diff options
| author | Tim Redfern <tim@eclectronics.org> | 2014-02-17 13:36:38 +0000 |
|---|---|---|
| committer | Tim Redfern <tim@eclectronics.org> | 2014-02-17 13:36:38 +0000 |
| commit | 22e28216336da876e1fd17f380ce42eaf1446769 (patch) | |
| tree | 444dad3dc7e2656992d29f34f7bce31970c122a5 /ffmpeg/libavcodec/alacenc.c | |
| parent | ae5e8541f6e06e64c28719467cdf366ac57aff31 (diff) | |
chasing indexing error
Diffstat (limited to 'ffmpeg/libavcodec/alacenc.c')
| -rw-r--r-- | ffmpeg/libavcodec/alacenc.c | 649 |
1 files changed, 0 insertions, 649 deletions
diff --git a/ffmpeg/libavcodec/alacenc.c b/ffmpeg/libavcodec/alacenc.c deleted file mode 100644 index bc68a06..0000000 --- a/ffmpeg/libavcodec/alacenc.c +++ /dev/null @@ -1,649 +0,0 @@ -/* - * ALAC audio encoder - * Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net> - * - * 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 - */ - -#include "avcodec.h" -#include "put_bits.h" -#include "internal.h" -#include "lpc.h" -#include "mathops.h" -#include "alac_data.h" - -#define DEFAULT_FRAME_SIZE 4096 -#define ALAC_EXTRADATA_SIZE 36 -#define ALAC_FRAME_HEADER_SIZE 55 -#define ALAC_FRAME_FOOTER_SIZE 3 - -#define ALAC_ESCAPE_CODE 0x1FF -#define ALAC_MAX_LPC_ORDER 30 -#define DEFAULT_MAX_PRED_ORDER 6 -#define DEFAULT_MIN_PRED_ORDER 4 -#define ALAC_MAX_LPC_PRECISION 9 -#define ALAC_MAX_LPC_SHIFT 9 - -#define ALAC_CHMODE_LEFT_RIGHT 0 -#define ALAC_CHMODE_LEFT_SIDE 1 -#define ALAC_CHMODE_RIGHT_SIDE 2 -#define ALAC_CHMODE_MID_SIDE 3 - -typedef struct RiceContext { - int history_mult; - int initial_history; - int k_modifier; - int rice_modifier; -} RiceContext; - -typedef struct AlacLPCContext { - int lpc_order; - int lpc_coeff[ALAC_MAX_LPC_ORDER+1]; - int lpc_quant; -} AlacLPCContext; - -typedef struct AlacEncodeContext { - int frame_size; /**< current frame size */ - int verbatim; /**< current frame verbatim mode flag */ - int compression_level; - int min_prediction_order; - int max_prediction_order; - int max_coded_frame_size; - int write_sample_size; - int extra_bits; - int32_t sample_buf[2][DEFAULT_FRAME_SIZE]; - int32_t predictor_buf[DEFAULT_FRAME_SIZE]; - int interlacing_shift; - int interlacing_leftweight; - PutBitContext pbctx; - RiceContext rc; - AlacLPCContext lpc[2]; - LPCContext lpc_ctx; - AVCodecContext *avctx; -} AlacEncodeContext; - - -static void init_sample_buffers(AlacEncodeContext *s, int channels, - uint8_t const *samples[2]) -{ - int ch, i; - int shift = av_get_bytes_per_sample(s->avctx->sample_fmt) * 8 - - s->avctx->bits_per_raw_sample; - -#define COPY_SAMPLES(type) do { \ - for (ch = 0; ch < channels; ch++) { \ - int32_t *bptr = s->sample_buf[ch]; \ - const type *sptr = (const type *)samples[ch]; \ - for (i = 0; i < s->frame_size; i++) \ - bptr[i] = sptr[i] >> shift; \ - } \ - } while (0) - - if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) - COPY_SAMPLES(int32_t); - else - COPY_SAMPLES(int16_t); -} - -static void encode_scalar(AlacEncodeContext *s, int x, - int k, int write_sample_size) -{ - int divisor, q, r; - - k = FFMIN(k, s->rc.k_modifier); - divisor = (1<<k) - 1; - q = x / divisor; - r = x % divisor; - - if (q > 8) { - // write escape code and sample value directly - put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE); - put_bits(&s->pbctx, write_sample_size, x); - } else { - if (q) - put_bits(&s->pbctx, q, (1<<q) - 1); - put_bits(&s->pbctx, 1, 0); - - if (k != 1) { - if (r > 0) - put_bits(&s->pbctx, k, r+1); - else - put_bits(&s->pbctx, k-1, 0); - } - } -} - -static