Mercurial > libavcodec.hg
view wmaprodec.c @ 10948:c80f0dfbf47d libavcodec
Only calculate the second chroma qp if it differs from the firstin the main
loop filter. (a little faster for the common case where they are equal)
| author | michael |
|---|---|
| date | Wed, 20 Jan 2010 01:49:24 +0000 |
| parents | 3d011a01a6a0 |
| children | 34a65026fa06 |
line wrap: on
line source
/* * 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. */ #include "avcodec.h" #include "internal.h" #include "get_bits.h" #include "put_bits.h" #include "wmaprodata.h" #include "dsputil.h" #include "wma.h" /** current decoder limitations */ #define WMAPRO_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 WMAPRO_BLOCK_MAX_BITS 12 ///< log2 of max block size #define WMAPRO_BLOCK_MAX_SIZE (1 << WMAPRO_BLOCK_MAX_BITS) ///< maximum block size #define WMAPRO_BLOCK_SIZES (WMAPRO_BLOCK_MAX_BITS - 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 VLC sf_vlc; ///< scale factor DPCM vlc static VLC sf_rl_vlc; ///< scale factor run length vlc static VLC vec4_vlc; ///< 4 coefficients per symbol static VLC vec2_vlc; ///< 2 coefficients per symbol static VLC vec1_vlc; ///< 1 coefficient per symbol static VLC coef_vlc[2]; ///< coefficient run length vlc codes 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 DECLARE_ALIGNED_16(float, out[WMAPRO_BLOCK_MAX_SIZE + WMAPRO_BLOCK_MAX_SIZE / 2]); ///< output buffer } WMAProChannelCtx; /** * @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[WMAPRO_MAX_CHANNELS*WMAPRO_MAX_CHANNELS]; float* channel_data[WMAPRO_MAX_CHANNELS]; ///< transformation coefficients } WMAProChannelGrp; /** * @brief main decoder context */ typedef struct WMAProDecodeCtx { /* 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[WMAPRO_BLOCK_SIZES]; ///< MDCT context per block size DECLARE_ALIGNED_16(float, tmp[WMAPRO_BLOCK_MAX_SIZE]); ///< IMDCT output buffer float* windows[WMAPRO_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[WMAPRO_BLOCK_SIZES]; ///< scale factor bands per block size int16_t sfb_offsets[WMAPRO_BLOCK_SIZES][MAX_BANDS]; ///< scale factor band offsets (multiples of 4) int8_t sf_offsets[WMAPRO_BLOCK_SIZES][WMAPRO_BLOCK_SIZES][MAX_BANDS]; ///< scale factor resample matrix int16_t subwoofer_cutoffs[WMAPRO_BLOCK_SIZES]; ///< subwoofer cutoff values /* packet decode state */ GetBitContext pgb; ///< bitstream reader context for the 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[WMAPRO_MAX_CHANNELS]; int8_t num_bands; ///< number of scale factor bands 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 WMAProChannelGrp chgroup[WMAPRO_MAX_CHANNELS]; ///< channel group information WMAProChannelCtx channel[WMAPRO_MAX_CHANNELS]; ///< per channel data } WMAProDecodeCtx; /** *@brief helper function to print the most important members of the context *@param s context */ static void av_cold dump_context(WMAProDecodeCtx *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); } /** *@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) { WMAProDecodeCtx *s = avctx->priv_data; int i; for (i = 0; i < WMAPRO_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) { WMAProDecodeCtx *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 = 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); if (!s->len_prefix) { av_log_ask_for_sample(avctx, "no length prefix\n"); return AVERROR_INVALIDDATA; } /** 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; if (s->max_num_subframes == 16) s->max_subframe_len_bit = 1; 