diff options
Diffstat (limited to 'contrib/ffmpeg/libavcodec/wmadec.c')
| -rw-r--r-- | contrib/ffmpeg/libavcodec/wmadec.c | 912 |
1 files changed, 912 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavcodec/wmadec.c b/contrib/ffmpeg/libavcodec/wmadec.c new file mode 100644 index 000000000..ef3cc7a33 --- /dev/null +++ b/contrib/ffmpeg/libavcodec/wmadec.c @@ -0,0 +1,912 @@ +/* + * WMA compatible decoder + * Copyright (c) 2002 The FFmpeg Project. + * + * 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 wmadec.c + * WMA compatible decoder. + * This decoder handles Microsoft Windows Media Audio data, versions 1 & 2. + * WMA v1 is identified by audio format 0x160 in Microsoft media files + * (ASF/AVI/WAV). WMA v2 is identified by audio format 0x161. + * + * To use this decoder, a calling application must supply the extra data + * bytes provided with the WMA data. These are the extra, codec-specific + * bytes at the end of a WAVEFORMATEX data structure. Transmit these bytes + * to the decoder using the extradata[_size] fields in AVCodecContext. There + * should be 4 extra bytes for v1 data and 6 extra bytes for v2 data. + */ + +#include "avcodec.h" +#include "wma.h" + +#undef NDEBUG +#include <assert.h> + +#define EXPVLCBITS 8 +#define EXPMAX ((19+EXPVLCBITS-1)/EXPVLCBITS) + +#define HGAINVLCBITS 9 +#define HGAINMAX ((13+HGAINVLCBITS-1)/HGAINVLCBITS) + +static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len); + +#ifdef TRACE +static void dump_shorts(WMADecodeContext *s, const char *name, const short *tab, int n) +{ + int i; + + tprintf(s->avctx, "%s[%d]:\n", name, n); + for(i=0;i<n;i++) { + if ((i & 7) == 0) + tprintf(s->avctx, "%4d: ", i); + tprintf(s->avctx, " %5d.0", tab[i]); + if ((i & 7) == 7) + tprintf(s->avctx, "\n"); + } +} + +static void dump_floats(WMADecodeContext *s, const char *name, int prec, const float *tab, int n) +{ + int i; + + tprintf(s->avctx, "%s[%d]:\n", name, n); + for(i=0;i<n;i++) { + if ((i & 7) == 0) + tprintf(s->avctx, "%4d: ", i); + tprintf(s->avctx, " %8.*f", prec, tab[i]); + if ((i & 7) == 7) + tprintf(s->avctx, "\n"); + } + if ((i & 7) != 0) + tprintf(s->avctx, "\n"); +} +#endif + +static int wma_decode_init(AVCodecContext * avctx) +{ + WMACodecContext *s = avctx->priv_data; + int i, flags1, flags2; + uint8_t *extradata; + + s->avctx = avctx; + + /* extract flag infos */ + flags1 = 0; + flags2 = 0; + extradata = avctx->extradata; + if (avctx->codec->id == CODEC_ID_WMAV1 && avctx->extradata_size >= 4) { + flags1 = extradata[0] | (extradata[1] << 8); + flags2 = extradata[2] | (extradata[3] << 8); + } else if (avctx->codec->id == CODEC_ID_WMAV2 && avctx->extradata_size >= 6) { + flags1 = extradata[0] | (extradata[1] << 8) | + (extradata[2] << 16) | (extradata[3] << 24); + flags2 = extradata[4] | (extradata[5] << 8); + } +// for(i=0; i<avctx->extradata_size; i++) +// av_log(NULL, AV_LOG_ERROR, "%02X ", extradata[i]); + + s->use_exp_vlc = flags2 & 0x0001; + s->use_bit_reservoir = flags2 & 0x0002; + s->use_variable_block_len = flags2 & 0x0004; + + ff_wma_init(avctx, flags2); + + /* init MDCT */ + for(i = 0; i < s->nb_block_sizes; i++) + ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 1); + + if (s->use_noise_coding) { + init_vlc(&s->hgain_vlc, HGAINVLCBITS, sizeof(ff_wma_hgain_huffbits), + ff_wma_hgain_huffbits, 1, 1, + ff_wma_hgain_huffcodes, 2, 2, 0); + } + + if (s->use_exp_vlc) { + init_vlc(&s->exp_vlc, EXPVLCBITS, sizeof(ff_wma_scale_huffbits), //FIXME move out of context + ff_wma_scale_huffbits, 1, 1, + ff_wma_scale_huffcodes, 4, 4, 0); + } else { + wma_lsp_to_curve_init(s, s->frame_len); + } + + return 0; +} + +/** + * interpolate values for a bigger or smaller block. The block must + * have multiple sizes + */ +static void interpolate_array(float *scale, int old_size, int new_size) +{ + int i, j, jincr, k; + float v; + + if (new_size > old_size) { + jincr = new_size / old_size; + j = new_size; + for(i = old_size - 1; i >=0; i--) { + v = scale[i]; + k = jincr; + do { + scale[--j] = v; + } while (--k); + } + } else if (new_size < old_size) { + j = 0; + jincr = old_size / new_size; + for(i = 0; i < new_size; i++) { + scale[i] = scale[j]; + j += jincr; + } + } +} + +/** + * compute x^-0.25 with an exponent and mantissa table. We use linear + * interpolation to reduce the mantissa table size at a small speed + * expense (linear interpolation approximately doubles the number of + * bits of precision). + */ +static inline float pow_m1_4(WMACodecContext *s, float x) +{ + union { + float f; + unsigned int v; + } u, t; + unsigned int e, m; + float a, b; + + u.f = x; + e = u.v >> 23; + m = (u.v >> (23 - LSP_POW_BITS)) & ((1 << LSP_POW_BITS) - 1); + /* build interpolation scale: 1 <= t < 2. */ + t.v = ((u.v << LSP_POW_BITS) & ((1 << 23) - 1)) | (127 << 23); + a = s->lsp_pow_m_table1[m]; + b = s->lsp_pow_m_table2[m]; + return s->lsp_pow_e_table[e] * (a + b * t.f); +} + +static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len) +{ + float wdel, a, b; + int i, e, m; + + wdel = M_PI / frame_len; + for(i=0;i<frame_len;i++) + s->lsp_cos_table[i] = 2.0f * cos(wdel * i); + + /* tables for x^-0.25 computation */ + for(i=0;i<256;i++) { + e = i - 126; + s->lsp_pow_e_table[i] = pow(2.0, e * -0.25); + } + + /* NOTE: these two tables are needed to avoid two operations in + pow_m1_4 */ + b = 1.0; + for(i=(1 << LSP_POW_BITS) - 1;i>=0;i--) { + m = (1 << LSP_POW_BITS) + i; + a = (float)m * (0.5 / (1 << LSP_POW_BITS)); + a = pow(a, -0.25); + s->lsp_pow_m_table1[i] = 2 * a - b; + s->lsp_pow_m_table2[i] = b - a; + b = a; + } +#if 0 + for(i=1;i<20;i++) { + float v, r1, r2; + v = 5.0 / i; + r1 = pow_m1_4(s, v); + r2 = pow(v,-0.25); + printf("%f^-0.25=%f e=%f\n", v, r1, r2 - r1); + } +#endif +} + +/** + * NOTE: We use the same code as Vorbis here + * @todo optimize it further with SSE/3Dnow + */ +static void wma_lsp_to_curve(WMACodecContext *s, + float *out, float *val_max_ptr, + int n, float *lsp) +{ + int i, j; + float p, q, w, v, val_max; + + val_max = 0; + for(i=0;i<n;i++) { + p = 0.5f; + q = 0.5f; + w = s->lsp_cos_table[i]; + for(j=1;j<NB_LSP_COEFS;j+=2){ + q *= w - lsp[j - 1]; + p *= w - lsp[j]; + } + p *= p * (2.0f - w); + q *= q * (2.0f + w); + v = p + q; + v = pow_m1_4(s, v); + if (v > val_max) + val_max = v; + out[i] = v; + } + *val_max_ptr = val_max; +} + +/** + * decode exponents coded with LSP coefficients (same idea as Vorbis) + */ +static void decode_exp_lsp(WMACodecContext *s, int ch) +{ + float lsp_coefs[NB_LSP_COEFS]; + int val, i; + + for(i = 0; i < NB_LSP_COEFS; i++) { + if (i == 0 || i >= 8) + val = get_bits(&s->gb, 3); + else + val = get_bits(&s->gb, 4); + lsp_coefs[i] = ff_wma_lsp_codebook[i][val]; + } + + wma_lsp_to_curve(s, s->exponents[ch], &s->max_exponent[ch], + s->block_len, lsp_coefs); +} + +/** + * decode exponents coded with VLC codes + */ +static int decode_exp_vlc(WMACodecContext *s, int ch) +{ + int last_exp, n, code; + const uint16_t *ptr, *band_ptr; + float v, *q, max_scale, *q_end; + + band_ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits]; + ptr = band_ptr; + q = s->exponents[ch]; + q_end = q + s->block_len; + max_scale = 0; + if (s->version == 1) { + last_exp = get_bits(&s->gb, 5) + 10; + /* XXX: use a table */ + v = pow(10, last_exp * (1.0 / 16.0)); + max_scale = v; + n = *ptr++; + do { + *q++ = v; + } while (--n); + }else + last_exp = 36; + + while (q < q_end) { + code = get_vlc2(&s->gb, s->exp_vlc.table, EXPVLCBITS, EXPMAX); + if (code < 0) + return -1; + /* NOTE: this offset is the same as MPEG4 AAC ! */ + last_exp += code - 60; + /* XXX: use a table */ + v = pow(10, last_exp * (1.0 / 16.0)); + if (v > max_scale) + max_scale = v; + n = *ptr++; + do { + *q++ = v; + } while (--n); + } + s->max_exponent[ch] = max_scale; + return 0; +} + + +/** + * Apply MDCT window and add into output. + * + * We ensure that when the windows overlap their squared sum + * is always 1 (MDCT reconstruction rule). + */ +static void wma_window(WMACodecContext *s, float *out) +{ + float *in = s->output; + int block_len, bsize, n; + + /* left part */ + if (s->block_len_bits <= s->prev_block_len_bits) { + block_len = s->block_len; + bsize = s->frame_len_bits - s->block_len_bits; + + s->dsp.vector_fmul_add_add(out, in, s->windows[bsize], + out, 0, block_len, 1); + + } else { + block_len = 1 << s->prev_block_len_bits; + n = (s->block_len - block_len) / 2; + bsize = s->frame_len_bits - s->prev_block_len_bits; + + s->dsp.vector_fmul_add_add(out+n, in+n, s->windows[bsize], + out+n, 0, block_len, 1); + + memcpy(out+n+block_len, in+n+block_len, n*sizeof(float)); + } + + out += s->block_len; + in += s->block_len; + + /* right part */ + if (s->block_len_bits <= s->next_block_len_bits) { + block_len = s->block_len; + bsize = s->frame_len_bits - s->block_len_bits; + + s->dsp.vector_fmul_reverse(out, in, s->windows[bsize], block_len); + + } else { + block_len = 1 << s->next_block_len_bits; + n = (s->block_len - block_len) / 2; + bsize = s->frame_len_bits - s->next_block_len_bits; + + memcpy(out, in, n*sizeof(float)); + + s->dsp.vector_fmul_reverse(out+n, in+n, s->windows[bsize], block_len); + + memset(out+n+block_len, 0, n*sizeof(float)); + } +} + + +/** + * @return 0 if OK. 1 if last block of frame. return -1 if + * unrecorrable error. + */ +static int wma_decode_block(WMACodecContext *s) +{ + int n, v, a, ch, code, bsize; + int coef_nb_bits, total_gain, parse_exponents; + int nb_coefs[MAX_CHANNELS]; + float mdct_norm; + +#ifdef TRACE + tprintf(s->avctx, "***decode_block: %d:%d\n", s->frame_count - 1, s->block_num); +#endif + + /* compute current block length */ + if (s->use_variable_block_len) { + n = av_log2(s->nb_block_sizes - 1) + 1; + + if (s->reset_block_lengths) { + s->reset_block_lengths = 0; + v = get_bits(&s->gb, n); + if (v >= s->nb_block_sizes) + return -1; + s->prev_block_len_bits = s->frame_len_bits - v; + v = get_bits(&s->gb, n); + if (v >= s->nb_block_sizes) + return -1; + s->block_len_bits = s->frame_len_bits - v; + } else { + /* update block lengths */ + s->prev_block_len_bits = s->block_len_bits; + s->block_len_bits = s->next_block_len_bits; + } + v = get_bits(&s->gb, n); + if (v >= s->nb_block_sizes) + return -1; + s->next_block_len_bits = s->frame_len_bits - v; + } else { + /* fixed block len */ + s->next_block_len_bits = s->frame_len_bits; + s->prev_block_len_bits = s->frame_len_bits; + s->block_len_bits = s->frame_len_bits; + } + + /* now check if the block length is coherent with the frame length */ + s->block_len = 1 << s->block_len_bits; + if ((s->block_pos + s->block_len) > s->frame_len) + return -1; + + if (s->nb_channels == 2) { + s->ms_stereo = get_bits(&s->gb, 1); + } + v = 0; + for(ch = 0; ch < s->nb_channels; ch++) { + a = get_bits(&s->gb, 1); + s->channel_coded[ch] = a; + v |= a; + } + /* if no channel coded, no need to go further */ + /* XXX: fix potential framing problems */ + if (!v) + goto next; + + bsize = s->frame_len_bits - s->block_len_bits; + + /* read total gain and extract corresponding number of bits for + coef escape coding */ + total_gain = 1; + for(;;) { + a = get_bits(&s->gb, 7); + total_gain += a; + if (a != 127) + break; + } + + coef_nb_bits= ff_wma_total_gain_to_bits(total_gain); + + /* compute number of coefficients */ + n = s->coefs_end[bsize] - s->coefs_start; + for(ch = 0; ch < s->nb_channels; ch++) + nb_coefs[ch] = n; + + /* complex coding */ + if (s->use_noise_coding) { + + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + int i, n, a; + n = s->exponent_high_sizes[bsize]; + for(i=0;i<n;i++) { + a = get_bits(&s->gb, 1); + s->high_band_coded[ch][i] = a; + /* if noise coding, the coefficients are not transmitted */ + if (a) + nb_coefs[ch] -= s->exponent_high_bands[bsize][i]; + } + } + } + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + int i, n, val, code; + + n = s->exponent_high_sizes[bsize]; + val = (int)0x80000000; + for(i=0;i<n;i++) { + if (s->high_band_coded[ch][i]) { + if (val == (int)0x80000000) { + val = get_bits(&s->gb, 7) - 19; + } else { + code = get_vlc2(&s->gb, s->hgain_vlc.table, HGAINVLCBITS, HGAINMAX); + if (code < 0) + return -1; + val += code - 18; + } + s->high_band_values[ch][i] = val; + } + } + } + } + } + + /* exposant can be interpolated in short blocks. */ + parse_exponents = 1; + if (s->block_len_bits != s->frame_len_bits) { + parse_exponents = get_bits(&s->gb, 1); + } + + if (parse_exponents) { + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + if (s->use_exp_vlc) { + if (decode_exp_vlc(s, ch) < 0) + return -1; + } else { + decode_exp_lsp(s, ch); + } + } + } + } else { + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + interpolate_array(s->exponents[ch], 1 << s->prev_block_len_bits, + s->block_len); + } + } + } + + /* parse spectral coefficients : just RLE encoding */ + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + VLC *coef_vlc; + int