Merge commit '5f3a081b42b84404a40a52c80ef7a354cf048c56'

[ffmpeg] / libavcodec / aacenc.c

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 796038c0dab0679f03bb7e67e5ef652cf47aaec4..2156fc03e1ec7878ef107e8d6656228b3a0ebf8e 100644 (file)
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -29,6 +29,7 @@
  * add sane pulse detection
  ***********************************/
 
+#include "libavutil/thread.h"
 #include "libavutil/float_dsp.h"
 #include "libavutil/opt.h"
 #include "avcodec.h"
@@ -46,6 +47,8 @@
 
 #include "psymodel.h"
 
+static AVOnce aac_table_init = AV_ONCE_INIT;
+
 /**
  * Make AAC audio config object.
  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@@ -292,7 +295,13 @@ static void apply_mid_side_stereo(ChannelElement *cpe)
         for (w2 =  0; w2 < ics->group_len[w]; w2++) {
             int start = (w+w2) * 128;
             for (g = 0; g < ics->num_swb; g++) {
-                if (!cpe->ms_mask[w*16 + g] && !cpe->is_mask[w*16 + g]) {
+                /* ms_mask can be used for other purposes in PNS and I/S,
+                 * so must not apply M/S if any band uses either, even if
+                 * ms_mask is set.
+                 */
+                if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
+                    || cpe->ch[0].band_type[w*16 + g] == NOISE_BT
+                    || cpe->ch[1].band_type[w*16 + g] == NOISE_BT) {
                     start += ics->swb_sizes[g];
                     continue;
                 }
@@ -669,14 +678,14 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
                 sce = &cpe->ch[ch];
                 s->cur_channel = start_ch + ch;
-                if (s->options.pns && s->coder->search_for_pns)
-                    s->coder->search_for_pns(s, avctx, sce);
                 if (s->options.tns && s->coder->search_for_tns)
                     s->coder->search_for_tns(s, sce);
                 if (s->options.tns && s->coder->apply_tns_filt)
                     s->coder->apply_tns_filt(s, sce);
                 if (sce->tns.present)
                     tns_mode = 1;
+                if (s->options.pns && s->coder->search_for_pns)
+                    s->coder->search_for_pns(s, avctx, sce);
             }
             s->cur_channel = start_ch;
             if (s->options.intensity_stereo) { /* Intensity Stereo */
@@ -784,7 +793,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             s->lambda = FFMIN(s->lambda * ratio, 65536.f);
 
             /* Keep iterating if we must reduce and lambda is in the sky */
-            if ((s->lambda < 300.f || ratio > 0.9f) && (s->lambda > 10.f || ratio < 1.1f)) {
+            if (ratio > 0.9f && ratio < 1.1f) {
                 break;
             } else {
                 if (is_mode || ms_mode || tns_mode || pred_mode) {
@@ -879,6 +888,11 @@ alloc_fail:
     return AVERROR(ENOMEM);
 }
 
+static av_cold void aac_encode_init_tables(void)
+{
+    ff_aac_tableinit();
+}
+
 static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
@@ -946,13 +960,17 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
         avctx->profile = FF_PROFILE_AAC_MAIN;
         WARN_IF(1,
                 "Chainging profile to \"aac_main\"\n");
-        ERROR_IF(s->options.pred,
+        ERROR_IF(s->options.ltp,
                  "LTP prediction unavailable in the \"aac_main\" profile\n");
     }
     s->profile = avctx->profile;
     s->coder = &ff_aac_coders[s->options.coder];
 
     if (s->options.coder != AAC_CODER_TWOLOOP) {
+        ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
+                 "Coders other than twoloop require -strict -2 and some may be removed in the future\n");
+        WARN_IF(s->options.coder == AAC_CODER_FAAC,
+                "The FAAC-like coder will be removed in the near future, please use twoloop!\n");
         s->options.intensity_stereo = 0;
         s->options.pns = 0;
     }
@@ -977,10 +995,11 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     s->psypp = ff_psy_preprocess_init(avctx);
     ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
 
-    if (HAVE_MIPSDSPR1)
+    if (HAVE_MIPSDSP)
         ff_aac_coder_init_mips(s);
 
-    ff_aac_tableinit();
+    if ((ret = ff_thread_once(&aac_table_init, &aac_encode_init_tables)) != 0)
+        return AVERROR_UNKNOWN;
 
     ff_af_queue_init(avctx, &s->afq);
 
@@ -1023,8 +1042,8 @@ AVCodec ff_aac_encoder = {
     .encode2        = aac_encode_frame,
     .close          = aac_encode_end,
     .supported_samplerates = mpeg4audio_sample_rates,
-    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
-                      AV_CODEC_CAP_EXPERIMENTAL,
+    .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE,
+    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
     .priv_class     = &aacenc_class,