From d5c5a1c7e2959a5174522b5ac1d73d5095b8c5a3 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Martin=20Storsj=C3=B6?= <martin@martin.st>
Date: Fri, 22 Apr 2011 00:45:06 +0300
Subject: [PATCH 01/20] doc: Mention that DragonFly BSD requires __BSD_VISIBLE
 set
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Setting this should fix the FATE build failure on DragonFly BSD,
since inet_aton only is visible if __BSD_VISIBLE is set.

Alternatively, a line defining __BSD_VISIBLE=1 could be
added at the top of os_support.c. For FreeBSD, similar lines
are required in libavdevice/bktr.c and libavdevice/oss_audio.c, too.

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 doc/general.texi | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/doc/general.texi b/doc/general.texi
index 423402ca4e087..edd5e60cc33ff 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -784,9 +784,9 @@ to configure.
 BSD make will not build Libav, you need to install and use GNU Make
 (@file{gmake}).
 
-@subsubsection FreeBSD
+@subsubsection FreeBSD, DragonFly BSD
 
-FreeBSD will not compile out-of-the-box due to broken system headers.
+These systems will not compile out-of-the-box due to broken system headers.
 Passing @code{--extra-cflags=-D__BSD_VISIBLE} to configure will work
 around the problem. This may have unexpected sideeffects, so use it at
 your own risk. If you care about FreeBSD, please make an attempt at

From 6e4f70a8de69884ce0caa8735d7a88915b4391d4 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Martin=20Storsj=C3=B6?= <martin@martin.st>
Date: Sun, 23 Jan 2011 01:31:41 +0200
Subject: [PATCH 02/20] Add a protocol handler for AES CBC decryption with
 PKCS7 padding
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This can later be extended to support other AES bit sizes,
encryption, other crypto algorithms, reading the key from a URL, etc.

In order to use it, the key and initialization vector has to be
passed via AVOptions. Since such options can't be passed to
protocols from the command line, the protocol is currently
only for libavformat internal use.

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 libavformat/Makefile     |   1 +
 libavformat/allformats.c |   1 +
 libavformat/crypto.c     | 170 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 172 insertions(+)
 create mode 100644 libavformat/crypto.c

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 5e029ed96bc89..e2e398212553e 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -312,6 +312,7 @@ OBJS+= avio.o aviobuf.o
 
