6101e5322f
* qatar/master: rtpdec_asf: Set the no_resync_search option for the chained asf demuxer asfdec: Add an option for not searching for the packet markers cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others cosmetics: Align codec declarations cosmetics: Convert mimic.c to utf-8 avconv: remove an unused function parameter. avconv: remove now pointless variables. avconv: drop support for building without libavfilter. nellymoserenc: fix crash due to memsetting the wrong area. libavformat: Only require first packet to be known for audio/video streams avplay: Don't try to scale timestamps if the tb isn't set Conflicts: Changelog configure ffmpeg.c libavcodec/aacenc.c libavcodec/bmpenc.c libavcodec/dnxhddec.c libavcodec/dnxhdenc.c libavcodec/ffv1.c libavcodec/flacenc.c libavcodec/fraps.c libavcodec/huffyuv.c libavcodec/libopenjpegdec.c libavcodec/mpeg12enc.c libavcodec/mpeg4videodec.c libavcodec/pamenc.c libavcodec/pgssubdec.c libavcodec/pngenc.c libavcodec/qtrleenc.c libavcodec/rawdec.c libavcodec/sgienc.c libavcodec/tiffenc.c libavcodec/v210dec.c libavcodec/wmv2dec.c libavformat/utils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
404 lines
14 KiB
C
404 lines
14 KiB
C
/*
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* Copyright (c) CMU 1993 Computer Science, Speech Group
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* Chengxiang Lu and Alex Hauptmann
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* Copyright (c) 2005 Steve Underwood <steveu at coppice.org>
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* Copyright (c) 2009 Kenan Gillet
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* Copyright (c) 2010 Martin Storsjo
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* G.722 ADPCM audio encoder
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*/
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#include "avcodec.h"
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#include "internal.h"
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#include "g722.h"
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#define FREEZE_INTERVAL 128
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/* This is an arbitrary value. Allowing insanely large values leads to strange
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problems, so we limit it to a reasonable value */
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#define MAX_FRAME_SIZE 32768
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/* We clip the value of avctx->trellis to prevent data type overflows and
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undefined behavior. Using larger values is insanely slow anyway. */
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#define MIN_TRELLIS 0
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#define MAX_TRELLIS 16
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static av_cold int g722_encode_close(AVCodecContext *avctx)
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{
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G722Context *c = avctx->priv_data;
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int i;
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for (i = 0; i < 2; i++) {
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av_freep(&c->paths[i]);
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av_freep(&c->node_buf[i]);
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av_freep(&c->nodep_buf[i]);
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}
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#if FF_API_OLD_ENCODE_AUDIO
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av_freep(&avctx->coded_frame);
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#endif
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return 0;
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}
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static av_cold int g722_encode_init(AVCodecContext * avctx)
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{
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G722Context *c = avctx->priv_data;
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int ret;
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if (avctx->channels != 1) {
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av_log(avctx, AV_LOG_ERROR, "Only mono tracks are allowed.\n");
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return AVERROR_INVALIDDATA;
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}
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c->band[0].scale_factor = 8;
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c->band[1].scale_factor = 2;
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c->prev_samples_pos = 22;
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if (avctx->trellis) {
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int frontier = 1 << avctx->trellis;
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int max_paths = frontier * FREEZE_INTERVAL;
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int i;
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for (i = 0; i < 2; i++) {
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c->paths[i] = av_mallocz(max_paths * sizeof(**c->paths));
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c->node_buf[i] = av_mallocz(2 * frontier * sizeof(**c->node_buf));
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c->nodep_buf[i] = av_mallocz(2 * frontier * sizeof(**c->nodep_buf));
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if (!c->paths[i] || !c->node_buf[i] || !c->nodep_buf[i]) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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}
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}
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if (avctx->frame_size) {
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/* validate frame size */
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if (avctx->frame_size & 1 || avctx->frame_size > MAX_FRAME_SIZE) {
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int new_frame_size;
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if (avctx->frame_size == 1)
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new_frame_size = 2;
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else if (avctx->frame_size > MAX_FRAME_SIZE)
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new_frame_size = MAX_FRAME_SIZE;
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else
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new_frame_size = avctx->frame_size - 1;
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av_log(avctx, AV_LOG_WARNING, "Requested frame size is not "
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"allowed. Using %d instead of %d\n", new_frame_size,
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avctx->frame_size);
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avctx->frame_size = new_frame_size;
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}
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} else {
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/* This is arbitrary. We use 320 because it's 20ms @ 16kHz, which is