level, run, sign, tindex; + int16_t *ptr, *eptr; + const uint16_t *level_table, *run_table; + + /* special VLC tables are used for ms stereo because + there is potentially less energy there */ + tindex = (ch == 1 && s->ms_stereo); + coef_vlc = &s->coef_vlc[tindex]; + run_table = s->run_table[tindex]; + level_table = s->level_table[tindex]; + /* XXX: optimize */ + ptr = &s->coefs1[ch][0]; + eptr = ptr + nb_coefs[ch]; + memset(ptr, 0, s->block_len * sizeof(int16_t)); + for(;;) { + code = get_vlc2(&s->gb, coef_vlc->table, VLCBITS, VLCMAX); + if (code < 0) + return -1; + if (code == 1) { + /* EOB */ + break; + } else if (code == 0) { + /* escape */ + level = get_bits(&s->gb, coef_nb_bits); + /* NOTE: this is rather suboptimal. reading + block_len_bits would be better */ + run = get_bits(&s->gb, s->frame_len_bits); + } else { + /* normal code */ + run = run_table[code]; + level = level_table[code]; + } + sign = get_bits(&s->gb, 1); + if (!sign) + level = -level; + ptr += run; + if (ptr >= eptr) + { + av_log(NULL, AV_LOG_ERROR, "overflow in spectral RLE, ignoring\n"); + break; + } + *ptr++ = level; + /* NOTE: EOB can be omitted */ + if (ptr >= eptr) + break; + } + } + if (s->version == 1 && s->nb_channels >= 2) { + align_get_bits(&s->gb); + } + } + + /* normalize */ + { + int n4 = s->block_len / 2; + mdct_norm = 1.0 / (float)n4; + if (s->version == 1) { + mdct_norm *= sqrt(n4); + } + } + + /* finally compute the MDCT coefficients */ + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + int16_t *coefs1; + float *coefs, *exponents, mult, mult1, noise, *exp_ptr; + int i, j, n, n1, last_high_band; + float exp_power[HIGH_BAND_MAX_SIZE]; + + coefs1 = s->coefs1[ch]; + exponents = s->exponents[ch]; + mult = pow(10, total_gain * 0.05) / s->max_exponent[ch]; + mult *= mdct_norm; + coefs = s->coefs[ch]; + if (s->use_noise_coding) { + mult1 = mult; + /* very low freqs : noise */ + for(i = 0;i < s->coefs_start; i++) { + *coefs++ = s->noise_table[s->noise_index] * (*exponents++) * mult1; + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + } + + n1 = s->exponent_high_sizes[bsize]; + + /* compute power of high bands */ + exp_ptr = exponents + + s->high_band_start[bsize] - + s->coefs_start; + last_high_band = 0; /* avoid warning */ + for(j=0;j<n1;j++) { + n = s->exponent_high_bands[s->frame_len_bits - + s->block_len_bits][j]; + if (s->high_band_coded[ch][j]) { + float e2, v; + e2 = 0; + for(i = 0;i < n; i++) { + v = exp_ptr[i]; + e2 += v * v; + } + exp_power[j] = e2 / n; + last_high_band = j; + tprintf(s->avctx, "%d: power=%f (%d)\n", j, exp_power[j], n); + } + exp_ptr += n; + } + + /* main freqs and high freqs */ + for(j=-1;j<n1;j++) { + if (j < 0) { + n = s->high_band_start[bsize] - + s->coefs_start; + } else { + n = s->exponent_high_bands[s->frame_len_bits - + s->block_len_bits][j]; + } + if (j >= 0 && s->high_band_coded[ch][j]) { + /* use noise with specified power */ + mult1 = sqrt(exp_power[j] / exp_power[last_high_band]); + /* XXX: use a table */ + mult1 = mult1 * pow(10, s->high_band_values[ch][j] * 0.05); + mult1 = mult1 / (s->max_exponent[ch] * s->noise_mult); + mult1 *= mdct_norm; + for(i = 0;i < n; i++) { + noise = s->noise_table[s->noise_index]; + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + *coefs++ = (*exponents++) * noise * mult1; + } + } else { + /* coded values + small noise */ + for(i = 0;i < n; i++) { + noise = s->noise_table[s->noise_index]; + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + *coefs++ = ((*coefs1++) + noise) * (*exponents++) * mult; + } + } + } + + /* very high freqs : noise */ + n = s->block_len - s->coefs_end[bsize]; + mult1 = mult * exponents[-1]; + for(i = 0; i < n; i++) { + *coefs++ = s->noise_table[s->noise_index] * mult1; + s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1); + } + } else { + /* XXX: optimize more */ + for(i = 0;i < s->coefs_start; i++) + *coefs++ = 0.0; + n = nb_coefs[ch]; + for(i = 0;i < n; i++) { + *coefs++ = coefs1[i] * exponents[i] * mult; + } + n = s->block_len - s->coefs_end[bsize]; + for(i = 0;i < n; i++) + *coefs++ = 0.0; + } + } + } + +#ifdef TRACE + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + dump_floats(s, "exponents", 3, s->exponents[ch], s->block_len); + dump_floats(s, "coefs", 1, s->coefs[ch], s->block_len); + } + } +#endif + + if (s->ms_stereo && s->channel_coded[1]) { + float a, b; + int i; + + /* nominal case for ms stereo: we do it before mdct */ + /* no need to optimize this case because it should almost + never happen */ + if (!s->channel_coded[0]) { + tprintf(s->avctx, "rare ms-stereo case happened\n"); + memset(s->coefs[0], 0, sizeof(float) * s->block_len); + s->channel_coded[0] = 1; + } + + for(i = 0; i < s->block_len; i++) { + a = s->coefs[0][i]; + b = s->coefs[1][i]; + s->coefs[0][i] = a + b; + s->coefs[1][i] = a - b; + } + } + + for(ch = 0; ch < s->nb_channels; ch++) { + if (s->channel_coded[ch]) { + int n4, index, n; + + n = s->block_len; + n4 = s->block_len / 2; + s->mdct_ctx[bsize].fft.imdct_calc(&s->mdct_ctx[bsize], + s->output, s->coefs[ch], s->mdct_tmp); + + /* multiply by the window and add in the frame */ + index = (s->frame_len / 2) + s->block_pos - n4; + wma_window(s, &s->frame_out[ch][index]); + + /* specific fast case for ms-stereo : add to second + channel if it is not coded */ + if (s->ms_stereo && !s->channel_coded[1]) { + wma_window(s, &s->frame_out[1][index]); + } + } + } + next: + /* update block number */ + s->block_num++; + s->block_pos += s->block_len; + if (s->block_pos >= s->frame_len) + return 1; + else + return 0; +} + +/* decode a frame of frame_len samples */ +static int wma_decode_frame(WMACodecContext *s, int16_t *samples) +{ + int ret, i, n, a, ch, incr; + int16_t *ptr; + float *iptr; + +#ifdef TRACE + tprintf(s->avctx, "***decode_frame: %d size=%d\n", s->frame_count++, s->frame_len); +#endif + + /* read each block */ + s->block_num = 0; + s->block_pos = 0; + for(;;) { + ret = wma_decode_block(s); + if (ret < 0) + return -1; + if (ret) + break; + } + + /* convert frame to integer */ + n = s->frame_len; + incr = s->nb_channels; + for(ch = 0; ch < s->nb_channels; ch++) { + ptr = samples + ch; + iptr = s->frame_out[ch]; + + for(i=0;i<n;i++) { + a = lrintf(*iptr++); + if (a > 32767) + a = 32767; + else if (a < -32768) + a = -32768; + *ptr = a; + ptr += incr; + } + /* prepare for next block */ + memmove(&s->frame_out[ch][0], &s->frame_out[ch][s->frame_len], + s->frame_len * sizeof(float)); + } + +#ifdef