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author | Justin Ruggles <justin.ruggles@gmail.com> | 2008-08-20 00:32:17 +0000 |
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committer | Justin Ruggles <justin.ruggles@gmail.com> | 2008-08-20 00:32:17 +0000 |
commit | bf09b550e638d3b70dac5c973a4a3f3b9415a0e2 (patch) | |
tree | ec7b72f398871f545e2d86d54f8e95ad0fe4259a /libavcodec/ac3dec.c | |
parent | e630027686a4798cf213b8f76a7bf34ee61ced80 (diff) | |
download | ffmpeg-bf09b550e638d3b70dac5c973a4a3f3b9415a0e2.tar.gz |
commit the OKed parts of the E-AC-3 decoder
Originally committed as revision 14860 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/ac3dec.c')
-rw-r--r-- | libavcodec/ac3dec.c | 62 |
1 files changed, 54 insertions, 8 deletions
diff --git a/libavcodec/ac3dec.c b/libavcodec/ac3dec.c index ddee8b2b70..32ad5b7231 100644 --- a/libavcodec/ac3dec.c +++ b/libavcodec/ac3dec.c @@ -1,8 +1,10 @@ /* * AC-3 Audio Decoder - * This code is developed as part of Google Summer of Code 2006 Program. + * This code was developed as part of Google Summer of Code 2006. + * E-AC-3 support was added as part of Google Summer of Code 2007. * * Copyright (c) 2006 Kartikey Mahendra BHATT (bhattkm at gmail dot com). + * Copyright (c) 2007-2008 Bartlomiej Wolowiec <bartek.wolowiec@gmail.com> * Copyright (c) 2007 Justin Ruggles <justin.ruggles@gmail.com> * * Portions of this code are derived from liba52 @@ -302,10 +304,23 @@ static int parse_frame_header(AC3DecodeContext *s) s->channel_in_cpl[s->lfe_ch] = 0; } - if(hdr.bitstream_id > 10) - return AC3_PARSE_ERROR_BSID; - + if (hdr.bitstream_id <= 10) { + s->eac3 = 0; + s->snr_offset_strategy = 2; + s->block_switch_syntax = 1; + s->dither_flag_syntax = 1; + s->bit_allocation_syntax = 1; + s->fast_gain_syntax = 0; + s->first_cpl_leak = 0; + s->dba_syntax = 1; + s->skip_syntax = 1; + memset(s->channel_uses_aht, 0, sizeof(s->channel_uses_aht)); return ac3_parse_header(s); + } else { + /*s->eac3 = 1; + return ff_eac3_parse_header(s);*/ + return AC3_PARSE_ERROR_BSID; + } } /** @@ -533,6 +548,25 @@ static void remove_dithering(AC3DecodeContext *s) { } } +#if 0 +static void get_transform_coeffs_ch(AC3DecodeContext *s, int blk, int ch, + mant_groups *m) +{ + if (!s->channel_uses_aht[ch]) { + ac3_get_transform_coeffs_ch(s, ch, m); + } else { + /* if AHT is used, mantissas for all blocks are encoded in the first + block of the frame. */ + int bin; + if (!blk) + ff_eac3_get_transform_coeffs_aht_ch(s, ch); + for (bin = s->start_freq[ch]; bin < s->end_freq[ch]; bin++) { + s->fixed_coeffs[ch][bin] = s->pre_mantissa[ch][bin][blk] >> s->dexps[ch][bin]; + } + } +} +#endif + /** * Get the transform coefficients. */ @@ -698,19 +732,23 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) /* block switch flags */ different_transforms = 0; + if (s->block_switch_syntax) { for (ch = 1; ch <= fbw_channels; ch++) { s->block_switch[ch] = get_bits1(gbc); if(ch > 1 && s->block_switch[ch] != s->block_switch[1]) different_transforms = 1; } + } /* dithering flags */ + if (s->dither_flag_syntax) { s->dither_all = 1; for (ch = 1; ch <= fbw_channels; ch++) { s->dither_flag[ch] = get_bits1(gbc); if(!s->dither_flag[ch]) s->dither_all = 0; } + } /* dynamic range */ i = !(s->channel_mode); @@ -870,6 +908,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) } /* bit allocation information */ + if (s->bit_allocation_syntax) { if (get_bits1(gbc)) { s->bit_alloc_params.slow_decay = ff_ac3_slow_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift; s->bit_alloc_params.fast_decay = ff_ac3_fast_decay_tab[get_bits(gbc, 2)] >> s->bit_alloc_params.sr_shift; @@ -882,6 +921,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) av_log(s->avctx, AV_LOG_ERROR, "new bit allocation info must be present in block 0\n"); return -1; } + } /* signal-to-noise ratio offsets and fast gains (signal-to-mask ratios) */ if (get_bits1(gbc)) { @@ -910,7 +950,7 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) } /* delta bit allocation information */ - if (get_bits1(gbc)) { + if (s->dba_syntax && get_bits1(gbc)) { /* delta bit allocation exists (strategy) */ for (ch = !cpl_in_use; ch <= fbw_channels; ch++) { s->dba_mode[ch] = get_bits(gbc, 2); @@ -959,16 +999,18 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) } if(bit_alloc_stages[ch] > 0) { /* Compute bit allocation */ + const uint8_t *bap_tab = s->channel_uses_aht[ch] ? + ff_eac3_hebap_tab : ff_ac3_bap_tab; ff_ac3_bit_alloc_calc_bap(s->mask[ch], s->psd[ch], s->start_freq[ch], s->end_freq[ch], s->snr_offset[ch], s->bit_alloc_params.floor, - ff_ac3_bap_tab, s->bap[ch]); + bap_tab, s->bap[ch]); } } /* unused dummy data */ - if (get_bits1(gbc)) { + if (s->skip_syntax && get_bits1(gbc)) { int skipl = get_bits(gbc, 9); while(skipl--) skip_bits(gbc, 8); @@ -978,6 +1020,10 @@ static int decode_audio_block(AC3DecodeContext *s, int blk) this also uncouples channels if coupling is in use. */ get_transform_coeffs(s); + /* TODO: generate enhanced coupling coordinates and uncouple */ + + /* TODO: apply spectral extension */ + /* recover coefficients if rematrixing is in use */ if(s->channel_mode == AC3_CHMODE_STEREO) do_rematrixing(s); @@ -1161,5 +1207,5 @@ AVCodec ac3_decoder = { .init = ac3_decode_init, .close = ac3_decode_end, .decode = ac3_decode_frame, - .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52 / AC-3"), + .long_name = NULL_IF_CONFIG_SMALL("ATSC A/52 (AC-3, E-AC-3)"), }; |