| /* |
| * 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 (Apple Lossless Audio Codec) decoder |
| * @author 2005 David Hammerton |
| * @see http://crazney.net/programs/itunes/alac.html |
| * |
| * Note: This decoder expects a 36-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: |
| * |
| * 32bit atom size |
| * 32bit tag ("alac") |
| * 32bit tag version (0) |
| * 32bit samples per frame (used when not set explicitly in the frames) |
| * 8bit compatible version (0) |
| * 8bit sample size |
| * 8bit history mult (40) |
| * 8bit initial history (14) |
| * 8bit kmodifier (10) |
| * 8bit channels |
| * 16bit maxRun (255) |
| * 32bit max coded frame size (0 means unknown) |
| * 32bit average bitrate (0 means unknown) |
| * 32bit samplerate |
| */ |
| |
| |
| #include "avcodec.h" |
| #include "get_bits.h" |
| #include "bytestream.h" |
| #include "unary.h" |
| #include "mathops.h" |
| |
| #define ALAC_EXTRADATA_SIZE 36 |
| #define MAX_CHANNELS 2 |
| |
| typedef struct { |
| |
| AVCodecContext *avctx; |
| AVFrame frame; |
| GetBitContext gb; |
| |
| int numchannels; |
| |
| /* buffers */ |
| int32_t *predicterror_buffer[MAX_CHANNELS]; |
| |
| int32_t *outputsamples_buffer[MAX_CHANNELS]; |
| |
| int32_t *extra_bits_buffer[MAX_CHANNELS]; |
| |
| /* stuff from setinfo */ |
| uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ |
| uint8_t setinfo_sample_size; /* 0x10 */ |
| uint8_t setinfo_rice_historymult; /* 0x28 */ |
| uint8_t setinfo_rice_initialhistory; /* 0x0a */ |
| uint8_t setinfo_rice_kmodifier; /* 0x0e */ |
| /* end setinfo stuff */ |
| |
| int extra_bits; /**< number of extra bits beyond 16-bit */ |
| } ALACContext; |
| |
| static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){ |
| /* read x - number of 1s before 0 represent the rice */ |
| int x = get_unary_0_9(gb); |
| |
| if (x > 8) { /* RICE THRESHOLD */ |
| /* use alternative encoding */ |
| x = get_bits(gb, readsamplesize); |
| } else { |
| if (k >= limit) |
| k = limit; |
| |
| if (k != 1) { |
| int extrabits = show_bits(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(gb, k); |
| } else |
| skip_bits(gb, k - 1); |
| } |
| } |
| return x; |
| } |
| |
| static int 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; |
| |
| /* standard rice encoding */ |
| int k; /* size of extra bits */ |
| |
| if(get_bits_left(&alac->gb) <= 0) |
| return -1; |
| |
| /* read k, that is bits as is */ |
| k = av_log2((history >> 9) + 3); |
| x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize); |
| |
| 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 k; |
| unsigned int block_size; |
| |
| sign_modifier = 1; |
| |
| k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */); |
| |
| block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16); |
| |
| if (block_size > 0) { |
| if(block_size >= output_size - output_count){ |
| av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count); |
| block_size= output_size - output_count - 1; |
| } |
| memset(&output_buffer[output_count+1], 0, block_size * 4); |
| output_count += block_size; |
| } |
| |
| if (block_size > 0xffff) |
| sign_modifier = 0; |
| |
| history = 0; |
| } |
| } |
| return 0; |
| } |
| |
| 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] = |
| sign_extend((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 = sign_extend(val, readsamplesize); |
| buffer_out[i+1] = val; |
| } |
| |
| /* 4 and 8 are very common cases (the only ones i've seen). these |
| * should be unrolled and optimized |
| */ |
| |
| /* 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_extend(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 decorrelate_stereo(int32_t *buffer[MAX_CHANNELS], |