TRACE + dump_shorts(s, "samples", samples, n * s->nb_channels); +#endif + return 0; +} + +static int wma_decode_superframe(AVCodecContext *avctx, + void *data, int *data_size, + uint8_t *buf, int buf_size) +{ + WMACodecContext *s = avctx->priv_data; + int nb_frames, bit_offset, i, pos, len; + uint8_t *q; + int16_t *samples; + + tprintf(avctx, "***decode_superframe:\n"); + + if(buf_size==0){ + s->last_superframe_len = 0; + return 0; + } + + samples = data; + + init_get_bits(&s->gb, buf, buf_size*8); + + if (s->use_bit_reservoir) { + /* read super frame header */ + get_bits(&s->gb, 4); /* super frame index */ + nb_frames = get_bits(&s->gb, 4) - 1; + + bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3); + + if (s->last_superframe_len > 0) { + // printf("skip=%d\n", s->last_bitoffset); + /* add bit_offset bits to last frame */ + if ((s->last_superframe_len + ((bit_offset + 7) >> 3)) > + MAX_CODED_SUPERFRAME_SIZE) + goto fail; + q = s->last_superframe + s->last_superframe_len; + len = bit_offset; + while (len > 7) { + *q++ = (get_bits)(&s->gb, 8); + len -= 8; + } + if (len > 0) { + *q++ = (get_bits)(&s->gb, len) << (8 - len); + } + + /* XXX: bit_offset bits into last frame */ + init_get_bits(&s->gb, s->last_superframe, MAX_CODED_SUPERFRAME_SIZE*8); + /* skip unused bits */ + if (s->last_bitoffset > 0) + skip_bits(&s->gb, s->last_bitoffset); + /* this frame is stored in the last superframe and in the + current one */ + if (wma_decode_frame(s, samples) < 0) + goto fail; + samples += s->nb_channels * s->frame_len; + } + + /* read each frame starting from bit_offset */ + pos = bit_offset + 4 + 4 + s->byte_offset_bits + 3; + init_get_bits(&s->gb, buf + (pos >> 3), (MAX_CODED_SUPERFRAME_SIZE - (pos >> 3))*8); + len = pos & 7; + if (len > 0) + skip_bits(&s->gb, len); + + s->reset_block_lengths = 1; + for(i=0;i<nb_frames;i++) { + if (wma_decode_frame(s, samples) < 0) + goto fail; + samples += s->nb_channels * s->frame_len; + } + + /* we copy the end of the frame in the last frame buffer */ + pos = get_bits_count(&s->gb) + ((bit_offset + 4 + 4 + s->byte_offset_bits + 3) & ~7); + s->last_bitoffset = pos & 7; + pos >>= 3; + len = buf_size - pos; + if (len > MAX_CODED_SUPERFRAME_SIZE || len < 0) { + goto fail; + } + s->last_superframe_len = len; + memcpy(s->last_superframe, buf + pos, len); + } else { + /* single frame decode */ + if (wma_decode_frame(s, samples) < 0) + goto fail; + samples += s->nb_channels * s->frame_len; + } + +//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d outbytes:%d eaten:%d\n", s->frame_len_bits, s->block_len_bits, s->frame_len, s->block_len, (int8_t *)samples - (int8_t *)data, s->block_align); + + *data_size = (int8_t *)samples - (int8_t *)data; + return s->block_align; + fail: + /* when error, we reset the bit reservoir */ + s->last_superframe_len = 0; + return -1; +} + +AVCodec wmav1_decoder = +{ + "wmav1", + CODEC_TYPE_AUDIO, + CODEC_ID_WMAV1, + sizeof(WMACodecContext), + wma_decode_init, + NULL, + ff_wma_end, + wma_decode_superframe, +}; + +AVCodec wmav2_decoder = +{ + "wmav2", + CODEC_TYPE_AUDIO, + CODEC_ID_WMAV2, + sizeof(WMACodecContext), + wma_decode_init, + NULL, + ff_wma_end, + wma_decode_superframe, +}; |