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); 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 || s->num_channels > WMAPRO_MAX_CHANNELS) { av_log_ask_for_sample(avctx, "invalid number of channels\n"); return AVERROR_NOTSUPP; } INIT_VLC_STATIC(&sf_vlc, SCALEVLCBITS, HUFF_SCALE_SIZE, scale_huffbits, 1, 1, scale_huffcodes, 2, 2, 616); INIT_VLC_STATIC(&sf_rl_vlc, VLCBITS, HUFF_SCALE_RL_SIZE, scale_rl_huffbits, 1, 1, scale_rl_huffcodes, 4, 4, 1406); INIT_VLC_STATIC(&coef_vlc[0], VLCBITS, HUFF_COEF0_SIZE, coef0_huffbits, 1, 1, coef0_huffcodes, 4, 4, 2108); INIT_VLC_STATIC(&coef_vlc[1], VLCBITS, HUFF_COEF1_SIZE, coef1_huffbits, 1, 1, coef1_huffcodes, 4, 4, 3912); INIT_VLC_STATIC(&vec4_vlc, VLCBITS, HUFF_VEC4_SIZE, vec4_huffbits, 1, 1, vec4_huffcodes, 2, 2, 604); INIT_VLC_STATIC(&vec2_vlc, VLCBITS, HUFF_VEC2_SIZE, vec2_huffbits, 1, 1, vec2_huffcodes, 2, 2, 562); INIT_VLC_STATIC(&vec1_vlc, VLCBITS, HUFF_VEC1_SIZE, vec1_huffbits, 1, 1, vec1_huffcodes, 2, 2, 562); /** calculate number of scale factor bands and their offsets for every possible block size */ for (i = 0; i < num_possible_block_sizes; i++) { int subframe_len = s->samples_per_frame >> i; int x; int band = 1; s->sfb_offsets[i][0] = 0; for (x = 0; x < MAX_BANDS-1 && s->sfb_offsets[i][band - 1] < subframe_len; x++) { int offset = (subframe_len * 2 * critical_freq[x]) / s->avctx->sample_rate + 2; offset &= ~3; if (offset > s->sfb_offsets[i][band - 1]) s->sfb_offsets[i][band++] = offset; } s->sfb_offsets[i][band - 1] = subframe_len; s->num_sfb[i] = band - 1; } /** Scale factors can be shared between blocks of different size as every block has a different scale factor band layout. The matrix sf_offsets is needed to find the correct scale factor. */ for (i = 0; i < num_possible_block_sizes; i++) { int b; for (b = 0; b < s->num_sfb[i]; b++) { int x; int offset = ((s->sfb_offsets[i][b] + s->sfb_offsets[i][b + 1] - 1) << i) >> 1; for (x = 0; x < num_possible_block_sizes; x++) { int v = 0; while (s->sfb_offsets[x][v + 1] << x < offset) ++v; s->sf_offsets[i][x][b] = v; } } } /** init MDCT, FIXME: only init needed sizes */ for (i = 0; i < WMAPRO_BLOCK_SIZES; i++) ff_mdct_init(&s->mdct_ctx[i], BLOCK_MIN_BITS+1+i, 1, 1.0 / (1 << (BLOCK_MIN_BITS + i - 1)) / (1 << (s->bits_per_sample - 1))); /** init MDCT windows: simple sinus window */ for (i = 0; i < WMAPRO_BLOCK_SIZES; i++) { const int win_idx = WMAPRO_BLOCK_MAX_BITS - i; ff_init_ff_sine_windows(win_idx); s->windows[WMAPRO_BLOCK_SIZES - i - 1] = ff_sine_windows[win_idx]; } /** calculate subwoofer cutoff values */ for (i = 0; i < num_possible_block_sizes; i++) { int block_size = s->samples_per_frame >> i; int cutoff = (440*block_size + 3 * (s->avctx->sample_rate >> 1) - 1) / s->avctx->sample_rate; s->subwoofer_cutoffs[i] = av_clip(cutoff, 4, block_size); } /** calculate sine values for the decorrelation matrix */ for (i = 0; i < 33; i++) sin64[i] = sin(i*M_PI / 64.0); if (avctx->debug & FF_DEBUG_BITSTREAM) dump_context(s); 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(WMAProDecodeCtx *s, int offset) { int frame_len_shift = 0; int subframe_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; /** 1 bit indicates if the subframe is of maximum length */ if (s->max_subframe_len_bit) { if (get_bits1(&s->gb)) frame_len_shift = 1 + get_bits(&s->gb, s->subframe_len_bits-1); } else frame_len_shift = get_bits(&s->gb, s->subframe_len_bits); subframe_len = s->samples_per_frame >> frame_len_shift; /** 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(WMAProDecodeCtx *s) { uint16_t num_samples[WMAPRO_MAX_CHANNELS]; /** sum of samples for all currently known subframes of a channel */ uint8_t contains_subframe[WMAPRO_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 subframe 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++) { WMAProChannelCtx* 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 > samples_per_frame\n"); 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++) { dprintf(s->avctx, "frame[%i] channel[%i] subframe[%i]" " len %i\n", s->frame_num, c, i, s->channel[c].subframe_len[i]); s->channel[c].subframe_offset[i] = offset; offset += s->channel[c].subframe_len[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(WMAProDecodeCtx *s, WMAProChannelGrp *chgroup) { int i; int offset = 0; int8_t rotation_offset[WMAPRO_MAX_CHANNELS * WMAPRO_MAX_CHANNELS]; memset(chgroup->decorrelation_matrix, 0, s->num_channels * s->num_channels * sizeof(*chgroup->decorrelation_matrix)); 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 Decode channel transformation parameters *@param s codec context *@return 0 in case of success, < 0 in case of bitstream errors */ static int decode_channel_transform(WMAProDecodeCtx* s) { int i; /* should never consume more than 1921 bits for the 8 channel case * 1 + MAX_CHANNELS * (MAX_CHANNELS + 2 + 3 * MAX_CHANNELS * MAX_CHANNELS * + MAX_CHANNELS + MAX_BANDS + 1) */ /** in the one channel case channel transforms are pointless */ s->num_chgroups = 0; if (s->num_channels > 1) { int remaining_channels = s->channels_for_cur_subframe; if (get_bits1(&s->gb)) { av_log_ask_for_sample(s->avctx, "unsupported channel transform bit\n"); return AVERROR_INVALIDDATA; } for (s->num_chgroups = 0; remaining_channels && s->num_chgroups < s->channels_for_cur_subframe; s->num_chgroups++) { WMAProChannelGrp* chgroup = &s->chgroup[s->num_chgroups]; float** channel_data = chgroup->channel_data; chgroup->num_channels = 0; chgroup->transform = 0; /** decode channel mask */ if (remaining_channels > 2) { for (i = 0; i < s->channels_for_cur_subframe; i++) { int channel_idx = s->channel_indexes_for_cur_subframe[i]; if (!s->channel[channel_idx].grouped && get_bits1(&s->gb)) { ++chgroup->num_channels; s->channel[channel_idx].grouped = 1; *channel_data++ = s->channel[channel_idx].coeffs; } } } else { chgroup->num_channels = remaining_channels; for (i = 0; i < s->channels_for_cur_subframe; i++) { int channel_idx = s->channel_indexes_for_cur_subframe[i]; if (!s->channel[channel_idx].grouped) *channel_data++ = s->channel[channel_idx].coeffs; s->channel[channel_idx].grouped = 1; } } /** decode transform type */ if (chgroup->num_channels == 2) { if (get_bits1(&s->gb)) { if (get_bits1(&s->gb)) { av_log_ask_for_sample(s->avctx, "unsupported channel transform type\n"); } } else { chgroup->transform = 1; if (s->num_channels == 2) { chgroup->decorrelation_matrix[0] = 1.0; chgroup->decorrelation_matrix[1] = -1.0; chgroup->decorrelation_matrix[2] = 1.0; chgroup->decorrelation_matrix[3] = 1.0; } else { /** cos(pi/4) */ chgroup->decorrelation_matrix[0] = 0.70703125; chgroup->decorrelation_matrix[1] = -0.70703125; chgroup->decorrelation_matrix[2] = 0.70703125; chgroup->decorrelation_matrix[3] = 0.70703125; } } } else if (chgroup->num_channels > 2) { if (get_bits1(&s->gb)) { chgroup->transform = 1; if (get_bits1(&s->gb)) { decode_decorrelation_matrix(s, chgroup); } else { /** FIXME: more than 6 coupled channels not supported */ if (chgroup->num_channels > 6) { av_log_ask_for_sample(s->avctx, "coupled channels > 