Merge remote-tracking branch 'qatar/master'

* qatar/master:
  doc/APIchanges: add an entry for codec descriptors.
  vorbisenc: set AVCodecContext.bit_rate to 0
  vorbisenc: fix quality parameter
  FATE: add ALAC encoding tests
  lpc: fix alignment of windowed samples for odd maximum LPC order
  alacenc: use s16p sample format as input
  alacenc: remove unneeded sample_fmt check
  alacenc: fix max_frame_size calculation for the final frame
  adpcm_swf: Use correct sample offsets when using trellis.
  rtmp: support strict rtmp servers
  mjpegdec: support AVRn interlaced
  x86: remove FASTDIV inline asm

Conflicts:
	doc/APIchanges
	libavcodec/mjpegdec.c
	libavcodec/vorbisenc.c
	libavutil/x86/intmath.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>

6 files changed, 35 insertions, 35 deletions

diff --git a/libavcodec/adpcmenc.c b/libavcodec/adpcmenc.c
index 3af81cbc21..a3194abb3c 100644
--- a/libavcodec/adpcmenc.c
+++ b/libavcodec/adpcmenc.c
@@ -616,10 +616,11 @@ static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
         if (avctx->trellis > 0) {
             FF_ALLOC_OR_GOTO(avctx, buf, 2 * n, error);
-            adpcm_compress_trellis(avctx, samples + 2, buf, &c->status[0], n);
+            adpcm_compress_trellis(avctx, samples + avctx->channels, buf,
+                                   &c->status[0], n);
             if (avctx->channels == 2)
-                adpcm_compress_trellis(avctx, samples + 3, buf + n,
-                                       &c->status[1], n);
+                adpcm_compress_trellis(avctx, samples + avctx->channels + 1,
+                                       buf + n, &c->status[1], n);
             for (i = 0; i < n; i++) {
                 put_bits(&pb, 4, buf[i]);
                 if (avctx->channels == 2)
diff --git a/libavcodec/alacenc.c b/libavcodec/alacenc.c
index 8b2b513cd7..79b6ba8dd8 100644
--- a/libavcodec/alacenc.c
+++ b/libavcodec/alacenc.c
@@ -78,17 +78,15 @@ typedef struct AlacEncodeContext {
 } AlacEncodeContext;
 
 
-static void init_sample_buffers(AlacEncodeContext *s,
-                                const int16_t *input_samples)
+static void init_sample_buffers(AlacEncodeContext *s, int16_t **input_samples)
 {
     int ch, i;
 
     for (ch = 0; ch < s->avctx->channels; ch++) {
-        const int16_t *sptr = input_samples + ch;
-        for (i = 0; i < s->frame_size; i++) {
-            s->sample_buf[ch][i] = *sptr;
-            sptr += s->avctx->channels;
-        }
+        int32_t       *bptr = s->sample_buf[ch];
+        const int16_t *sptr = input_samples[ch];
+        for (i = 0; i < s->frame_size; i++)
+            bptr[i] = sptr[i];
     }
 }
 
@@ -347,8 +345,7 @@ static void alac_entropy_coder(AlacEncodeContext *s)
     }
 }
 
-static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
-                       const int16_t *samples)
+static int write_frame(AlacEncodeContext *s, AVPacket *avpkt, int16_t **samples)
 {
     int i, j;
     int prediction_type = 0;
@@ -358,8 +355,10 @@ static int write_frame(AlacEncodeContext *s, AVPacket *avpkt,
 
     if (s->verbatim) {
         write_frame_header(s);
-        for (i = 0; i < s->frame_size * s->avctx->channels; i++)
-            put_sbits(pb, 16, *samples++);
+        /* samples are channel-interleaved in verbatim mode */
+        for (i = 0; i < s->frame_size; i++)
+            for (j = 0; j < s->avctx->channels; j++)
+                put_sbits(pb, 16, samples[j][i]);
     } else {
         init_sample_buffers(s, samples);
         write_frame_header(s);
@@ -426,11 +425,6 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
 
     avctx->frame_size = s->frame_size = DEFAULT_FRAME_SIZE;
 
-    if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
-        av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
-        return -1;
-    }
-
     /* TODO: Correctly implement multi-channel ALAC.
              It is similar to multi-channel AAC, in that it has a series of
              single-channel (SCE), channel-pair (CPE), and LFE elements. */
@@ -542,11 +536,11 @@ static int alac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 {
     AlacEncodeContext *s = avctx->priv_data;
     int out_bytes, max_frame_size, ret;
-    const int16_t *samples = (const int16_t *)frame->data[0];
+    int16_t **samples = (int16_t **)frame->extended_data;
 
     s->frame_size = frame->nb_samples;
 
