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>

1 files changed, 13 insertions, 19 deletions

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)"),
 };