 OBJS-$(CONFIG_APPLEHTTP_PROTOCOL)        += applehttpproto.o
 OBJS-$(CONFIG_CONCAT_PROTOCOL)           += concat.o
+OBJS-$(CONFIG_CRYPTO_PROTOCOL)           += crypto.o
 OBJS-$(CONFIG_FILE_PROTOCOL)             += file.o
 OBJS-$(CONFIG_GOPHER_PROTOCOL)           += gopher.o
 OBJS-$(CONFIG_HTTP_PROTOCOL)             += http.o httpauth.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 931947d60a965..f1c3d3b768115 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -235,6 +235,7 @@ void av_register_all(void)
     /* protocols */
     REGISTER_PROTOCOL (APPLEHTTP, applehttp);
     REGISTER_PROTOCOL (CONCAT, concat);
+    REGISTER_PROTOCOL (CRYPTO, crypto);
     REGISTER_PROTOCOL (FILE, file);
     REGISTER_PROTOCOL (GOPHER, gopher);
     REGISTER_PROTOCOL (HTTP, http);
diff --git a/libavformat/crypto.c b/libavformat/crypto.c
new file mode 100644
index 0000000000000..ea6012ad09fc9
--- /dev/null
+++ b/libavformat/crypto.c
@@ -0,0 +1,170 @@
+/*
+ * Decryption protocol handler
+ * Copyright (c) 2011 Martin Storsjo
+ *
+ * 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
+ */
+
+#include "avformat.h"
+#include "libavutil/aes.h"
+#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "internal.h"
+#include "url.h"
+
+#define MAX_BUFFER_BLOCKS 150
+#define BLOCKSIZE 16
+
+typedef struct {
+    const AVClass *class;
+    URLContext *hd;
+    uint8_t inbuffer [BLOCKSIZE*MAX_BUFFER_BLOCKS],
+            outbuffer[BLOCKSIZE*MAX_BUFFER_BLOCKS];
+    uint8_t *outptr;
+    int indata, indata_used, outdata;
+    int eof;
+    uint8_t *key;
+    int keylen;
+    uint8_t *iv;
+    int ivlen;
+    struct AVAES *aes;
+} CryptoContext;
+
+#define OFFSET(x) offsetof(CryptoContext, x)
+static const AVOption options[] = {
+    {"key", "AES decryption key", OFFSET(key), FF_OPT_TYPE_BINARY },
+    {"iv",  "AES decryption initialization vector", OFFSET(iv),  FF_OPT_TYPE_BINARY },
+    { NULL }
+};
+
+static const AVClass crypto_class = {
+    "crypto", av_default_item_name, options, LIBAVUTIL_VERSION_INT
+};
+
+static int crypto_open(URLContext *h, const char *uri, int flags)
+{
+    const char *nested_url;
+    int ret;
+    CryptoContext *c = h->priv_data;
+
+    if (!av_strstart(uri, "crypto+", &nested_url) &&
+        !av_strstart(uri, "crypto:", &nested_url)) {
+        av_log(h, AV_LOG_ERROR, "Unsupported url %s\n", uri);
+        ret = AVERROR(EINVAL);
+        goto err;
+    }
+
+    if (c->keylen < BLOCKSIZE || c->ivlen < BLOCKSIZE) {
+        av_log(h, AV_LOG_ERROR, "Key or IV not set\n");
+        ret = AVERROR(EINVAL);
+        goto err;
+    }
+    if (flags & AVIO_FLAG_WRITE) {
+        av_log(h, AV_LOG_ERROR, "Only decryption is supported currently\n");
+        ret = AVERROR(ENOSYS);
+        goto err;
+    }
+    if ((ret = ffurl_open(&c->hd, nested_url, AVIO_FLAG_READ)) < 0) {
+        av_log(h, AV_LOG_ERROR, "Unable to open input\n");
+        goto err;
+    }
+    c->aes = av_mallocz(av_aes_size);
+    if (!c->aes) {
+        ret = AVERROR(ENOMEM);
+        goto err;
+    }
+
+    av_aes_init(c->aes, c->key, 128, 1);
+
+    h->is_streamed = 1;
+
+    return 0;
+err:
+    av_free(c->key);
+    av_free(c->iv);
+    return ret;
+}
+
+static int crypto_read(URLContext *h, uint8_t *buf, int size)
+{
+    CryptoContext *c = h->priv_data;
+    int blocks;
+retry:
+    if (c->outdata > 0) {
+        size = FFMIN(size, c->outdata);
+        memcpy(buf, c->outptr, size);
+        c->outptr  += size;
+        c->outdata -= size;
+        return size;
+    }
+    // We avoid using the last block until we've found EOF,
+    // since we'll remove PKCS7 padding at the end. So make
+    // sure we've got at least 2 blocks, so we can decrypt
+    // at least one.
+    while (c->indata - c->indata_used < 2*BLOCKSIZE) {
+        int n = ffurl_read(c->hd, c->inbuffer + c->indata,
+                           sizeof(c->inbuffer) - c->indata);
+        if (n <= 0) {
+            c->eof = 1;
+            break;
+        }
+        c->indata += n;
+    }
+    blocks = (c->indata - c->indata_used) / BLOCKSIZE;
+    if (!blocks)
+        return AVERROR_EOF;
+    if (!c->eof)
+        blocks--;
+    av_aes_crypt(c->aes, c->outbuffer, c->inbuffer + c->indata_used, blocks,
+                 c->iv, 1);
+    c->outdata      = BLOCKSIZE * blocks;
+    c->outptr       = c->outbuffer;
+    c->indata_used += BLOCKSIZE * blocks;
+    if (c->indata_used >= sizeof(c->inbuffer)/2) {
+        memmove(c->inbuffer, c->inbuffer + c->indata_used,
+                c->indata - c->indata_used);
+        c->indata     -= c->indata_used;
+        c->indata_used = 0;
+    }
+    if (c->eof) {
+        // Remove PKCS7 padding at the end
+        int padding = c->outbuffer[c->outdata - 1];
+        c->outdata -= padding;
+    }
+    goto retry;
+}
+
+static int crypto_close(URLContext *h)
+{
+    CryptoContext *c = h->priv_data;
+    if (c->hd)
+        ffurl_close(c->hd);
+    av_free(c->aes);
+    av_free(c->key);
+    av_free(c->iv);
+    return 0;
+}
+
+URLProtocol ff_crypto_protocol = {
+    .name            = "crypto",
+    .url_open        = crypto_open,
+    .url_read        = crypto_read,
+    .url_close       = crypto_close,
+    .priv_data_size  = sizeof(CryptoContext),
+    .priv_data_class = &crypto_class,
+    .flags           = URL_PROTOCOL_FLAG_NESTED_SCHEME,
+};