void write_element_header(AlacEncodeContext *s, - enum AlacRawDataBlockType element, - int instance) -{ - int encode_fs = 0; - - if (s->frame_size < DEFAULT_FRAME_SIZE) - encode_fs = 1; - - put_bits(&s->pbctx, 3, element); // element type - put_bits(&s->pbctx, 4, instance); // element instance - put_bits(&s->pbctx, 12, 0); // unused header bits - put_bits(&s->pbctx, 1, encode_fs); // Sample count is in the header - put_bits(&s->pbctx, 2, s->extra_bits >> 3); // Extra bytes (for 24-bit) - put_bits(&s->pbctx, 1, s->verbatim); // Audio block is verbatim - if (encode_fs) - put_bits32(&s->pbctx, s->frame_size); // No. of samples in the frame -} - -static void calc_predictor_params(AlacEncodeContext *s, int ch) -{ - int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER]; - int shift[MAX_LPC_ORDER]; - int opt_order; - - if (s->compression_level == 1) { - s->lpc[ch].lpc_order = 6; - s->lpc[ch].lpc_quant = 6; - s->lpc[ch].lpc_coeff[0] = 160; - s->lpc[ch].lpc_coeff[1] = -190; - s->lpc[ch].lpc_coeff[2] = 170; - s->lpc[ch].lpc_coeff[3] = -130; - s->lpc[ch].lpc_coeff[4] = 80; - s->lpc[ch].lpc_coeff[5] = -25; - } else { - opt_order = ff_lpc_calc_coefs(&s->lpc_ctx, s->sample_buf[ch], - s->frame_size, - s->min_prediction_order, - s->max_prediction_order, - ALAC_MAX_LPC_PRECISION, coefs, shift, - FF_LPC_TYPE_LEVINSON, 0, - ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1); - - s->lpc[ch].lpc_order = opt_order; - s->lpc[ch].lpc_quant = shift[opt_order-1]; - memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int)); - } -} - -static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n) -{ - int i, best; - int32_t lt, rt; - uint64_t sum[4]; - uint64_t score[4]; - - /* calculate sum of 2nd order residual for each channel */ - sum[0] = sum[1] = sum[2] = sum[3] = 0; - for (i = 2; i < n; i++) { - lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2]; - rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2]; - sum[2] += FFABS((lt + rt) >> 1); - sum[3] += FFABS(lt - rt); - sum[0] += FFABS(lt); - sum[1] += FFABS(rt); - } - - /* calculate score for each mode */ - score[0] = sum[0] + sum[1]; - score[1] = sum[0] + sum[3]; - score[2] = sum[1] + sum[3]; - score[3] = sum[2] + sum[3]; - - /* return mode with lowest score */ - best = 0; - for (i = 1; i < 4; i++) { - if (score[i] < score[best]) - best = i; - } - return best; -} - -static void alac_stereo_decorrelation(AlacEncodeContext *s) -{ - int32_t *left = s->sample_buf[0], *right = s->sample_buf[1]; - int i, mode, n = s->frame_size; - int32_t tmp; - - mode = estimate_stereo_mode(left, right, n); - - switch (mode) { - case ALAC_CHMODE_LEFT_RIGHT: - s->interlacing_leftweight = 0; - s->interlacing_shift = 0; - break; - case ALAC_CHMODE_LEFT_SIDE: - for (i = 0; i < n; i++) - right[i] = left[i] - right[i]; - s->interlacing_leftweight = 1; - s->interlacing_shift = 0; - break; - case ALAC_CHMODE_RIGHT_SIDE: - for (i = 0; i < n; i++) { - tmp = right[i]; - right[i] = left[i] - right[i]; - left[i] = tmp + (right[i] >> 31); - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 31; - break; - default: - for (i = 0; i < n; i++) { - tmp = left[i]; - left[i] = (tmp + right[i]) >> 1; - right[i] = tmp - right[i]; - } - s->interlacing_leftweight = 1; - s->interlacing_shift = 1; - break; - } -} - -static void alac_linear_predictor(AlacEncodeContext *s, int ch) -{ - int i; - AlacLPCContext lpc = s->lpc[ch]; - - if (lpc.lpc_order == 31) { - s->predictor_buf[0] = s->sample_buf[ch][0]; - - for (i = 1; i < s->frame_size; i++) { - s->predictor_buf[i] = s->sample_buf[ch][i ] - - s->sample_buf[ch][i - 1]; - } - - return; - } - - // generalised linear predictor - - if (lpc.lpc_order > 0) { - int32_t *samples = s->sample_buf[ch]; - int32_t *residual = s->predictor_buf; - - // generate warm-up samples - residual[0] = samples[0]; - for (i = 1; i <= lpc.lpc_order; i++) - residual[i] = sign_extend(samples[i] - samples[i-1], s->write_sample_size); - - // perform lpc on remaining samples - for (i = lpc.lpc_order + 1; i < s->frame_size; i++) { - int sum = 1 << (lpc.lpc_quant - 1), res_val, j; - - for (j = 0; j < lpc.lpc_order; j++) { - sum += (samples[lpc.lpc_order-j] - samples[0]) * - lpc.lpc_coeff[j]; - } - - sum >>= lpc.lpc_quant; - sum += samples[0]; - residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum, - s->write_sample_size); - res_val = residual[i]; - - if (res_val) { - int index = lpc.lpc_order - 1; - int neg = (res_val < 0); - - while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) { - int val = samples[0] - samples[lpc.lpc_order - index]; - int sign = (val ? FFSIGN(val) : 0); - - if (neg) - sign *= -1; - - lpc.lpc_coeff[index] -= sign; - val *= sign; - res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index); - index--; - } - } - samples++; - } - } -} - -static void alac_entropy_coder(AlacEncodeContext *s) -{ - unsigned int history = s->rc.initial_history; - int sign_modifier = 0, i, k; - int32_t *samples = s->predictor_buf; - - for (i = 0; i < s->frame_size;) { - int x; - - k = av_log2((history >> 9) + 3); - - x = -2 * (*samples) -1; - x ^= x >> 31; - - samples++; - i++; - - encode_scalar(s, x - sign_modifier, k, s->write_sample_size); - - history += x * s->rc.history_mult - - ((history * s->rc.history_mult) >> 9); - - sign_modifier = 0; - if (x > 0xFFFF) - history = 0xFFFF; - - if (history < 128 && i < s->frame_size) { - unsigned int block_size = 0; - - k = 7 - av_log2(history) + ((history + 16) >> 6); - - while (*samples == 0 && i < s->frame_size) { - samples++; - i++; - block_size++; - } - encode_scalar(s, block_size, k, 16); - sign_modifier = (block_size <= 0xFFFF); - history = 0; - } - - } -} - -static void write_element(AlacEncodeContext *s, - enum AlacRawDataBlockType element, int instance, - const uint8_t *samples0, const uint8_t *samples1) -{ - uint8_t const *samples[2] = { samples0, samples1 }; - int i, j, channels; - int prediction_type = 0; - PutBitContext *pb = &s->pbctx; - - channels = element == TYPE_CPE ? 2 : 1; - - if (s->verbatim) { - write_element_header(s, element, instance); - /* samples are channel-interleaved in verbatim mode */ - if (s->avctx->sample_fmt == AV_SAMPLE_FMT_S32P) { - int shift = 32 - s->avctx->bits_per_raw_sample; - int32_t const *samples_s32[2] = { (const int32_t *)samples0, - (const int32_t *)samples1 }; - for (i = 0; i < s->frame_size; i++) - for (j = 0; j < channels; j++) - put_sbits(pb, s->avctx->bits_per_raw_sample, - samples_s32[j][i] >> shift); - } else { - int16_t const *samples_s16[2] = { (const int16_t *)samples0, - (const int16_t *)samples1 }; - for (i = 0; i < s->frame_size; i++) - for (j = 0; j < channels; j++) - put_sbits(pb, s->avctx->bits_per_raw_sample, - samples_s16[j][i]); - } - } else { - s->write_sample_size = s->avctx->bits_per_raw_sample - s->extra_bits + - channels - 1; - - init_sample_buffers(s, channels, samples); - write_element_header(s, element, instance); - - if (channels == 2) - alac_stereo_decorrelation(s); - else - s->interlacing_shift = s->interlacing_leftweight = 0; - put_bits(pb, 