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47687a2f8a
Explicitly identify decoder/encoder wrappers with a common name. This saves API users from guessing by the name suffix. For example, they don't have to guess that "h264_qsv" is the h264 QSV implementation, and instead they can just check the AVCodec .codec and .wrapper_name fields. Explicitly mark AVCodec entries that are hardware decoders or most likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing API users listing hardware decoders in a more generic way. The proposed AVCodecHWConfig does not provide this information fully, because it's concerned with decoder configuration, not information about the fact whether the hardware is used or not. AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software implementations in case the hardware is not capable. Based on a patch by Philip Langdale <philipl@overt.org>. Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
159 lines
5.0 KiB
C
159 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 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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#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 "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 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) {
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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(avpkt, MAX_PACKET_SIZE))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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return ret;
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}
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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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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 = &class,
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.wrapper_name = "libvo_amrwbenc",
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};
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