradata_size; i++)
dprintf(avctx, "[%x] ", avctx->extradata[i]);
dprintf(avctx, "\n");
} else {
av_log_ask_for_sample(avctx, "Unknown extradata size\n");
return AVERROR_INVALIDDATA;
}
/** generic init */
s->log2_frame_size = av_log2(avctx->block_align) + 4;
/** frame info */
s->skip_frame = 1; /* skip first frame */
s->packet_loss = 1;
s->len_prefix = (s->decode_flags & 0x40);
/** get frame len */
s->samples_per_frame = 1 << ff_wma_get_frame_len_bits(avctx->sample_rate,
3, s->decode_flags);
/** init previous block len */
for (i = 0; i < avctx->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;
s->max_subframe_len_bit = 0;
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);
s->bV3RTM = s->decode_flags & 0x100;
if (s->max_num_subframes > MAX_SUBFRAMES) {
av_log(avctx, AV_LOG_ERROR, "invalid number of subframes %i\n",
s->max_num_subframes);
return AVERROR_INVALIDDATA;
}
s->num_channels = avctx->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) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels %d\n", s->num_channels);
return AVERROR_INVALIDDATA;
} else if (s->num_channels > WMALL_MAX_CHANNELS) {
av_log_ask_for_sample(avctx, "unsupported number of channels\n");
return AVERROR_PATCHWELCOME;
}
avctx->channel_layout = channel_mask;
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(WmallDecodeCtx *s, int offset)
{
int frame_len_ratio;
int subframe_len, 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;
len = av_log2(s->max_num_subframes - 1) + 1;
frame_len_ratio = get_bits(&s->gb, len);
subframe_len = s->min_samples_per_subframe * (frame_len_ratio + 1);
/** sanity check the length */
if (subframe_len < s->min_samples_per_subframe ||
subframe_len > s->samples_per_frame) {
av_log(s->avctx, AV_LOG_ERROR, "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(WmallDecodeCtx *s)
{
uint16_t num_samples[WMALL_MAX_CHANNELS]; /**< sum of samples for all currently known subframes of a channel */
uint8_t contains_subframe[WMALL_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 subfra2me 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++) {
WmallChannelCtx* chan = &s->channel[c];
if (contains_subframe[c]) {
if (chan->num_subframes >= MAX_SUBFRAMES) {
av_log(s->avctx, AV_LOG_ERROR,
"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) {
av_log(s->avctx, AV_LOG_ERROR, "broken frame: "
"channel len(%d) > samples_per_frame(%d)\n",
num_samples[c], s->samples_per_frame);
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++) {
s->channel[c].subframe_offset[i] = offset;
offset += s->channel[c].subframe_len[i];
}
}
return 0;
}
static int my_log2(unsigned int i)
{
unsigned int iLog2 = 0;
while ((i >> iLog2) > 1)
iLog2++;
return iLog2;
}
/**
*
*/
static void decode_ac_filter(WmallDecodeCtx *s)
{
int i;
s->acfilter_order = get_bits(&s->gb, 4) + 1;
s->acfilter_scaling = get_bits(&s->gb, 4);
for(i = 0; i < s->acfilter_order; i++) {
s->acfilter_coeffs[i] = get_bits(&s->gb, s->acfilter_scaling) + 1;
}