6\n"); } else { memcpy(chgroup->decorrelation_matrix, default_decorrelation[chgroup->num_channels], chgroup->num_channels * chgroup->num_channels * sizeof(*chgroup->decorrelation_matrix)); } } } } /** decode transform on / off */ if (chgroup->transform) { if (!get_bits1(&s->gb)) { int i; /** transform can be enabled for individual bands */ for (i = 0; i < s->num_bands; i++) { chgroup->transform_band[i] = get_bits1(&s->gb); } } else { memset(chgroup->transform_band, 1, s->num_bands); } } remaining_channels -= chgroup->num_channels; } } return 0; } /** *@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(WMAProDecodeCtx *s, int c) { /* Integers 0..15 as single-precision floats. The table saves a costly int to float conversion, and storing the values as integers allows fast sign-flipping. */ static const int fval_tab[16] = { 0x00000000, 0x3f800000, 0x40000000, 0x40400000, 0x40800000, 0x40a00000, 0x40c00000, 0x40e00000, 0x41000000, 0x41100000, 0x41200000, 0x41300000, 0x41400000, 0x41500000, 0x41600000, 0x41700000, }; int vlctable; VLC* vlc; WMAProChannelCtx* ci = &s->channel[c]; int rl_mode = 0; int cur_coeff = 0; int num_zeros = 0; const uint16_t* run; const float* 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) { int v0, v1; v0 = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH); if (v0 == HUFF_VEC1_SIZE - 1) v0 += ff_wma_get_large_val(&s->gb); v1 = get_vlc2(&s->gb, vec1_vlc.table, VLCBITS, VEC1MAXDEPTH); if (v1 == HUFF_VEC1_SIZE - 1) v1 += ff_wma_get_large_val(&s->gb); ((float*)vals)[i ] = v0; ((float*)vals)[i+1] = v1; } else { vals[i] = fval_tab[symbol_to_vec2[idx] >> 4 ]; vals[i+1] = fval_tab[symbol_to_vec2[idx] & 0xF]; } } } else { vals[0] = fval_tab[ symbol_to_vec4[idx] >> 12 ]; vals[1] = fval_tab[(symbol_to_vec4[idx] >> 8) & 0xF]; vals[2] = fval_tab[(symbol_to_vec4[idx] >> 4) & 0xF]; vals[3] = fval_tab[ symbol_to_vec4[idx] & 0xF]; } /** decode sign */ for (i = 0; i < 4; i++) { if (vals[i]) { int sign = get_bits1(&s->gb) - 1; *(uint32_t*)&ci->coeffs[cur_coeff] = vals[i] ^ sign<<31; num_zeros = 0; } else { ci->coeffs[cur_coeff] = 0; /** 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) { memset(&ci->coeffs[cur_coeff], 0, sizeof(*ci->coeffs) * (s->subframe_len - cur_coeff)); 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 Extract scale factors from the bitstream. *@param s codec context *@return 0 on success, < 0 in case of bitstream errors */ static int decode_scale_factors(WMAProDecodeCtx* s) { int i; /** should never consume more than 5344 bits * MAX_CHANNELS * (1 + MAX_BANDS * 23) */ for (i = 0; i < s->channels_for_cur_subframe; i++) { int c = s->channel_indexes_for_cur_subframe[i]; int* sf; int* sf_end; s->channel[c].scale_factors = s->channel[c].saved_scale_factors[!s->channel[c].scale_factor_idx]; sf_end = s->channel[c].scale_factors + s->num_bands; /** resample scale factors for the new block size * as the scale factors might need to be resampled several times * before some new values are transmitted, a backup of the last * transmitted scale factors is kept in saved_scale_factors */ if (s->channel[c].reuse_sf) { const int8_t* sf_offsets = s->sf_offsets[s->table_idx][s->channel[c].table_idx]; int b; for (b = 0; b < s->num_bands; b++) s->channel[c].scale_factors[b] = s->channel[c].saved_scale_factors[s->channel[c].scale_factor_idx][*sf_offsets++]; } if (!s->channel[c].cur_subframe || get_bits1(&s->gb)) { if (!s->channel[c].reuse_sf) { int val; /** decode DPCM coded scale factors */ s->channel[c].scale_factor_step = get_bits(&s->gb, 2) + 1; val = 45 / s->channel[c].scale_factor_step; for (sf = s->channel[c].scale_factors; sf < sf_end; sf++) { val += get_vlc2(&s->gb, sf_vlc.table, SCALEVLCBITS, SCALEMAXDEPTH) - 60; *sf = val; } } else { int i; /** run level decode differences to the resampled factors */ for (i = 0; i < s->num_bands; i++) { int idx; int skip; int val; int sign; idx = get_vlc2(&s->gb, sf_rl_vlc.table, VLCBITS, SCALERLMAXDEPTH); if (!idx) { uint32_t code = get_bits(&s->gb, 14); val = code >> 6; sign = (code & 1) - 1; skip = (code & 0x3f) >> 1; } else if (idx == 1) { break; } else { skip = scale_rl_run[idx]; val = scale_rl_level[idx]; sign = get_bits1(&s->gb)-1; } i += skip; if (i >= s->num_bands) { av_log(s->avctx, AV_LOG_ERROR, "invalid scale factor coding\n"); return AVERROR_INVALIDDATA; } s->channel[c].scale_factors[i] += (val ^ sign) - sign; } } /** swap buffers */ s->channel[c].scale_factor_idx = !s->channel[c].scale_factor_idx; s->channel[c].table_idx = s->table_idx; s->channel[c].reuse_sf = 1; } /** calculate new scale factor maximum */ s->channel[c].max_scale_factor = s->channel[c].scale_factors[0]; for (sf = s->channel[c].scale_factors + 1; sf < sf_end; sf++) { s->channel[c].max_scale_factor = FFMAX(s->channel[c].max_scale_factor, *sf); } } return 0; } /** *@brief Reconstruct the individual channel data. *@param s codec context */ static void inverse_channel_transform(WMAProDecodeCtx *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) { int len = FFMIN(sfb[1], s->subframe_len) - sfb[0]; s->dsp.vector_fmul_scalar(ch_data[0] + sfb[0], ch_data[0] + sfb[0], 181.0 / 128, len); s->dsp.vector_fmul_scalar(ch_data[1] + sfb[0], ch_data[1] + sfb[0], 181.0 / 128, len); } } } } } /** *@brief Apply sine window and reconstruct the output buffer. *@param s codec context */ static void wmapro_window(WMAProDecodeCtx *s) { int i; for (i = 0; i < s->channels_for_cur_subframe; i++) { int c = s->channel_indexes_for_cur_subframe[i]; float* window; int winlen = s->channel[c].prev_block_len; float* start = s->channel[c].coeffs - (winlen >> 1); if (s->subframe_len < winlen) { start += (winlen - s->subframe_len) >> 1; winlen = s->subframe_len; } window = s->windows[av_log2(winlen) - BLOCK_MIN_BITS]; winlen >>= 1; s->dsp.vector_fmul_window(start, start, start + winlen, window, 0, winlen); s->channel[c].prev_block_len = s->subframe_len; } } /** *@brief Decode a single subframe (block). *@param s codec context *@return 0 on success, < 0 when decoding failed */ static int decode_subframe(WMAProDecodeCtx *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 transmit_coeffs = 0; int cur_subwoofer_cutoff; 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]; } } dprintf(s->avctx, "processing subframe with offset %i len %i\n", offset, subframe_len); /** 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; dprintf(s->avctx, "subframe is part of %i channels\n", s->channels_for_cur_subframe); /** calculate number of scale factor bands and their offsets */ s->table_idx = av_log2(s->samples_per_frame/subframe_len); s->num_bands = s->num_sfb[s->table_idx]; s->cur_sfb_offsets = s->sfb_offsets[s->table_idx]; cur_subwoofer_cutoff = s->subwoofer_cutoffs[s->table_idx]; /** configure the decoder for the current subframe */ for (i = 0; i < s->channels_for_cur_subframe; i++) { int c = s->channel_indexes_for_cur_subframe[i]; s->channel[c].coeffs = &s->channel[c].out[(s->samples_per_frame >> 1) + offset]; } s->subframe_len = subframe_len; s->esc_len = av_log2(s->subframe_len - 1) + 1; /** skip extended header if any */ if (get_bits1(&s->gb)) { int num_fill_bits; if (!