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Diffstat (limited to 'contrib/ffmpeg/libavcodec/alac.c')
| -rw-r--r-- | contrib/ffmpeg/libavcodec/alac.c | 662 |
1 files changed, 662 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavcodec/alac.c b/contrib/ffmpeg/libavcodec/alac.c new file mode 100644 index 000000000..0689a46ef --- /dev/null +++ b/contrib/ffmpeg/libavcodec/alac.c @@ -0,0 +1,662 @@ +/* + * ALAC (Apple Lossless Audio Codec) decoder + * Copyright (c) 2005 David Hammerton + * + * 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" +#include "bytestream.h" +#include "unary.h" + +#define ALAC_EXTRADATA_SIZE 36 +#define MAX_CHANNELS 2 + +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[MAX_CHANNELS]; + + int32_t *outputsamples_buffer[MAX_CHANNELS]; + + /* 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 */ /* max sample size?? */ + uint32_t setinfo_86; /* 0x00069fe4 */ /* bit rate (average)?? */ + uint32_t setinfo_8a_rate; /* 0x0000ac44 */ + /* end setinfo stuff */ + +} ALACContext; + +static void allocate_buffers(ALACContext *alac) +{ + int chan; + for (chan = 0; chan < MAX_CHANNELS; chan++) { + alac->predicterror_buffer[chan] = + av_malloc(alac->setinfo_max_samples_per_frame * 4); + + alac->outputsamples_buffer[chan] = + av_malloc(alac->setinfo_max_samples_per_frame * 4); + } +} + +static int alac_set_info(ALACContext *alac) +{ + const 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; + } + + /* buffer size / 2 ? */ + alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr); + alac->setinfo_7a = *ptr++; + alac->setinfo_sample_size = *ptr++; + alac->setinfo_rice_historymult = *ptr++; + alac->setinfo_rice_initialhistory = *ptr++; + alac->setinfo_rice_kmodifier = *ptr++; + /* channels? */ + alac->setinfo_7f = *ptr++; + alac->setinfo_80 = bytestream_get_be16(&ptr); + /* max coded frame size */ + alac->setinfo_82 = bytestream_get_be32(&ptr); + /* bitrate ? */ + alac->setinfo_86 = bytestream_get_be32(&ptr); + /* samplerate */ + alac->setinfo_8a_rate = bytestream_get_be32(&ptr); + + allocate_buffers(alac); + + return 0; +} + +static inline int count_leading_zeros(int32_t input) +{ + return 31-av_log2(input); +} + +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; + int32_t x_modified; + int32_t final_val; + + /* read x - number of 1s before 0 represent the rice */ + x = get_unary_0_9(&alac->gb); + + 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; + skip_bits(&alac->gb, k); + } else + skip_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 = get_unary_0_9(&alac->gb); + + 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; + skip_bits(&alac->gb, k - 1); + } else { + skip_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; + } + } +} + +static inline int32_t extend_sign32(int32_t val, int bits) +{ + return (val << (32 - bits)) >> (32 - bits); +} + +static inline int sign_only(int v) +{ + return v ? FFSIGN(v) : 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] = + extend_sign32((prev_value + error_value), readsamplesize); + } + return; + } + + /* read warm-up samples */ + if (predictor_coef_num > 0) + for (i = 0; i < predictor_coef_num; i++) { + int32_t val; + + val = buffer_out[i] + error_buffer[i+1]; + val = extend_sign32(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 optimized + */ + if (predictor_coef_num == 4) { + /* FIXME: optimized general case */ + return; + } + + if (predictor_coef_table == 8) { + /* FIXME: optimized 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 = extend_sign32(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 reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS], + 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 a, b; + + a = buffer[0][i]; + b = buffer[1][i]; + + a -= (b * interlacing_leftweight) >> interlacing_shift; + b += a; + + buffer_out[i*numchannels] = b; + buffer_out[i*numchannels + 1] = a; + } + + return; + } + + /* otherwise basic interlacing took place */ + for (i = 0; i < numsamples; i++) { + int16_t left, right; + + left = buffer[0][i]; + right = buffer[1][i]; + + buffer_out[i*numchannels] = left; + buffer_out[i*numchannels + 1] = right; + } +} + +static int alac_decode_frame(AVCodecContext *avctx, + void *outbuffer, int *outputsize, + const uint8_t *inbuffer, int input_buffer_size) +{ + ALACContext *alac = avctx->priv_data; + + int channels; + int32_t outputsamples; + int hassize; + int readsamplesize; + int wasted_bytes; + int isnotcompressed; + uint8_t