}
/**
*
*/
static void decode_mclms(WmallDecodeCtx *s)
{
s->mclms_order = (get_bits(&s->gb, 4) + 1) * 2;
s->mclms_scaling = get_bits(&s->gb, 4);
if(get_bits1(&s->gb)) {
// mclms_send_coef
int i;
int send_coef_bits;
int cbits = av_log2(s->mclms_scaling + 1);
assert(cbits == my_log2(s->mclms_scaling + 1));
if(1 << cbits < s->mclms_scaling + 1)
cbits++;
send_coef_bits = (cbits ? get_bits(&s->gb, cbits) : 0) + 2;
for(i = 0; i < s->mclms_order * s->num_channels * s->num_channels; i++) {
s->mclms_coeffs[i] = get_bits(&s->gb, send_coef_bits);
}
for(i = 0; i < s->num_channels; i++) {
int c;
for(c = 0; c < i; c++) {
s->mclms_coeffs_cur[i * s->num_channels + c] = get_bits(&s->gb, send_coef_bits);
}
}
}
}
/**
*
*/
static void decode_cdlms(WmallDecodeCtx *s)
{
int c, i;
int cdlms_send_coef = get_bits1(&s->gb);
for(c = 0; c < s->num_channels; c++) {
s->cdlms_ttl[c] = get_bits(&s->gb, 3) + 1;
for(i = 0; i < s->cdlms_ttl[c]; i++) {
s->cdlms[c][i].order = (get_bits(&s->gb, 7) + 1) * 8;
}
for(i = 0; i < s->cdlms_ttl[c]; i++) {
s->cdlms[c][i].scaling = get_bits(&s->gb, 4);
}
if(cdlms_send_coef) {
for(i = 0; i < s->cdlms_ttl[c]; i++) {
int cbits, shift_l, shift_r, j;
cbits = av_log2(s->cdlms[c][i].order);
if(1 << cbits < s->cdlms[c][i].order)
cbits++;
s->cdlms[c][i].coefsend = get_bits(&s->gb, cbits) + 1;
cbits = av_log2(s->cdlms[c][i].scaling + 1);
if(1 << cbits < s->cdlms[c][i].scaling + 1)
cbits++;
s->cdlms[c][i].bitsend = get_bits(&s->gb, cbits) + 2;
shift_l = 32 - s->cdlms[c][i].bitsend;
shift_r = 32 - 2 - s->cdlms[c][i].scaling;
for(j = 0; j < s->cdlms[c][i].coefsend; j++) {
s->cdlms[c][i].coefs[j] =
(get_bits(&s->gb, s->cdlms[c][i].bitsend) << shift_l) >> shift_r;
}
}
}
}
}
/**
*
*/
static int decode_channel_residues(WmallDecodeCtx *s, int ch, int tile_size)
{
int i = 0;
unsigned int ave_mean;
s->transient[ch] = get_bits1(&s->gb);
if(s->transient[ch]) {
s->transient_pos[ch] = get_bits(&s->gb, av_log2(tile_size));
if (s->transient_pos[ch])
s->transient[ch] = 0;
s->channel[ch].transient_counter =
FFMAX(s->channel[ch].transient_counter, s->samples_per_frame / 2);
} else if (s->channel[ch].transient_counter)
s->transient[ch] = 1;
if(s->seekable_tile) {
ave_mean = get_bits(&s->gb, s->bits_per_sample);
s->ave_sum[ch] = ave_mean << (s->movave_scaling + 1);
// s->ave_sum[ch] *= 2;
}
if(s->seekable_tile) {
if(s->do_inter_ch_decorr)
s->channel_residues[ch][0] = get_sbits(&s->gb, s->bits_per_sample + 1);
else
s->channel_residues[ch][0] = get_sbits(&s->gb, s->bits_per_sample);
i++;
}
//av_log(0, 0, "%8d: ", num_logged_tiles++);
for(; i < tile_size; i++) {
int quo = 0, rem, rem_bits, residue;
while(get_bits1(&s->gb))
quo++;
if(quo >= 32)
quo += get_bits_long(&s->gb, get_bits(&s->gb, 5) + 1);
ave_mean = (s->ave_sum[ch] + (1 << s->movave_scaling)) >> (s->movave_scaling + 1);
rem_bits = av_ceil_log2(ave_mean);
rem = rem_bits ? get_bits(&s->gb, rem_bits) : 0;
residue = (quo << rem_bits) + rem;
