/* * Wmall compatible decoder * Copyright (c) 2007 Baptiste Coudurier, Benjamin Larsson, Ulion * Copyright (c) 2008 - 2011 Sascha Sommer, Benjamin Larsson * Copyright (c) 2011 Andreas Öman * * 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 wmall decoder implementation * Wmall 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 wmall 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 wmall 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 wmall 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 wmall 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 "dsputil.h" #include "wma.h" /** current decoder limitations */ #define WMALL_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 WMALL_BLOCK_MIN_BITS 6 ///< log2 of min block size #define WMALL_BLOCK_MAX_BITS 12 ///< log2 of max block size #define WMALL_BLOCK_MAX_SIZE (1 << WMALL_BLOCK_MAX_BITS) ///< maximum block size #define WMALL_BLOCK_SIZES (WMALL_BLOCK_MAX_BITS - WMALL_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 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 uint16_t num_vec_coeffs; ///< number of vector coded coefficients DECLARE_ALIGNED(16, float, out)[WMALL_BLOCK_MAX_SIZE + WMALL_BLOCK_MAX_SIZE / 2]; ///< output buffer int transient_counter; ///< number of transient samples from the beginning of transient zone } WmallChannelCtx; /** * @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[WMALL_MAX_CHANNELS*WMALL_MAX_CHANNELS]; float* channel_data[WMALL_MAX_CHANNELS]; ///< transformation coefficients } WmallChannelGrp; /** * @brief main decoder context */ typedef struct WmallDecodeCtx { /* 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[WMALL_BLOCK_SIZES]; ///< MDCT context per block size DECLARE_ALIGNED(16, float, tmp)[WMALL_BLOCK_MAX_SIZE]; ///< IMDCT output buffer float* windows[WMALL_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[WMALL_BLOCK_SIZES]; ///< scale factor bands per block size int16_t sfb_offsets[WMALL_BLOCK_SIZES][MAX_BANDS]; ///< scale factor band offsets (multiples of 4) int8_t sf_offsets[WMALL_BLOCK_SIZES][WMALL_BLOCK_SIZES][MAX_BANDS]; ///< scale factor resample matrix int16_t subwoofer_cutoffs[WMALL_BLOCK_SIZES]; ///< subwoofer cutoff values /* packet decode state */ GetBitContext pgb; ///< bitstream reader context for the packet int next_packet_start; ///< start offset of the next wma packet in the demuxer 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[WMALL_MAX_CHANNELS]; int8_t num_bands; ///< number of scale factor bands int8_t transmit_num_vec_coeffs; ///< number of vector coded coefficients is part of the bitstream 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 WmallChannelGrp chgroup[WMALL_MAX_CHANNELS]; ///< channel group information WmallChannelCtx channel[WMALL_MAX_CHANNELS]; ///< per channel data // WMA lossless uint8_t do_arith_coding; uint8_t do_ac_filter; uint8_t do_inter_ch_decorr; uint8_t do_mclms; uint8_t do_lpc; int8_t acfilter_order; int8_t acfilter_scaling; int acfilter_coeffs[16]; int8_t mclms_order; int8_t mclms_scaling; int16_t mclms_coeffs[128]; int16_t mclms_coeffs_cur[4]; int mclms_prevvalues[64]; // FIXME: should be 32-bit / 16-bit depending on bit-depth int16_t mclms_updates[64]; int mclms_recent; int movave_scaling; int quant_stepsize; struct { int order; int scaling; int coefsend; int bitsend; int16_t coefs[256]; int lms_prevvalues[512]; // FIXME: see above int16_t lms_updates[512]; // and here too int recent; } cdlms[2][9]; /* XXX: Here, 2 is the max. no. of channels allowed, 9 is the maximum no. of filters per channel. Question is, why 2 if WMALL_MAX_CHANNELS == 8 */ int cdlms_ttl[2]; int bV3RTM; int is_channel_coded[2]; // XXX: same question as above applies here too (and below) int update_speed[2]; int transient[2]; int transient_pos[2]; int seekable_tile; int ave_sum[2]; int channel_residues[2][2048]; int lpc_coefs[2][40]; int