Mercurial > libavcodec.hg
view alac.c @ 5230:2a340aea762d libavcodec
simplify ff_copy_bits: merge 2 test branches
author | aurel |
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date | Fri, 06 Jul 2007 14:14:41 +0000 |
parents | 05e932ddaaa9 |
children | 8256f8ec2fac |
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/* * ALAC (Apple Lossless Audio Codec) decoder * Copyright (c) 2005 David Hammerton * All rights reserved. * * 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 alac.c * ALAC (Apple Lossless Audio Codec) decoder * @author 2005 David Hammerton * * For more information on the ALAC format, visit: * http://crazney.net/programs/itunes/alac.html * * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be * passed through the extradata[_size] fields. This atom is tacked onto * the end of an 'alac' stsd atom and has the following format: * bytes 0-3 atom size (0x24), big-endian * bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd) * bytes 8-35 data bytes needed by decoder * * Extradata: * 32bit size * 32bit tag (=alac) * 32bit zero? * 32bit max sample per frame * 8bit ?? (zero?) * 8bit sample size * 8bit history mult * 8bit initial history * 8bit kmodifier * 8bit channels? * 16bit ?? * 32bit max coded frame size * 32bit bitrate? * 32bit samplerate */ #include "avcodec.h" #include "bitstream.h" #define ALAC_EXTRADATA_SIZE 36 typedef struct { AVCodecContext *avctx; GetBitContext gb; /* init to 0; first frame decode should initialize from extradata and * set this to 1 */ int context_initialized; int samplesize; int numchannels; int bytespersample; /* buffers */ int32_t *predicterror_buffer_a; int32_t *predicterror_buffer_b; int32_t *outputsamples_buffer_a; int32_t *outputsamples_buffer_b; /* stuff from setinfo */ uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ uint8_t setinfo_7a; /* 0x00 */ uint8_t setinfo_sample_size; /* 0x10 */ uint8_t setinfo_rice_historymult; /* 0x28 */ uint8_t setinfo_rice_initialhistory; /* 0x0a */ uint8_t setinfo_rice_kmodifier; /* 0x0e */ uint8_t setinfo_7f; /* 0x02 */ uint16_t setinfo_80; /* 0x00ff */ uint32_t setinfo_82; /* 0x000020e7 */ uint32_t setinfo_86; /* 0x00069fe4 */ uint32_t setinfo_8a_rate; /* 0x0000ac44 */ /* end setinfo stuff */ } ALACContext; static void allocate_buffers(ALACContext *alac) { alac->predicterror_buffer_a = av_malloc(alac->setinfo_max_samples_per_frame * 4); alac->predicterror_buffer_b = av_malloc(alac->setinfo_max_samples_per_frame * 4); alac->outputsamples_buffer_a = av_malloc(alac->setinfo_max_samples_per_frame * 4); alac->outputsamples_buffer_b = av_malloc(alac->setinfo_max_samples_per_frame * 4); } static int alac_set_info(ALACContext *alac) { unsigned char *ptr = alac->avctx->extradata; ptr += 4; /* size */ ptr += 4; /* alac */ ptr += 4; /* 0 ? */ if(AV_RB32(ptr) >= UINT_MAX/4){ av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n"); return -1; } alac->setinfo_max_samples_per_frame = AV_RB32(ptr); /* buffer size / 2 ? */ ptr += 4; alac->setinfo_7a = *ptr++; alac->setinfo_sample_size = *ptr++; alac->setinfo_rice_historymult = *ptr++; alac->setinfo_rice_initialhistory = *ptr++; alac->setinfo_rice_kmodifier = *ptr++; alac->setinfo_7f = *ptr++; // channels? alac->setinfo_80 = AV_RB16(ptr); ptr += 2; alac->setinfo_82 = AV_RB32(ptr); // max coded frame size ptr += 4; alac->setinfo_86 = AV_RB32(ptr); // bitrate ? ptr += 4; alac->setinfo_8a_rate = AV_RB32(ptr); // samplerate ptr += 4; allocate_buffers(alac); return 0; } /* hideously inefficient. could use a bitmask search, * alternatively bsr on x86, */ static int count_leading_zeros(int32_t input) { int i = 0; while (!