s->ave_sum[ch] = residue + s->ave_sum[ch] - (s->ave_sum[ch] >> s->movave_scaling);
if(residue & 1)
residue = -(residue >> 1) - 1;
else
residue = residue >> 1;
s->channel_residues[ch][i] = residue;
/*if (num_logged_tiles < 1)
av_log(0, 0, "%4d ", residue); */
}
dump_int_buffer(s->channel_residues[ch], tile_size, 16);
return 0;
}
/**
*
*/
static void
decode_lpc(WmallDecodeCtx *s)
{
int ch, i, cbits;
s->lpc_order = get_bits(&s->gb, 5) + 1;
s->lpc_scaling = get_bits(&s->gb, 4);
s->lpc_intbits = get_bits(&s->gb, 3) + 1;
cbits = s->lpc_scaling + s->lpc_intbits;
for(ch = 0; ch < s->num_channels; ch++) {
for(i = 0; i < s->lpc_order; i++) {
s->lpc_coefs[ch][i] = get_sbits(&s->gb, cbits);
}
}
}
static void clear_codec_buffers(WmallDecodeCtx *s)
{
int ich, ilms;
memset(s->acfilter_coeffs, 0, 16 * sizeof(int));
memset(s->lpc_coefs , 0, 40 * 2 * sizeof(int));
memset(s->mclms_coeffs , 0, 128 * sizeof(int16_t));
memset(s->mclms_coeffs_cur, 0, 4 * sizeof(int16_t));
memset(s->mclms_prevvalues, 0, 64 * sizeof(int));
memset(s->mclms_updates , 0, 64 * sizeof(int16_t));
for (ich = 0; ich < s->num_channels; ich++) {
for (ilms = 0; ilms < s->cdlms_ttl[ich]; ilms++) {
memset(s->cdlms[ich][ilms].coefs , 0, 256 * sizeof(int16_t));
memset(s->cdlms[ich][ilms].lms_prevvalues, 0, 512 * sizeof(int));
memset(s->cdlms[ich][ilms].lms_updates , 0, 512 * sizeof(int16_t));
}
s->ave_sum[ich] = 0;
}
}
/**
*@brief Resets filter parameters and transient area at new seekable tile
*/
static void reset_codec(WmallDecodeCtx *s)
{
int ich, ilms;
s->mclms_recent = s->mclms_order * s->num_channels;
for (ich = 0; ich < s->num_channels; ich++) {
for (ilms = 0; ilms < s->cdlms_ttl[ich]; ilms++)
s->cdlms[ich][ilms].recent = s->cdlms[ich][ilms].order;
/* first sample of a seekable subframe is considered as the starting of
a transient area which is samples_per_frame samples long */
s->channel[ich].transient_counter = s->samples_per_frame;
s->transient[ich] = 1;
}
}
static int lms_predict(WmallDecodeCtx *s, int ich, int ilms)
{
int32_t pred = 0, icoef;
int recent = s->cdlms[ich][ilms].recent;
for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
pred += s->cdlms[ich][ilms].coefs[icoef] *
s->cdlms[ich][ilms].lms_prevvalues[icoef + recent];
pred += (1 << (s->cdlms[ich][ilms].scaling - 1));
/* XXX: Table 29 has:
iPred >= cdlms[iCh][ilms].scaling;
seems to me like a missing > */
pred >>= s->cdlms[ich][ilms].scaling;
return pred;
}
static void lms_update(WmallDecodeCtx *s, int ich, int ilms, int32_t input, int32_t pred)
{
int icoef;
int recent = s->cdlms[ich][ilms].recent;
int range = 1 << (s->bits_per_sample - 1);
int bps = s->bits_per_sample > 16 ? 4 : 2; // bytes per sample
if (input > pred) {
for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
s->cdlms[ich][ilms].coefs[icoef] +=
s->cdlms[ich][ilms].lms_updates[icoef + recent];
} else {
for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
s->cdlms[ich][ilms].coefs[icoef] -=
s->cdlms[ich][ilms].lms_updates[icoef]; // XXX: [icoef + recent] ?