lpc_order; int lpc_scaling; int lpc_intbits; int channel_coeffs[2][2048]; } WmallDecodeCtx; #undef dprintf #define dprintf(pctx, ...) av_log(pctx, AV_LOG_DEBUG, __VA_ARGS__) static int num_logged_tiles = 0; /** *@brief helper function to print the most important members of the context *@param s context */ static void av_cold dump_context(WmallDecodeCtx *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); } static int dump_int_buffer(int *buffer, int length, int delimiter) { int i; for (i=0 ; ipriv_data; int i; for (i = 0; i < WMALL_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) { WmallDecodeCtx *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; s->max_subframe_len_bit = 0; 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); s->bV3RTM = s->decode_flags & 0x100; 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 > WMALL_MAX_CHANNELS) { av_log_ask_for_sample(avctx, "unsupported number of channels\n"); return AVERROR_PATCHWELCOME; } 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(WmallDecodeCtx *s, int offset) { int frame_len_ratio; int subframe_len, 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; len = av_log2(s->max_num_subframes - 1) + 1; frame_len_ratio = get_bits(&s->gb, len); subframe_len = s->min_samples_per_subframe * (frame_len_ratio + 1); /** 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(WmallDecodeCtx *s) { uint16_t num_samples[WMALL_MAX_CHANNELS]; /**< sum of samples for all currently known subframes of a channel */ uint8_t contains_subframe[WMALL_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 subfra2me 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++) { WmallChannelCtx* 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(%d) > samples_per_frame(%d)\n", num_samples[c], s->samples_per_frame); 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++) { s->channel[c].subframe_offset[i] = offset; offset += s->channel[c].subframe_len[i]; } } return 0; } static int my_log2(unsigned int i) { unsigned int iLog2 = 0; while ((i >> iLog2) > 1) iLog2++; return iLog2; } /** * */ static void decode_ac_filter(WmallDecodeCtx *s) { int i; s->acfilter_order = get_bits(&s->gb, 4) + 1; s->acfilter_scaling = get_bits(&s->gb, 4); for(i = 0; i < s->acfilter_order; i++) { s->acfilter_coeffs[i] = get_bits(&s->gb, s->acfilter_scaling) + 1; } } /** * */ static void decode_mclms(WmallDecodeCtx *s) { s->mclms_order = (get_bits(&s->gb, 4) + 1) * 2; s->mclms_scaling = get_bits(&s->gb, 4); if(get_bits1(&s->gb)) { // mclms_send_coef int i; int send_coef_bits; int cbits = av_log2(s->mclms_scaling + 1); assert(cbits == my_log2(s->mclms_scaling + 1)); if(1 << cbits < s->mclms_scaling + 1) cbits++; send_coef_bits = (cbits ? get_bits(&s->gb, cbits) : 0) + 2; for(i = 0; i < s->mclms_order * s->num_channels * s->num_channels; i++) { s->mclms_coeffs[i] = get_bits(&s->gb, send_coef_bits); } for(i = 0; i < s->num_channels; i++) { int c; for(c = 0; c < i; c++) { s->mclms_coeffs_cur[i * s->num_channels + c] = get_bits(&s->gb, send_coef_bits); } } } } /** * */ static void decode_cdlms(WmallDecodeCtx *s) { int c, i; int cdlms_send_coef = get_bits1(&s->gb); for(c = 0; c < s->num_channels; c++) { s->cdlms_ttl[c] = get_bits(&s->gb, 3) + 1; for(i = 0; i < s->cdlms_ttl[c]; i++) { s->cdlms[c][i].order = (get_bits(&s->gb, 7) + 1) * 8; } for(i = 0; i < s->cdlms_ttl[c]; i++) { s->cdlms[c][i].scaling = get_bits(&s->gb, 4); } if(cdlms_send_coef) { for(i = 0; i < s->cdlms_ttl[c]; i++) { int cbits, shift_l, shift_r, j; cbits = av_log2(s->cdlms[c][i].order); if(1 << cbits < s->cdlms[c][i].order) cbits++; s->cdlms[c][i].coefsend = get_bits(&s->gb, cbits) + 1; cbits = av_log2(s->cdlms[c][i].scaling + 1); if(1 << cbits < s->cdlms[c][i].scaling + 1) cbits++; s->cdlms[c][i].bitsend = get_bits(&s->gb, cbits) + 2; shift_l = 32 - s->cdlms[c][i].bitsend; shift_r = 32 - 2 - s->cdlms[c][i].scaling; for(j = 