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a common packet size for VoIP applications */
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avctx->frame_size = 320;
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}
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avctx->delay = 22;
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if (avctx->trellis) {
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/* validate trellis */
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if (avctx->trellis < MIN_TRELLIS || avctx->trellis > MAX_TRELLIS) {
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int new_trellis = av_clip(avctx->trellis, MIN_TRELLIS, MAX_TRELLIS);
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av_log(avctx, AV_LOG_WARNING, "Requested trellis value is not "
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"allowed. Using %d instead of %d\n", new_trellis,
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avctx->trellis);
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avctx->trellis = new_trellis;
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}
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}
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#if FF_API_OLD_ENCODE_AUDIO
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avctx->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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#endif
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return 0;
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error:
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g722_encode_close(avctx);
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return ret;
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}
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static const int16_t low_quant[33] = {
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35, 72, 110, 150, 190, 233, 276, 323,
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370, 422, 473, 530, 587, 650, 714, 786,
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858, 940, 1023, 1121, 1219, 1339, 1458, 1612,
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1765, 1980, 2195, 2557, 2919
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};
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static inline void filter_samples(G722Context *c, const int16_t *samples,
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int *xlow, int *xhigh)
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{
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int xout1, xout2;
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c->prev_samples[c->prev_samples_pos++] = samples[0];
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c->prev_samples[c->prev_samples_pos++] = samples[1];
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ff_g722_apply_qmf(c->prev_samples + c->prev_samples_pos - 24, &xout1, &xout2);
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*xlow = xout1 + xout2 >> 14;
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*xhigh = xout1 - xout2 >> 14;
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if (c->prev_samples_pos >= PREV_SAMPLES_BUF_SIZE) {
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memmove(c->prev_samples,
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c->prev_samples + c->prev_samples_pos - 22,
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22 * sizeof(c->prev_samples[0]));
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c->prev_samples_pos = 22;
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}
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}
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static inline int encode_high(const struct G722Band *state, int xhigh)
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{
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int diff = av_clip_int16(xhigh - state->s_predictor);
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int pred = 141 * state->scale_factor >> 8;
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/* = diff >= 0 ? (diff < pred) + 2 : diff >= -pred */
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return ((diff ^ (diff >> (sizeof(diff)*8-1))) < pred) + 2*(diff >= 0);
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}
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static inline int encode_low(const struct G722Band* state, int xlow)
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{
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int diff = av_clip_int16(xlow - state->s_predictor);
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/* = diff >= 0 ? diff : -(diff + 1) */
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int limit = diff ^ (diff >> (sizeof(diff)*8-1));
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int i = 0;
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limit = limit + 1 << 10;
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if (limit > low_quant[8] * state->scale_factor)
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i = 9;
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while (i < 29 && limit > low_quant[i] * state->scale_factor)
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i++;
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return (diff < 0 ? (i < 2 ? 63 : 33) : 61) - i;
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}
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static void g722_encode_trellis(G722Context *c, int trellis,
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uint8_t *dst, int nb_samples,
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const int16_t *samples)
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{
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int i, j, k;
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int frontier = 1 << trellis;
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struct TrellisNode **nodes[2];
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struct TrellisNode **nodes_next[2];
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int pathn[2] = {0, 0}, froze = -1;
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struct TrellisPath *p[2];
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for (i = 0; i < 2; i++) {
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nodes[i] = c->nodep_buf[i];
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nodes_next[i] = c->nodep_buf[i] + frontier;
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memset(c->nodep_buf[i], 0, 2 * frontier * sizeof(*c->nodep_buf));
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nodes[i][0] = c->node_buf[i] + frontier;
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nodes[i][0]->ssd = 0;
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nodes[i][0]->path = 0;
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nodes[i][0]->state = c->band[i];
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}
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for (i = 0; i < nb_samples >> 1; i++) {
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int xlow, xhigh;