From 84465f2180308a3e998089517e76586563fd6162 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Martin=20Storsj=C3=B6?= <martin@martin.st>
Date: Sun, 23 Jan 2011 23:42:18 +0200
Subject: [PATCH 03/20] applehttp: Handle AES-128 encrypted streams
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This should hopefully fix roundup issue 2586.

This commit only implements it in the demuxer, not in the
protocol handler. If desired, some of the code could be
refactored to be shared by both implementations.

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 libavformat/applehttp.c | 108 ++++++++++++++++++++++++++++++++++++++--
 libavformat/version.h   |   2 +-
 2 files changed, 106 insertions(+), 4 deletions(-)

diff --git a/libavformat/applehttp.c b/libavformat/applehttp.c
index 90b86a8733d6d..2e0e8a1b3ad98 100644
--- a/libavformat/applehttp.c
+++ b/libavformat/applehttp.c
@@ -27,6 +27,8 @@
 
 #define _XOPEN_SOURCE 600
 #include "libavutil/avstring.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/opt.h"
 #include "avformat.h"
 #include "internal.h"
 #include <unistd.h>
@@ -47,9 +49,17 @@
  * one anonymous toplevel variant for this, to maintain the structure.
  */
 
+enum KeyType {
+    KEY_NONE,
+    KEY_AES_128,
+};
+
 struct segment {
     int duration;
     char url[MAX_URL_SIZE];
+    char key[MAX_URL_SIZE];
+    enum KeyType key_type;
+    uint8_t iv[16];
 };
 
 /*
@@ -77,6 +87,9 @@ struct variant {
     int needed, cur_needed;
     int cur_seq_no;
     int64_t last_load_time;
+
+    char key_url[MAX_URL_SIZE];
+    uint8_t key[16];
 };
 
 typedef struct AppleHTTPContext {
@@ -160,10 +173,35 @@ static void handle_variant_args(struct variant_info *info, const char *key,
     }
 }
 