(0x80000000 & input) && i < 32) { i++; input = input << 1; } return i; } static void bastardized_rice_decompress(ALACContext *alac, int32_t *output_buffer, int output_size, int readsamplesize, /* arg_10 */ int rice_initialhistory, /* arg424->b */ int rice_kmodifier, /* arg424->d */ int rice_historymult, /* arg424->c */ int rice_kmodifier_mask /* arg424->e */ ) { int output_count; unsigned int history = rice_initialhistory; int sign_modifier = 0; for (output_count = 0; output_count < output_size; output_count++) { int32_t x = 0; int32_t x_modified; int32_t final_val; /* read x - number of 1s before 0 represent the rice */ while (x <= 8 && get_bits1(&alac->gb)) { x++; } if (x > 8) { /* RICE THRESHOLD */ /* use alternative encoding */ int32_t value; value = get_bits(&alac->gb, readsamplesize); /* mask value to readsamplesize size */ if (readsamplesize != 32) value &= (0xffffffff >> (32 - readsamplesize)); x = value; } else { /* standard rice encoding */ int extrabits; int k; /* size of extra bits */ /* read k, that is bits as is */ k = 31 - rice_kmodifier - count_leading_zeros((history >> 9) + 3); if (k < 0) k += rice_kmodifier; else k = rice_kmodifier; if (k != 1) { extrabits = show_bits(&alac->gb, k); /* multiply x by 2^k - 1, as part of their strange algorithm */ x = (x << k) - x; if (extrabits > 1) { x += extrabits - 1; get_bits(&alac->gb, k); } else { get_bits(&alac->gb, k - 1); } } } x_modified = sign_modifier + x; final_val = (x_modified + 1) / 2; if (x_modified & 1) final_val *= -1; output_buffer[output_count] = final_val; sign_modifier = 0; /* now update the history */ history += (x_modified * rice_historymult) - ((history * rice_historymult) >> 9); if (x_modified > 0xffff) history = 0xffff; /* special case: there may be compressed blocks of 0 */ if ((history < 128) && (output_count+1 < output_size)) { int block_size; sign_modifier = 1; x = 0; while (x <= 8 && get_bits1(&alac->gb)) { x++; } if (x > 8) { block_size = get_bits(&alac->gb, 16); block_size &= 0xffff; } else { int k; int extrabits; k = count_leading_zeros(history) + ((history + 16) >> 6 /* / 64 */) - 24; extrabits = show_bits(&alac->gb, k); block_size = (((1 << k) - 1) & rice_kmodifier_mask) * x + extrabits - 1; if (extrabits < 2) { x = 1 - extrabits; block_size += x; get_bits(&alac->gb, k - 1); } else { get_bits(&alac->gb, k); } } if (block_size > 0) { memset(&output_buffer[output_count+1], 0, block_size * 4); output_count += block_size; } if (block_size > 0xffff) sign_modifier = 0; history = 0; } } } #define SIGN_EXTENDED32(val, bits) ((val << (32 - bits)) >> (32 - bits)) #define SIGN_ONLY(v) \ ((v < 0) ? (-1) : \ ((v > 0) ? (1) : \ (0))) static void predictor_decompress_fir_adapt(int32_t *error_buffer, int32_t *buffer_out, int output_size, int readsamplesize, int16_t *predictor_coef_table, int predictor_coef_num, int predictor_quantitization) { int i; /* first sample always copies */ *buffer_out = *error_buffer; if (!predictor_coef_num) { if (output_size <= 1) return; memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4); return; } if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */ /* second-best case scenario for fir decompression, * error describes a small difference from the previous sample only */ if (output_size <= 1) return; for (i = 0; i < output_size - 1; i++) { int32_t prev_value; int32_t error_value; prev_value = buffer_out[i]; error_value = error_buffer[i+1]; buffer_out[i+1] = SIGN_EXTENDED32((prev_value + error_value), readsamplesize); } return; } /* read warm-up samples */ if (predictor_coef_num > 0) { int i; for (i = 0; i < predictor_coef_num; i++) { int32_t val; val = buffer_out[i] + error_buffer[i+1]; val = SIGN_EXTENDED32(val, readsamplesize); buffer_out[i+1] = val; } } #if 0 /* 4 and 8 are very common cases (the only ones i've seen). these * should be unrolled and optimised */ if (predictor_coef_num == 4) { /* FIXME: optimised general case */ return; } if (predictor_coef_table == 8) { /* FIXME: optimised general case */ return; } #endif /* general case */ if (predictor_coef_num > 0) { for (i = predictor_coef_num + 1; i < output_size; i++) { int j; int sum = 0; int outval; int error_val = error_buffer[i]; for (j = 0; j < predictor_coef_num; j++) { sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) * predictor_coef_table[j]; } outval = (1 << (predictor_quantitization-1)) + sum; outval = outval >> predictor_quantitization; outval = outval + buffer_out[0] + error_val; outval = SIGN_EXTENDED32(outval, readsamplesize); buffer_out[predictor_coef_num+1] = outval; if (error_val > 0) { int predictor_num = predictor_coef_num - 1; while (predictor_num >= 0 && error_val > 0) { int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; int sign = SIGN_ONLY(val); predictor_coef_table[predictor_num] -= sign; val *= sign; /* absolute value */ error_val -= ((val >> predictor_quantitization) * (predictor_coef_num - predictor_num)); predictor_num--; } } else if (error_val < 0) { int predictor_num = predictor_coef_num - 1; while (predictor_num >= 0 && error_val < 0) { int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; int sign = - SIGN_ONLY(val); predictor_coef_table[predictor_num] -= sign; val *= sign; /* neg value */ error_val -= ((val >> predictor_quantitization) * (predictor_coef_num - predictor_num)); predictor_num--; } } buffer_out++; } } } static void deinterlace_16(int32_t *buffer_a, int32_t *buffer_b, int16_t *buffer_out, int numchannels, int numsamples, uint8_t interlacing_shift, uint8_t interlacing_leftweight) { int i; if (numsamples <= 0) return; /* weighted interlacing */ if (interlacing_leftweight) { for (i = 0; i < numsamples; i++) { int32_t difference, midright; int16_t left; int16_t right; midright = buffer_a[i]; difference = buffer_b[i]; right = midright - ((difference * interlacing_leftweight) >> interlacing_shift); left = (midright - ((difference * interlacing_leftweight) >> interlacing_shift)) + difference; buffer_out[i*numchannels] = left; buffer_out[i*numchannels + 1] = right; } return; } /* otherwise basic interlacing took place */ for (i = 0; i < numsamples; i++) { int16_t left, right; left = buffer_a[i]; right = buffer_b[i]; buffer_out[i*numchannels] = left; buffer_out[i*numchannels + 1] = right; } } static int alac_decode_frame(AVCodecContext *avctx, void *outbuffer, int *outputsize, uint8_t *inbuffer, int input_buffer_size) { ALACContext *alac = avctx->priv_data; int channels; int32_t outputsamples; /* short-circuit null buffers */ if (!inbuffer || !input_buffer_size) return input_buffer_size; /* initialize from the extradata */ if (!alac->context_initialized) { if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) { av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n", ALAC_EXTRADATA_SIZE); return input_buffer_size; } if (alac_set_info(alac)) { av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); return input_buffer_size; } alac->context_initialized = 1; } outputsamples = alac->setinfo_max_samples_per_frame; init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); channels = get_bits(&alac->gb, 3); *outputsize = outputsamples * alac->bytespersample; switch(channels) { case 0: { /* 1 channel */ int hassize; int isnotcompressed; int readsamplesize; int wasted_bytes; int ricemodifier; /* 2^result = something to do with output waiting. * perhaps matters if we read > 1 frame in a pass? */ get_bits(&alac->gb, 4); get_bits(&alac->gb, 12); /* unknown, skip 12 bits */ hassize = get_bits(&alac->gb, 1); /* the output sample size is stored soon */ wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */ isnotcompressed = get_bits(&alac->gb, 1); /* whether the frame is compressed */ if (hassize) { /* now read the number of samples, * as a 32bit integer */ outputsamples = get_bits(&alac->gb, 32); *outputsize = outputsamples * alac->bytespersample; } readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8); if (!isnotcompressed) { /* so it is compressed */ int16_t predictor_coef_table[32]; int predictor_coef_num; int prediction_type; int prediction_quantitization; int i; /* FIXME: skip 16 bits, not sure what they are. seem to be used in * two channel case */ get_bits(&alac->gb, 8); get_bits(&alac->gb, 