Merge remote-tracking branch 'qatar/master'

* qatar/master:
  Remove ffmpeg.
  aacenc: Simplify windowing
  aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples.
  aacenc: Deinterleave input samples before processing.
  aacenc: Store channel count in AACEncContext.
  aacenc: Move Q^3/4 calculation to it's own table
  aacenc: Request normalized float samples instead of converting s16 samples to float.
  aacpsy: Replace an if with FFMAX in LAME windowing.
  aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated.
  aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons
  aacenc: cosmetics: move init() and end() to the bottom of the file.
  aacenc: aac_encode_init() cleanup
  XWD encoder and decoder
  vc1: don't read the interpfrm and bfraction elements for interlaced frames
  mxfdec: fix memleak on mxf_read_close()
  westwood: split the AUD and VQA demuxers into separate files.

Conflicts:
	.gitignore
	Changelog
	Makefile
	configure
	doc/ffmpeg.texi
	ffmpeg.c
	libavcodec/Makefile
	libavcodec/aacenc.c
	libavcodec/allcodecs.c
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/Makefile
	libavformat/img2.c

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

4 files changed, 179 insertions, 148 deletions

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 7c7cea5aad..e4e32afba2 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -326,8 +326,8 @@ OBJS-$(CONFIG_WC3_DEMUXER)               += wc3movie.o
 OBJS-$(CONFIG_WEBM_MUXER)                += matroskaenc.o matroska.o \
                                             riff.o isom.o avc.o \
                                             flacenc_header.o avlanguage.o
-OBJS-$(CONFIG_WSAUD_DEMUXER)             += westwood.o
-OBJS-$(CONFIG_WSVQA_DEMUXER)             += westwood.o
+OBJS-$(CONFIG_WSAUD_DEMUXER)             += westwood_aud.o
+OBJS-$(CONFIG_WSVQA_DEMUXER)             += westwood_vqa.o
 OBJS-$(CONFIG_WTV_DEMUXER)               += wtvdec.o wtv.o asfdec.o asf.o asfcrypt.o \
                                             avlanguage.o mpegts.o isom.o riff.o
 OBJS-$(CONFIG_WTV_MUXER)                 += wtvenc.o wtv.o asf.o asfenc.o riff.o
diff --git a/libavformat/img2.c b/libavformat/img2.c
index 775f5e7a05..8adba0ac5f 100644
--- a/libavformat/img2.c
+++ b/libavformat/img2.c
@@ -91,6 +91,7 @@ static const IdStrMap img_tags[] = {
     { CODEC_ID_JPEG2000  , "jpc"},
     { CODEC_ID_DPX       , "dpx"},
     { CODEC_ID_PICTOR    , "pic"},
+    { CODEC_ID_XWD       , "xwd"},
     { CODEC_ID_NONE      , NULL}
 };
 