+struct key_info {
+     char uri[MAX_URL_SIZE];
+     char method[10];
+     char iv[35];
+};
+
+static void handle_key_args(struct key_info *info, const char *key,
+                            int key_len, char **dest, int *dest_len)
+{
+    if (!strncmp(key, "METHOD=", key_len)) {
+        *dest     =        info->method;
+        *dest_len = sizeof(info->method);
+    } else if (!strncmp(key, "URI=", key_len)) {
+        *dest     =        info->uri;
+        *dest_len = sizeof(info->uri);
+    } else if (!strncmp(key, "IV=", key_len)) {
+        *dest     =        info->iv;
+        *dest_len = sizeof(info->iv);
+    }
+}
+
 static int parse_playlist(AppleHTTPContext *c, const char *url,
                           struct variant *var, AVIOContext *in)
 {
     int ret = 0, duration = 0, is_segment = 0, is_variant = 0, bandwidth = 0;
+    enum KeyType key_type = KEY_NONE;
+    uint8_t iv[16] = "";
+    int has_iv = 0;
+    char key[MAX_URL_SIZE];
     char line[1024];
     const char *ptr;
     int close_in = 0;
@@ -192,6 +230,19 @@ static int parse_playlist(AppleHTTPContext *c, const char *url,
             ff_parse_key_value(ptr, (ff_parse_key_val_cb) handle_variant_args,
                                &info);
             bandwidth = atoi(info.bandwidth);
+        } else if (av_strstart(line, "#EXT-X-KEY:", &ptr)) {
+            struct key_info info = {{0}};
+            ff_parse_key_value(ptr, (ff_parse_key_val_cb) handle_key_args,
+                               &info);
+            key_type = KEY_NONE;
+            has_iv = 0;
+            if (!strcmp(info.method, "AES-128"))
+                key_type = KEY_AES_128;
+            if (!strncmp(info.iv, "0x", 2) || !strncmp(info.iv, "0X", 2)) {
+                ff_hex_to_data(iv, info.iv + 2);
+                has_iv = 1;
+            }
+            av_strlcpy(key, info.uri, sizeof(key));
         } else if (av_strstart(line, "#EXT-X-TARGETDURATION:", &ptr)) {
             if (!var) {
                 var = new_variant(c, 0, url, NULL);
@@ -242,6 +293,15 @@ static int parse_playlist(AppleHTTPContext *c, const char *url,
                     goto fail;
                 }
                 seg->duration = duration;
+                seg->key_type = key_type;
+                if (has_iv) {
+                    memcpy(seg->iv, iv, sizeof(iv));
+                } else {
+                    int seq = var->start_seq_no + var->n_segments;
+                    memset(seg->iv, 0, sizeof(seg->iv));
+                    AV_WB32(seg->iv + 12, seq);
+                }
+                ff_make_absolute_url(seg->key, sizeof(seg->key), url, key);
                 ff_make_absolute_url(seg->url, sizeof(seg->url), url, line);
                 dynarray_add(&var->segments, &var->n_segments, seg);
                 is_segment = 0;
@@ -257,6 +317,50 @@ static int parse_playlist(AppleHTTPContext *c, const char *url,
     return ret;
 }
 
+static int open_input(struct variant *var)
+{
+    struct segment *seg = var->segments[var->cur_seq_no - var->start_seq_no];
+    if (seg->key_type == KEY_NONE) {
+        return ffurl_open(&var->input, seg->url, AVIO_FLAG_READ);
+    } else if (seg->key_type == KEY_AES_128) {
+        char iv[33], key[33], url[MAX_URL_SIZE];
+        int ret;
+        if (strcmp(seg->key, var->key_url)) {
+            URLContext *uc;
+            if (ffurl_open(&uc, seg->key, AVIO_FLAG_READ) == 0) {
+                if (ffurl_read_complete(uc, var->key, sizeof(var->key))
+                    != sizeof(var->key)) {
+                    av_log(NULL, AV_LOG_ERROR, "Unable to read key file %s\n",
+                           seg->key);
+                }
+                ffurl_close(uc);
+            } else {
+                av_log(NULL, AV_LOG_ERROR, "Unable to open key file %s\n",
+                       seg->key);
+            }
+            av_strlcpy(var->key_url, seg->key, sizeof(var->key_url));
+        }
+        ff_data_to_hex(iv, seg->iv, sizeof(seg->iv), 0);
+        ff_data_to_hex(key, var->key, sizeof(var->key), 0);
+        iv[32] = key[32] = '\0';
+        if (strstr(seg->url, "://"))
+            snprintf(url, sizeof(url), "crypto+%s", seg->url);
+        else
+            snprintf(url, sizeof(url), "crypto:%s", seg->url);
+        if ((ret = ffurl_alloc(&var->input, url, AVIO_FLAG_READ)) < 0)
+            return ret;
+        av_set_string3(var->input->priv_data, "key", key, 0, NULL);
+        av_set_string3(var->input->priv_data, "iv", iv, 0, NULL);
+        if ((ret = ffurl_connect(var->input)) < 0) {
+            ffurl_close(var->input);
+            var->input = NULL;
+            return ret;
+        }
+        return 0;
+    }
+    return AVERROR(ENOSYS);
+}
+
 static int read_data(void *opaque, uint8_t *buf, int buf_size)
 {
     struct variant *v = opaque;
@@ -291,9 +395,7 @@ static int read_data(void *opaque, uint8_t *buf, int buf_size)
             goto reload;
         }
 
