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FFmpeg/libavcodec/wmaprodec.c
Sascha Sommer 75e4efd2a7 add support for blocksize 64 
fixes issue 2537 and 2538

Originally committed as revision 26328 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-14 16:36:57 +00:00

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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
* @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 "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "put_bits.h"
#include "wmaprodata.h"
#include "dsputil.h"
#include "wma.h"
/** current decoder limitations */
#define WMAPRO_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 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 - WMAPRO_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 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
/**
* @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
DECLARE_ALIGNED(16, float, out)[WMAPRO_BLOCK_MAX_SIZE + WMAPRO_BLOCK_MAX_SIZE / 2]; ///< 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];
float* channel_data[WMAPRO_MAX_CHANNELS]; ///< transformation coefficients
} WMAProChannelGrp;
/**
* @brief main decoder context
*/
typedef struct WMAProDecodeCtx {
/* 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[WMAPRO_BLOCK_SIZES]; ///< MDCT context per block size
DECLARE_ALIGNED(16, float, tmp)[WMAPRO_BLOCK_MAX_SIZE]; ///< IMDCT output buffer
float* windows[WMAPRO_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[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 (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[WMAPRO_MAX_CHANNELS];
int8_t num_bands; ///< number of scale factor bands
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;
/**
*@brief helper function to print the most important members of the context
*@param s context
*/
static void av_cold dump_context(WMAProDecodeCtx *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);
}
/**
*@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)
{
WMAProDecodeCtx *s = avctx->priv_data;
int i;
for (i = 0; i < WMAPRO_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)
{
WMAProDecodeCtx *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;
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) {
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 > WMAPRO_MAX_CHANNELS) {
av_log_ask_for_sample(avctx, "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);
INIT_VLC_STATIC(&sf_rl_vlc, VLCBITS, HUFF_SCALE_RL_SIZE,
scale_rl_huffbits, 1, 1,
scale_rl_huffcodes, 4, 4, 1406);
INIT_VLC_STATIC(&coef_vlc[0], VLCBITS, HUFF_COEF0_SIZE,
coef0_huffbits, 1, 1,
coef0_huffcodes, 4, 4, 2108);
INIT_VLC_STATIC(&coef_vlc[1], VLCBITS, HUFF_COEF1_SIZE,
coef1_huffbits, 1, 1,
coef1_huffcodes, 4, 4, 3912);
INIT_VLC_STATIC(&vec4_vlc, VLCBITS, HUFF_VEC4_SIZE,
vec4_huffbits, 1, 1,
vec4_huffcodes, 2, 2, 604);
INIT_VLC_STATIC(&vec2_vlc, VLCBITS, HUFF_VEC2_SIZE,
vec2_huffbits, 1, 1,
vec2_huffcodes, 2, 2, 562);
INIT_VLC_STATIC(&vec1_vlc, VLCBITS, HUFF_VEC1_SIZE,
vec1_huffbits, 1, 1,
vec1_huffcodes, 2, 2, 562);
/** 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])
/ s->avctx->sample_rate + 2;
offset &= ~3;
if (offset > s->sfb_offsets[i][band - 1])
s->sfb_offsets[i][band++] = offset;
}
s->sfb_offsets[i][band - 1] = subframe_len;
s->num_sfb[i] = band - 1;
}
/** 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;
}
}
}
/** init MDCT, FIXME: only init needed sizes */
for (i = 0; i < WMAPRO_BLOCK_SIZES; i++)
ff_mdct_init(&s->mdct_ctx[i], WMAPRO_BLOCK_MIN_BITS+1+i, 1,
1.0 / (1 << (WMAPRO_BLOCK_MIN_BITS + i - 1))
/ (1 << (s->bits_per_sample - 1)));
/** init MDCT windows: simple sinus window */
for (i = 0; i < WMAPRO_BLOCK_SIZES; i++) {
const int win_idx = WMAPRO_BLOCK_MAX_BITS - i;
ff_init_ff_sine_windows(win_idx);
s->windows[WMAPRO_BLOCK_SIZES - i - 1] = ff_sine_windows[win_idx];
}
/** 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 * (s->avctx->sample_rate >> 1) - 1)
/ s->avctx->sample_rate;
s->subwoofer_cutoffs[i] = av_clip(cutoff, 4, block_size);
}
/** calculate sine values for the decorrelation matrix */
for (i = 0; i < 33; i++)
sin64[i] = sin(i*M_PI / 64.0);
if (avctx->debug & FF_DEBUG_BITSTREAM)
dump_context(s);
