hevc: change the stride of the MC buffer to be in bytes instead of elements

[ffmpeg] / libavcodec / aacenc.c

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 6f582ca8aa2518ae85ea532dda46d7f3bcd8cf6f..00261c095e07e5a0f328caec24e883dc1be50e76 100644 (file)
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -34,7 +34,6 @@
 #include "libavutil/opt.h"
 #include "avcodec.h"
 #include "put_bits.h"
-#include "dsputil.h"
 #include "internal.h"
 #include "mpeg4audio.h"
 #include "kbdwin.h"
@@ -183,7 +182,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
 }
 
 #define WINDOW_FUNC(type) \
-static void apply_ ##type ##_window(DSPContext *dsp, AVFloatDSPContext *fdsp, \
+static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
                                     SingleChannelElement *sce, \
                                     const float *audio)
 
@@ -193,8 +192,8 @@ WINDOW_FUNC(only_long)
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     float *out = sce->ret_buf;
 
-    fdsp->vector_fmul       (out,        audio,        lwindow, 1024);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
+    fdsp->vector_fmul        (out,        audio,        lwindow, 1024);
+    fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
 }
 
 WINDOW_FUNC(long_start)
@@ -205,7 +204,7 @@ WINDOW_FUNC(long_start)
 
     fdsp->vector_fmul(out, audio, lwindow, 1024);
     memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
+    fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
     memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
 }
 
@@ -218,7 +217,7 @@ WINDOW_FUNC(long_stop)
     memset(out, 0, sizeof(out[0]) * 448);
     fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
     memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
-    dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
+    fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
 }
 
 WINDOW_FUNC(eight_short)
@@ -230,15 +229,15 @@ WINDOW_FUNC(eight_short)
     int w;
 
     for (w = 0; w < 8; w++) {
-        fdsp->vector_fmul       (out, in, w ? pwindow : swindow, 128);
+        fdsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
         out += 128;
         in  += 128;
-        dsp->vector_fmul_reverse(out, in, swindow, 128);
+        fdsp->vector_fmul_reverse(out, in, swindow, 128);
         out += 128;
     }
 }
 
-static void (*const apply_window[4])(DSPContext *dsp, AVFloatDSPContext *fdsp,
+static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
                                      SingleChannelElement *sce,
                                      const float *audio) = {
     [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
@@ -253,7 +252,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
     int i;
     float *output = sce->ret_buf;
 
-    apply_window[sce->ics.window_sequence[0]](&s->dsp, &s->fdsp, sce, audio);
+    apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
 
     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
@@ -582,7 +581,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
         init_put_bits(&s->pb, avpkt->data, avpkt->size);
 
-        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
+        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
             put_bitstream_info(s, LIBAVCODEC_IDENT);
         start_ch = 0;
         memset(chan_el_counter, 0, sizeof(chan_el_counter));
@@ -598,7 +597,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                 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 * 2 + ch;
+                s->cur_channel = start_ch + ch;
                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
             }
             cpe->common_window = 0;
@@ -614,7 +613,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                     }
                 }
             }
-            s->cur_channel = start_ch * 2;
+            s->cur_channel = start_ch;
             if (s->options.stereo_mode && cpe->common_window) {
                 if (s->options.stereo_mode > 0) {
                     IndividualChannelStream *ics = &cpe->ch[0].ics;
@@ -655,7 +654,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
     avctx->frame_bits = put_bits_count(&s->pb);
 
     // rate control stuff
-    if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
+    if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
         float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
         s->lambda *= ratio;
         s->lambda = FFMIN(s->lambda, 65536.f);
@@ -684,9 +683,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
     av_freep(&s->buffer.samples);
     av_freep(&s->cpe);
     ff_af_queue_close(&s->afq);
-#if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avctx->coded_frame);
-#endif
     return 0;
 }
 
@@ -694,8 +690,7 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
     int ret = 0;
 
-    ff_dsputil_init(&s->dsp, avctx);
-    avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+    avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
 
     // window init
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
@@ -716,16 +711,11 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
     int ch;
     FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
     FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
 
     for(ch = 0; ch < s->channels; ch++)
         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
 
-#if FF_API_OLD_ENCODE_AUDIO
-    if (!(avctx->coded_frame = avcodec_alloc_frame()))
-        goto alloc_fail;
-#endif
-
     return 0;
 alloc_fail:
     return AVERROR(ENOMEM);
@@ -754,16 +744,18 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
              "Unsupported profile %d\n", avctx->profile);
     ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
-             "Too many bits per frame requested\n");
+             "Too many bits %f > %d per frame requested\n",
+             1024.0 * avctx->bit_rate / avctx->sample_rate,
+             6144 * s->channels);
 
     s->samplerate_index = i;
 
     s->chan_map = aac_chan_configs[s->channels-1];
 
-    if (ret = dsp_init(avctx, s))
+    if ((ret = dsp_init(avctx, s)) < 0)
         goto fail;
 
-    if (ret = alloc_buffers(avctx, s))
+    if ((ret = alloc_buffers(avctx, s)) < 0)
         goto fail;
 
     avctx->extradata_size = 5;
@@ -775,7 +767,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     lengths[1] = ff_aac_num_swb_128[i];
     for (i = 0; i < s->chan_map[0]; i++)
         grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
-    if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
+    if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
+                           s->chan_map[0], grouping)) < 0)
         goto fail;
     s->psypp = ff_psy_preprocess_init(avctx);
     s->coder = &ff_aac_coders[2];
@@ -787,7 +780,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     for (i = 0; i < 428; i++)
         ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
 
-    avctx->delay = 1024;
+    avctx->initial_padding = 1024;
     ff_af_queue_init(avctx, &s->afq);
 
     return 0;
@@ -814,16 +807,16 @@ static const AVClass aacenc_class = {
 
 AVCodec ff_aac_encoder = {
     .name           = "aac",
+    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_AAC,
     .priv_data_size = sizeof(AACEncContext),
     .init           = aac_encode_init,
     .encode2        = aac_encode_frame,
     .close          = aac_encode_end,
-    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
-                      CODEC_CAP_EXPERIMENTAL,
+    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
+                      AV_CODEC_CAP_EXPERIMENTAL,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
-    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
     .priv_class     = &aacenc_class,
 };