-    if (avctx->frame_size < DEFAULT_FRAME_SIZE)
+    if (frame->nb_samples < DEFAULT_FRAME_SIZE)
         max_frame_size = get_max_frame_size(s->frame_size, avctx->channels,
                                             DEFAULT_SAMPLE_SIZE);
     else
@@ -580,7 +574,7 @@ AVCodec ff_alac_encoder = {
     .encode2        = alac_encode_frame,
     .close          = alac_encode_close,
     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P,
                                                      AV_SAMPLE_FMT_NONE },
     .long_name      = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
 };
diff --git a/libavcodec/lpc.c b/libavcodec/lpc.c
index 3e5e3d63ad..ac8984a583 100644
--- a/libavcodec/lpc.c
+++ b/libavcodec/lpc.c
@@ -179,11 +179,9 @@ int ff_lpc_calc_coefs(LPCContext *s,
     }
 
     if (lpc_type == FF_LPC_TYPE_LEVINSON) {
-        double *windowed_samples = s->windowed_samples + max_order;
+        s->lpc_apply_welch_window(samples, blocksize, s->windowed_samples);
 
-        s->lpc_apply_welch_window(samples, blocksize, windowed_samples);
-
-        s->lpc_compute_autocorr(windowed_samples, blocksize, max_order, autoc);
+        s->lpc_compute_autocorr(s->windowed_samples, blocksize, max_order, autoc);
 
         compute_lpc_coefs(autoc, max_order, &lpc[0][0], MAX_LPC_ORDER, 0, 1);
 
@@ -252,10 +250,11 @@ av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order,
     s->lpc_type  = lpc_type;
 
     if (lpc_type == FF_LPC_TYPE_LEVINSON) {
-        s->windowed_samples = av_mallocz((blocksize + max_order + 2) *
-                                         sizeof(*s->windowed_samples));
-        if (!s->windowed_samples)
+        s->windowed_buffer = av_mallocz((blocksize + 2 + FFALIGN(max_order, 4)) *
+                                        sizeof(*s->windowed_samples));
+        if (!s->windowed_buffer)
             return AVERROR(ENOMEM);
+        s->windowed_samples = s->windowed_buffer + FFALIGN(max_order, 4);
     } else {
         s->windowed_samples = NULL;
     }
@@ -271,5 +270,5 @@ av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order,
 
 av_cold void ff_lpc_end(LPCContext *s)
 {
-    av_freep(&s->windowed_samples);
+    av_freep(&s->windowed_buffer);
 }
diff --git a/libavcodec/lpc.h b/libavcodec/lpc.h
index c58e37c4bc..680e102713 100644
--- a/libavcodec/lpc.h
+++ b/libavcodec/lpc.h
@@ -51,6 +51,7 @@ typedef struct LPCContext {
     int blocksize;
     int max_order;
     enum FFLPCType lpc_type;
+    double *windowed_buffer;
     double *windowed_samples;
 
     /**
diff --git a/libavcodec/mjpegdec.c b/libavcodec/mjpegdec.c
index 5fbda48d96..29d47b6fef 100644
--- a/libavcodec/mjpegdec.c
+++ b/libavcodec/mjpegdec.c
@@ -1235,6 +1235,7 @@ int ff_mjpeg_decode_sos(MJpegDecodeContext *s, const uint8_t *mb_bitmask,
     /* mjpeg-b can have padding bytes between sos and image data, skip them */
     for (i = s->mjpb_skiptosod; i > 0; i--)
         skip_bits(&s->gb, 8);
+
 next_field:
     for (i = 0; i < nb_components; i++)
         s->last_dc[i] = 1024;
@@ -1271,11 +1272,14 @@ next_field:
                 return ret;
         }
     }
-    if(s->interlaced && get_bits_left(&s->gb) > 32 && show_bits(&s->gb, 8) == 0xFF) {
-        GetBitContext bak= s->gb;
+
+    if (s->interlaced &&
+        get_bits_left(&s->gb) > 32 &&
+        show_bits(&s->gb, 8) == 0xFF) {
+        GetBitContext bak = s->gb;
         align_get_bits(&bak);
-        if(show_bits(&bak, 16) == 0xFFD1) {
-            av_log(s->avctx, AV_LOG_DEBUG, "AVRn ingterlaced picture\n");
+        if (show_bits(&bak, 16) == 0xFFD1) {
+            av_log(s->avctx, AV_LOG_DEBUG, "AVRn interlaced picture marker found\n");
             s->gb = bak;
             skip_bits(&s->gb, 16);
             s->bottom_field ^= 1;
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index b53fd371f2..b88791f274 100644
--- a/libavcodec/vorbisenc.c
+++ b/libavcodec/vorbisenc.c
@@ -1177,8 +1177,9 @@ static av_cold int vorbis_encode_init(AVCodecContext *avccontext)
     if ((ret = create_vorbis_context(venc, avccontext)) < 0)
         goto error;
 
+    avccontext->bit_rate = 0;
     if (avccontext->flags & CODEC_FLAG_QSCALE)
-        venc->quality = avccontext->global_quality / (float)FF_QP2LAMBDA / 10.;
+        venc->quality = avccontext->global_quality / (float)FF_QP2LAMBDA;
     else
         venc->quality = 8;
     venc->quality *= venc->quality;