mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-07 11:13:41 +02:00
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>
388 lines
12 KiB
C
388 lines
12 KiB
C
/*
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* AMR Audio decoder 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 <inttypes.h>
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "encode.h"
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#include "internal.h"
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#if CONFIG_LIBOPENCORE_AMRNB_DECODER || CONFIG_LIBOPENCORE_AMRWB_DECODER
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static int amr_decode_fix_avctx(AVCodecContext *avctx)
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{
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const int is_amr_wb = 1 + (avctx->codec_id == AV_CODEC_ID_AMR_WB);
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if (!avctx->sample_rate)
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avctx->sample_rate = 8000 * is_amr_wb;
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if (avctx->channels > 1) {
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avpriv_report_missing_feature(avctx, "multi-channel AMR");
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return AVERROR_PATCHWELCOME;
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}
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avctx->channels = 1;
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avctx->channel_layout = AV_CH_LAYOUT_MONO;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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return 0;
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}
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#endif
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#if CONFIG_LIBOPENCORE_AMRNB
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#include <opencore-amrnb/interf_dec.h>
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#include <opencore-amrnb/interf_enc.h>
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typedef struct AMRContext {
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AVClass *av_class;
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void *dec_state;
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void *enc_state;
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int enc_bitrate;
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int enc_mode;
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int enc_dtx;
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int enc_last_frame;
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AudioFrameQueue afq;
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} AMRContext;
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#if CONFIG_LIBOPENCORE_AMRNB_DECODER
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static av_cold int amr_nb_decode_init(AVCodecContext *avctx)
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{
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AMRContext *s = avctx->priv_data;
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int ret;
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if ((ret = amr_decode_fix_avctx(avctx)) < 0)
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return ret;
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s->dec_state = Decoder_Interface_init();
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if (!s->dec_state) {
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av_log(avctx, AV_LOG_ERROR, "Decoder_Interface_init error\n");
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return -1;
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}
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return 0;
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}
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static av_cold int amr_nb_decode_close(AVCodecContext *avctx)
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{
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AMRContext *s = avctx->priv_data;
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Decoder_Interface_exit(s->dec_state);
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return 0;
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}
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static int amr_nb_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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AVFrame *frame = data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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AMRContext *s = avctx->priv_data;
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static const uint8_t block_size[16] = { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };
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enum Mode dec_mode;
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int packet_size, ret;
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ff_dlog(avctx, "amr_decode_frame buf=%p buf_size=%d frame_count=%d!!\n",
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buf, buf_size, avctx->frame_number);
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/* get output buffer */
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frame->nb_samples = 160;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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dec_mode = (buf[0] >> 3) & 0x000F;
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packet_size = block_size[dec_mode] + 1;
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if (packet_size > buf_size) {
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av_log(avctx, AV_LOG_ERROR, "AMR frame too short (%d, should be %d)\n",
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buf_size, packet_size);
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return AVERROR_INVALIDDATA;
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}
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ff_dlog(avctx, "packet_size=%d buf= 0x%"PRIx8" %"PRIx8" %"PRIx8" %"PRIx8"\n",
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packet_size, buf[0], buf[1], buf[2], buf[3]);
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/* call decoder */
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Decoder_Interface_Decode(s->dec_state, buf, (short *)frame->data[0], 0);
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*got_frame_ptr = 1;
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return packet_size;
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}
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const AVCodec ff_libopencore_amrnb_decoder = {
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.name = "libopencore_amrnb",
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.long_name = NULL_IF_CONFIG_SMALL("OpenCORE AMR-NB (Adaptive Multi-Rate Narrow-Band)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_AMR_NB,
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.priv_data_size = sizeof(AMRContext),
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.init = amr_nb_decode_init,
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.close = amr_nb_decode_close,
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.decode = amr_nb_decode_frame,
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.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
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};
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#endif /* CONFIG_LIBOPENCORE_AMRNB_DECODER */
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#if CONFIG_LIBOPENCORE_AMRNB_ENCODER
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/* Common code for fixed and float version*/
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typedef struct AMR_bitrates {
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int rate;
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enum Mode mode;
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} AMR_bitrates;
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/* Match desired bitrate */
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static int get_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 AMR_bitrates rates[] = {
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{ 4750, MR475 }, { 5150, MR515 }, { 5900, MR59 }, { 6700, MR67 },
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{ 7400, MR74 }, { 7950, MR795 }, { 10200, MR102 }, { 12200, MR122 }
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};
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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 < 8; i++) {
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if (rates[i].rate == bitrate)
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return rates[i].mode;
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if (best < 0 || abs(rates[i].rate - bitrate) < min_diff) {
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best = i;
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min_diff = abs(rates[i].rate - 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 < 8; i++)
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av_strlcatf(log_buf, sizeof(log_buf), "%.2fk, ", rates[i].rate / 1000.f);
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av_strlcatf(log_buf, sizeof(log_buf), "using %.2fk", rates[best].rate / 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 const AVOption options[] = {
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{ "dtx", "Allow DTX (generate comfort noise)", offsetof(AMRContext, enc_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 amrnb_class = {
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.class_name = "libopencore_amrnb",
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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 av_cold int amr_nb_encode_init(AVCodecContext *avctx)
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{
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AMRContext *s = avctx->priv_data;
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if (avctx->sample_rate != 8000 && avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL) {
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av_log(avctx, AV_LOG_ERROR, "Only 8000Hz 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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avctx->frame_size = 160;
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avctx->initial_padding = 50;
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ff_af_queue_init(avctx, &s->afq);
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s->enc_state = Encoder_Interface_init(s->enc_dtx);
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if (!s->enc_state) {
