Mercurial > libavcodec.hg
view wmaprodec.c @ 10094:a1f5f022adba libavcodec
dsputil: declare some arrays 16-byte aligned
These arrays are passed to functions for which we promise 16-byte
alignment, so make sure we fulfil that promise.
author | mru |
---|---|
date | Wed, 26 Aug 2009 13:17:06 +0000 |
parents | bf63f070fbf7 |
children | 071d6d272a77 |
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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 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 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 == 1) { /** M/S stereo decoding */ int16_t* sfb_offsets = s->cur_sfb_offsets; float* ch0 = *sfb_offsets + s->channel[0].coeffs; float* ch1 = *sfb_offsets++ + s->channel[1].coeffs; const char* tb = s->chgroup[i].transform_band; const char* tb_end = tb + s->num_bands; while (tb < tb_end) { const float* ch0_end = s->channel[0].coeffs + FFMIN(*sfb_offsets, s->subframe_len); if (*tb++ == 1) { while (ch0 < ch0_end) { const float v1 = *ch0; const float v2 = *ch1; *ch0++ = v1 - v2; *ch1++ = v1 + v2; } } else { while (ch0 < ch0_end) { *ch0++ *= 181.0 / 128; *ch1++ *= 181.0 / 128; } } ++sfb_offsets; } } else 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++) { if (*tb++ == 1) { int y; /** 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; } } } } } } }