8, s->interlacing_shift); - put_bits(pb, 8, s->interlacing_leftweight); - - for (i = 0; i < channels; i++) { - calc_predictor_params(s, i); - - put_bits(pb, 4, prediction_type); - put_bits(pb, 4, s->lpc[i].lpc_quant); - - put_bits(pb, 3, s->rc.rice_modifier); - put_bits(pb, 5, s->lpc[i].lpc_order); - // predictor coeff. table - for (j = 0; j < s->lpc[i].lpc_order; j++) - put_sbits(pb, 16, s->lpc[i].lpc_coeff[j]); - } - - // write extra bits if needed - if (s->extra_bits) { - uint32_t mask = (1 << s->extra_bits) - 1; - for (i = 0; i < s->frame_size; i++) { - for (j = 0; j < channels; j++) { - put_bits(pb, s->extra_bits, s->sample_buf[j][i] & mask); - s->sample_buf[j][i] >>= s->extra_bits; - } - } - } - - // apply lpc and entropy coding to audio samples - for (i = 0; i < channels; i++) { - alac_linear_predictor(s, i); - - // TODO: determine when this will actually help. for now it's not used. - if (prediction_type == 15) { - // 2nd pass 1st order filter - for (j = s->frame_size - 1; j > 0; j--) - s->predictor_buf[j] -= s->predictor_buf[j - 1]; - } - alac_entropy_coder(s); - } - } -} - -static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, - uint8_t * const *samples) -{ - PutBitContext *pb = &s->pbctx; - const enum AlacRawDataBlockType *ch_elements = ff_alac_channel_elements[s->avctx->channels - 1]; - const uint8_t *ch_map = ff_alac_channel_layout_offsets[s->avctx->channels - 1]; - int ch, element, sce, cpe; - - init_put_bits(pb, avpkt->data, avpkt->size); - - ch = element = sce = cpe = 0; - while (ch < s->avctx->channels) { - if (ch_elements[element] == TYPE_CPE) { - write_element(s, TYPE_CPE, cpe, samples[ch_map[ch]], - samples[ch_map[ch + 1]]); - cpe++; - ch += 2; - } else { - write_element(s, TYPE_SCE, sce, samples[ch_map[ch]], NULL); - sce++; - ch++; - } - element++; - } - - put_bits(pb, 3, TYPE_END); - flush_put_bits(pb); - - return put_bits_count(pb) >> 3; -} - -static av_always_inline int get_max_frame_size(int frame_size, int ch, int bps) -{ - int header_bits = 23 + 32 * (frame_size < DEFAULT_FRAME_SIZE); - return FFALIGN(header_bits + bps * ch * frame_size + 3, 8) / 8; -} - -static av_cold int alac_encode_close(AVCodecContext *avctx) -{ - AlacEncodeContext *s = avctx->priv_data; - ff_lpc_end(&s->lpc_ctx); - av_freep(&avctx->extradata); - avctx->extradata_size = 0; - return 0; -} - -static av_cold int alac_encode_init(AVCodecContext *avctx) -{ - AlacEncodeContext *s = avctx->priv_data; - int ret; - uint8_t *alac_extradata; - - avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE; - - if (avctx->sample_fmt == AV_SAMPLE_FMT_S32P) { - if (avctx->bits_per_raw_sample != 24) - av_log(avctx, AV_LOG_WARNING, "encoding as 24 bits-per-sample\n"); - avctx->bits_per_raw_sample = 24; - } else { - avctx->bits_per_raw_sample = 16; - s->extra_bits = 0; - } - - // Set default compression level - if (avctx->compression_level == FF_COMPRESSION_DEFAULT) - s->compression_level = 2; - else - s->compression_level = av_clip(avctx->compression_level, 0, 2); - - // Initialize default Rice parameters - s->rc.history_mult = 40; - s->rc.initial_history = 10; - s->rc.k_modifier = 14; - s->rc.rice_modifier = 4; - - s->max_coded_frame_size = get_max_frame_size(avctx->frame_size, - avctx->channels, - avctx->bits_per_raw_sample); - - avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); - if (!avctx->extradata) { - ret = AVERROR(ENOMEM); - goto error; - } - avctx->extradata_size = ALAC_EXTRADATA_SIZE; - - alac_extradata = avctx->extradata; - AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE); - AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c')); - AV_WB32(alac_extradata+12, avctx->frame_size); - AV_WB8 (alac_extradata+17, avctx->bits_per_raw_sample); - AV_WB8 (alac_extradata+21, avctx->channels); - AV_WB32(alac_extradata+24, s->max_coded_frame_size); - AV_WB32(alac_extradata+28, - avctx->sample_rate * avctx->channels * avctx->bits_per_raw_sample); // average bitrate - AV_WB32(alac_extradata+32, avctx->sample_rate); - - // Set relevant extradata fields - if (s->compression_level > 0) { - AV_WB8(alac_extradata+18, s->rc.history_mult); - AV_WB8(alac_extradata+19, s->rc.initial_history); - AV_WB8(alac_extradata+20, s->rc.k_modifier); - } - - s->min_prediction_order = DEFAULT_MIN_PRED_ORDER; - if (avctx->min_prediction_order >= 0) { - if (avctx->min_prediction_order < MIN_LPC_ORDER || - avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) { - av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", - avctx->min_prediction_order); - ret = AVERROR(EINVAL); - goto error; - } - - s->min_prediction_order = avctx->min_prediction_order; - } - - s->max_prediction_order = DEFAULT_MAX_PRED_ORDER; - if (avctx->max_prediction_order >= 0) { - if (avctx->max_prediction_order < MIN_LPC_ORDER || - avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) { - av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", - avctx->max_prediction_order); - ret = AVERROR(EINVAL); - goto error; - } - - s->max_prediction_order = avctx->max_prediction_order; - } - - if (s->max_prediction_order < s->min_prediction_order) { - av_log(avctx, AV_LOG_ERROR, - "invalid prediction orders: min=%d max=%d\n", - s->min_prediction_order, s->max_prediction_order); - ret = AVERROR(EINVAL); - goto error; - } - - s->avctx = avctx; - - if ((ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, - s->max_prediction_order, - FF_LPC_TYPE_LEVINSON)) < 0) { - goto error; - } - - return 0; -error: - alac_encode_close(avctx); - return ret; -} - -static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, - const AVFrame *frame, int *got_packet_ptr) -{ - AlacEncodeContext *s = avctx->priv_data; - int out_bytes, max_frame_size, ret; - - s->frame_size = frame->nb_samples; - - if (frame->nb_samples < DEFAULT_FRAME_SIZE) - max_frame_size = get_max_frame_size(s->frame_size, avctx->channels, - avctx->bits_per_raw_sample); - else - max_frame_size = s->max_coded_frame_size; - - if ((ret = ff_alloc_packet2(avctx, avpkt, 2 * max_frame_size)) < 0) - return ret; - - /* use verbatim mode for compression_level 0 */ - if (s->compression_level) { - s->verbatim = 0; - s->extra_bits = avctx->bits_per_raw_sample - 16; - } else { - s->verbatim = 1; - s->extra_bits = 0; - } - - out_bytes = write_frame(s, avpkt, frame->extended_data); - - if (out_bytes > max_frame_size) { - /* frame too large. use verbatim mode */ - s->verbatim = 1; - s->extra_bits = 0; - out_bytes = write_frame(s, avpkt, frame->extended_data); - } - - avpkt->size = out_bytes; - *got_packet_ptr = 1; - return 0; -} - -AVCodec ff_alac_encoder = { - .name = "alac", - .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_ALAC, - .priv_data_size = sizeof(AlacEncodeContext), - .init = alac_encode_init, - .encode2 = alac_encode_frame, - .close = alac_encode_close, - .capabilities = CODEC_CAP_SMALL_LAST_FRAME, - .channel_layouts = ff_alac_channel_layouts, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P, - AV_SAMPLE_FMT_S16P, - AV_SAMPLE_FMT_NONE }, -}; 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