@@ -528,7 +529,7 @@ AVOutputFormat ff_image2_muxer = {
     .name           = "image2",
     .long_name      = NULL_IF_CONFIG_SMALL("image2 sequence"),
     .extensions     = "bmp,dpx,jls,jpeg,jpg,ljpg,pam,pbm,pcx,pgm,pgmyuv,png,"
-                      "ppm,sgi,tga,tif,tiff,jp2,j2c",
+                      "ppm,sgi,tga,tif,tiff,jp2,j2c,xwd",
     .priv_data_size = sizeof(VideoData),
     .video_codec    = CODEC_ID_MJPEG,
     .write_header   = write_header,
diff --git a/libavformat/westwood_aud.c b/libavformat/westwood_aud.c
new file mode 100644
index 0000000000..b1eb768016
--- /dev/null
+++ b/libavformat/westwood_aud.c
@@ -0,0 +1,173 @@
+/*
+ * Westwood Studios AUD Format Demuxer
+ * Copyright (c) 2003 The ffmpeg Project
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Westwood Studios AUD file demuxer
+ * by Mike Melanson (melanson@pcisys.net)
+ * for more information on the Westwood file formats, visit:
+ *   http://www.pcisys.net/~melanson/codecs/
+ *   http://www.geocities.com/SiliconValley/8682/aud3.txt
+ *
+ * Implementation note: There is no definite file signature for AUD files.
+ * The demuxer uses a probabilistic strategy for content detection. This
+ * entails performing sanity checks on certain header values in order to
+ * qualify a file. Refer to wsaud_probe() for the precise parameters.
+ */
+
+#include "libavutil/intreadwrite.h"
+#include "avformat.h"
+#include "internal.h"
+
+#define AUD_HEADER_SIZE 12
+#define AUD_CHUNK_PREAMBLE_SIZE 8
+#define AUD_CHUNK_SIGNATURE 0x0000DEAF
+
+typedef struct WsAudDemuxContext {
+    int audio_samplerate;
+    int audio_channels;
+    int audio_bits;
+    enum CodecID audio_type;
+    int audio_stream_index;
+    int64_t audio_frame_counter;
+} WsAudDemuxContext;
+
+static int wsaud_probe(AVProbeData *p)
+{
+    int field;
+
+    /* Probabilistic content detection strategy: There is no file signature
+     * so perform sanity checks on various header parameters:
+     *   8000 <= sample rate (16 bits) <= 48000  ==> 40001 acceptable numbers
+     *   flags <= 0x03 (2 LSBs are used)         ==> 4 acceptable numbers
+     *   compression type (8 bits) = 1 or 99     ==> 2 acceptable numbers
+     *   first audio chunk signature (32 bits)   ==> 1 acceptable number
+     * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
+     * 320008 acceptable number combinations.
+     */
+
+    if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
+        return 0;
+
+    /* check sample rate */
+    field = AV_RL16(&p->buf[0]);
+    if ((field < 8000) || (field > 48000))
+        return 0;
+
+    /* enforce the rule that the top 6 bits of this flags field are reserved (0);
+     * this might not be true, but enforce it until deemed unnecessary */
+    if (p->buf[10] & 0xFC)
+        return 0;
+
+    /* note: only check for WS IMA (type 99) right now since there is no
+     * support for type 1 */
+    if (p->buf[11] != 99)
+        return 0;
+
+    /* read ahead to the first audio chunk and validate the first header signature */
+    if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
+        return 0;
+
+    /* return 1/2 certainty since this file check is a little sketchy */
+    return AVPROBE_SCORE_MAX / 2;
+}
+
+static int wsaud_read_header(AVFormatContext *s,
+                             AVFormatParameters *ap)
+{
+    WsAudDemuxContext *wsaud = s->priv_data;
+    AVIOContext *pb = s->pb;
+    AVStream *st;
+    unsigned char header[AUD_HEADER_SIZE];
+
+    if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
+        return AVERROR(EIO);
+    wsaud->audio_samplerate = AV_RL16(&header[0]);
+    if (header[11] == 99)
+        wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
+    else
+        return AVERROR_INVALIDDATA;
+
+    /* flag 0 indicates stereo */
+    wsaud->audio_channels = (header[10] & 0x1) + 1;
+    /* flag 1 indicates 16 bit audio */
+    wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
+
+    /* initialize the audio decoder stream */
+    st = avformat_new_stream(s, NULL);
+    if (!st)
+        return AVERROR(ENOMEM);
+    avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
+    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+    st->codec->codec_id = wsaud->audio_type;
+    st->codec->codec_tag = 0;  /* no tag */
+    st->codec->channels = wsaud->audio_channels;
+    st->codec->sample_rate = wsaud->audio_samplerate;
+    st->codec->bits_per_coded_sample = wsaud->audio_bits;
+    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
+        st->codec->bits_per_coded_sample / 4;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
+
+    wsaud->audio_stream_index = st->index;
+    wsaud->audio_frame_counter = 0;
+
+    return 0;
+}
+
+static int wsaud_read_packet(AVFormatContext *s,
+                             AVPacket *pkt)
+{
+    WsAudDemuxContext *wsaud = s->priv_data;
+    AVIOContext *pb = s->pb;
+    unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
+    unsigned int chunk_size;
+    int ret = 0;
+
+    if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
+        AUD_CHUNK_PREAMBLE_SIZE)
+        return AVERROR(EIO);
+
+    /* validate the chunk */
+    if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
+        return AVERROR_INVALIDDATA;
+
+    chunk_size = AV_RL16(&preamble[0]);
+    ret= av_get_packet(pb, pkt, chunk_size);
+    if (ret != chunk_size)
+        return AVERROR(EIO);
+    pkt->stream_index = wsaud->audio_stream_index;
+    pkt->pts = wsaud->audio_frame_counter;
+    pkt->pts /= wsaud->audio_samplerate;
+
+    /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
+    wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
+
+    return ret;
+}
+
+AVInputFormat ff_wsaud_demuxer = {
+    .name           = "wsaud",
+    .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
+    .priv_data_size = sizeof(WsAudDemuxContext),
+    .read_probe     = wsaud_probe,
+    .read_header    = wsaud_read_header,
+    .read_packet    = wsaud_read_packet,
+};
diff --git a/libavformat/westwood.c b/libavformat/westwood_vqa.c
index 4cbd5be396..4a851473b2 100644
--- a/libavformat/westwood.c
+++ b/libavformat/westwood_vqa.c
@@ -1,5 +1,5 @@
 /*
- * Westwood Studios Multimedia Formats Demuxer (VQA, AUD)
+ * Westwood Studios VQA Format Demuxer
  * Copyright (c) 2003 The ffmpeg Project
  *
  * This file is part of FFmpeg.
@@ -21,26 +21,17 @@
 