| int numsamples, uint8_t interlacing_shift, |
| uint8_t interlacing_leftweight) |
| { |
| int i; |
| |
| 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[0][i] = b; |
| buffer[1][i] = a; |
| } |
| } |
| |
| static void append_extra_bits(int32_t *buffer[MAX_CHANNELS], |
| int32_t *extra_bits_buffer[MAX_CHANNELS], |
| int extra_bits, int numchannels, int numsamples) |
| { |
| int i, ch; |
| |
| for (ch = 0; ch < numchannels; ch++) |
| for (i = 0; i < numsamples; i++) |
| buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i]; |
| } |
| |
| static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS], |
| int16_t *buffer_out, int numsamples) |
| { |
| int i; |
| |
| for (i = 0; i < numsamples; i++) { |
| *buffer_out++ = buffer[0][i]; |
| *buffer_out++ = buffer[1][i]; |
| } |
| } |
| |
| static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS], |
| int32_t *buffer_out, int numsamples) |
| { |
| int i; |
| |
| for (i = 0; i < numsamples; i++) { |
| *buffer_out++ = buffer[0][i] << 8; |
| *buffer_out++ = buffer[1][i] << 8; |
| } |
| } |
| |
| static void interleave_stereo_32(int32_t *buffer[MAX_CHANNELS], |
| int32_t *buffer_out, int numsamples) |
| { |
| int i; |
| |
| for (i = 0; i < numsamples; i++) { |
| *buffer_out++ = buffer[0][i]; |
| *buffer_out++ = buffer[1][i]; |
| } |
| } |
| |
| static int alac_decode_frame(AVCodecContext *avctx, void *data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| const uint8_t *inbuffer = avpkt->data; |
| int input_buffer_size = avpkt->size; |
| ALACContext *alac = avctx->priv_data; |
| |
| int channels; |
| unsigned int outputsamples; |
| int hassize; |
| unsigned int readsamplesize; |
| int isnotcompressed; |
| uint8_t interlacing_shift; |
| uint8_t interlacing_leftweight; |
| int i, ch, ret; |
| |
| init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); |
| |
| channels = get_bits(&alac->gb, 3) + 1; |
| if (channels != avctx->channels) { |
| av_log(avctx, AV_LOG_ERROR, "frame header channel count mismatch\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| /* 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); |
| |
| alac->extra_bits = get_bits(&alac->gb, 2) << 3; |
| |
| /* 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_long(&alac->gb, 32); |
| if(outputsamples > alac->setinfo_max_samples_per_frame){ |
| av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame); |
| return -1; |
| } |
| } else |
| outputsamples = alac->setinfo_max_samples_per_frame; |
| |
| /* get output buffer */ |
| if (outputsamples > INT32_MAX) { |
| av_log(avctx, AV_LOG_ERROR, "unsupported block size: %u\n", outputsamples); |
| return AVERROR_INVALIDDATA; |
| } |
| alac->frame.nb_samples = outputsamples; |
| if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) { |
| av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); |
| return ret; |
| } |
| |
| readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1; |
| if (readsamplesize > MIN_CACHE_BITS) { |
| av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize); |
| return -1; |
| } |
| |
| if (!isnotcompressed) { |
| /* so it is compressed */ |
| int16_t predictor_coef_table[MAX_CHANNELS][32]; |
| int predictor_coef_num[MAX_CHANNELS]; |
| int prediction_type[MAX_CHANNELS]; |
| int prediction_quantitization[MAX_CHANNELS]; |
| int ricemodifier[MAX_CHANNELS]; |
| |
| interlacing_shift = get_bits(&alac->gb, 8); |
| interlacing_leftweight = get_bits(&alac->gb, 8); |
| |
| for (ch = 0; ch < channels; ch++) { |
| prediction_type[ch] = get_bits(&alac->gb, 4); |
| prediction_quantitization[ch] = get_bits(&alac->gb, 4); |
| |
| ricemodifier[ch] = get_bits(&alac->gb, 3); |
| predictor_coef_num[ch] = get_bits(&alac->gb, 5); |