0; j < s->cdlms[c][i].coefsend; j++) { s->cdlms[c][i].coefs[j] = (get_bits(&s->gb, s->cdlms[c][i].bitsend) << shift_l) >> shift_r; } } } } } /** * */ static int decode_channel_residues(WmallDecodeCtx *s, int ch, int tile_size) { int i = 0; unsigned int ave_mean; s->transient[ch] = get_bits1(&s->gb); if(s->transient[ch]) { s->transient_pos[ch] = get_bits(&s->gb, av_log2(tile_size)); if (s->transient_pos[ch]) s->transient[ch] = 0; s->channel[ch].transient_counter = FFMAX(s->channel[ch].transient_counter, s->samples_per_frame / 2); } else if (s->channel[ch].transient_counter) s->transient[ch] = 1; if(s->seekable_tile) { ave_mean = get_bits(&s->gb, s->bits_per_sample); s->ave_sum[ch] = ave_mean << (s->movave_scaling + 1); // s->ave_sum[ch] *= 2; } if(s->seekable_tile) { if(s->do_inter_ch_decorr) s->channel_residues[ch][0] = get_sbits(&s->gb, s->bits_per_sample + 1); else s->channel_residues[ch][0] = get_sbits(&s->gb, s->bits_per_sample); i++; } //av_log(0, 0, "%8d: ", num_logged_tiles++); for(; i < tile_size; i++) { int quo = 0, rem, rem_bits, residue; while(get_bits1(&s->gb)) quo++; if(quo >= 32) quo += get_bits_long(&s->gb, get_bits(&s->gb, 5) + 1); ave_mean = (s->ave_sum[ch] + (1 << s->movave_scaling)) >> (s->movave_scaling + 1); rem_bits = av_ceil_log2(ave_mean); rem = rem_bits ? get_bits(&s->gb, rem_bits) : 0; residue = (quo << rem_bits) + rem; s->ave_sum[ch] = residue + s->ave_sum[ch] - (s->ave_sum[ch] >> s->movave_scaling); if(residue & 1) residue = -(residue >> 1) - 1; else residue = residue >> 1; s->channel_residues[ch][i] = residue; /*if (num_logged_tiles < 1) av_log(0, 0, "%4d ", residue); */ } dump_int_buffer(s->channel_residues[ch], tile_size, 16); return 0; } /** * */ static void decode_lpc(WmallDecodeCtx *s) { int ch, i, cbits; s->lpc_order = get_bits(&s->gb, 5) + 1; s->lpc_scaling = get_bits(&s->gb, 4); s->lpc_intbits = get_bits(&s->gb, 3) + 1; cbits = s->lpc_scaling + s->lpc_intbits; for(ch = 0; ch < s->num_channels; ch++) { for(i = 0; i < s->lpc_order; i++) { s->lpc_coefs[ch][i] = get_sbits(&s->gb, cbits); } } } static void clear_codec_buffers(WmallDecodeCtx *s) { int ich, ilms; memset(s->acfilter_coeffs, 0, 16 * sizeof(int)); memset(s->lpc_coefs , 0, 40 * 2 * sizeof(int)); memset(s->mclms_coeffs , 0, 128 * sizeof(int16_t)); memset(s->mclms_coeffs_cur, 0, 4 * sizeof(int16_t)); memset(s->mclms_prevvalues, 0, 64 * sizeof(int)); memset(s->mclms_updates , 0, 64 * sizeof(int16_t)); for (ich = 0; ich < s->num_channels; ich++) { for (ilms = 0; ilms < s->cdlms_ttl[ich]; ilms++) { memset(s->cdlms[ich][ilms].coefs , 0, 256 * sizeof(int16_t)); memset(s->cdlms[ich][ilms].lms_prevvalues, 0, 512 * sizeof(int)); memset(s->cdlms[ich][ilms].lms_updates , 0, 512 * sizeof(int16_t)); } s->ave_sum[ich] = 0; } } /** *@brief Resets filter parameters and transient area at new seekable tile */ static void reset_codec(WmallDecodeCtx *s) { int ich, ilms; s->mclms_recent = s->mclms_order * s->num_channels; for (ich = 0; ich < s->num_channels; ich++) { for (ilms = 0; ilms < s->cdlms_ttl[ich]; ilms++) s->cdlms[ich][ilms].recent = s->cdlms[ich][ilms].order; /* first sample of a seekable subframe is considered as the starting of a transient area which is samples_per_frame samples long */ s->channel[ich].transient_counter = s->samples_per_frame; s->transient[ich] = 1; } } static int lms_predict(WmallDecodeCtx *s, int ich, int ilms) { int32_t pred = 0, icoef; int recent = s->cdlms[ich][ilms].recent; for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++) pred += s->cdlms[ich][ilms].coefs[icoef] * s->cdlms[ich][ilms].lms_prevvalues[icoef + recent]; pred += (1 << (s->cdlms[ich][ilms].scaling - 1)); /* XXX: Table 29 has: iPred >= cdlms[iCh][ilms].scaling; seems to me like a missing > */ pred >>= s->cdlms[ich][ilms].scaling; return pred; } static void lms_update(WmallDecodeCtx *s, int ich, int ilms, int32_t input, int32_t pred) { int icoef; int recent = s->cdlms[ich][ilms].recent; int range = 1 << (s->bits_per_sample - 1); int bps = s->bits_per_sample > 16 ? 