 /**
  * @file
- * Westwood Studios VQA & AUD file demuxers
+ * Westwood Studios VQA file demuxer
  * by Mike Melanson (melanson@pcisys.net)
  * for more information on the Westwood file formats, visit:
  *   http://www.pcisys.net/~melanson/codecs/
  *   http://www.geocities.com/SiliconValley/8682/aud3.txt
- *
- * Implementation note: There is no definite file signature for AUD files.
- * The demuxer uses a probabilistic strategy for content detection. This
- * entails performing sanity checks on certain header values in order to
- * qualify a file. Refer to wsaud_probe() for the precise parameters.
  */
 
 #include "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
 
-#define AUD_HEADER_SIZE 12
-#define AUD_CHUNK_PREAMBLE_SIZE 8
-#define AUD_CHUNK_SIGNATURE 0x0000DEAF
-
 #define FORM_TAG MKBETAG('F', 'O', 'R', 'M')
 #define WVQA_TAG MKBETAG('W', 'V', 'Q', 'A')
 #define VQHD_TAG MKBETAG('V', 'Q', 'H', 'D')
@@ -63,15 +54,6 @@
 #define VQA_FRAMERATE 15
 #define VQA_PREAMBLE_SIZE 8
 
-typedef struct WsAudDemuxContext {
-    int audio_samplerate;
-    int audio_channels;
-    int audio_bits;
-    enum CodecID audio_type;
-    int audio_stream_index;
-    int64_t audio_frame_counter;
-} WsAudDemuxContext;
-
 typedef struct WsVqaDemuxContext {
     int audio_samplerate;
     int audio_channels;
@@ -83,119 +65,6 @@ typedef struct WsVqaDemuxContext {
     int64_t audio_frame_counter;
 } WsVqaDemuxContext;
 
