X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fra144enc.c;h=5b5de76434f24d9cb0934901f94370a12a44bb42;hb=015da965a68bdb48819dc98317888fc84eced599;hp=3f8694eb8f7759273ad96d99e29133edb7a95b51;hpb=56f8952b252f85281317ecd3e0b04c4cae93fd72;p=ffmpeg diff --git a/libavcodec/ra144enc.c b/libavcodec/ra144enc.c index 3f8694eb8f7..5b5de76434f 100644 --- a/libavcodec/ra144enc.c +++ b/libavcodec/ra144enc.c @@ -2,20 +2,20 @@ * Real Audio 1.0 (14.4K) encoder * Copyright (c) 2010 Francesco Lavra * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -28,32 +28,61 @@ #include #include "avcodec.h" +#include "audio_frame_queue.h" +#include "internal.h" #include "put_bits.h" #include "celp_filters.h" #include "ra144.h" +static av_cold int ra144_encode_close(AVCodecContext *avctx) +{ + RA144Context *ractx = avctx->priv_data; + ff_lpc_end(&ractx->lpc_ctx); + ff_af_queue_close(&ractx->afq); +#if FF_API_OLD_ENCODE_AUDIO + av_freep(&avctx->coded_frame); +#endif + return 0; +} + + static av_cold int ra144_encode_init(AVCodecContext * avctx) { RA144Context *ractx; + int ret; - if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) { - av_log(avctx, AV_LOG_ERROR, "invalid sample format\n"); - return -1; - } if (avctx->channels != 1) { av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels); return -1; } avctx->frame_size = NBLOCKS * BLOCKSIZE; + avctx->delay = avctx->frame_size; avctx->bit_rate = 8000; ractx = avctx->priv_data; ractx->lpc_coef[0] = ractx->lpc_tables[0]; ractx->lpc_coef[1] = ractx->lpc_tables[1]; ractx->avctx = avctx; - ff_lpc_init(&ractx->lpc_ctx); + ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER, + FF_LPC_TYPE_LEVINSON); + if (ret < 0) + goto error; + + ff_af_queue_init(avctx, &ractx->afq); + +#if FF_API_OLD_ENCODE_AUDIO + avctx->coded_frame = avcodec_alloc_frame(); + if (!avctx->coded_frame) { + ret = AVERROR(ENOMEM); + goto error; + } +#endif + return 0; +error: + ra144_encode_close(avctx); + return ret; } @@ -204,7 +233,7 @@ static int adaptive_cb_search(const int16_t *adapt_cb, float *work, ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER); for (i = 0; i < BLOCKSIZE; i++) data[i] -= best_gain * work[i]; - return (best_vect - BLOCKSIZE / 2 + 1); + return best_vect - BLOCKSIZE / 2 + 1; } @@ -313,7 +342,7 @@ static void ra144_encode_subblock(RA144Context *ractx, const int16_t *lpc_coefs, unsigned int rms, PutBitContext *pb) { - float data[BLOCKSIZE], work[LPC_ORDER + BLOCKSIZE]; + float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE]; float coefs[LPC_ORDER]; float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE]; int16_t cba_vect[BLOCKSIZE]; @@ -331,7 +360,6 @@ static void ra144_encode_subblock(RA144Context *ractx, * Calculate the zero-input response of the LPC filter and subtract it from * input data. */ - memset(data, 0, sizeof(data)); ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE, LPC_ORDER); for (i = 0; i < BLOCKSIZE; i++) { @@ -409,12 +437,12 @@ static void ra144_encode_subblock(RA144Context *ractx, } -static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame, - int buf_size, void *data) +static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) { static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4}; static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; - RA144Context *ractx; + RA144Context *ractx = avctx->priv_data; PutBitContext pb; int32_t lpc_data[NBLOCKS * BLOCKSIZE]; int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER]; @@ -422,14 +450,17 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame, int16_t block_coefs[NBLOCKS][LPC_ORDER]; int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */ unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */ + const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL; int energy = 0; - int i, idx; + int i, idx, ret; - if (buf_size < FRAMESIZE) { - av_log(avctx, AV_LOG_ERROR, "output buffer too small\n"); + if (ractx->last_frame) return 0; + + if ((ret = ff_alloc_packet(avpkt, FRAMESIZE))) { + av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + return ret; } - ractx = avctx->priv_data; /** * Since the LPC coefficients are calculated on a frame centered over the @@ -442,16 +473,20 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame, lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i]; energy += (lpc_data[i] * lpc_data[i]) >> 4; } - for (i = 2 * BLOCKSIZE + BLOCKSIZE / 2; i < NBLOCKS * BLOCKSIZE; i++) { - lpc_data[i] = *((int16_t *)data + i - 2 * BLOCKSIZE - BLOCKSIZE / 2) >> - 2; - energy += (lpc_data[i] * lpc_data[i]) >> 4; + if (frame) { + int j; + for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) { + lpc_data[i] = samples[j] >> 2; + energy += (lpc_data[i] * lpc_data[i]) >> 4; + } } + if (i < NBLOCKS * BLOCKSIZE) + memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data)); energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab, 32)]; ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER, - LPC_ORDER, 16, lpc_coefs, shift, AV_LPC_TYPE_LEVINSON, + LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON, 0, ORDER_METHOD_EST, 12, 0); for (i = 0; i < LPC_ORDER; i++) block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] << @@ -467,9 +502,12 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame, * The filter is unstable: use the coefficients of the previous frame. */ ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]); - ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx); + if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) { + /* the filter is still unstable. set reflection coeffs to zero. */ + memset(lpc_refl, 0, sizeof(lpc_refl)); + } } - init_put_bits(&pb, frame, buf_size); + init_put_bits(&pb, avpkt->data, avpkt->size); for (i = 0; i < LPC_ORDER; i++) { idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]); put_bits(&pb, bit_sizes[i], idx); @@ -492,19 +530,40 @@ static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame, ractx->old_energy = energy; ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0]; FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); - for (i = 0; i < NBLOCKS * BLOCKSIZE; i++) - ractx->curr_block[i] = *((int16_t *)data + i) >> 2; - return FRAMESIZE; + + /* copy input samples to current block for processing in next call */ + i = 0; + if (frame) { + for (; i < frame->nb_samples; i++) + ractx->curr_block[i] = samples[i] >> 2; + + if ((ret = ff_af_queue_add(&ractx->afq, frame) < 0)) + return ret; + } else + ractx->last_frame = 1; + memset(&ractx->curr_block[i], 0, + (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block)); + + /* Get the next frame pts/duration */ + ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts, + &avpkt->duration); + + avpkt->size = FRAMESIZE; + *got_packet_ptr = 1; + return 0; } -AVCodec ra_144_encoder = -{ - "real_144", - AVMEDIA_TYPE_AUDIO, - CODEC_ID_RA_144, - sizeof(RA144Context), - ra144_encode_init, - ra144_encode_frame, - .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K) encoder"), +AVCodec ff_ra_144_encoder = { + .name = "real_144", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_RA_144, + .priv_data_size = sizeof(RA144Context), + .init = ra144_encode_init, + .encode2 = ra144_encode_frame, + .close = ra144_encode_close, + .capabilities = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, + .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), };