mirror of https://github.com/FFmpeg/FFmpeg.git
* qatar/master: build: Fix Ogg demuxer dependencies build: Fix FLAC demuxer dependencies flac: Move flac functions shared between libraries to flac common code build: Fix CAF demuxer dependencies build: Fix MP2 muxer dependencies build: Add missing build rules for the ISMV muxer configure: Drop redundant mxf_d10 test dependency declaration Support AAC encoding via the external library fdk-aac libavcodec: Add more AAC profiles dct/fft-test: use a replacement getopt() if the system has none present. Conflicts: Changelog libavcodec/Makefile libavcodec/version.h Merged-by: Michael Niedermayer <michaelni@gmx.at>pull/59/head
Michael Niedermayer
13 years ago
commit
620c6292b1
15 changed files with 602 additions and 91 deletions
@ -0,0 +1,84 @@ |
||||
/*
|
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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 |
||||
* License as published by the Free Software Foundation; either |
||||
* version 2.1 of the License, or (at your option) any later version. |
||||
* |
||||
* FFmpeg is distributed in the hope that it will be useful, |
||||
* but WITHOUT ANY WARRANTY; without even the implied warranty of |
||||
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
||||
* Lesser General Public License for more details. |
||||
* |
||||
* You should have received a copy of the GNU Lesser General Public |
||||
* 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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/*
|
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* This file was copied from the following newsgroup posting: |
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* |
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* Newsgroups: mod.std.unix |
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* Subject: public domain AT&T getopt source |
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* Date: 3 Nov 85 19:34:15 GMT |
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* |
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* Here's something you've all been waiting for: the AT&T public domain |
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* source for getopt(3). It is the code which was given out at the 1985 |
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* UNIFORUM conference in Dallas. I obtained it by electronic mail |
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* directly from AT&T. The people there assure me that it is indeed |
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* in the public domain. |
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*/ |
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|
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#define EOF (-1) |
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|
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static int opterr = 1; |
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static int optind = 1; |
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static int optopt; |
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static char *optarg; |
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|
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#undef fprintf |
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|
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static int getopt(int argc, char *argv[], char *opts) |
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{ |
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static int sp = 1; |
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int c; |
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char *cp; |
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|
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if (sp == 1) |
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if (optind >= argc || |
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argv[optind][0] != '-' || argv[optind][1] == '\0') |
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return EOF; |
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else if (!strcmp(argv[optind], "--")) { |
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optind++; |
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return EOF; |
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} |
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optopt = c = argv[optind][sp]; |
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if (c == ':' || (cp = strchr(opts, c)) == NULL) { |
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fprintf(stderr, ": illegal option -- %c\n", c); |
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if (argv[optind][++sp] == '\0') { |
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optind++; |
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sp = 1; |
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} |
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return '?'; |
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} |
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if (*++cp == ':') { |
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if (argv[optind][sp+1] != '\0') |
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optarg = &argv[optind++][sp+1]; |
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else if(++optind >= argc) { |
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fprintf(stderr, ": option requires an argument -- %c\n", c); |
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sp = 1; |
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return '?'; |
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} else |
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optarg = argv[optind++]; |
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sp = 1; |
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} else { |
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if (argv[optind][++sp] == '\0') { |
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sp = 1; |
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optind++; |
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} |
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optarg = NULL; |
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} |
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|
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return c; |
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} |
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@ -0,0 +1,384 @@ |
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/*
|
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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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|
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#include <fdk-aac/aacenc_lib.h> |
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|
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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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|
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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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|
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AudioFrameQueue afq; |
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} AACContext; |
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|
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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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{ NULL } |