4 : 2; // bytes per sample if (input > pred) { for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++) s->cdlms[ich][ilms].coefs[icoef] += s->cdlms[ich][ilms].lms_updates[icoef + recent]; } else { for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++) s->cdlms[ich][ilms].coefs[icoef] -= s->cdlms[ich][ilms].lms_updates[icoef]; // XXX: [icoef + recent] ? } s->cdlms[ich][ilms].recent--; s->cdlms[ich][ilms].lms_prevvalues[recent] = av_clip(input, -range, range - 1); if (input > pred) s->cdlms[ich][ilms].lms_updates[recent] = s->update_speed[ich]; else if (input < pred) s->cdlms[ich][ilms].lms_updates[recent] = -s->update_speed[ich]; /* XXX: spec says: cdlms[iCh][ilms].updates[iRecent + cdlms[iCh][ilms].order >> 4] >>= 2; lms_updates[iCh][ilms][iRecent + cdlms[iCh][ilms].order >> 3] >>= 1; Questions is - are cdlms[iCh][ilms].updates[] and lms_updates[][][] two seperate buffers? Here I've assumed that the two are same which makes more sense to me. */ s->cdlms[ich][ilms].lms_updates[recent + s->cdlms[ich][ilms].order >> 4] >>= 2; s->cdlms[ich][ilms].lms_updates[recent + s->cdlms[ich][ilms].order >> 3] >>= 1; /* XXX: recent + (s->cdlms[ich][ilms].order >> 4) ? */ if (s->cdlms[ich][ilms].recent == 0) { /* XXX: This memcpy()s will probably fail if a fixed 32-bit buffer is used. follow kshishkov's suggestion of using a union. */ memcpy(s->cdlms[ich][ilms].lms_prevvalues + s->cdlms[ich][ilms].order, s->cdlms[ich][ilms].lms_prevvalues, bps * s->cdlms[ich][ilms].order); memcpy(s->cdlms[ich][ilms].lms_updates + s->cdlms[ich][ilms].order, s->cdlms[ich][ilms].lms_updates, bps * s->cdlms[ich][ilms].order); s->cdlms[ich][ilms].recent = s->cdlms[ich][ilms].order; } } static void use_high_update_speed(WmallDecodeCtx *s, int ich) { int ilms, recent, icoef; s->update_speed[ich] = 16; for (ilms = s->cdlms_ttl[ich] - 1; ilms >= 0; ilms--) { recent = s->cdlms[ich][ilms].recent; if (s->bV3RTM) { for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++) s->cdlms[ich][ilms].lms_updates[icoef + recent] *= 2; } else { for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++) s->cdlms[ich][ilms].lms_updates[icoef] *= 2; } } } static void use_normal_update_speed(WmallDecodeCtx *s, int ich) { int ilms, recent, icoef; s->update_speed[ich] = 8; for (ilms = s->cdlms_ttl[ich] - 1; ilms >= 0; ilms--) { recent = s->cdlms[ich][ilms].recent; if (s->bV3RTM) { for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++) s->cdlms[ich][ilms].lms_updates[icoef + recent] /= 2; } else { for (icoef = 0; icoef < s->cdlms[ich][ilms].order; icoef++) s->cdlms[ich][ilms].lms_updates[icoef] /= 2; } } } static void revert_cdlms(WmallDecodeCtx *s, int tile_size) { int icoef, ich; int32_t pred, channel_coeff; int ilms, num_lms; for (ich = 0; ich < s->num_channels; ich++) { if (!s->is_channel_coded[ich]) continue; for (icoef = 0; icoef < tile_size; icoef++) { num_lms = s->cdlms_ttl[ich]; channel_coeff = s->channel_residues[ich][icoef]; if (icoef == s->transient_pos[ich]) { s->transient[ich] = 1; use_high_update_speed(s, ich); } for (ilms = num_lms - 1; ilms >= 0; ilms--) { pred = lms_predict(s, ich, ilms); channel_coeff += pred; lms_update(s, ich, ilms, channel_coeff, pred); } if (s->transient[ich]) { --s->channel[ich].transient_counter; if(!s->channel[ich].transient_counter) use_normal_update_speed(s, ich); } s->channel_coeffs[ich][icoef] = channel_coeff; } } } /** *@brief Decode a single subframe (block). *@param s codec context *@return 0 on success, < 0 when decoding failed */ static int decode_subframe(WmallDecodeCtx *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 rawpcm_tile; int padding_zeroes; 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]; } } /** 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; s->seekable_tile = get_bits1(&s->gb); if(s->seekable_tile) { clear_codec_buffers(s); s->do_arith_coding = get_bits1(&s->gb); if(s->do_arith_coding) { dprintf(s->avctx, "do_arith_coding == 1"); abort(); } s->do_ac_filter = get_bits1(&s->gb); s->do_inter_ch_decorr = get_bits1(&s->gb); s->do_mclms = get_bits1(&s->gb); if(s->do_ac_filter) decode_ac_filter(s); if(s->do_mclms) decode_mclms(s); decode_cdlms(s); s->movave_scaling = get_bits(&s->gb, 3); s->quant_stepsize = get_bits(&s->gb, 8) + 1; reset_codec(s); } rawpcm_tile = get_bits1(&s->gb); for(i = 0; i < s->num_channels; i++) { s->is_channel_coded[i] = 1; } if(!rawpcm_tile) { for(i = 0; i < s->num_channels; i++) { s->is_channel_coded[i] = get_bits1(&s->gb); } if(s->bV3RTM) { // LPC s->do_lpc = get_bits1(&s->gb); if(s->do_lpc) { decode_lpc(s); } } else { s->do_lpc = 0; } } if(get_bits1(&s->gb)) { padding_zeroes = get_bits(&s->gb, 5); } else { padding_zeroes = 0; } if(rawpcm_tile) { int bits = s->bits_per_sample - padding_zeroes; int j; dprintf(s->avctx, "RAWPCM %d bits per sample. total %d bits, remain=%d\n", bits, bits * s->num_channels * subframe_len, get_bits_count(&s->gb)); for(i = 0; i < s->num_channels; i++) { for(j = 0; j < subframe_len; j++) { s->channel_coeffs[i][j] = get_sbits(&s->gb, bits); // dprintf(s->avctx, "PCM[%d][%d] = 0x%04x\n", i, j, s->channel_coeffs[i][j]); } } } else { for(i = 0; i < s->num_channels; i++) if(s->is_channel_coded[i]) decode_channel_residues(s, i, subframe_len); } revert_cdlms(s, subframe_len); /** 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(WmallDecodeCtx *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); /** decode tile information */ if (decode_tilehdr(s)) { s->packet_loss = 1; return 0; } /** read drc info */ if (s->dynamic_range_compression) { s->drc_gain = get_bits(gb, 8); } /** 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); } } /** 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; } } dprintf(s->avctx, "Frame done\n"); 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) { dprintf(s->avctx, "skip1\n"); } */ } /** 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(WmallDecodeCtx *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(WmallDecodeCtx *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) { avpriv_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; avpriv_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) { WmallDecodeCtx *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; /** sanity check for the buffer length */ if (buf_size < avctx->block_align) return 0; s->next_packet_start = buf_size - avctx->block_align; buf_size = avctx->block_align; s->buf_bit_size = buf_size << 3; /** parse packet header */ init_get_bits(gb, buf, s->buf_bit_size); packet_sequence_number = get_bits(gb, 4); int seekable_frame_in_packet = get_bits1(gb); int spliced_packet = get_bits1(gb); /** get number of bits that need to be added to the previous frame */ num_bits_prev_frame = get_bits(gb, s->log2_frame_size); /** 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) { int remaining_packet_bits = s->buf_bit_size - get_bits_count(gb); if (num_bits_prev_frame >= remaining_packet_bits) { num_bits_prev_frame = remaining_packet_bits; s->packet_done = 1; } /** 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); /** 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 - s->next_packet_start) << 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 = 0; // (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) { WmallDecodeCtx *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 wmall decoder */ AVCodec ff_wmalossless_decoder = { "wmalossless", AVMEDIA_TYPE_AUDIO, CODEC_ID_WMALOSSLESS, sizeof(WmallDecodeCtx), decode_init, NULL, decode_end, decode_packet, .capabilities = CODEC_CAP_SUBFRAMES, .flush= flush, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Lossless"), };