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(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) {
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(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) {
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 > 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++) {
dprintf(s->avctx, "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;
}
/**
*@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;
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];
int n = rotation_offset[offset + x];
float sinv;
float cosv;
if (n < 32) {
sinv = sin64[n];
cosv = sin64[32 - n];
} else {
sinv = sin64[64 - n];
cosv = -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);
}
}
offset += i;
}
}
/**
*@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)) {
av_log_ask_for_sample(s->avctx,
"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];
float** 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)) {
av_log_ask_for_sample(s->avctx,
"unsupported channel transform type\n");
}
} else {
chgroup->transform = 1;
if (s->num_channels == 2) {
chgroup->decorrelation_matrix[0] = 1.0;
chgroup->decorrelation_matrix[1] = -1.0;
chgroup->decorrelation_matrix[2] = 1.0;
chgroup->decorrelation_matrix[3] = 1.0;
} else {
/** cos(pi/4) */
chgroup->decorrelation_matrix[0] = 0.70703125;
chgroup->decorrelation_matrix[1] = -0.70703125;
chgroup->decorrelation_matrix[2] = 0.70703125;
chgroup->decorrelation_matrix[3] = 0.70703125;
}
}
} else if (chgroup->num_channels > 2) {
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));
}
}
}
}
/** 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)
{
/* Integers 0..15 as single-precision floats. The table saves a
costly int to float conversion, and storing the values as
integers allows fast sign-flipping. */
static const int fval_tab[16] = {
0x00000000, 0x3f800000, 0x40000000, 0x40400000,
0x40800000, 0x40a00000, 0x40c00000, 0x40e00000,
0x41000000, 0x41100000, 0x41200000, 0x41300000,
0x41400000, 0x41500000, 0x41600000, 0x41700000,
};
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 float* level;
dprintf(s->avctx, "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 (!rl_mode && cur_coeff + 3 < s->subframe_len) {
int 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);
((float*)vals)[i ] = v0;
((float*)vals)[i+1] = v1;
} else {
vals[i] = fval_tab[symbol_to_vec2[idx] >> 4 ];
vals[i+1] = fval_tab[symbol_to_vec2[idx] & 0xF];
}
}
} else {
vals[0] = fval_tab[ symbol_to_vec4[idx] >> 12 ];
vals[1] = fval_tab[(symbol_to_vec4[idx] >> 8) & 0xF];
vals[2] = fval_tab[(symbol_to_vec4[idx] >> 4) & 0xF];
vals[3] = fval_tab[ symbol_to_vec4[idx] & 0xF];
}
/** decode sign */
for (i = 0; i < 4; i++) {
if (vals[i]) {
int sign = get_bits1(&s->gb) - 1;
*(uint32_t*)&ci->coeffs[cur_coeff] = vals[i] ^ sign<<31;
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 (rl_mode) {
memset(&ci->coeffs[cur_coeff], 0,
sizeof(*ci->coeffs) * (s->subframe_len - cur_coeff));
if (ff_wma_run_level_decode(s->avctx, &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) {
av_log(s->avctx, AV_LOG_ERROR,
"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 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) {
float data[WMAPRO_MAX_CHANNELS];
const int num_channels = s->chgroup[i].num_channels;
float** ch_data = s->chgroup[i].channel_data;
float** 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) {
/** multiply values with the decorrelation_matrix */
for (y = sfb[0]; y < FFMIN(sfb[1], s->subframe_len); y++) {
const float* mat = s->chgroup[i].decorrelation_matrix;
const float* data_end = data + num_channels;
float* data_ptr = data;
float** ch;
for (ch = ch_data; ch < ch_end; ch++)
*data_ptr++ = (*ch)[y];
for (ch = ch_data; ch < ch_end; ch++) {
float sum = 0;
data_ptr = data;
while (data_ptr < data_end)
sum += *data_ptr++ * *mat++;
(*ch)[y] = sum;
}
}
} else if (s->num_channels == 2) {
int len = FFMIN(sfb[1], s->subframe_len) - sfb[0];
s->dsp.vector_fmul_scalar(ch_data[0] + sfb[0],
ch_data[0] + sfb[0],
181.0 / 128, len);
s->dsp.vector_fmul_scalar(ch_data[1] + sfb[0],
ch_data[1] + sfb[0],
181.0 / 128, 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];
float* window;
