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https://github.com/FFmpeg/FFmpeg.git
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f7d4e26c6a
* qatar/master: rtmp: Add a new option 'rtmp_pageurl' doc: Update the description of the rtmp_tcurl option rtmp: Make the description of the rtmp_tcurl option more generic libfdk-aacenc: add LATM/LOAS encapsulation support sctp: add port missing error message tcp: add port missing error message avfilter: Fix printf format string conversion specifier Conflicts: libavcodec/version.h libavfilter/avfilter.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
410 lines
15 KiB
C
410 lines
15 KiB
C
/*
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* AAC encoder wrapper
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* Copyright (c) 2012 Martin Storsjo
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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 <fdk-aac/aacenc_lib.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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#include "libavutil/audioconvert.h"
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#include "libavutil/opt.h"
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typedef struct AACContext {
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const AVClass *class;
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HANDLE_AACENCODER handle;
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int afterburner;
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int eld_sbr;
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int signaling;
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int latm;
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int header_period;
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AudioFrameQueue afq;
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} AACContext;
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static const AVOption aac_enc_options[] = {
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{ "afterburner", "Afterburner (improved quality)", offsetof(AACContext, afterburner), AV_OPT_TYPE_INT, { 1 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
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{ "eld_sbr", "Enable SBR for ELD (for SBR in other configurations, use the -profile parameter)", offsetof(AACContext, eld_sbr), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
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{ "signaling", "SBR/PS signaling style", offsetof(AACContext, signaling), AV_OPT_TYPE_INT, { -1 }, -1, 2, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
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{ "default", "Choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)", 0, AV_OPT_TYPE_CONST, { -1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
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{ "implicit", "Implicit backwards compatible signaling", 0, AV_OPT_TYPE_CONST, { 0 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
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{ "explicit_sbr", "Explicit SBR, implicit PS signaling", 0, AV_OPT_TYPE_CONST, { 1 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
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{ "explicit_hierarchical", "Explicit hierarchical signaling", 0, AV_OPT_TYPE_CONST, { 2 }, 0, 0, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM, "signaling" },
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{ "latm", "Output LATM/LOAS encapsulated data", offsetof(AACContext, latm), AV_OPT_TYPE_INT, { 0 }, 0, 1, AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM },
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{ "header_period", "StreamMuxConfig and PCE repetition period (in frames)", offsetof(AACContext, header_period), AV_OPT_TYPE_INT, { 0 }, 0, 0xffff, 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 aac_enc_class = {
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"libfdk_aac", av_default_item_name, aac_enc_options, LIBAVUTIL_VERSION_INT
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};
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static const char *aac_get_error(AACENC_ERROR err)
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{
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switch (err) {
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case AACENC_OK:
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return "No error";
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case AACENC_INVALID_HANDLE:
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return "Invalid handle";
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case AACENC_MEMORY_ERROR:
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return "Memory allocation error";
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case AACENC_UNSUPPORTED_PARAMETER:
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return "Unsupported parameter";
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case AACENC_INVALID_CONFIG:
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return "Invalid config";
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case AACENC_INIT_ERROR:
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return "Initialization error";
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case AACENC_INIT_AAC_ERROR:
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return "AAC library initialization error";
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case AACENC_INIT_SBR_ERROR:
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return "SBR library initialization error";
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case AACENC_INIT_TP_ERROR:
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return "Transport library initialization error";
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case AACENC_INIT_META_ERROR:
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return "Metadata library initialization error";
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case AACENC_ENCODE_ERROR:
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return "Encoding error";
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case AACENC_ENCODE_EOF:
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return "End of file";
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default:
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return "Unknown error";
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}
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}
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static int aac_encode_close(AVCodecContext *avctx)
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{
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AACContext *s = avctx->priv_data;
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if (s->handle)
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aacEncClose(&s->handle);
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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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return 0;
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}
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static av_cold int aac_encode_init(AVCodecContext *avctx)
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{
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AACContext *s = avctx->priv_data;
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int ret = AVERROR(EINVAL);
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AACENC_InfoStruct info = { 0 };
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CHANNEL_MODE mode;
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AACENC_ERROR err;
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int aot = FF_PROFILE_AAC_LOW + 1;
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int sce = 0, cpe = 0;
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if ((err = aacEncOpen(&s->handle, 0, avctx->channels)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to open the encoder: %s\n",
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aac_get_error(err));
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goto error;
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}
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if (avctx->profile != FF_PROFILE_UNKNOWN)
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aot = avctx->profile + 1;
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if ((err = aacEncoder_SetParam(s->handle, AACENC_AOT, aot)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set the AOT %d: %s\n",
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aot, aac_get_error(err));
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goto error;
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}