-        ret = ffurl_open(&v->input,
-                         v->segments[v->cur_seq_no - v->start_seq_no]->url,
-                         AVIO_FLAG_READ);
+        ret = open_input(v);
         if (ret < 0)
             return ret;
     }
diff --git a/libavformat/version.h b/libavformat/version.h
index 04c5d73f50220..9041f92626f1e 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -25,7 +25,7 @@
 
 #define LIBAVFORMAT_VERSION_MAJOR 53
 #define LIBAVFORMAT_VERSION_MINOR  0
-#define LIBAVFORMAT_VERSION_MICRO  1
+#define LIBAVFORMAT_VERSION_MICRO  2
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
                                                LIBAVFORMAT_VERSION_MINOR, \

From 6ffd7eddb832a21f51661038620ef655640ee83f Mon Sep 17 00:00:00 2001
From: Carl Eugen Hoyos <cehoyos@ag.or.at>
Date: Fri, 22 Apr 2011 19:24:17 +0200
Subject: [PATCH 04/20] doc: FLV muxing is supported.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 doc/general.texi | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/doc/general.texi b/doc/general.texi
index edd5e60cc33ff..303159eba7d3f 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -96,7 +96,7 @@ library:
     @tab Only embedded audio is decoded.
 @item FLI/FLC/FLX animation     @tab   @tab X
     @tab .fli/.flc files
-@item Flash Video (FLV)         @tab   @tab X
+@item Flash Video (FLV)         @tab X @tab X
     @tab Macromedia Flash video files
 @item framecrc testing format   @tab X @tab
 @item FunCom ISS                @tab   @tab X

From 15919ee48f507a4ca17147e7247f5cb68f752c2c Mon Sep 17 00:00:00 2001
From: Jason Garrett-Glaser <jason@x264.com>
Date: Fri, 22 Apr 2011 17:59:55 -0400
Subject: [PATCH 05/20] bswap: use native types for av_bwap16().

This prevents a call to bytestream_get_be16() using a movzwl both before
and after the ror instruction, which is obviously inefficient. Arm uses
the same trick also.

Sintel decoding goes from (avg+SD) 9.856 +/- 0.003 to 9.797 +/- 0.003 sec.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
---
 libavutil/x86/bswap.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/libavutil/x86/bswap.h b/libavutil/x86/bswap.h
index 28e3fecb1b0a0..b60d9ccd6cff9 100644
--- a/libavutil/x86/bswap.h
+++ b/libavutil/x86/bswap.h
@@ -29,9 +29,9 @@
 #include "libavutil/attributes.h"
 
 #define av_bswap16 av_bswap16
-static av_always_inline av_const uint16_t av_bswap16(uint16_t x)
+static av_always_inline av_const unsigned av_bswap16(unsigned x)
 {
-    __asm__("rorw $8, %0" : "+r"(x));
+    __asm__("rorw $8, %w0" : "+r"(x));
     return x;
 }
 

From 3583d7cffe5173cc02c5476451800efdc767ce15 Mon Sep 17 00:00:00 2001
From: Alex Converse <aconverse@google.com>
Date: Wed, 20 Apr 2011 17:25:47 +0000
Subject: [PATCH 06/20] Fix mov debug (u)int64_t format strings.