| |
| /* read the predictor table */ |
| for (i = 0; i < predictor_coef_num[ch]; i++) |
| predictor_coef_table[ch][i] = (int16_t)get_bits(&alac->gb, 16); |
| } |
| |
| if (alac->extra_bits) { |
| for (i = 0; i < outputsamples; i++) { |
| if(get_bits_left(&alac->gb) <= 0) |
| return -1; |
| for (ch = 0; ch < channels; ch++) |
| alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits); |
| } |
| } |
| for (ch = 0; ch < channels; ch++) { |
| int ret = bastardized_rice_decompress(alac, |
| alac->predicterror_buffer[ch], |
| outputsamples, |
| readsamplesize, |
| alac->setinfo_rice_initialhistory, |
| alac->setinfo_rice_kmodifier, |
| ricemodifier[ch] * alac->setinfo_rice_historymult / 4, |
| (1 << alac->setinfo_rice_kmodifier) - 1); |
| if(ret<0) |
| return ret; |
| |
| /* adaptive FIR filter */ |
| if (prediction_type[ch] == 15) { |
| /* Prediction type 15 runs the adaptive FIR twice. |
| * The first pass uses the special-case coef_num = 31, while |
| * the second pass uses the coefs from the bitstream. |
| * |
| * However, this prediction type is not currently used by the |
| * reference encoder. |
| */ |
| predictor_decompress_fir_adapt(alac->predicterror_buffer[ch], |
| alac->predicterror_buffer[ch], |
| outputsamples, readsamplesize, |
| NULL, 31, 0); |
| } else if (prediction_type[ch] > 0) { |
| av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n", |
| prediction_type[ch]); |
| } |
| predictor_decompress_fir_adapt(alac->predicterror_buffer[ch], |
| alac->outputsamples_buffer[ch], |
| outputsamples, readsamplesize, |
| predictor_coef_table[ch], |
| predictor_coef_num[ch], |
| prediction_quantitization[ch]); |
| } |
| } else { |
| /* not compressed, easy case */ |
| for (i = 0; i < outputsamples; i++) { |
| if(get_bits_left(&alac->gb) <= 0) |
| return -1; |
| for (ch = 0; ch < channels; ch++) { |
| alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb, |
| alac->setinfo_sample_size); |
| } |
| } |
| alac->extra_bits = 0; |
| interlacing_shift = 0; |
| interlacing_leftweight = 0; |
| } |
| if (get_bits(&alac->gb, 3) != 7) |
| av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n"); |
| |
| if (channels == 2 && interlacing_leftweight) { |
| decorrelate_stereo(alac->outputsamples_buffer, outputsamples, |
| interlacing_shift, interlacing_leftweight); |
| } |
| |
| if (alac->extra_bits) { |
| append_extra_bits(alac->outputsamples_buffer, alac->extra_bits_buffer, |
| alac->extra_bits, alac->numchannels, outputsamples); |
| } |
| |
| switch(alac->setinfo_sample_size) { |
| case 16: |
| if (channels == 2) { |
| interleave_stereo_16(alac->outputsamples_buffer, |
| (int16_t *)alac->frame.data[0], outputsamples); |
| } else { |
| int16_t *outbuffer = (int16_t *)alac->frame.data[0]; |
| for (i = 0; i < outputsamples; i++) { |
| outbuffer[i] = alac->outputsamples_buffer[0][i]; |
| } |
| } |
| break; |
| case 24: |
| if (channels == 2) { |
| interleave_stereo_24(alac->outputsamples_buffer, |
| (int32_t *)alac->frame.data[0], outputsamples); |
| } else { |
| int32_t *outbuffer = (int32_t *)alac->frame.data[0]; |
| for (i = 0; i < outputsamples; i++) |
| outbuffer[i] = alac->outputsamples_buffer[0][i] << 8; |
| } |
| break; |
| case 32: |
| if (channels == 2) { |
| interleave_stereo_32(alac->outputsamples_buffer, |
| (int32_t *)alac->frame.data[0], outputsamples); |
| } else { |
| int32_t *outbuffer = (int32_t *)alac->frame.data[0]; |
| for (i = 0; i < outputsamples; i++) |
| outbuffer[i] = alac->outputsamples_buffer[0][i]; |
| } |
| break; |
| } |
| |
| if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8) |
| av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb)); |
| |
| *got_frame_ptr = 1; |
