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https://gitee.com/openharmony/third_party_ffmpeg
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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>
238 lines
7.3 KiB
C
238 lines
7.3 KiB
C
/*
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* Interface to libfaac for aac encoding
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* Copyright (c) 2002 Gildas Bazin <gbazin@netcourrier.com>
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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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/**
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* @file
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* Interface to libfaac for aac encoding.
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*/
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#include <faac.h>
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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/* libfaac has an encoder delay of 1024 samples */
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#define FAAC_DELAY_SAMPLES 1024
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typedef struct FaacAudioContext {
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faacEncHandle faac_handle;
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AudioFrameQueue afq;
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} FaacAudioContext;
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static const int channel_maps[][6] = {
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{ 2, 0, 1 }, //< C L R
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{ 2, 0, 1, 3 }, //< C L R Cs
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{ 2, 0, 1, 3, 4 }, //< C L R Ls Rs
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{ 2, 0, 1, 4, 5, 3 }, //< C L R Ls Rs LFE
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};
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static av_cold int Faac_encode_close(AVCodecContext *avctx)
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{
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FaacAudioContext *s = avctx->priv_data;
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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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av_freep(&avctx->extradata);
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ff_af_queue_close(&s->afq);
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if (s->faac_handle)
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faacEncClose(s->faac_handle);
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return 0;
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}
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static av_cold int Faac_encode_init(AVCodecContext *avctx)
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{
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FaacAudioContext *s = avctx->priv_data;
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faacEncConfigurationPtr faac_cfg;
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unsigned long samples_input, max_bytes_output;
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int ret;
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/* number of channels */
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if (avctx->channels < 1 || avctx->channels > 6) {
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av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed\n", avctx->channels);
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ret = AVERROR(EINVAL);
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goto error;
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}
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s->faac_handle = faacEncOpen(avctx->sample_rate,
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avctx->channels,
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&samples_input, &max_bytes_output);
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if (!s->faac_handle) {
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av_log(avctx, AV_LOG_ERROR, "error in faacEncOpen()\n");
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ret = AVERROR_UNKNOWN;
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goto error;
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}
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/* check faac version */
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faac_cfg = faacEncGetCurrentConfiguration(s->faac_handle);
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if (faac_cfg->version != FAAC_CFG_VERSION) {
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av_log(avctx, AV_LOG_ERROR, "wrong libfaac version (compiled for: %d, using %d)\n", FAAC_CFG_VERSION, faac_cfg->version);
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ret = AVERROR(EINVAL);
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goto error;
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}
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/* put the options in the configuration struct */
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switch(avctx->profile) {
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case FF_PROFILE_AAC_MAIN:
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faac_cfg->aacObjectType = MAIN;
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break;
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case FF_PROFILE_UNKNOWN:
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case FF_PROFILE_AAC_LOW:
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faac_cfg->aacObjectType = LOW;
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break;
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case FF_PROFILE_AAC_SSR:
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faac_cfg->aacObjectType = SSR;
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break;
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case FF_PROFILE_AAC_LTP:
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faac_cfg->aacObjectType = LTP;
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break;
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default:
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av_log(avctx, AV_LOG_ERROR, "invalid AAC profile\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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faac_cfg->mpegVersion = MPEG4;
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faac_cfg->useTns = 0;
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faac_cfg->allowMidside = 1;
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faac_cfg->bitRate = avctx->bit_rate / avctx->channels;
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faac_cfg->bandWidth = avctx->cutoff;
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if(avctx->flags & CODEC_FLAG_QSCALE) {
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faac_cfg->bitRate = 0;
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faac_cfg->quantqual = avctx->global_quality / FF_QP2LAMBDA;
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}
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faac_cfg->outputFormat = 1;
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faac_cfg->inputFormat = FAAC_INPUT_16BIT;
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if (avctx->channels > 2)
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memcpy(faac_cfg->channel_map, channel_maps[avctx->channels-3],
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avctx->channels * sizeof(int));
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avctx->frame_size = samples_input / avctx->channels;
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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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/* Set decoder specific info */
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avctx->extradata_size = 0;
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if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
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unsigned char *buffer = NULL;
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unsigned long decoder_specific_info_size;
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if (!faacEncGetDecoderSpecificInfo(s->faac_handle, &buffer,
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&decoder_specific_info_size)) {
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avctx->extradata = av_malloc(decoder_specific_info_size + FF_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata) {
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ret = AVERROR(ENOMEM);
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goto error;
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}
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avctx->extradata_size = decoder_specific_info_size;
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memcpy(avctx->extradata, buffer, avctx->extradata_size);
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faac_cfg->outputFormat = 0;
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}
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#undef free
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free(buffer);
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#define free please_use_av_free
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}
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if (!faacEncSetConfiguration(s->faac_handle, faac_cfg)) {
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av_log(avctx, AV_LOG_ERROR, "libfaac doesn't support this output format!\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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avctx->delay = FAAC_DELAY_SAMPLES;
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ff_af_queue_init(avctx, &s->afq);
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return 0;
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error:
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Faac_encode_close(avctx);
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return ret;
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}
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static int Faac_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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FaacAudioContext *s = avctx->priv_data;
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int bytes_written, ret;
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int num_samples = frame ? frame->nb_samples : 0;
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void *samples = frame ? frame->data[0] : NULL;
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if ((ret = ff_alloc_packet2(avctx, avpkt, (7 + 768) * avctx->channels)))
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return ret;
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bytes_written = faacEncEncode(s->faac_handle, samples,
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num_samples * avctx->channels,
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avpkt->data, avpkt->size);
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if (bytes_written < 0) {
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av_log(avctx, AV_LOG_ERROR, "faacEncEncode() error\n");
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return bytes_written;
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}
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/* add current frame to the queue */
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if (frame) {
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if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
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return ret;
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}
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if (!bytes_written)
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return 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 = bytes_written;
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*got_packet_ptr = 1;
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return 0;
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}
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static const AVProfile profiles[] = {
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{ FF_PROFILE_AAC_MAIN, "Main" },
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{ FF_PROFILE_AAC_LOW, "LC" },
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{ FF_PROFILE_AAC_SSR, "SSR" },
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{ FF_PROFILE_AAC_LTP, "LTP" },
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{ FF_PROFILE_UNKNOWN },
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};
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AVCodec ff_libfaac_encoder = {
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.name = "libfaac",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_AAC,
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.priv_data_size = sizeof(FaacAudioContext),
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.init = Faac_encode_init,
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.encode2 = Faac_encode_frame,
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.close = Faac_encode_close,
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.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
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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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.long_name = NULL_IF_CONFIG_SMALL("libfaac AAC (Advanced Audio Codec)"),
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.profiles = NULL_IF_CONFIG_SMALL(profiles),
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};
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