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https://git.videolan.org/git/ffmpeg.git
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229 lines
8.9 KiB
C
229 lines
8.9 KiB
C
/*
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* Wmapro compatible decoder
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* Copyright (c) 2007 Baptiste Coudurier, Benjamin Larsson, Ulion
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* Copyright (c) 2008 - 2009 Sascha Sommer, Benjamin Larsson
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file libavcodec/wmaprodec.c
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* @brief wmapro decoder implementation
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* Wmapro is an MDCT based codec comparable to wma standard or AAC.
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* The decoding therefore consists of the following steps:
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* - bitstream decoding
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* - reconstruction of per-channel data
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* - rescaling and inverse quantization
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* - IMDCT
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* - windowing and overlapp-add
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*
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* The compressed wmapro bitstream is split into individual packets.
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* Every such packet contains one or more wma frames.
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* The compressed frames may have a variable length and frames may
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* cross packet boundaries.
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* Common to all wmapro frames is the number of samples that are stored in
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* a frame.
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* The number of samples and a few other decode flags are stored
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* as extradata that has to be passed to the decoder.
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*
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* The wmapro frames themselves are again split into a variable number of
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* subframes. Every subframe contains the data for 2^N time domain samples
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* where N varies between 7 and 12.
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*
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* Example wmapro bitstream (in samples):
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*
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* || packet 0 || packet 1 || packet 2 packets
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* ---------------------------------------------------
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* || frame 0 || frame 1 || frame 2 || frames
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* ---------------------------------------------------
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* || | | || | | | || || subframes of channel 0
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* ---------------------------------------------------
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* || | | || | | | || || subframes of channel 1
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* ---------------------------------------------------
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*
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* The frame layouts for the individual channels of a wma frame does not need
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* to be the same.
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*
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* However, if the offsets and lengths of several subframes of a frame are the
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* same, the subframes of the channels can be grouped.
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* Every group may then use special coding techniques like M/S stereo coding
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* to improve the compression ratio. These channel transformations do not
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* need to be applied to a whole subframe. Instead, they can also work on
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* individual scale factor bands (see below).
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* The coefficients that carry the audio signal in the frequency domain
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* are transmitted as huffman-coded vectors with 4, 2 and 1 elements.
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* In addition to that, the encoder can switch to a runlevel coding scheme
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* by transmitting subframe_length / 128 zero coefficients.
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*
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* Before the audio signal can be converted to the time domain, the
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* coefficients have to be rescaled and inverse quantized.
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* A subframe is therefore split into several scale factor bands that get
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* scaled individually.
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* Scale factors are submitted for every frame but they might be shared
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* between the subframes of a channel. Scale factors are initially DPCM-coded.
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* Once scale factors are shared, the differences are transmitted as runlevel
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* codes.
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* Every subframe length and offset combination in the frame layout shares a
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* common quantization factor that can be adjusted for every channel by a
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* modifier.
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* After the inverse quantization, the coefficients get processed by an IMDCT.
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* The resulting values are then windowed with a sine window and the first half
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* of the values are added to the second half of the output from the previous
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* subframe in order to reconstruct the output samples.
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*/
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/**
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*@brief Uninitialize the decoder and free all resources.
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*@param avctx codec context
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*@return 0 on success, < 0 otherwise
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*/
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static av_cold int decode_end(AVCodecContext *avctx)
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{
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WMA3DecodeContext *s = avctx->priv_data;
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int i;
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for (i = 0 ; i < WMAPRO_BLOCK_SIZES ; i++)
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ff_mdct_end(&s->mdct_ctx[i]);
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return 0;
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}
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/**
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*@brief Calculate a decorrelation matrix from the bitstream parameters.