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}; |
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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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|
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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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|
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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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|
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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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|
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return 0; |
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} |
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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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|
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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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|
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if (avctx->profile != FF_PROFILE_UNKNOWN) |
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aot = avctx->profile + 1; |
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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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 : 2)) != AACENC_OK) { |
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av_log(avctx, AV_LOG_ERROR, "Unable to set the transmux format: %s\n", |
||||
aac_get_error(err)); |
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goto error; |
||||
} |
||||
|
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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) |
||||
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", |
||||
s->signaling, aac_get_error(err)); |
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goto error; |
||||
} |
||||
|
||||
if ((err = aacEncoder_SetParam(s->handle, AACENC_AFTERBURNER, |
||||
s->afterburner)) != AACENC_OK) { |
||||
av_log(avctx, AV_LOG_ERROR, "Unable to set afterburner to %d: %s\n", |
||||
s->afterburner, aac_get_error(err)); |
||||
goto error; |
||||
} |
||||
|
||||
if ((err = aacEncEncode(s->handle, NULL, NULL, NULL, NULL)) != AACENC_OK) { |
||||
av_log(avctx, AV_LOG_ERROR, "Unable to initialize the encoder: %s\n", |
||||
aac_get_error(err)); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
|
||||
if ((err = aacEncInfo(s->handle, &info)) != AACENC_OK) { |
||||
av_log(avctx, AV_LOG_ERROR, "Unable to get encoder info: %s\n", |
||||
aac_get_error(err)); |
||||
goto error; |
||||
} |
||||
|
||||
#if FF_API_OLD_ENCODE_AUDIO |
||||
avctx->coded_frame = avcodec_alloc_frame(); |
||||
if (!avctx->coded_frame) { |
||||
ret = AVERROR(ENOMEM); |
||||
goto error; |
||||
} |
||||
#endif |
||||
avctx->frame_size = info.frameLength; |
||||
avctx->delay = info.encoderDelay; |
||||
ff_af_queue_init(avctx, &s->afq); |
||||
|
||||
if (avctx->flags & CODEC_FLAG_GLOBAL_HEADER) { |
||||
avctx->extradata_size = info.confSize; |
||||
avctx->extradata = av_mallocz(avctx->extradata_size + |
||||
FF_INPUT_BUFFER_PADDING_SIZE); |
||||
if (!avctx->extradata) { |
||||
ret = AVERROR(ENOMEM); |
||||
goto error; |
||||
} |
||||
|
||||
memcpy(avctx->extradata, info.confBuf, info.confSize); |
||||
} |
||||
return 0; |
||||
error: |
||||
aac_encode_close(avctx); |
||||
return ret; |
||||
} |
||||
|
||||
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, |
||||
const AVFrame *frame, int *got_packet_ptr) |
||||
{ |
||||
AACContext *s = avctx->priv_data; |
||||
AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 }; |
||||
AACENC_InArgs in_args = { 0 }; |
||||
AACENC_OutArgs out_args = { 0 }; |
||||
int in_buffer_identifier = IN_AUDIO_DATA; |
||||
int in_buffer_size, in_buffer_element_size; |
||||
int out_buffer_identifier = OUT_BITSTREAM_DATA; |
||||
int out_buffer_size, out_buffer_element_size; |
||||
void *in_ptr, *out_ptr; |
||||
int ret; |
||||
AACENC_ERROR err; |
||||
|
||||
/* handle end-of-stream small frame and flushing */ |
||||
if (!frame) { |
||||
in_args.numInSamples = -1; |
||||
} else { |
||||
in_ptr = frame->data[0]; |
||||
in_buffer_size = 2 * avctx->channels * frame->nb_samples; |
||||
in_buffer_element_size = 2; |
||||
|
||||
in_args.numInSamples = avctx->channels * frame->nb_samples; |
||||
in_buf.numBufs = 1; |
||||
in_buf.bufs = &in_ptr; |
||||
in_buf.bufferIdentifiers = &in_buffer_identifier; |
||||
in_buf.bufSizes = &in_buffer_size; |
||||
in_buf.bufElSizes = &in_buffer_element_size; |
||||
|
||||
/* add current frame to the queue */ |
||||
if ((ret = ff_af_queue_add(&s->afq, frame) < 0)) |
||||
return ret; |
||||
} |
||||
|
||||
/* The maximum packet size is 6144 bits aka 768 bytes per channel. */ |
||||
if ((ret = ff_alloc_packet(avpkt, FFMAX(8192, 768 * avctx->channels)))) { |
||||
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); |
||||
return ret; |
||||
} |
||||
|
||||
out_ptr = avpkt->data; |
||||
out_buffer_size = avpkt->size; |
||||
out_buffer_element_size = 1; |
||||
out_buf.numBufs = 1; |
||||
out_buf.bufs = &out_ptr; |
||||
out_buf.bufferIdentifiers = &out_buffer_identifier; |
||||
out_buf.bufSizes = &out_buffer_size; |
||||
out_buf.bufElSizes = &out_buffer_element_size; |
||||
|
||||
if ((err = aacEncEncode(s->handle, &in_buf, &out_buf, &in_args, |
||||
&out_args)) != AACENC_OK) { |
||||
if (!frame && err == AACENC_ENCODE_EOF) |
||||
return 0; |
||||
av_log(avctx, AV_LOG_ERROR, "Unable to encode frame: %s\n", |
||||
aac_get_error(err)); |
||||
return AVERROR(EINVAL); |
||||
} |
||||
|
||||
if (!out_args.numOutBytes) |
||||
return 0; |
||||
|
||||
/* Get the next frame pts & duration */ |
||||
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, |
||||
&avpkt->duration); |
||||
|
||||
avpkt->size = out_args.numOutBytes; |
||||
*got_packet_ptr = 1; |
||||
return 0; |
||||
} |
||||
|
||||
static const AVProfile profiles[] = { |
||||
{ FF_PROFILE_AAC_LOW, "LC" }, |
||||
{ FF_PROFILE_AAC_HE, "HE-AAC" }, |
||||
{ FF_PROFILE_AAC_HE_V2, "HE-AACv2" }, |
||||
{ FF_PROFILE_AAC_LD, "LD" }, |
||||
{ FF_PROFILE_AAC_ELD, "ELD" }, |
||||
{ FF_PROFILE_UNKNOWN }, |
||||
}; |
||||
|
||||
static const AVCodecDefault aac_encode_defaults[] = { |
||||
{ "b", "0" }, |
||||
{ NULL } |
||||
}; |
||||
|
||||
static const uint64_t aac_channel_layout[] = { |
||||
AV_CH_LAYOUT_MONO, |
||||
AV_CH_LAYOUT_STEREO, |
||||
AV_CH_LAYOUT_SURROUND, |
||||
AV_CH_LAYOUT_4POINT0, |
||||
AV_CH_LAYOUT_5POINT0_BACK, |
||||
AV_CH_LAYOUT_5POINT1_BACK, |
||||
0, |
||||
}; |
||||
|
||||
AVCodec ff_libfdk_aac_encoder = { |
||||
.name = "libfdk_aac", |
||||
.type = AVMEDIA_TYPE_AUDIO, |
||||
.id = CODEC_ID_AAC, |
||||
.priv_data_size = sizeof(AACContext), |
||||
.init = aac_encode_init, |
||||
.encode2 = aac_encode_frame, |
||||
.close = aac_encode_close, |
||||
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY, |
||||
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, |
||||
AV_SAMPLE_FMT_NONE }, |
||||
.long_name = NULL_IF_CONFIG_SMALL("Fraunhofer FDK AAC"), |
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.priv_class = &aac_enc_class, |
||||
.defaults = aac_encode_defaults, |
||||
.profiles = profiles, |
||||
.channel_layouts = aac_channel_layout, |
||||
}; |
||||
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Reference in new issue