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av_log(avctx, AV_LOG_ERROR, "Encoder_Interface_init error\n");
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return -1;
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}
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s->enc_mode = get_bitrate_mode(avctx->bit_rate, avctx);
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s->enc_bitrate = avctx->bit_rate;
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return 0;
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}
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static av_cold int amr_nb_encode_close(AVCodecContext *avctx)
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{
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AMRContext *s = avctx->priv_data;
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Encoder_Interface_exit(s->enc_state);
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ff_af_queue_close(&s->afq);
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return 0;
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}
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static int amr_nb_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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AMRContext *s = avctx->priv_data;
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int written, ret;
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int16_t *flush_buf = NULL;
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const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
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if (s->enc_bitrate != avctx->bit_rate) {
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s->enc_mode = get_bitrate_mode(avctx->bit_rate, avctx);
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s->enc_bitrate = avctx->bit_rate;
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}
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if ((ret = ff_alloc_packet(avctx, avpkt, 32)) < 0)
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return ret;
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if (frame) {
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if (frame->nb_samples < avctx->frame_size) {
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flush_buf = av_mallocz_array(avctx->frame_size, sizeof(*flush_buf));
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if (!flush_buf)
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return AVERROR(ENOMEM);
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memcpy(flush_buf, samples, frame->nb_samples * sizeof(*flush_buf));
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samples = flush_buf;
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if (frame->nb_samples < avctx->frame_size - avctx->initial_padding)
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s->enc_last_frame = -1;
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}
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if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) {
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av_freep(&flush_buf);
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return ret;
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}
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} else {
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if (s->enc_last_frame < 0)
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return 0;
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flush_buf = av_mallocz_array(avctx->frame_size, sizeof(*flush_buf));
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if (!flush_buf)
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return AVERROR(ENOMEM);
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samples = flush_buf;
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s->enc_last_frame = -1;
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}
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written = Encoder_Interface_Encode(s->enc_state, s->enc_mode, samples,
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avpkt->data, 0);
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ff_dlog(avctx, "amr_nb_encode_frame encoded %u bytes, bitrate %u, first byte was %#02x\n",
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written, s->enc_mode, avpkt->data[0]);
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/* Get the next frame pts/duration */
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ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
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&avpkt->duration);
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avpkt->size = written;
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*got_packet_ptr = 1;
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av_freep(&flush_buf);
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return 0;
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}
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const AVCodec ff_libopencore_amrnb_encoder = {
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.name = "libopencore_amrnb",
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.long_name = NULL_IF_CONFIG_SMALL("OpenCORE AMR-NB (Adaptive Multi-Rate Narrow-Band)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_AMR_NB,
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.priv_data_size = sizeof(AMRContext),
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.init = amr_nb_encode_init,
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.encode2 = amr_nb_encode_frame,
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.close = amr_nb_encode_close,
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.capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
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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 = &amrnb_class,
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};
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#endif /* CONFIG_LIBOPENCORE_AMRNB_ENCODER */
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#endif /* CONFIG_LIBOPENCORE_AMRNB */
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/* -----------AMR wideband ------------*/
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#if CONFIG_LIBOPENCORE_AMRWB_DECODER
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#include <opencore-amrwb/dec_if.h>
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#include <opencore-amrwb/if_rom.h>
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typedef struct AMRWBContext {
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void *state;
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} AMRWBContext;
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static av_cold int amr_wb_decode_init(AVCodecContext *avctx)
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{
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AMRWBContext *s = avctx->priv_data;
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int ret;
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if ((ret = amr_decode_fix_avctx(avctx)) < 0)
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return ret;
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s->state = D_IF_init();
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return 0;
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}
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static int amr_wb_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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AVFrame *frame = data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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AMRWBContext *s = avctx->priv_data;
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int mode, ret;
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int packet_size;
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static const uint8_t block_size[16] = {18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 6, 0, 0, 0, 1, 1};
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/* get output buffer */
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frame->nb_samples = 320;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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mode = (buf[0] >> 3) & 0x000F;
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packet_size = block_size[mode];
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if (packet_size > buf_size) {
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av_log(avctx, AV_LOG_ERROR, "AMR frame too short (%d, should be %d)\n",
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buf_size, packet_size + 1);
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return AVERROR_INVALIDDATA;
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}
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if (!packet_size) {
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av_log(avctx, AV_LOG_ERROR, "amr packet_size invalid\n");
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return AVERROR_INVALIDDATA;
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}
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D_IF_decode(s->state, buf, (short *)frame->data[0], _good_frame);
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*got_frame_ptr = 1;
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return packet_size;
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}
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static int amr_wb_decode_close(AVCodecContext *avctx)
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{
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AMRWBContext *s = avctx->priv_data;
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D_IF_exit(s->state);
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return 0;
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}
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const AVCodec ff_libopencore_amrwb_decoder = {
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.name = "libopencore_amrwb",
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.long_name = NULL_IF_CONFIG_SMALL("OpenCORE AMR-WB (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_decode_init,
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.close = amr_wb_decode_close,
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.decode = amr_wb_decode_frame,
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.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
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.wrapper_name = "libopencore_amrwb",
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};
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#endif /* CONFIG_LIBOPENCORE_AMRWB_DECODER */
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