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228 lines
8.9 KiB
228 lines
8.9 KiB
/* |
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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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