---
 libavformat/mov.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/libavformat/mov.c b/libavformat/mov.c
index 76dc8a13af8de..65fcf1987037e 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -588,7 +588,7 @@ static int mov_read_moov(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 static int mov_read_moof(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 {
     c->fragment.moof_offset = avio_tell(pb) - 8;
-    av_dlog(c->fc, "moof offset %llx\n", c->fragment.moof_offset);
+    av_dlog(c->fc, "moof offset %"PRIx64"\n", c->fragment.moof_offset);
     return mov_read_default(c, pb, atom);
 }
 
@@ -2367,7 +2367,7 @@ static int mov_read_header(AVFormatContext *s, AVFormatParameters *ap)
         av_log(s, AV_LOG_ERROR, "moov atom not found\n");
         return -1;
     }
-    av_dlog(mov->fc, "on_parse_exit_offset=%lld\n", avio_tell(pb));
+    av_dlog(mov->fc, "on_parse_exit_offset=%"PRId64"\n", avio_tell(pb));
 
     if (pb->seekable && mov->chapter_track > 0)
         mov_read_chapters(s);
@@ -2416,7 +2416,7 @@ static int mov_read_packet(AVFormatContext *s, AVPacket *pkt)
             mov_read_default(mov, s->pb, (MOVAtom){ AV_RL32("root"), INT64_MAX }) < 0 ||
             s->pb->eof_reached)
             return AVERROR_EOF;
-        av_dlog(s, "read fragments, offset 0x%llx\n", avio_tell(s->pb));
+        av_dlog(s, "read fragments, offset 0x%"PRIx64"\n", avio_tell(s->pb));
         goto retry;
     }
     sc = st->priv_data;

From 785c4418289e5aace9823823a8d19974e40d4b64 Mon Sep 17 00:00:00 2001
From: Alex Converse <aconverse@google.com>
Date: Wed, 20 Apr 2011 17:47:28 +0000
Subject: [PATCH 07/20] Add some debug log messages to AAC extradata
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

On Wed, Apr 20, 2011 at 11:39 AM, Justin Ruggles
<justin.ruggles@gmail.com> wrote:
> On 04/20/2011 02:26 PM, Alex Converse wrote:
>
>> ---
>>  libavcodec/aacdec.c |   10 +++++++++-
>>  1 files changed, 9 insertions(+), 1 deletions(-)
>>
>>
>>
>> 0002-Add-some-Debug-log-messages-to-AAC-extradata.patch
>>
>>
>> diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
>> index c9761a1..3ec274f 100644
>> --- a/libavcodec/aacdec.c
>> +++ b/libavcodec/aacdec.c
>> @@ -79,7 +79,6 @@
>>             Parametric Stereo.
>>   */
>>
>> -
>>  #include "avcodec.h"
>>  #include "internal.h"
>>  #include "get_bits.h"
>
>
> stray whitespace change
>

oops, fixed

>From 94e8d0eea77480630f84368c97646cabc0f50628 Mon Sep 17 00:00:00 2001
From: Alex Converse <aconverse@google.com>
Date: Wed, 20 Apr 2011 11:23:34 -0700
Subject: [PATCH] Add some debug log messages to AAC extradata
MIME-Version: 1.0
Content-Type: multipart/mixed; boundary="------------1"

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---
 libavcodec/aacdec.c | 9 +++++++++
 1 file changed, 9 insertions(+)

diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index 3160c66d4498a..0c6312fb015f0 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -463,6 +463,11 @@ static int decode_audio_specific_config(AACContext *ac,
     GetBitContext gb;
     int i;
 
+    av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
+    for (i = 0; i < avctx->extradata_size; i++)
+         av_dlog(avctx, "%02x ", avctx->extradata[i]);
+    av_dlog(avctx, "\n");
+
     init_get_bits(&gb, data, data_size * 8);
 
     if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
@@ -489,6 +494,10 @@ static int decode_audio_specific_config(AACContext *ac,
         return -1;
     }
 
+    av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
+            m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
+            m4ac->sample_rate, m4ac->sbr, m4ac->ps);
+
     return get_bits_count(&gb);
 }
 

From cfc2a0cf848e71fcec0861a73b26c2c96a201357 Mon Sep 17 00:00:00 2001
From: Alex Converse <aconverse@google.com>
Date: Thu, 21 Apr 2011 18:39:04 +0000
Subject: [PATCH 08/20] Remove RDFT dependency from AAC decoder.