8); prediction_type = get_bits(&alac->gb, 4); prediction_quantitization = get_bits(&alac->gb, 4); ricemodifier = get_bits(&alac->gb, 3); predictor_coef_num = get_bits(&alac->gb, 5); /* read the predictor table */ for (i = 0; i < predictor_coef_num; i++) { predictor_coef_table[i] = (int16_t)get_bits(&alac->gb, 16); } if (wasted_bytes) { /* these bytes seem to have something to do with * > 2 channel files. */ av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n"); } bastardized_rice_decompress(alac, alac->predicterror_buffer_a, outputsamples, readsamplesize, alac->setinfo_rice_initialhistory, alac->setinfo_rice_kmodifier, ricemodifier * alac->setinfo_rice_historymult / 4, (1 << alac->setinfo_rice_kmodifier) - 1); if (prediction_type == 0) { /* adaptive fir */ predictor_decompress_fir_adapt(alac->predicterror_buffer_a, alac->outputsamples_buffer_a, outputsamples, readsamplesize, predictor_coef_table, predictor_coef_num, prediction_quantitization); } else { av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type); /* i think the only other prediction type (or perhaps this is just a * boolean?) runs adaptive fir twice.. like: * predictor_decompress_fir_adapt(predictor_error, tempout, ...) * predictor_decompress_fir_adapt(predictor_error, outputsamples ...) * little strange.. */ } } else { /* not compressed, easy case */ if (readsamplesize <= 16) { int i; for (i = 0; i < outputsamples; i++) { int32_t audiobits = get_bits(&alac->gb, readsamplesize); audiobits = SIGN_EXTENDED32(audiobits, readsamplesize); alac->outputsamples_buffer_a[i] = audiobits; } } else { int i; for (i = 0; i < outputsamples; i++) { int32_t audiobits; audiobits = get_bits(&alac->gb, 16); /* special case of sign extension.. * as we'll be ORing the low 16bits into this */ audiobits = audiobits << 16; audiobits = audiobits >> (32 - readsamplesize); audiobits |= get_bits(&alac->gb, readsamplesize - 16); alac->outputsamples_buffer_a[i] = audiobits; } } /* wasted_bytes = 0; // unused */ } switch(alac->setinfo_sample_size) { case 16: { int i; for (i = 0; i < outputsamples; i++) { int16_t sample = alac->outputsamples_buffer_a[i]; ((int16_t*)outbuffer)[i * alac->numchannels] = sample; } break; } case 20: case 24: case 32: av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size); break; default: break; } break; } case 1: { /* 2 channels */ int hassize; int isnotcompressed; int readsamplesize; int wasted_bytes; uint8_t interlacing_shift; uint8_t interlacing_leftweight; /* 2^result = something to do with output waiting. * perhaps matters if we read > 1 frame in a pass? */ get_bits(&alac->gb, 4); get_bits(&alac->gb, 12); /* unknown, skip 12 bits */ hassize = get_bits(&alac->gb, 1); /* the output sample size is stored soon */ wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */ isnotcompressed = get_bits(&alac->gb, 1); /* whether the frame is compressed */ if (hassize) { /* now read the number of samples, * as a 32bit integer */ outputsamples = get_bits(&alac->gb, 32); *outputsize = outputsamples * alac->bytespersample; } readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + 1; if (!isnotcompressed) { /* compressed */ int16_t predictor_coef_table_a[32]; int predictor_coef_num_a; int prediction_type_a; int prediction_quantitization_a; int ricemodifier_a; int16_t predictor_coef_table_b[32]; int predictor_coef_num_b; int prediction_type_b; int prediction_quantitization_b; int ricemodifier_b; int i; interlacing_shift = get_bits(&alac->gb, 8); interlacing_leftweight = get_bits(&alac->gb, 8); /******** channel 1 ***********/ prediction_type_a = get_bits(&alac->gb, 4); prediction_quantitization_a = get_bits(&alac->gb, 4); ricemodifier_a = get_bits(&alac->gb, 3); predictor_coef_num_a = get_bits(&alac->gb, 5); /* read the predictor table */ for (i = 0; i < predictor_coef_num_a; i++) { predictor_coef_table_a[i] = (int16_t)get_bits(&alac->gb, 16); } /******** channel 2 *********/ prediction_type_b = get_bits(&alac->gb, 4); prediction_quantitization_b = get_bits(&alac->gb, 4); ricemodifier_b = get_bits(&alac->gb, 3); predictor_coef_num_b = get_bits(&alac->gb, 5); /* read the predictor table */ for (i = 0; i < predictor_coef_num_b; i++) { predictor_coef_table_b[i] = (int16_t)get_bits(&alac->gb, 16); } /*********************/ if (wasted_bytes) { /* see mono case */ av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n"); } /* channel 1 */ bastardized_rice_decompress(alac, alac->predicterror_buffer_a, outputsamples, readsamplesize, alac->setinfo_rice_initialhistory, alac->setinfo_rice_kmodifier, ricemodifier_a * alac->setinfo_rice_historymult / 4, (1 << alac->setinfo_rice_kmodifier) - 1); if (prediction_type_a == 0) { /* adaptive fir */ predictor_decompress_fir_adapt(alac->predicterror_buffer_a, alac->outputsamples_buffer_a, outputsamples, readsamplesize, predictor_coef_table_a, predictor_coef_num_a, prediction_quantitization_a); } else { /* see mono case */ av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type_a); } /* channel 2 */ bastardized_rice_decompress(alac, alac->predicterror_buffer_b, outputsamples, readsamplesize, alac->setinfo_rice_initialhistory, alac->setinfo_rice_kmodifier, ricemodifier_b * alac->setinfo_rice_historymult / 4, (1 << alac->setinfo_rice_kmodifier) - 1); if (prediction_type_b == 0) { /* adaptive fir */ predictor_decompress_fir_adapt(alac->predicterror_buffer_b, alac->outputsamples_buffer_b, outputsamples, readsamplesize, predictor_coef_table_b, predictor_coef_num_b, prediction_quantitization_b); } else { av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type_b); } } else { /* not compressed, easy case */ if (alac->setinfo_sample_size <= 16) { int i; for (i = 0; i < outputsamples; i++) { int32_t audiobits_a, audiobits_b; audiobits_a = get_bits(&alac->gb, alac->setinfo_sample_size); audiobits_b = get_bits(&alac->gb, alac->setinfo_sample_size); audiobits_a = SIGN_EXTENDED32(audiobits_a, alac->setinfo_sample_size); audiobits_b = SIGN_EXTENDED32(audiobits_b, alac->setinfo_sample_size); alac->outputsamples_buffer_a[i] = audiobits_a; alac->outputsamples_buffer_b[i] = audiobits_b; } } else { int i; for (i = 0; i < outputsamples; i++) { int32_t audiobits_a, audiobits_b; audiobits_a = get_bits(&alac->gb, 16); audiobits_a = audiobits_a << 16; audiobits_a = audiobits_a >> (32 - alac->setinfo_sample_size); audiobits_a |= get_bits(&alac->gb, alac->setinfo_sample_size - 16); audiobits_b = get_bits(&alac->gb, 16); audiobits_b = audiobits_b << 16; audiobits_b = audiobits_b >> (32 - alac->setinfo_sample_size); audiobits_b |= get_bits(&alac->gb, alac->setinfo_sample_size - 16); alac->outputsamples_buffer_a[i] = audiobits_a; alac->outputsamples_buffer_b[i] = audiobits_b; } } /* wasted_bytes = 0; */ interlacing_shift = 0; interlacing_leftweight = 0; } switch(alac->setinfo_sample_size) { case 16: { deinterlace_16(alac->outputsamples_buffer_a, alac->outputsamples_buffer_b, (int16_t*)outbuffer, alac->numchannels, outputsamples, interlacing_shift, interlacing_leftweight); break; } case 20: case 24: case 32: av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size); break; default: break; } break; } } return input_buffer_size; } static int alac_decode_init(AVCodecContext * avctx) { ALACContext *alac = avctx->priv_data; alac->avctx = avctx; alac->context_initialized = 0; alac->samplesize = alac->avctx->bits_per_sample; alac->numchannels = alac->avctx->channels; alac->bytespersample = (alac->samplesize / 8) * alac->numchannels; return 0; } static int alac_decode_close(AVCodecContext *avctx) { ALACContext *alac = avctx->priv_data; av_free(alac->predicterror_buffer_a); av_free(alac->predicterror_buffer_b); av_free(alac->outputsamples_buffer_a); av_free(alac->outputsamples_buffer_b); return 0; } AVCodec alac_decoder = { "alac", CODEC_TYPE_AUDIO, CODEC_ID_ALAC, sizeof(ALACContext), alac_decode_init, NULL, alac_decode_close, alac_decode_frame, };