int winlen = s->channel[c].prev_block_len;
float* start = s->channel[c].coeffs - (winlen >> 1);
if (s->subframe_len < winlen) {
start += (winlen - s->subframe_len) >> 1;
winlen = s->subframe_len;
}
window = s->windows[av_log2(winlen) - WMAPRO_BLOCK_MIN_BITS];
winlen >>= 1;
s->dsp.vector_fmul_window(start, start, start + winlen,
window, 0, 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];
}
}
dprintf(s->avctx,
"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;
dprintf(s->avctx, "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 = get_bits(&s->gb, len) + 1;
}
if (num_fill_bits >= 0) {
if (get_bits_count(&s->gb) + num_fill_bits > s->num_saved_bits) {
av_log(s->avctx, AV_LOG_ERROR, "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)) {
av_log_ask_for_sample(s->avctx, "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;
if ((get_bits1(&s->gb))) {
/** FIXME: might change run level mode decision */
av_log_ask_for_sample(s->avctx, "unsupported quant step coding\n");
return AVERROR_INVALIDDATA;
}
/** 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) {
av_log(s->avctx, AV_LOG_DEBUG, "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;
}
dprintf(s->avctx, "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);
}
dprintf(s->avctx, "BITSTREAM: subframe length was %i\n",
get_bits_count(&s->gb) - s->subframe_offset);
if (transmit_coeffs) {
/** 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;
const float quant = pow(10.0, exp / 20.0);
int start = s->cur_sfb_offsets[b];
s->dsp.vector_fmul_scalar(s->tmp + start,
s->channel[c].coeffs + start,
quant, end - start);
}
/** apply imdct (ff_imdct_half == DCTIV with reverse) */
ff_imdct_half(&s->mdct_ctx[av_log2(subframe_len) - WMAPRO_BLOCK_MIN_BITS],
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) {
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(WMAProDecodeCtx *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);
dprintf(s->avctx, "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)) {
if (get_bits1(gb)) {
for (i = 0; i < s->num_channels * s->num_channels; i++)
skip_bits(gb, 4);
}
}
/** read drc info */
if (s->dynamic_range_compression) {
s->drc_gain = get_bits(gb, 8);
dprintf(s->avctx, "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)) {
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);
}
}
dprintf(s->avctx, "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;
}
}
/** interleave samples and write them to the output buffer */
for (i = 0; i < s->num_channels; i++) {
float* ptr = s->samples + i;
int incr = s->num_channels;
float* iptr = s->channel[i].out;
float* iend = iptr + s->samples_per_frame;
// FIXME should create/use a DSP function here
while (iptr < iend) {
*ptr = *iptr++;
ptr += incr;
}
/** reuse second half of the IMDCT output for the next frame */
memcpy(&s->channel[i].out[0],
&s->channel[i].out[s->samples_per_frame],
s->samples_per_frame * sizeof(*s->channel[i].out) >> 1);
}
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) {
}
}
/** 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 = (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) {
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
*/
static int decode_packet(AVCodecContext *avctx,
void *data, int *data_size, AVPacket* avpkt)
{
WMAProDecodeCtx *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;
s->buf_bit_size = buf_size << 3;
/** sanity check for the buffer length */
if (buf_size < avctx->block_align)
return 0;
buf_size = avctx->block_align;
/** 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);
dprintf(avctx, "packet[%d]: nbpf %x\n", avctx->frame_number,
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;
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) {
/** 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);
dprintf(avctx, "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) {
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 << 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 = (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)
{
WMAProDecodeCtx *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 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"),
};
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