}
s->cdlms[ich][ilms].recent--;
s->cdlms[ich][ilms].lms_prevvalues[recent] = av_clip(input, -range, range - 1);
if (input > pred)
s->cdlms[ich][ilms].lms_updates[recent] = s->update_speed[ich];
else if (input < pred)
s->cdlms[ich][ilms].lms_updates[recent] = -s->update_speed[ich];
/* XXX: spec says:
cdlms[iCh][ilms].updates[iRecent + cdlms[iCh][ilms].order >> 4] >>= 2;
lms_updates[iCh][ilms][iRecent + cdlms[iCh][ilms].order >> 3] >>= 1;
Questions is - are cdlms[iCh][ilms].updates[] and lms_updates[][][] two
seperate buffers? Here I've assumed that the two are same which makes
more sense to me.
*/
s->cdlms[ich][ilms].lms_updates[recent + s->cdlms[ich][ilms].order >> 4] >>= 2;
s->cdlms[ich][ilms].lms_updates[recent + s->cdlms[ich][ilms].order >> 3] >>= 1;
/* XXX: recent + (s->cdlms[ich][ilms].order >> 4) ? */
if (s->cdlms[ich][ilms].recent == 0) {
/* XXX: This memcpy()s will probably fail if a fixed 32-bit buffer is used.
follow kshishkov's suggestion of using a union. */
memcpy(s->cdlms[ich][ilms].lms_prevvalues + s->cdlms[ich][ilms].order,
s->cdlms[ich][ilms].lms_prevvalues,
bps * s->cdlms[ich][ilms].order);
memcpy(s->cdlms[ich][ilms].lms_updates + s->cdlms[ich][ilms].order,
s->cdlms[ich][ilms].lms_updates,
bps * s->cdlms[ich][ilms].order);
s->cdlms[ich][ilms].recent = s->cdlms[ich][ilms].order;
}
}
static void use_high_update_speed(WmallDecodeCtx *s, int ich)
{
int ilms, recent, icoef;
s->update_speed[ich] = 16;
for (ilms = s->cdlms_ttl[ich] - 1; ilms >= 0; ilms--) {
recent = s->cdlms[ich][ilms].recent;
if (s->bV3RTM) {
for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
s->cdlms[ich][ilms].lms_updates[icoef + recent] *= 2;
} else {
for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
s->cdlms[ich][ilms].lms_updates[icoef] *= 2;
}
}
}
static void use_normal_update_speed(WmallDecodeCtx *s, int ich)
{
int ilms, recent, icoef;
s->update_speed[ich] = 8;
for (ilms = s->cdlms_ttl[ich] - 1; ilms >= 0; ilms--) {
recent = s->cdlms[ich][ilms].recent;
if (s->bV3RTM) {
for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
s->cdlms[ich][ilms].lms_updates[icoef + recent] /= 2;
} else {
for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++)
s->cdlms[ich][ilms].lms_updates[icoef] /= 2;
}
}
}
static void revert_cdlms(WmallDecodeCtx *s, int tile_size)
{
int icoef, ich;
int32_t pred, channel_coeff;
int ilms, num_lms;
for (ich = 0; ich < s->num_channels; ich++) {
if (!s->is_channel_coded[ich])
continue;
for (icoef = 0; icoef < tile_size; icoef++) {
num_lms = s->cdlms_ttl[ich];
channel_coeff = s->channel_residues[ich][icoef];
if (icoef == s->transient_pos[ich]) {
s->transient[ich] = 1;
use_high_update_speed(s, ich);
}
for (ilms = num_lms - 1; ilms >= 0; ilms--) {
pred = lms_predict(s, ich, ilms);
channel_coeff += pred;
lms_update(s, ich, ilms, channel_coeff, pred);
}
if (s->transient[ich]) {
--s->channel[ich].transient_counter;
if(!s->channel[ich].transient_counter)
use_normal_update_speed(s, ich);
}
s->channel_coeffs[ich][icoef] = channel_coeff;
}
}
}
/**
*@brief Decode a single subframe (block).
