Merge remote-tracking branch 'qatar/master'

* qatar/master: (22 commits)
  H.264: fix filter_mb_fast with 4:4:4 + 8x8dct
  alsa: limit buffer_size to 32768 frames.
  alsa: fallback to buffer_size/4 for period_size.
  doc: replace @pxref by @ref where appropriate
  mpeg1video: don't abort if thread_count is too high.
  segafilm: add support for videos with cri adx adpcm
  gxf: Fix 25 fps DV material in GXF being misdetected as 50 fps
  libxvid: Add const qualifier to silence compiler warning.
  H.264: improve qp_thresh check
  H.264: use fill_rectangle in CABAC decoding
  H.264: Remove redundant hl_motion_16/8 code
  H.264: merge fill_rectangle into P-SKIP MV prediction, to match B-SKIP
  H.264: faster P-SKIP decoding
  H.264: av_always_inline some more functions
  H.264: Add x86 assembly for 10-bit H.264 predict functions
  swscale: rename uv_off/uv_off2 to uv_off_px/byte.
  swscale: implement error dithering in planarCopyWrapper.
  swscale: error dithering for 16/9/10-bit to 8-bit.
  swscale: fix overflow in 16-bit vertical scaling.
  swscale: fix crash in 8-bpc bilinear output without alpha.
  ...

Conflicts:
	doc/developer.texi
	libavdevice/alsa-audio.h
	libavformat/gxf.c
	libswscale/swscale.c
	libswscale/swscale_internal.h
	libswscale/swscale_unscaled.c
	libswscale/x86/swscale_template.c
	tests/ref/lavfi/pixdesc
	tests/ref/lavfi/pixfmts_copy
	tests/ref/lavfi/pixfmts_crop
	tests/ref/lavfi/pixfmts_hflip
	tests/ref/lavfi/pixfmts_null
	tests/ref/lavfi/pixfmts_scale
	tests/ref/lavfi/pixfmts_vflip

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

1 files changed, 21 insertions, 7 deletions

diff --git a/libavformat/segafilm.c b/libavformat/segafilm.c
index ae194d496b..2251a893dd 100644
--- a/libavformat/segafilm.c
+++ b/libavformat/segafilm.c
@@ -111,7 +111,9 @@ static int film_read_header(AVFormatContext *s,
         film->audio_samplerate = AV_RB16(&scratch[24]);
         film->audio_channels = scratch[21];
         film->audio_bits = scratch[22];
-        if (film->audio_bits == 8)
+        if (scratch[23] == 2)
+            film->audio_type = CODEC_ID_ADPCM_ADX;
+        else if (film->audio_bits == 8)
             film->audio_type = CODEC_ID_PCM_S8;
         else if (film->audio_bits == 16)
             film->audio_type = CODEC_ID_PCM_S16BE;
@@ -149,12 +151,19 @@ static int film_read_header(AVFormatContext *s,
         st->codec->codec_id = film->audio_type;
         st->codec->codec_tag = 1;
         st->codec->channels = film->audio_channels;
-        st->codec->bits_per_coded_sample = film->audio_bits;
         st->codec->sample_rate = film->audio_samplerate;
+
+        if (film->audio_type == CODEC_ID_ADPCM_ADX) {
+            st->codec->bits_per_coded_sample = 18 * 8 / 32;
+            st->codec->block_align = st->codec->channels * 18;
+        } else {
+            st->codec->bits_per_coded_sample = film->audio_bits;
+            st->codec->block_align = st->codec->channels *
+                st->codec->bits_per_coded_sample / 8;
+        }
+
         st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
             st->codec->bits_per_coded_sample;
-        st->codec->block_align = st->codec->channels *
-            st->codec->bits_per_coded_sample / 8;
     }
 
     /* load the sample table */
@@ -187,8 +196,12 @@ static int film_read_header(AVFormatContext *s,
             film->sample_table[i].pts *= film->base_clock;
             film->sample_table[i].pts /= film->audio_samplerate;
 
-            audio_frame_counter += (film->sample_table[i].sample_size /
-                (film->audio_channels * film->audio_bits / 8));
+            if (film->audio_type == CODEC_ID_ADPCM_ADX)
+                audio_frame_counter += (film->sample_table[i].sample_size * 32 /
+                    (18 * film->audio_channels));
+            else
+                audio_frame_counter += (film->sample_table[i].sample_size /
+                    (film->audio_channels * film->audio_bits / 8));
         } else {
             film->sample_table[i].stream = film->video_stream_index;
             film->sample_table[i].pts = AV_RB32(&scratch[8]) & 0x7FFFFFFF;
@@ -227,7 +240,8 @@ static int film_read_packet(AVFormatContext *s,
             return AVERROR(ENOMEM);
         avio_read(pb, pkt->data, sample->sample_size);
     } else if ((sample->stream == film->audio_stream_index) &&
-        (film->audio_channels == 2)) {
+        (film->audio_channels == 2) &&
+        (film->audio_type != CODEC_ID_ADPCM_ADX)) {
         /* stereo PCM needs to be interleaved */
 
         if (av_new_packet(pkt, sample->sample_size))