interlacing_shift; + uint8_t interlacing_leftweight; + + /* 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; + } + + init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); + + channels = get_bits(&alac->gb, 3) + 1; + if (channels > MAX_CHANNELS) { + av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n", + MAX_CHANNELS); + return input_buffer_size; + } + + /* 2^result = something to do with output waiting. + * perhaps matters if we read > 1 frame in a pass? + */ + skip_bits(&alac->gb, 4); + + skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */ + + /* the output sample size is stored soon */ + hassize = get_bits1(&alac->gb); + + wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */ + + /* whether the frame is compressed */ + isnotcompressed = get_bits1(&alac->gb); + + if (hassize) { + /* now read the number of samples as a 32bit integer */ + outputsamples = get_bits(&alac->gb, 32); + } else + outputsamples = alac->setinfo_max_samples_per_frame; + + *outputsize = outputsamples * alac->bytespersample; + readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1; + + if (!isnotcompressed) { + /* so it is compressed */ + int16_t predictor_coef_table[channels][32]; + int predictor_coef_num[channels]; + int prediction_type[channels]; + int prediction_quantitization[channels]; + int ricemodifier[channels]; + int i, chan; + + interlacing_shift = get_bits(&alac->gb, 8); + interlacing_leftweight = get_bits(&alac->gb, 8); + + for (chan = 0; chan < channels; chan++) { + prediction_type[chan] = get_bits(&alac->gb, 4); + prediction_quantitization[chan] = get_bits(&alac->gb, 4); + + ricemodifier[chan] = get_bits(&alac->gb, 3); + predictor_coef_num[chan] = get_bits(&alac->gb, 5); + + /* read the predictor table */ + for (i = 0; i < predictor_coef_num[chan]; i++) + predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); + } + + if (wasted_bytes) + av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n"); + + for (chan = 0; chan < channels; chan++) { + bastardized_rice_decompress(alac, + alac->predicterror_buffer[chan], + outputsamples, + readsamplesize, + alac->setinfo_rice_initialhistory, + alac->setinfo_rice_kmodifier, + ricemodifier[chan] * alac->setinfo_rice_historymult / 4, + (1 << alac->setinfo_rice_kmodifier) - 1); + + if (prediction_type[chan] == 0) { + /* adaptive fir */ + predictor_decompress_fir_adapt(alac->predicterror_buffer[chan], + alac->outputsamples_buffer[chan], + outputsamples, + readsamplesize, + predictor_coef_table[chan], + predictor_coef_num[chan], + prediction_quantitization[chan]); + } else { + av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]); + /* 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 (alac->setinfo_sample_size <= 16) { + int i, chan; + for (chan = 0; chan < channels; chan++) + for (i = 0; i < outputsamples; i++) { + int32_t audiobits; + + audiobits = get_bits(&alac->gb, alac->setinfo_sample_size); + audiobits = extend_sign32(audiobits, readsamplesize); + + alac->outputsamples_buffer[chan][i] = audiobits; + } + } else { + int i, chan; + for (chan = 0; chan < channels; chan++) + 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 - alac->setinfo_sample_size); + audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16); + + alac->outputsamples_buffer[chan][i] = audiobits; + } + } + /* wasted_bytes = 0; */ + interlacing_shift = 0; + interlacing_leftweight = 0; + } + + switch(alac->setinfo_sample_size) { + case 16: + if (channels == 2) { + reconstruct_stereo_16(alac->outputsamples_buffer, + (int16_t*)outbuffer, + alac->numchannels, + outputsamples, + interlacing_shift, + interlacing_leftweight); + } else { + int i; + for (i = 0; i < outputsamples; i++) { + int16_t sample = alac->outputsamples_buffer[0][i]; + ((int16_t*)outbuffer)[i * alac->numchannels] = sample; + } + } + break; + case 20: + case 24: + // It is not clear if there exist any encoder that creates 24 bit ALAC + // files. iTunes convert 24 bit raw files to 16 bit before encoding. + case 32: + av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size); + break; + default: + 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; + + int chan; + for (chan = 0; chan < MAX_CHANNELS; chan++) { + av_free(alac->predicterror_buffer[chan]); + av_free(alac->outputsamples_buffer[chan]); + } + + return 0; +} + +AVCodec alac_decoder = { + "alac", + CODEC_TYPE_AUDIO, + CODEC_ID_ALAC, + sizeof(ALACContext), + alac_decode_init, + NULL, + alac_decode_close, + alac_decode_frame, +}; |