*@param s codec context
*@return 0 on success, < 0 when decoding failed
*/
static int decode_subframe(WmallDecodeCtx *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 rawpcm_tile;
int padding_zeroes;
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];
}
}
/** 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;
s->seekable_tile = get_bits1(&s->gb);
if(s->seekable_tile) {
clear_codec_buffers(s);
s->do_arith_coding = get_bits1(&s->gb);
if(s->do_arith_coding) {
dprintf(s->avctx, "do_arith_coding == 1");
abort();
}
s->do_ac_filter = get_bits1(&s->gb);
s->do_inter_ch_decorr = get_bits1(&s->gb);
s->do_mclms = get_bits1(&s->gb);
if(s->do_ac_filter)
decode_ac_filter(s);
if(s->do_mclms)
decode_mclms(s);
decode_cdlms(s);
s->movave_scaling = get_bits(&s->gb, 3);
s->quant_stepsize = get_bits(&s->gb, 8) + 1;
reset_codec(s);
}
rawpcm_tile = get_bits1(&s->gb);
for(i = 0; i < s->num_channels; i++) {
s->is_channel_coded[i] = 1;
}
if(!rawpcm_tile) {
for(i = 0; i < s->num_channels; i++) {
s->is_channel_coded[i] = get_bits1(&s->gb);
}
if(s->bV3RTM) {
// LPC
s->do_lpc = get_bits1(&s->gb);
if(s->do_lpc) {
decode_lpc(s);
}
} else {
s->do_lpc = 0;
}
}
if(get_bits1(&s->gb)) {
padding_zeroes = get_bits(&s->gb, 5);
} else {
padding_zeroes = 0;
}
if(rawpcm_tile) {
int bits = s->bits_per_sample - padding_zeroes;
int j;
dprintf(s->avctx, "RAWPCM %d bits per sample. total %d bits, remain=%d\n", bits,
bits * s->num_channels * subframe_len, get_bits_count(&s->gb));
for(i = 0; i < s->num_channels; i++) {
for(j = 0; j < subframe_len; j++) {
s->channel_coeffs[i][j] = get_sbits(&s->gb, bits);
// dprintf(s->avctx, "PCM[%d][%d] = 0x%04x\n", i, j, s->channel_coeffs[i][j]);
}
}
} else {
for(i = 0; i < s->num_channels; i++)
if(s->is_channel_coded[i])
decode_channel_residues(s, i, subframe_len);
}
revert_cdlms(s, subframe_len);
/** 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) {
av_log(s->avctx, AV_LOG_ERROR, "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(WmallDecodeCtx *s)
{
GetBitContext* gb = &s->gb;
int more_frames = 0;
int len = 0;
int i;
/** check for potential output buffer overflow */
if (s->num_channels * s->samples_per_frame > s->samples_end - s->samples) {
/** return an error if no frame could be decoded at all */
av_log(s->avctx, AV_LOG_ERROR,
"not enough space for the output samples\n");
s->packet_loss = 1;
return 0;
}
/** get frame length */
if (s->len_prefix)
len = get_bits(gb, s->log2_frame_size);
/** decode tile information */
if (decode_tilehdr(s)) {
s->packet_loss = 1;
return 0;
}
/** read drc info */
if (s->dynamic_range_compression) {
s->drc_gain = get_bits(gb, 8);
}
/** 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)) {
int skip;
/** usually true for the first frame */
if (get_bits1(gb)) {
skip = get_bits(gb, av_log2(s->samples_per_frame * 2));
dprintf(s->avctx, "start skip: %i\n", skip);
}
/** sometimes true for the last frame */
if (get_bits1(gb)) {
skip = get_bits(gb, av_log2(s->samples_per_frame * 2));
dprintf(s->avctx, "end skip: %i\n", skip);
}
}
/** 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;
}
}
dprintf(s->avctx, "Frame done\n");
if (s->skip_frame) {
s->skip_frame = 0;
} else
s->samples += s->num_channels * s->samples_per_frame;
if (s->len_prefix) {
if (len != (get_bits_count(gb) - s->frame_offset) + 2) {
/** FIXME: not sure if this is always an error */
av_log(s->avctx, AV_LOG_ERROR,
"frame[%i] would have to skip %i bits\n", 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);
} else {
/*
while (get_bits_count(gb) < s->num_saved_bits && get_bits1(gb) == 0) {
dprintf(s->avctx, "skip1\n");
}
*/
}