-static int wsaud_probe(AVProbeData *p)
-{
-    int field;
-
-    /* Probabilistic content detection strategy: There is no file signature
-     * so perform sanity checks on various header parameters:
-     *   8000 <= sample rate (16 bits) <= 48000  ==> 40001 acceptable numbers
-     *   flags <= 0x03 (2 LSBs are used)         ==> 4 acceptable numbers
-     *   compression type (8 bits) = 1 or 99     ==> 2 acceptable numbers
-     *   first audio chunk signature (32 bits)   ==> 1 acceptable number
-     * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
-     * 320008 acceptable number combinations.
-     */
-
-    if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
-        return 0;
-
-    /* check sample rate */
-    field = AV_RL16(&p->buf[0]);
-    if ((field < 8000) || (field > 48000))
-        return 0;
-
-    /* enforce the rule that the top 6 bits of this flags field are reserved (0);
-     * this might not be true, but enforce it until deemed unnecessary */
-    if (p->buf[10] & 0xFC)
-        return 0;
-
-    /* note: only check for WS IMA (type 99) right now since there is no
-     * support for type 1 */
-    if (p->buf[11] != 99)
-        return 0;
-
-    /* read ahead to the first audio chunk and validate the first header signature */
-    if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
-        return 0;
-
-    /* return 1/2 certainty since this file check is a little sketchy */
-    return AVPROBE_SCORE_MAX / 2;
-}
-
-static int wsaud_read_header(AVFormatContext *s,
-                             AVFormatParameters *ap)
-{
-    WsAudDemuxContext *wsaud = s->priv_data;
-    AVIOContext *pb = s->pb;
-    AVStream *st;
-    unsigned char header[AUD_HEADER_SIZE];
-
-    if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
-        return AVERROR(EIO);
-    wsaud->audio_samplerate = AV_RL16(&header[0]);
-    if (header[11] == 99)
-        wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
-    else
-        return AVERROR_INVALIDDATA;
-
-    /* flag 0 indicates stereo */
-    wsaud->audio_channels = (header[10] & 0x1) + 1;
-    /* flag 1 indicates 16 bit audio */
-    wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
-
-    /* initialize the audio decoder stream */
-    st = avformat_new_stream(s, NULL);
-    if (!st)
-        return AVERROR(ENOMEM);
-    avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
-    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-    st->codec->codec_id = wsaud->audio_type;
-    st->codec->codec_tag = 0;  /* no tag */
-    st->codec->channels = wsaud->audio_channels;
-    st->codec->sample_rate = wsaud->audio_samplerate;
-    st->codec->bits_per_coded_sample = wsaud->audio_bits;
-    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
-        st->codec->bits_per_coded_sample / 4;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
-
-    wsaud->audio_stream_index = st->index;
-    wsaud->audio_frame_counter = 0;
-
-    return 0;
-}
-
-static int wsaud_read_packet(AVFormatContext *s,
-                             AVPacket *pkt)
-{
-    WsAudDemuxContext *wsaud = s->priv_data;
-    AVIOContext *pb = s->pb;
-    unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
-    unsigned int chunk_size;
-    int ret = 0;
-
-    if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
-        AUD_CHUNK_PREAMBLE_SIZE)
-        return AVERROR(EIO);
-
-    /* validate the chunk */
-    if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
-        return AVERROR_INVALIDDATA;
-
-    chunk_size = AV_RL16(&preamble[0]);
-    ret= av_get_packet(pb, pkt, chunk_size);
-    if (ret != chunk_size)
-        return AVERROR(EIO);
-    pkt->stream_index = wsaud->audio_stream_index;
-    pkt->pts = wsaud->audio_frame_counter;
-    pkt->pts /= wsaud->audio_samplerate;
-
-    /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
-    wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
-
-    return ret;
-}
-
 static int wsvqa_probe(AVProbeData *p)
 {
     /* need 12 bytes to qualify */
@@ -367,17 +236,6 @@ static int wsvqa_read_packet(AVFormatContext *s,
     return ret;
 }
 
-#if CONFIG_WSAUD_DEMUXER
-AVInputFormat ff_wsaud_demuxer = {
-    .name           = "wsaud",
-    .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
-    .priv_data_size = sizeof(WsAudDemuxContext),
-    .read_probe     = wsaud_probe,
-    .read_header    = wsaud_read_header,
-    .read_packet    = wsaud_read_packet,
-};
-#endif
-#if CONFIG_WSVQA_DEMUXER
 AVInputFormat ff_wsvqa_demuxer = {
     .name           = "wsvqa",
     .long_name      = NULL_IF_CONFIG_SMALL("Westwood Studios VQA format"),
@@ -386,4 +244,3 @@ AVInputFormat ff_wsvqa_demuxer = {
     .read_header    = wsvqa_read_header,
     .read_packet    = wsvqa_read_packet,
 };
-#endif