ffmpeg/libavcodec/wmaprodec.c
Diego Biurrun ae9253151f whitespace cosmetics
Originally committed as revision 19725 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-08-26 23:12:41 +00:00

295 lines
11 KiB
C

/*
* 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.
*/
/**
*@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)
{
WMA3DecodeContext *s = avctx->priv_data;
int i;
for (i = 0; i < WMAPRO_BLOCK_SIZES; i++)
ff_mdct_end(&s->mdct_ctx[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(WMA3DecodeContext *s,
WMA3ChannelGroup *chgroup)
{
int i;
int offset = 0;
int8_t rotation_offset[WMAPRO_MAX_CHANNELS * WMAPRO_MAX_CHANNELS];
memset(chgroup->decorrelation_matrix, 0,
sizeof(float) *s->num_channels * s->num_channels);
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 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(WMA3DecodeContext *s, int c)
{
int vlctable;
VLC* vlc;
WMA3ChannelCtx* ci = &s->channel[c];
int rl_mode = 0;
int cur_coeff = 0;
int num_zeros = 0;
const uint16_t* run;
const uint16_t* 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) {
vals[i] = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH);
if (vals[i] == HUFF_VEC1_SIZE - 1)
vals[i] += ff_wma_get_large_val(&s->gb);
vals[i+1] = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH);
if (vals[i+1] == HUFF_VEC1_SIZE - 1)
vals[i+1] += ff_wma_get_large_val(&s->gb);
} 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;
}
/** decode sign */
for (i = 0; i < 4; i++) {
if (vals[i]) {
int sign = get_bits1(&s->gb) - 1;
ci->coeffs[cur_coeff] = (vals[i]^sign) - sign;
num_zeros = 0;
} else {
/** 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) {
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 Reconstruct the individual channel data.
*@param s codec context
*/
static void inverse_channel_transform(WMA3DecodeContext *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) {
for (y = sfb[0]; y < FFMIN(sfb[1], s->subframe_len); y++) {
ch_data[0][y] *= 181.0 / 128;
ch_data[1][y] *= 181.0 / 128;
}
}
}
}
}
}