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struct TrellisNode *next[2];
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int heap_pos[2] = {0, 0};
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for (j = 0; j < 2; j++) {
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next[j] = c->node_buf[j] + frontier*(i & 1);
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memset(nodes_next[j], 0, frontier * sizeof(**nodes_next));
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}
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filter_samples(c, &samples[2*i], &xlow, &xhigh);
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for (j = 0; j < frontier && nodes[0][j]; j++) {
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/* Only k >> 2 affects the future adaptive state, therefore testing
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* small steps that don't change k >> 2 is useless, the original
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* value from encode_low is better than them. Since we step k
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* in steps of 4, make sure range is a multiple of 4, so that
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* we don't miss the original value from encode_low. */
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int range = j < frontier/2 ? 4 : 0;
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struct TrellisNode *cur_node = nodes[0][j];
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int ilow = encode_low(&cur_node->state, xlow);
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for (k = ilow - range; k <= ilow + range && k <= 63; k += 4) {
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int decoded, dec_diff, pos;
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uint32_t ssd;
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struct TrellisNode* node;
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if (k < 0)
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continue;
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decoded = av_clip((cur_node->state.scale_factor *
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ff_g722_low_inv_quant6[k] >> 10)
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+ cur_node->state.s_predictor, -16384, 16383);
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dec_diff = xlow - decoded;
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#define STORE_NODE(index, UPDATE, VALUE)\
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ssd = cur_node->ssd + dec_diff*dec_diff;\
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/* Check for wraparound. Using 64 bit ssd counters would \
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* be simpler, but is slower on x86 32 bit. */\
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if (ssd < cur_node->ssd)\
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continue;\
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if (heap_pos[index] < frontier) {\
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pos = heap_pos[index]++;\
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assert(pathn[index] < FREEZE_INTERVAL * frontier);\
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node = nodes_next[index][pos] = next[index]++;\
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node->path = pathn[index]++;\
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} else {\
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/* Try to replace one of the leaf nodes with the new \
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* one, but not always testing the same leaf position */\
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pos = (frontier>>1) + (heap_pos[index] & ((frontier>>1) - 1));\
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if (ssd >= nodes_next[index][pos]->ssd)\
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continue;\
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heap_pos[index]++;\
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node = nodes_next[index][pos];\
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}\
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node->ssd = ssd;\
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node->state = cur_node->state;\
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UPDATE;\
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c->paths[index][node->path].value = VALUE;\
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c->paths[index][node->path].prev = cur_node->path;\
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/* Sift the newly inserted node up in the heap to restore \
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* the heap property */\
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while (pos > 0) {\
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int parent = (pos - 1) >> 1;\
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if (nodes_next[index][parent]->ssd <= ssd)\
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break;\
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FFSWAP(struct TrellisNode*, nodes_next[index][parent],\
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nodes_next[index][pos]);\
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pos = parent;\
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}
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STORE_NODE(0, ff_g722_update_low_predictor(&node->state, k >> 2), k);
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}
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}
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for (j = 0; j < frontier && nodes[1][j]; j++) {
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int ihigh;
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struct TrellisNode *cur_node = nodes[1][j];
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/* We don't try to get any initial guess for ihigh via
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* encode_high - since there's only 4 possible values, test
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* them all. Testing all of these gives a much, much larger
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* gain than testing a larger range around ilow. */
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for (ihigh = 0; ihigh < 4; ihigh++) {
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int dhigh, decoded, dec_diff, pos;
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uint32_t ssd;
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struct TrellisNode* node;
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dhigh = cur_node->state.scale_factor *
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ff_g722_high_inv_quant[ihigh] >> 10;
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decoded = av_clip(dhigh + cur_node->state.s_predictor,
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-16384, 16383);
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dec_diff = xhigh - decoded;
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STORE_NODE(1, ff_g722_update_high_predictor(&node->state, dhigh, ihigh), ihigh);
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}
