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📄 alac.c

📁 mediastreamer2是开源的网络传输媒体流的库
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    }    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,};

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