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>

doc/APIchanges

@@ -77,22 +77,31 @@ API changes, most recent first:
 2012-03-26 - a67d9cf - lavfi 2.66.100
   Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions.
 
-2012-08-08 - xxxxxxx - lavu 51.38 - dict.h
+2012-08-18 - lavc 54.26 - avcodec.h
+  Add codec descriptors for accessing codec properties without having
+  to refer to a specific decoder or encoder.
+
+  c223d79 - Add an AVCodecDescriptor struct and functions
+            avcodec_descriptor_get() and avcodec_descriptor_next().
+  51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY.
+  91e59fe - Add avcodec_descriptor_get_by_name().
+
+2012-08-08 - 987170c - lavu 51.38 - dict.h
   Add av_dict_count().
 
-2012-08-xx - xxxxxxx - lavc 54.25 - avcodec.h
+2012-08-07 - 104e10f - lavc 54.25 - avcodec.h
   Rename CodecID to AVCodecID and all CODEC_ID_* to AV_CODEC_ID_*.
   To provide backwards compatibility, CodecID is now #defined as AVCodecID.
   Note that this can break user code that includes avcodec.h and uses the
   'CodecID' identifier. Such code should either #undef CodecID or stop using the
   CodecID name.
 
-2012-08-03 - xxxxxxx - lavu 51.37.1 - cpu.h
+2012-08-03 - 239fdf1 - lavu 51.37.1 - cpu.h
                        lsws 2.1.1   - swscale.h
   Rename AV_CPU_FLAG_MMX2  ---> AV_CPU_FLAG_MMXEXT.
   Rename SWS_CPU_CAPS_MMX2 ---> SWS_CPU_CAPS_MMXEXT.
 
-2012-07-xx - xxxxxxx - lavf 54.13.0 - avformat.h
+2012-07-29 - 681ed00 - lavf 54.13.0 - avformat.h
   Add AVFMT_FLAG_NOBUFFER for low latency use cases.
 
 2012-07-10 - 5fade8a - lavu 51.37.0

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)

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)"),
 };

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);
 }

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;
 
     /**

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;

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;

libavformat/rtmpproto.c

@@ -549,7 +549,7 @@ static int gen_release_stream(URLContext *s, RTMPContext *rt)
     ff_amf_write_null(&p);
     ff_amf_write_string(&p, rt->playpath);
 
-    return rtmp_send_packet(rt, &pkt, 0);
+    return rtmp_send_packet(rt, &pkt, 1);
 }
 
 /**
@@ -573,7 +573,7 @@ static int gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
     ff_amf_write_null(&p);
     ff_amf_write_string(&p, rt->playpath);
 
-    return rtmp_send_packet(rt, &pkt, 0);
+    return rtmp_send_packet(rt, &pkt, 1);
 }
 
 /**
@@ -1525,8 +1525,11 @@ static int handle_invoke_error(URLContext *s, RTMPPacket *pkt)
 
     if (!ff_amf_get_field_value(pkt->data + 9, data_end,
                                 "description", tmpstr, sizeof(tmpstr))) {
-        if (tracked_method && !strcmp(tracked_method, "_checkbw")) {
-            /* Ignore _checkbw errors. */
+        if (tracked_method && (!strcmp(tracked_method, "_checkbw")      ||
+                               !strcmp(tracked_method, "releaseStream") ||
+                               !strcmp(tracked_method, "FCSubscribe")   ||
+                               !strcmp(tracked_method, "FCPublish"))) {
+            /* Gracefully ignore Adobe-specific historical artifact errors. */
             level = AV_LOG_WARNING;
             ret = 0;
         } else

tests/Makefile

@@ -38,6 +38,7 @@ include $(SRC_PATH)/tests/fate/vcodec.mak
 include $(SRC_PATH)/tests/fate/aac.mak
 include $(SRC_PATH)/tests/fate/ac3.mak
 include $(SRC_PATH)/tests/fate/adpcm.mak
+include $(SRC_PATH)/tests/fate/alac.mak
 include $(SRC_PATH)/tests/fate/als.mak
 include $(SRC_PATH)/tests/fate/amrnb.mak
 include $(SRC_PATH)/tests/fate/amrwb.mak

tests/fate/alac.mak

+FATE_ALAC += fate-alac-level-0                                          \
+             fate-alac-level-1                                          \
+             fate-alac-level-2                                          \
+             fate-alac-lpc-orders                                       \
+
+fate-alac-level-%: OPTS = -compression_level $(@:fate-alac-level-%=%)
+fate-alac-lpc-orders: OPTS = -min_prediction_order 1 -max_prediction_order 30
+
+fate-alac-%: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
+fate-alac-%: CMD = enc_dec_pcm mov wav s16le $(REF) -c alac $(OPTS)
+fate-alac-%: CMP = oneoff
+fate-alac-%: FUZZ = 0
+
+FATE_SAMPLES_AVCONV += $(FATE_ALAC)
+fate-alac: $(FATE_ALAC)