]> git.sesse.net Git - ffmpeg/blobdiff - libavcodec/aacenc.c
doc/ffmpeg: s/avserver/ffserver/
[ffmpeg] / libavcodec / aacenc.c
index 0de6622389ef90eb7fbad67f32a584ced63fec70..e60a778c6d1dce45c6305274cd79717c9e75e2e6 100644 (file)
@@ -174,6 +174,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     AACEncContext *s = avctx->priv_data;
     int i;
     const uint8_t *sizes[2];
+    uint8_t grouping[AAC_MAX_CHANNELS];
     int lengths[2];
 
     avctx->frame_size = 1024;
@@ -208,8 +209,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     ff_init_ff_sine_windows(10);
     ff_init_ff_sine_windows(7);
 
+    s->chan_map           = aac_chan_configs[avctx->channels-1];
     s->samples            = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
-    s->cpe                = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
+    s->cpe                = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]);
     avctx->extradata      = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE);
     avctx->extradata_size = 5;
     put_audio_specific_config(avctx);
@@ -218,7 +220,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     sizes[1]   = swb_size_128[i];
     lengths[0] = ff_aac_num_swb_1024[i];
     lengths[1] = ff_aac_num_swb_128[i];
-    ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
+    for (i = 0; i < s->chan_map[0]; i++)
+        grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
+    ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
     s->psypp = ff_psy_preprocess_init(avctx);
     s->coder = &ff_aac_coders[2];
 
@@ -372,7 +376,7 @@ static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, in
         if (msc == 0 || ics0->max_sfb == 0)
             cpe->ms_mode = 0;
         else
-            cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
+            cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
     }
 }
 
@@ -500,7 +504,6 @@ static int aac_encode_frame(AVCodecContext *avctx,
     int16_t *samples = s->samples, *samples2, *la;
     ChannelElement *cpe;
     int i, ch, w, g, chans, tag, start_ch;
-    const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
     int chan_el_counter[4];
     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
 
@@ -521,8 +524,8 @@ static int aac_encode_frame(AVCodecContext *avctx,
         } else {
             start_ch = 0;
             samples2 = s->samples + 1024 * avctx->channels;
-            for (i = 0; i < chan_map[0]; i++) {
-                tag = chan_map[i+1];
+            for (i = 0; i < s->chan_map[0]; i++) {
+                tag = s->chan_map[i+1];
                 chans = tag == TYPE_CPE ? 2 : 1;
                 ff_psy_preprocess(s->psypp,
                                   (uint16_t*)data + channel_maps[avctx->channels-1][start_ch],
@@ -538,9 +541,9 @@ static int aac_encode_frame(AVCodecContext *avctx,
     }
 
     start_ch = 0;
-    for (i = 0; i < chan_map[0]; i++) {
+    for (i = 0; i < s->chan_map[0]; i++) {
         FFPsyWindowInfo* wi = windows + start_ch;
-        tag      = chan_map[i+1];
+        tag      = s->chan_map[i+1];
         chans    = tag == TYPE_CPE ? 2 : 1;
         cpe      = &s->cpe[i];
         for (ch = 0; ch < chans; ch++) {
@@ -555,6 +558,12 @@ static int aac_encode_frame(AVCodecContext *avctx,
                 wi[ch].window_shape   = 0;
                 wi[ch].num_windows    = 1;
                 wi[ch].grouping[0]    = 1;
+
+                /* Only the lowest 12 coefficients are used in a LFE channel.
+                 * The expression below results in only the bottom 8 coefficients
+                 * being used for 11.025kHz to 16kHz sample rates.
+                 */
+                ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
             } else {
                 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
                                               ics->window_sequence[0]);
@@ -565,7 +574,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
             ics->use_kb_window[0]   = wi[ch].window_shape;
             ics->num_windows        = wi[ch].num_windows;
             ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
-            ics->num_swb            = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8];
+            ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
             for (w = 0; w < ics->num_windows; w++)
                 ics->group_len[w] = wi[ch].grouping[w];
 
@@ -580,16 +589,19 @@ static int aac_encode_frame(AVCodecContext *avctx,
             put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
         start_ch = 0;
         memset(chan_el_counter, 0, sizeof(chan_el_counter));
-        for (i = 0; i < chan_map[0]; i++) {
+        for (i = 0; i < s->chan_map[0]; i++) {
             FFPsyWindowInfo* wi = windows + start_ch;
-            tag      = chan_map[i+1];
+            const float *coeffs[2];
+            tag      = s->chan_map[i+1];
             chans    = tag == TYPE_CPE ? 2 : 1;
             cpe      = &s->cpe[i];
             put_bits(&s->pb, 3, tag);
             put_bits(&s->pb, 4, chan_el_counter[tag]++);
+            for (ch = 0; ch < chans; ch++)
+                coeffs[ch] = cpe->ch[ch].coeffs;
+            s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
             for (ch = 0; ch < chans; ch++) {
-                s->cur_channel = start_ch + ch;
-                s->psy.model->analyze(&s->psy, s->cur_channel, cpe->ch[ch].coeffs, &wi[ch]);
+                s->cur_channel = start_ch * 2 + ch;
                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
             }
             cpe->common_window = 0;
@@ -605,7 +617,7 @@ static int aac_encode_frame(AVCodecContext *avctx,
                     }
                 }
             }
-            s->cur_channel = start_ch;
+            s->cur_channel = start_ch * 2;
             if (s->options.stereo_mode && cpe->common_window) {
                 if (s->options.stereo_mode > 0) {
                     IndividualChannelStream *ics = &cpe->ch[0].ics;
@@ -689,13 +701,13 @@ static const AVClass aacenc_class = {
 };
 
 AVCodec ff_aac_encoder = {
-    "aac",
-    AVMEDIA_TYPE_AUDIO,
-    CODEC_ID_AAC,
-    sizeof(AACEncContext),
-    aac_encode_init,
-    aac_encode_frame,
-    aac_encode_end,
+    .name           = "aac",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_AAC,
+    .priv_data_size = sizeof(AACEncContext),
+    .init           = aac_encode_init,
+    .encode         = aac_encode_frame,
+    .close          = aac_encode_end,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
     .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),