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305e4b35ea
* qatar/master: (34 commits) mlp_parser: fix the channel mask value used for the top surround channel vorbisenc: check all allocations for failure roqaudioenc: return AVERROR codes instead of -1 roqaudioenc: set correct bit rate roqaudioenc: use AVCodecContext.frame_size correctly. roqaudioenc: remove unneeded sample_fmt check ra144enc: use int16_t* for input samples rather than void* ra144enc: set AVCodecContext.coded_frame ra144enc: remove unneeded sample_fmt check nellymoserenc: set AVCodecContext.coded_frame nellymoserenc: improve error checking in encode_init() nellymoserenc: return AVERROR codes instead of -1 libvorbis: improve error checking in oggvorbis_encode_init() mpegaudioenc: return AVERROR codes instead of -1 libfaac: improve error checking and handling in Faac_encode_init() avutil: add AVERROR_UNKNOWN check for coded_frame allocation failure in several audio encoders audio encoders: do not set coded_frame->key_frame. g722enc: check for trellis data allocation error libspeexenc: export encoder delay through AVCodecContext.delay ... Conflicts: doc/APIchanges libavcodec/avcodec.h libavcodec/fraps.c libavcodec/kgv1dec.c libavcodec/libfaac.c libavcodec/libgsm.c libavcodec/libvorbis.c libavcodec/mlp_parser.c libavcodec/roqaudioenc.c libavcodec/vorbisenc.c libavutil/avutil.h libavutil/error.c libavutil/error.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
135 lines
4.2 KiB
C
135 lines
4.2 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 "avcodec.h"
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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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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, { 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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"libvo_amrwbenc", av_default_item_name, options, 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 correspondance 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->coded_frame = avcodec_alloc_frame();
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if (!avctx->coded_frame)
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return AVERROR(ENOMEM);
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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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av_freep(&avctx->coded_frame);
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return 0;
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}
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static int amr_wb_encode_frame(AVCodecContext *avctx,
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unsigned char *frame/*out*/,
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int buf_size, void *data/*in*/)
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{
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AMRWBContext *s = avctx->priv_data;
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int size;
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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, data, frame, s->allow_dtx);
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return size;
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}
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AVCodec ff_libvo_amrwbenc_encoder = {
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.name = "libvo_amrwbenc",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = 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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.encode = 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,AV_SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn Adaptive Multi-Rate "
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"(AMR) Wide-Band"),
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.priv_class = &class,
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};
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