aacenc: make a better estimate for the audio bitrate if not provided
aacenc: temporarily disable Mid/Side coding with multichannel files from Brad (maintainer)
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ajacoutot
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386887e461
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@ -1,11 +1,11 @@
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# $OpenBSD: Makefile,v 1.143 2016/01/23 14:12:28 ajacoutot Exp $
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# $OpenBSD: Makefile,v 1.144 2016/02/25 03:53:53 ajacoutot Exp $
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COMMENT= audio/video converter and streamer
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V= 20160113
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DISTNAME= ffmpeg-git-${V}
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PKGNAME= ffmpeg-${V}
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REVISION= 3
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REVISION= 4
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CATEGORIES= graphics multimedia
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MASTER_SITES= http://comstyle.com/source/
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EXTRACT_SUFX= .tar.xz
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@ -1,4 +1,4 @@
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$OpenBSD: patch-libavcodec_aacenc_c,v 1.12 2016/01/22 07:05:01 ajacoutot Exp $
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$OpenBSD: patch-libavcodec_aacenc_c,v 1.13 2016/02/25 03:53:53 ajacoutot Exp $
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aacenc: copy PRNG from the decoder
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@ -70,8 +70,12 @@ aacenc: remove FAAC-like coder
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avcodec/aacenc: Check all coefficients for finiteness
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aacenc: make a better estimate for the audio bitrate if not provided
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aacenc: temporarily disable Mid/Side coding with multichannel files
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--- libavcodec/aacenc.c.orig Wed Jan 13 15:27:48 2016
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+++ libavcodec/aacenc.c Wed Jan 20 21:30:40 2016
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+++ libavcodec/aacenc.c Mon Feb 15 17:12:33 2016
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@@ -29,6 +29,7 @@
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* add sane pulse detection
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***********************************/
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@ -436,50 +440,62 @@ avcodec/aacenc: Check all coefficients for finiteness
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ff_mdct_end(&s->mdct1024);
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ff_mdct_end(&s->mdct128);
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ff_psy_end(&s->psy);
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@@ -796,76 +900,109 @@ static av_cold int aac_encode_init(AVCodecContext *avc
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@@ -796,76 +900,123 @@ static av_cold int aac_encode_init(AVCodecContext *avc
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uint8_t grouping[AAC_MAX_CHANNELS];
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int lengths[2];
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+ s->channels = avctx->channels;
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+ s->chan_map = aac_chan_configs[s->channels-1];
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+ s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
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+ /* Constants */
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+ s->last_frame_pb_count = 0;
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+ avctx->extradata_size = 5;
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avctx->frame_size = 1024;
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+ avctx->initial_padding = 1024;
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+ avctx->bit_rate = (int)FFMIN(
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+ 6144 * s->channels / 1024.0 * avctx->sample_rate,
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+ avctx->bit_rate);
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+ avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
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+ avctx->profile;
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+ s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
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+ /* Channel map and unspecified bitrate guessing */
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+ s->channels = avctx->channels;
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+ ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
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+ "Unsupported number of channels: %d\n", s->channels);
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+ s->chan_map = aac_chan_configs[s->channels-1];
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+ if (!avctx->bit_rate) {
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+ for (i = 1; i <= s->chan_map[0]; i++) {
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+ avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
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+ s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
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+ 69000 ; /* SCE */
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+ }
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+ }
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+
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+ /* Samplerate */
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for (i = 0; i < 16; i++)
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if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
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break;
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+ s->samplerate_index = i;
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-
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- s->channels = avctx->channels;
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-
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- ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
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+ s->samplerate_index = i;
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+ ERROR_IF(s->samplerate_index == 16 ||
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+ s->samplerate_index >= ff_aac_swb_size_1024_len ||
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+ s->samplerate_index >= ff_aac_swb_size_128_len,
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"Unsupported sample rate %d\n", avctx->sample_rate);
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- ERROR_IF(s->channels > AAC_MAX_CHANNELS,
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+ ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
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"Unsupported number of channels: %d\n", s->channels);
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- "Unsupported number of channels: %d\n", s->channels);
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+
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+ /* Bitrate limiting */
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WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
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- "Too many bits per frame requested, clamping to max\n");
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- if (avctx->profile == FF_PROFILE_AAC_MAIN) {
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+ "Too many bits %f > %d per frame requested, clamping to max\n",
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+ 1024.0 * avctx->bit_rate / avctx->sample_rate,
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+ 6144 * s->channels);
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+
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+ avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
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+ avctx->bit_rate);
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+
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+ /* Profile and option setting */
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+ avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
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+ avctx->profile;
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+ for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
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+ if (avctx->profile == aacenc_profiles[i])
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+ break;
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+ ERROR_IF(i == FF_ARRAY_ELEMS(aacenc_profiles),
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+ "Unsupported encoding profile: %d\n", avctx->profile);
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+ if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
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+ avctx->profile = FF_PROFILE_AAC_LOW;
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+ ERROR_IF(s->options.pred,
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@ -520,21 +536,25 @@ avcodec/aacenc: Check all coefficients for finiteness
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+ "LTP prediction unavailable in the \"aac_main\" profile\n");
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}
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+ s->profile = avctx->profile;
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+ s->coder = &ff_aac_coders[s->options.coder];
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- if (s->options.aac_coder != AAC_CODER_TWOLOOP) {
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+ /* Coder limitations */
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+ s->coder = &ff_aac_coders[s->options.coder];
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+ if (s->options.coder != AAC_CODER_TWOLOOP) {
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+ ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
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+ "Coders other than twoloop require -strict -2 and some may be removed in the future\n");
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s->options.intensity_stereo = 0;
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s->options.pns = 0;
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}
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+ ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
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+ "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
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- avctx->bit_rate = (int)FFMIN(
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- 6144 * s->channels / 1024.0 * avctx->sample_rate,
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- avctx->bit_rate);
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+ ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
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+ "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
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+ /* M/S introduces horrible artifacts with multichannel files, this is temporary */
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+ if (s->channels > 3)
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+ s->options.mid_side = 0;
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- s->samplerate_index = i;
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-
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@ -578,7 +598,7 @@ avcodec/aacenc: Check all coefficients for finiteness
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ff_af_queue_init(avctx, &s->afq);
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return 0;
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@@ -876,27 +1013,16 @@ fail:
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@@ -876,27 +1027,16 @@ fail:
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#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
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static const AVOption aacenc_options[] = {
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@ -616,9 +636,23 @@ avcodec/aacenc: Check all coefficients for finiteness
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{NULL}
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};
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@@ -917,8 +1043,7 @@ AVCodec ff_aac_encoder = {
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@@ -907,6 +1047,11 @@ static const AVClass aacenc_class = {
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LIBAVUTIL_VERSION_INT,
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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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+
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AVCodec ff_aac_encoder = {
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.name = "aac",
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.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
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@@ -916,9 +1061,9 @@ AVCodec ff_aac_encoder = {
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.init = aac_encode_init,
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.encode2 = aac_encode_frame,
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.close = aac_encode_end,
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+ .defaults = aac_encode_defaults,
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.supported_samplerates = mpeg4audio_sample_rates,
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- .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
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- AV_CODEC_CAP_EXPERIMENTAL,
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