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if (aot == FF_PROFILE_AAC_ELD + 1 && s->eld_sbr) {
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if ((err = aacEncoder_SetParam(s->handle, AACENC_SBR_MODE,
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1)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to enable SBR for ELD: %s\n",
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aac_get_error(err));
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goto error;
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}
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_SAMPLERATE,
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avctx->sample_rate)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set the sample rate %d: %s\n",
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avctx->sample_rate, aac_get_error(err));
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goto error;
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}
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switch (avctx->channels) {
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case 1: mode = MODE_1; sce = 1; cpe = 0; break;
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case 2: mode = MODE_2; sce = 0; cpe = 1; break;
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case 3: mode = MODE_1_2; sce = 1; cpe = 1; break;
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case 4: mode = MODE_1_2_1; sce = 2; cpe = 1; break;
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case 5: mode = MODE_1_2_2; sce = 1; cpe = 2; break;
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case 6: mode = MODE_1_2_2_1; sce = 2; cpe = 2; break;
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default:
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av_log(avctx, AV_LOG_ERROR,
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"Unsupported number of channels %d\n", avctx->channels);
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goto error;
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELMODE,
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mode)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR,
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"Unable to set channel mode %d: %s\n", mode, aac_get_error(err));
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goto error;
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_CHANNELORDER,
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1)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR,
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"Unable to set wav channel order %d: %s\n",
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mode, aac_get_error(err));
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goto error;
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}
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if (avctx->flags & CODEC_FLAG_QSCALE) {
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int mode = avctx->global_quality;
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if (mode < 1 || mode > 5) {
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av_log(avctx, AV_LOG_WARNING,
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"VBR quality %d out of range, should be 1-5\n", mode);
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mode = av_clip(mode, 1, 5);
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATEMODE,
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mode)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set the VBR bitrate mode %d: %s\n",
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mode, aac_get_error(err));
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goto error;
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}
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} else {
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if (avctx->bit_rate <= 0) {
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if (avctx->profile == FF_PROFILE_AAC_HE_V2) {
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sce = 1;
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cpe = 0;
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}
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avctx->bit_rate = (96*sce + 128*cpe) * avctx->sample_rate / 44;
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if (avctx->profile == FF_PROFILE_AAC_HE ||
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avctx->profile == FF_PROFILE_AAC_HE_V2 ||
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s->eld_sbr)
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avctx->bit_rate /= 2;
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_BITRATE,
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avctx->bit_rate)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set the bitrate %d: %s\n",
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avctx->bit_rate, aac_get_error(err));
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goto error;
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}
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}
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/* Choose bitstream format - if global header is requested, use
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* raw access units, otherwise use ADTS. */
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if ((err = aacEncoder_SetParam(s->handle, AACENC_TRANSMUX,
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avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 0 : s->latm ? 10 : 2)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n",
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aac_get_error(err));
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goto error;
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}
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if (s->latm && s->header_period) {
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if ((err = aacEncoder_SetParam(s->handle, AACENC_HEADER_PERIOD,
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s->header_period)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set header period: %s\n",
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aac_get_error(err));
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goto error;
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}
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}
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/* If no signaling mode is chosen, use explicit hierarchical signaling
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* if using mp4 mode (raw access units, with global header) and
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* implicit signaling if using ADTS. */
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if (s->signaling < 0)
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s->signaling = avctx->flags & CODEC_FLAG_GLOBAL_HEADER ? 2 : 0;
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if ((err = aacEncoder_SetParam(s->handle, AACENC_SIGNALING_MODE,
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s->signaling)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set signaling mode %d: %s\n",
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s->signaling, aac_get_error(err));
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goto error;
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER,
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s->afterburner)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n",
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s->afterburner, aac_get_error(err));
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goto error;
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}
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if (avctx->cutoff > 0) {
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if (avctx->cutoff < (avctx->sample_rate + 255) >> 8) {
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av_log(avctx, AV_LOG_ERROR, "cutoff valid range is %d-20000\n",
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(avctx->sample_rate + 255) >> 8);
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goto error;
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}
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if ((err = aacEncoder_SetParam(s->handle, AACENC_BANDWIDTH,
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avctx->cutoff)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to set the encoder bandwith to %d: %s\n",
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avctx->cutoff, aac_get_error(err));
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goto error;
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}
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}
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if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n",