| *(AVFrame *)data = alac->frame; |
| |
| return input_buffer_size; |
| } |
| |
| static av_cold int alac_decode_close(AVCodecContext *avctx) |
| { |
| ALACContext *alac = avctx->priv_data; |
| |
| int ch; |
| for (ch = 0; ch < alac->numchannels; ch++) { |
| av_freep(&alac->predicterror_buffer[ch]); |
| av_freep(&alac->outputsamples_buffer[ch]); |
| av_freep(&alac->extra_bits_buffer[ch]); |
| } |
| |
| return 0; |
| } |
| |
| static int allocate_buffers(ALACContext *alac) |
| { |
| int ch; |
| for (ch = 0; ch < alac->numchannels; ch++) { |
| int buf_size = alac->setinfo_max_samples_per_frame * sizeof(int32_t); |
| |
| FF_ALLOC_OR_GOTO(alac->avctx, alac->predicterror_buffer[ch], |
| buf_size, buf_alloc_fail); |
| |
| FF_ALLOC_OR_GOTO(alac->avctx, alac->outputsamples_buffer[ch], |
| buf_size, buf_alloc_fail); |
| |
| FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch], |
| buf_size, buf_alloc_fail); |
| } |
| return 0; |
| buf_alloc_fail: |
| alac_decode_close(alac->avctx); |
| return AVERROR(ENOMEM); |
| } |
| |
| static int alac_set_info(ALACContext *alac) |
| { |
| const unsigned char *ptr = alac->avctx->extradata; |
| |
| ptr += 4; /* size */ |
| ptr += 4; /* alac */ |
| ptr += 4; /* version */ |
| |
| 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); |
| ptr++; /* compatible version */ |
| alac->setinfo_sample_size = *ptr++; |
| alac->setinfo_rice_historymult = *ptr++; |
| alac->setinfo_rice_initialhistory = *ptr++; |
| alac->setinfo_rice_kmodifier = *ptr++; |
| alac->numchannels = *ptr++; |
| bytestream_get_be16(&ptr); /* maxRun */ |
| bytestream_get_be32(&ptr); /* max coded frame size */ |
| bytestream_get_be32(&ptr); /* average bitrate */ |
| bytestream_get_be32(&ptr); /* samplerate */ |
| |
| return 0; |
| } |
| |
| static av_cold int alac_decode_init(AVCodecContext * avctx) |
| { |
| int ret; |
| ALACContext *alac = avctx->priv_data; |
| alac->avctx = avctx; |
| |
| /* initialize from the extradata */ |
| if (alac->avctx->extradata_size < ALAC_EXTRADATA_SIZE) { |
| av_log(avctx, AV_LOG_ERROR, "alac: extradata is too small\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| if (alac_set_info(alac)) { |
| av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); |
| return -1; |
| } |
| |
| switch (alac->setinfo_sample_size) { |
| case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
| break; |
| case 32: |
| case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32; |
| break; |
| default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n", |
| alac->setinfo_sample_size); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| if (alac->numchannels < 1) { |
| av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n"); |
| alac->numchannels = avctx->channels; |
| } else { |
| if (alac->numchannels > MAX_CHANNELS) |
| alac->numchannels = avctx->channels; |
| else |
| avctx->channels = alac->numchannels; |
| } |
| if (avctx->channels > MAX_CHANNELS) { |
| av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n", |
| avctx->channels); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| if ((ret = allocate_buffers(alac)) < 0) { |
| av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n"); |
| return ret; |
| } |
| |
| avcodec_get_frame_defaults(&alac->frame); |
| avctx->coded_frame = &alac->frame; |
| |
| return 0; |
| } |
| |
| AVCodec ff_alac_decoder = { |
| .name = "alac", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = CODEC_ID_ALAC, |
| .priv_data_size = sizeof(ALACContext), |
| .init = alac_decode_init, |
| .close = alac_decode_close, |
| .decode = alac_decode_frame, |
| .capabilities = CODEC_CAP_DR1, |
| .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), |
| }; |