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}
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for (j = 0; j < 2; j++) {
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FFSWAP(struct TrellisNode**, nodes[j], nodes_next[j]);
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if (nodes[j][0]->ssd > (1 << 16)) {
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for (k = 1; k < frontier && nodes[j][k]; k++)
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nodes[j][k]->ssd -= nodes[j][0]->ssd;
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nodes[j][0]->ssd = 0;
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}
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}
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if (i == froze + FREEZE_INTERVAL) {
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p[0] = &c->paths[0][nodes[0][0]->path];
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p[1] = &c->paths[1][nodes[1][0]->path];
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for (j = i; j > froze; j--) {
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dst[j] = p[1]->value << 6 | p[0]->value;
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p[0] = &c->paths[0][p[0]->prev];
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p[1] = &c->paths[1][p[1]->prev];
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}
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froze = i;
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pathn[0] = pathn[1] = 0;
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memset(nodes[0] + 1, 0, (frontier - 1)*sizeof(**nodes));
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memset(nodes[1] + 1, 0, (frontier - 1)*sizeof(**nodes));
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}
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}
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p[0] = &c->paths[0][nodes[0][0]->path];
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p[1] = &c->paths[1][nodes[1][0]->path];
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for (j = i; j > froze; j--) {
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dst[j] = p[1]->value << 6 | p[0]->value;
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p[0] = &c->paths[0][p[0]->prev];
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p[1] = &c->paths[1][p[1]->prev];
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}
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c->band[0] = nodes[0][0]->state;
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c->band[1] = nodes[1][0]->state;
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}
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static av_always_inline void encode_byte(G722Context *c, uint8_t *dst,
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const int16_t *samples)
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{
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int xlow, xhigh, ilow, ihigh;
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filter_samples(c, samples, &xlow, &xhigh);
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ihigh = encode_high(&c->band[1], xhigh);
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ilow = encode_low (&c->band[0], xlow);
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ff_g722_update_high_predictor(&c->band[1], c->band[1].scale_factor *
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ff_g722_high_inv_quant[ihigh] >> 10, ihigh);
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ff_g722_update_low_predictor(&c->band[0], ilow >> 2);
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*dst = ihigh << 6 | ilow;
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}
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static void g722_encode_no_trellis(G722Context *c,
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uint8_t *dst, int nb_samples,
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const int16_t *samples)
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{
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int i;
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for (i = 0; i < nb_samples; i += 2)
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encode_byte(c, dst++, &samples[i]);
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}
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static int g722_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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G722Context *c = avctx->priv_data;
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const int16_t *samples = (const int16_t *)frame->data[0];
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int nb_samples, out_size, ret;
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out_size = (frame->nb_samples + 1) / 2;
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if ((ret = ff_alloc_packet2(avctx, avpkt, out_size)))
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return ret;
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nb_samples = frame->nb_samples - (frame->nb_samples & 1);
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if (avctx->trellis)
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g722_encode_trellis(c, avctx->trellis, avpkt->data, nb_samples, samples);
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else
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g722_encode_no_trellis(c, avpkt->data, nb_samples, samples);
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/* handle last frame with odd frame_size */
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if (nb_samples < frame->nb_samples) {
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int16_t last_samples[2] = { samples[nb_samples], samples[nb_samples] };
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encode_byte(c, &avpkt->data[nb_samples >> 1], last_samples);
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}
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if (frame->pts != AV_NOPTS_VALUE)
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avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
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*got_packet_ptr = 1;
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return 0;
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}
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AVCodec ff_adpcm_g722_encoder = {
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.name = "g722",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_ADPCM_G722,
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.priv_data_size = sizeof(G722Context),
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.init = g722_encode_init,
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.close = g722_encode_close,
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.encode2 = g722_encode_frame,
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.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
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.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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