(num_fill_bits = get_bits(&s->gb, 2))) { int len = get_bits(&s->gb, 4); num_fill_bits = get_bits(&s->gb, len) + 1; } if (num_fill_bits >= 0) { if (get_bits_count(&s->gb) + num_fill_bits > s->num_saved_bits) { av_log(s->avctx, AV_LOG_ERROR, "invalid number of fill bits\n"); return AVERROR_INVALIDDATA; } skip_bits_long(&s->gb, num_fill_bits); } } /** no idea for what the following bit is used */ if (get_bits1(&s->gb)) { av_log_ask_for_sample(s->avctx, "reserved bit set\n"); return AVERROR_INVALIDDATA; } if (decode_channel_transform(s) < 0) return AVERROR_INVALIDDATA; for (i = 0; i < s->channels_for_cur_subframe; i++) { int c = s->channel_indexes_for_cur_subframe[i]; if ((s->channel[c].transmit_coefs = get_bits1(&s->gb))) transmit_coeffs = 1; } if (transmit_coeffs) { int step; int quant_step = 90 * s->bits_per_sample >> 4; if ((get_bits1(&s->gb))) { /** FIXME: might change run level mode decision */ av_log_ask_for_sample(s->avctx, "unsupported quant step coding\n"); return AVERROR_INVALIDDATA; } /** decode quantization step */ step = get_sbits(&s->gb, 6); quant_step += step; if (step == -32 || step == 31) { const int sign = (step == 31) - 1; int quant = 0; while (get_bits_count(&s->gb) + 5 < s->num_saved_bits && (step = get_bits(&s->gb, 5)) == 31) { quant += 31; } quant_step += ((quant + step) ^ sign) - sign; } if (quant_step < 0) { av_log(s->avctx, AV_LOG_DEBUG, "negative quant step\n"); } /** decode quantization step modifiers for every channel */ if (s->channels_for_cur_subframe == 1) { s->channel[s->channel_indexes_for_cur_subframe[0]].quant_step = quant_step; } else { int modifier_len = get_bits(&s->gb, 3); for (i = 0; i < s->channels_for_cur_subframe; i++) { int c = s->channel_indexes_for_cur_subframe[i]; s->channel[c].quant_step = quant_step; if (get_bits1(&s->gb)) { if (modifier_len) { s->channel[c].quant_step += get_bits(&s->gb, modifier_len) + 1; } else ++s->channel[c].quant_step; } } } /** decode scale factors */ if (decode_scale_factors(s) < 0) return AVERROR_INVALIDDATA; } dprintf(s->avctx, "BITSTREAM: subframe header length was %i\n", get_bits_count(&s->gb) - s->subframe_offset); /** parse coefficients */ for (i = 0; i < s->channels_for_cur_subframe; i++) { int c = s->channel_indexes_for_cur_subframe[i]; if (s->channel[c].transmit_coefs && get_bits_count(&s->gb) < s->num_saved_bits) { decode_coeffs(s, c); } else memset(s->channel[c].coeffs, 0, sizeof(*s->channel[c].coeffs) * subframe_len); } dprintf(s->avctx, "BITSTREAM: subframe length was %i\n", get_bits_count(&s->gb) - s->subframe_offset); if (transmit_coeffs) { /** reconstruct the per channel data */ inverse_channel_transform(s); for (i = 0; i < s->channels_for_cur_subframe; i++) { int c = s->channel_indexes_for_cur_subframe[i]; const int* sf = s->channel[c].scale_factors; int b; if (c == s->lfe_channel) memset(&s->tmp[cur_subwoofer_cutoff], 0, sizeof(*s->tmp) * (subframe_len - cur_subwoofer_cutoff)); /** inverse quantization and rescaling */ for (b = 0; b < s->num_bands; b++) { const int end = FFMIN(s->cur_sfb_offsets[b+1], s->subframe_len); const int exp = s->channel[c].quant_step - (s->channel[c].max_scale_factor - *sf++) * s->channel[c].scale_factor_step; const float quant = pow(10.0, exp / 20.0); int start = s->cur_sfb_offsets[b]; s->dsp.vector_fmul_scalar(s->tmp + start, s->channel[c].coeffs + start, quant, end - start); } /** apply imdct (ff_imdct_half == DCTIV with