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aac_get_error(err));
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return AVERROR(EINVAL);
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}
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if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) {
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av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n",
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aac_get_error(err));
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goto error;
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}
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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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avctx->frame_size = info.frameLength;
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avctx->delay = info.encoderDelay;
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ff_af_queue_init(avctx, &s->afq);
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if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) {
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avctx->extradata_size = info.confSize;
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avctx->extradata = av_mallocz(avctx->extradata_size +
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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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memcpy(avctx->extradata, info.confBuf, info.confSize);
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}
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return 0;
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error:
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aac_encode_close(avctx);
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return ret;
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}
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static int aac_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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AACContext *s = avctx->priv_data;
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AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
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AACENC_InArgs in_args = { 0 };
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AACENC_OutArgs out_args = { 0 };
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int in_buffer_identifier = IN_AUDIO_DATA;
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int in_buffer_size, in_buffer_element_size;
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int out_buffer_identifier = OUT_BITSTREAM_DATA;
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int out_buffer_size, out_buffer_element_size;
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void *in_ptr, *out_ptr;
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int ret;
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AACENC_ERROR err;
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/* handle end-of-stream small frame and flushing */
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if (!frame) {
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in_args.numInSamples = -1;
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} else {
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in_ptr = frame->data[0];
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in_buffer_size = 2 * avctx->channels * frame->nb_samples;
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in_buffer_element_size = 2;
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in_args.numInSamples = avctx->channels * frame->nb_samples;
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in_buf.numBufs = 1;
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in_buf.bufs = &in_ptr;
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in_buf.bufferIdentifiers = &in_buffer_identifier;
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in_buf.bufSizes = &in_buffer_size;
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in_buf.bufElSizes = &in_buffer_element_size;
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/* add current frame to the queue */
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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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/* The maximum packet size is 6144 bits aka 768 bytes per channel. */
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if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels))))
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return ret;
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out_ptr = avpkt->data;
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out_buffer_size = avpkt->size;
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out_buffer_element_size = 1;
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out_buf.numBufs = 1;
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out_buf.bufs = &out_ptr;
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out_buf.bufferIdentifiers = &out_buffer_identifier;
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out_buf.bufSizes = &out_buffer_size;
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out_buf.bufElSizes = &out_buffer_element_size;
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if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args,
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&out_args)) != AACENC_OK) {
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if (!frame && err == AACENC_ENCODE_EOF)
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return 0;
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av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n",
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aac_get_error(err));
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return AVERROR(EINVAL);
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}
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if (!out_args.numOutBytes)
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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 = out_args.numOutBytes;
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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_LOW, "LC" },
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{ FF_PROFILE_AAC_HE, "HE-AAC" },
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{ FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
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{ FF_PROFILE_AAC_LD, "LD" },
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{ FF_PROFILE_AAC_ELD, "ELD" },
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{ FF_PROFILE_UNKNOWN },
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};
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static const AVCodecDefault aac_encode_defaults[] = {
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{ "b", "0" },
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{ NULL }
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};
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static const uint64_t aac_channel_layout[] = {
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AV_CH_LAYOUT_MONO,
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AV_CH_LAYOUT_STEREO,
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AV_CH_LAYOUT_SURROUND,
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AV_CH_LAYOUT_4POINT0,
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AV_CH_LAYOUT_5POINT0_BACK,
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AV_CH_LAYOUT_5POINT1_BACK,
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0,
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};
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AVCodec ff_libfdk_aac_encoder = {
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.name = "libfdk_aac",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_AAC,
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.priv_data_size = sizeof(AACContext),
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.init = aac_encode_init,
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.encode2 = aac_encode_frame,
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.close = aac_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("Fraunhofer FDK AAC"),
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.priv_class = &aac_enc_class,
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.defaults = aac_encode_defaults,
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.profiles = profiles,
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.channel_layouts = aac_channel_layout,
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};
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