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56e9e0273a
Up until now, ff_alloc_packet2() has a min_size parameter: It is supposed to be a lower bound on the final size of the packet to allocate. If it is not too far from the upper bound (namely, if it is at least half the upper bound), then ff_alloc_packet2() already allocates the final, already refcounted packet; if it is not, then the packet is not refcounted and its data only points to a buffer owned by the AVCodecContext (in this case, the packet will be made refcounted in encode_simple_internal() in libavcodec/encode.c). The goal of this was to avoid data copies and intermediate buffers if one has a precise lower bound. Yet those encoders for which precise lower bounds exist have recently been switched to ff_get_encode_buffer() (which automatically allocates final buffers), leaving only two encoders to actually set the min_size to something else than zero (namely aliaspixenc and hapenc). Both of these encoders use a very low lower bound that is not helpful in any nontrivial case. This commit therefore removes the min_size parameter as well as the codepath in ff_alloc_packet2() for the allocation of final buffers. Furthermore, the function has been renamed to ff_alloc_packet() and moved to encode.h alongside ff_get_encode_buffer(). Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
158 lines
5.0 KiB
C
158 lines
5.0 KiB
C
/*
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* AMR Audio encoder stub
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* Copyright (c) 2003 The FFmpeg project
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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#include <vo-amrwbenc/enc_if.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include "libavutil/avstring.h"
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#include "libavutil/internal.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "encode.h"
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#include "internal.h"
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#define MAX_PACKET_SIZE (1 + (477 + 7) / 8)
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typedef struct AMRWBContext {
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AVClass *av_class;
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void *state;
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int mode;
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int last_bitrate;
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int allow_dtx;
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} AMRWBContext;
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static const AVOption options[] = {
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{ "dtx", "Allow DTX (generate comfort noise)", offsetof(AMRWBContext, allow_dtx), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
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{ NULL }
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};
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static const AVClass amrwb_class = {
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.class_name = "libvo_amrwbenc",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static int get_wb_bitrate_mode(int bitrate, void *log_ctx)
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{
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/* make the correspondence between bitrate and mode */
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static const int rates[] = { 6600, 8850, 12650, 14250, 15850, 18250,
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19850, 23050, 23850 };
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int i, best = -1, min_diff = 0;
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char log_buf[200];
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for (i = 0; i < 9; i++) {
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if (rates[i] == bitrate)
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return i;
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if (best < 0 || abs(rates[i] - bitrate) < min_diff) {
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best = i;
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min_diff = abs(rates[i] - bitrate);
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}
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}
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/* no bitrate matching exactly, log a warning */
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snprintf(log_buf, sizeof(log_buf), "bitrate not supported: use one of ");
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for (i = 0; i < 9; i++)
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av_strlcatf(log_buf, sizeof(log_buf), "%.2fk, ", rates[i] / 1000.f);
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av_strlcatf(log_buf, sizeof(log_buf), "using %.2fk", rates[best] / 1000.f);
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av_log(log_ctx, AV_LOG_WARNING, "%s\n", log_buf);
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return best;
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}
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static av_cold int amr_wb_encode_init(AVCodecContext *avctx)
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{
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AMRWBContext *s = avctx->priv_data;
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if (avctx->sample_rate != 16000 && avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL) {
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av_log(avctx, AV_LOG_ERROR, "Only 16000Hz sample rate supported\n");
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return AVERROR(ENOSYS);
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}
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if (avctx->channels != 1) {
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av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
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return AVERROR(ENOSYS);
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}
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s->mode = get_wb_bitrate_mode(avctx->bit_rate, avctx);
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s->last_bitrate = avctx->bit_rate;
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avctx->frame_size = 320;
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avctx->initial_padding = 80;
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s->state = E_IF_init();
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return 0;
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}
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static int amr_wb_encode_close(AVCodecContext *avctx)
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{
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AMRWBContext *s = avctx->priv_data;
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E_IF_exit(s->state);
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return 0;
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}
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static int amr_wb_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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AMRWBContext *s = avctx->priv_data;
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const int16_t *samples = (const int16_t *)frame->data[0];
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int size, ret;
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if ((ret = ff_alloc_packet(avctx, avpkt, MAX_PACKET_SIZE)) < 0)
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return ret;
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if (s->last_bitrate != avctx->bit_rate) {
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s->mode = get_wb_bitrate_mode(avctx->bit_rate, avctx);
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s->last_bitrate = avctx->bit_rate;
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}
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size = E_IF_encode(s->state, s->mode, samples, avpkt->data, s->allow_dtx);
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if (size <= 0 || size > MAX_PACKET_SIZE) {
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av_log(avctx, AV_LOG_ERROR, "Error encoding frame\n");
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return AVERROR(EINVAL);
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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->initial_padding);
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avpkt->size = size;
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*got_packet_ptr = 1;
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return 0;
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}
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const AVCodec ff_libvo_amrwbenc_encoder = {
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.name = "libvo_amrwbenc",
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.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AMR-WB "
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"(Adaptive Multi-Rate Wide-Band)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_AMR_WB,
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.priv_data_size = sizeof(AMRWBContext),
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.init = amr_wb_encode_init,
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.encode2 = amr_wb_encode_frame,
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.close = amr_wb_encode_close,
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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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.priv_class = &amrwb_class,
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.wrapper_name = "libvo_amrwbenc",
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};
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