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*@param s codec context
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*@param chgroup channel group for which the matrix needs to be calculated
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*/
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static void decode_decorrelation_matrix(WMA3DecodeContext *s,
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WMA3ChannelGroup *chgroup)
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{
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int i;
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int offset = 0;
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int8_t rotation_offset[WMAPRO_MAX_CHANNELS * WMAPRO_MAX_CHANNELS];
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memset(chgroup->decorrelation_matrix,0,
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sizeof(float) *s->num_channels * s->num_channels);
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for (i = 0; i < chgroup->num_channels * (chgroup->num_channels - 1) >> 1; i++)
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rotation_offset[i] = get_bits(&s->gb,6);
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for (i = 0; i < chgroup->num_channels; i++)
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chgroup->decorrelation_matrix[chgroup->num_channels * i + i] =
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get_bits1(&s->gb) ? 1.0 : -1.0;
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for (i = 1; i < chgroup->num_channels; i++) {
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int x;
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for (x = 0; x < i; x++) {
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int y;
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for (y = 0; y < i + 1 ; y++) {
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float v1 = chgroup->decorrelation_matrix[x * chgroup->num_channels + y];
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float v2 = chgroup->decorrelation_matrix[i * chgroup->num_channels + y];
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int n = rotation_offset[offset + x];
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float sinv;
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float cosv;
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if (n < 32) {
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sinv = sin64[n];
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cosv = sin64[32-n];
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} else {
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sinv = sin64[64-n];
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cosv = -sin64[n-32];
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}
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chgroup->decorrelation_matrix[y + x * chgroup->num_channels] =
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(v1 * sinv) - (v2 * cosv);
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chgroup->decorrelation_matrix[y + i * chgroup->num_channels] =
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(v1 * cosv) + (v2 * sinv);
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}
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}
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offset += i;
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}
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}
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/**
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*@brief Reconstruct the individual channel data.
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*@param s codec context
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*/
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static void inverse_channel_transform(WMA3DecodeContext *s)
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{
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int i;
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for (i = 0; i < s->num_chgroups; i++) {
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if (s->chgroup[i].transform == 1) {
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/** M/S stereo decoding */
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int16_t* sfb_offsets = s->cur_sfb_offsets;
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float* ch0 = *sfb_offsets + s->channel[0].coeffs;
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float* ch1 = *sfb_offsets++ + s->channel[1].coeffs;
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const char* tb = s->chgroup[i].transform_band;
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const char* tb_end = tb + s->num_bands;
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while (tb < tb_end) {
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const float* ch0_end = s->channel[0].coeffs +
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FFMIN(*sfb_offsets,s->subframe_len);
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if (*tb++ == 1) {
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while (ch0 < ch0_end) {
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const float v1 = *ch0;
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const float v2 = *ch1;
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*ch0++ = v1 - v2;
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*ch1++ = v1 + v2;
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}
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} else {
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while (ch0 < ch0_end) {
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*ch0++ *= 181.0 / 128;
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*ch1++ *= 181.0 / 128;
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}
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}
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++sfb_offsets;
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}
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} else if (s->chgroup[i].transform) {
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float data[WMAPRO_MAX_CHANNELS];
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const int num_channels = s->chgroup[i].num_channels;
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float** ch_data = s->chgroup[i].channel_data;
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float** ch_end = ch_data + num_channels;
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const int8_t* tb = s->chgroup[i].transform_band;
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int16_t* sfb;
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/** multichannel decorrelation */
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for (sfb = s->cur_sfb_offsets ;
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sfb < s->cur_sfb_offsets + s->num_bands;sfb++) {
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if (*tb++ == 1) {
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int y;
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/** multiply values with the decorrelation_matrix */
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for (y = sfb[0]; y < FFMIN(sfb[1], s->subframe_len); y++) {
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const float* mat = s->chgroup[i].decorrelation_matrix;
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const float* data_end = data + num_channels;
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float* data_ptr = data;
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float** ch;
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for (ch = ch_data;ch < ch_end; ch++)
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*data_ptr++ = (*ch)[y];
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for (ch = ch_data; ch < ch_end; ch++) {
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float sum = 0;
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data_ptr = data;
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while (data_ptr < data_end)
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sum += *data_ptr++ * *mat++;
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(*ch)[y] = sum;
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}
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}
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}
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}
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}
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}
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}
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