/** 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(WmallDecodeCtx *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(WmallDecodeCtx *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 = (s->num_saved_bits + len + 8) >> 3;
if (len <= 0 || buflen > MAX_FRAMESIZE) {
av_log_ask_for_sample(s->avctx, "input buffer too small\n");
s->packet_loss = 1;
return;
}
s->num_saved_bits += len;
if (!append) {
avpriv_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;
avpriv_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
*/
static int decode_packet(AVCodecContext *avctx,
void *data, int *data_size, AVPacket* avpkt)
{
WmallDecodeCtx *s = avctx->priv_data;
GetBitContext* gb = &s->pgb;
const uint8_t* buf = avpkt->data;
int buf_size = avpkt->size;
int num_bits_prev_frame;
int packet_sequence_number;
s->samples = data;
s->samples_end = (float*)((int8_t*)data + *data_size);
*data_size = 0;
if (s->packet_done || s->packet_loss) {
s->packet_done = 0;
/** sanity check for the buffer length */
if (buf_size < avctx->block_align)
return 0;
s->next_packet_start = buf_size - avctx->block_align;
buf_size = avctx->block_align;
s->buf_bit_size = buf_size << 3;
/** parse packet header */
init_get_bits(gb, buf, s->buf_bit_size);
packet_sequence_number = get_bits(gb, 4);
int seekable_frame_in_packet = get_bits1(gb);
int spliced_packet = get_bits1(gb);
/** get number of bits that need to be added to the previous frame */
num_bits_prev_frame = get_bits(gb, s->log2_frame_size);
/** check for packet loss */
if (!s->packet_loss &&
((s->packet_sequence_number + 1) & 0xF) != packet_sequence_number) {
s->packet_loss = 1;
av_log(avctx, AV_LOG_ERROR, "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);
/** 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) {
dprintf(avctx, "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 = (avpkt->size - s->next_packet_start) << 3;
init_get_bits(gb, avpkt->data, s->buf_bit_size);
skip_bits(gb, s->packet_offset);
if (s->len_prefix && 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);
s->packet_done = !decode_frame(s);
} else if (!s->len_prefix
&& s->num_saved_bits > get_bits_count(&s->gb)) {
/** when the frames do not have a length prefix, we don't know
the compressed length of the individual frames
however, we know what part of a new packet belongs to the
previous frame
therefore we save the incoming packet first, then we append
the "previous frame" data from the next packet so that
we get a buffer that only contains full frames */
s->packet_done = !decode_frame(s);
} else {
s->packet_done = 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);
}
*data_size = 0; // (int8_t *)s->samples - (int8_t *)data;
s->packet_offset = get_bits_count(gb) & 7;
return (s->packet_loss) ? AVERROR_INVALIDDATA : get_bits_count(gb) >> 3;
}
/**
*@brief Clear decoder buffers (for seeking).
*@param avctx codec context
*/
static void flush(AVCodecContext *avctx)
{
WmallDecodeCtx *s = avctx->priv_data;
int i;
/** reset output buffer as a part of it is used during the windowing of a
new frame */
for (i = 0; i < s->num_channels; i++)
memset(s->channel[i].out, 0, s->samples_per_frame *
sizeof(*s->channel[i].out));
s->packet_loss = 1;
}
/**
*@brief wmall decoder
*/
AVCodec ff_wmalossless_decoder = {
"wmalossless",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_WMALOSSLESS,
sizeof(WmallDecodeCtx),
decode_init,
NULL,
decode_end,
decode_packet,
.capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_EXPERIMENTAL,
.flush= flush,
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Lossless"),
};
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