reverse) */ ff_imdct_half(&s->mdct_ctx[av_log2(subframe_len) - BLOCK_MIN_BITS], s->channel[c].coeffs, s->tmp); } } /** window and overlapp-add */ wmapro_window(s); /** 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(WMAProDecodeCtx *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); dprintf(s->avctx, "decoding frame with length %x\n", len); /** decode tile information */ if (decode_tilehdr(s)) { s->packet_loss = 1; return 0; } /** read postproc transform */ if (s->num_channels > 1 && get_bits1(gb)) { av_log_ask_for_sample(s->avctx, "Unsupported postproc transform found\n"); s->packet_loss = 1; return 0; } /** read drc info */ if (s->dynamic_range_compression) { s->drc_gain = get_bits(gb, 8); dprintf(s->avctx, "drc_gain %i\n", s->drc_gain); } /** 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); } } dprintf(s->avctx, "BITSTREAM: frame header length was %i\n", get_bits_count(gb) - s->frame_offset); /** 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; } } /** interleave samples and write them to the output buffer */ for (i = 0; i < s->num_channels; i++) { float* ptr; int incr = s->num_channels; float* iptr = s->channel[i].out; int x; ptr = s->samples + i; for (x = 0; x < s->samples_per_frame; x++) { *ptr = av_clipf(*iptr++, -1.0, 32767.0 / 32768.0); ptr += incr; } /** reuse second half of the IMDCT output for the next frame */ memcpy(&s->channel[i].out[0], &s->channel[i].out[s->samples_per_frame], s->samples_per_frame * sizeof(*s->channel[i].out) >> 1); } if (s->skip_frame) { s->skip_frame = 0; } else s->samples += s->num_channels * s->samples_per_frame; 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); /** 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(WMAProDecodeCtx *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(WMAProDecodeCtx *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) { ff_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; ff_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) { WMAProDecodeCtx *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; s->buf_bit_size = buf_size << 3; /** sanity check for the buffer length */ if (buf_size < avctx->block_align) return 0; buf_size = avctx->block_align; /** parse packet header */ init_get_bits(gb, buf, s->buf_bit_size); packet_sequence_number = get_bits(gb, 4); skip_bits(gb, 2); /** get number of bits that need to be added to the previous frame */ num_bits_prev_frame = get_bits(gb, s->log2_frame_size); dprintf(avctx, "packet[%d]: nbpf %x\n", avctx->frame_number, num_bits_prev_frame); /** 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) { /** 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); dprintf(avctx, "accumulated %x bits of frame data\n", s->num_saved_bits - s->frame_offset); /** 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); } s->packet_loss = 0; } else { int frame_size; s->buf_bit_size = avpkt->size << 3; init_get_bits(gb, avpkt->data, s->buf_bit_size); skip_bits(gb, s->packet_offset); if (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 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 = (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) { WMAProDecodeCtx *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 wmapro decoder */ AVCodec wmapro_decoder = { "wmapro", CODEC_TYPE_AUDIO, CODEC_ID_WMAPRO, sizeof(WMAProDecodeCtx), decode_init, NULL, decode_end, decode_packet, .capabilities = CODEC_CAP_SUBFRAMES, .flush= flush, .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 9 Professional"), };