$subj

>From 557176d961c70604c2a96d81aff4bd6faa670d8a Mon Sep 17 00:00:00 2001
From: Alex Converse <aconverse@google.com>
Date: Thu, 21 Apr 2011 12:11:42 -0700
Subject: [PATCH] Remove RDFT dependency from AAC decoder.
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It was used for an old implementation of the SBR filterbank.

./configure --disable-everything --disable-ffplay --enable-decoder=aac works.
---
 configure | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/configure b/configure
index dd44ba47396ae..c01c551cf5491 100755
--- a/configure
+++ b/configure
@@ -1237,7 +1237,7 @@ mdct_select="fft"
 rdft_select="fft"
 
 # decoders / encoders / hardware accelerators
-aac_decoder_select="mdct rdft sinewin"
+aac_decoder_select="mdct sinewin"
 aac_encoder_select="mdct sinewin"
 aac_latm_decoder_select="aac_decoder aac_latm_parser"
 ac3_decoder_select="mdct ac3dsp ac3_parser"

From 230c1a90754860719b017a9ea4627dc81f67ff6e Mon Sep 17 00:00:00 2001
From: Nathan Caldwell <saintdev@gmail.com>
Date: Wed, 5 Jan 2011 01:32:16 -0700
Subject: [PATCH 09/20] aacenc: Finish 3GPP psymodel analysis for non mid/side
 cases.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

There is still are still a few sections missing relating to TNS (not present)
and mid/side (contains other bugs).

Overall this improves quality, and vastly improves rate-control.

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 libavcodec/aacenc.c |   4 +-
 libavcodec/aacpsy.c | 305 +++++++++++++++++++++++++++++++++++++++++++-
 2 files changed, 303 insertions(+), 6 deletions(-)

diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index d4b61126bdefc..4ec76d063a9a5 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -606,8 +606,10 @@ static int aac_encode_frame(AVCodecContext *avctx,
         }
 
         frame_bits = put_bits_count(&s->pb);
-        if (frame_bits <= 6144 * avctx->channels - 3)
+        if (frame_bits <= 6144 * avctx->channels - 3) {
+            s->psy.bitres.bits = frame_bits / avctx->channels;
             break;
+        }
 
         s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
 
diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index 4250a5d76386f..98be176c0f2bc 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -30,7 +30,6 @@
 
 /***********************************
  *              TODOs:
- * thresholds linearization after their modifications for attaining given bitrate
  * try other bitrate controlling mechanism (maybe use ratecontrol.c?)
  * control quality for quality-based output
  **********************************/
@@ -41,10 +40,51 @@
  */
 #define PSY_3GPP_THR_SPREAD_HI   1.5f // spreading factor for low-to-hi threshold spreading  (15 dB/Bark)
 #define PSY_3GPP_THR_SPREAD_LOW  3.0f // spreading factor for hi-to-low threshold spreading  (30 dB/Bark)
+/* spreading factor for low-to-hi energy spreading, long block, > 22kbps/channel (20dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_HI_L1 2.0f
+/* spreading factor for low-to-hi energy spreading, long block, <= 22kbps/channel (15dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_HI_L2 1.5f
+/* spreading factor for low-to-hi energy spreading, short block (15 dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_HI_S  1.5f
+/* spreading factor for hi-to-low energy spreading, long block (30dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_LOW_L 3.0f
+/* spreading factor for hi-to-low energy spreading, short block (20dB/Bark) */
+#define PSY_3GPP_EN_SPREAD_LOW_S 2.0f
 
 #define PSY_3GPP_RPEMIN      0.01f
 #define PSY_3GPP_RPELEV      2.0f
 
+#define PSY_3GPP_C1          3.0f           /* log2(8) */
+#define PSY_3GPP_C2          1.3219281f     /* log2(2.5) */
+#define PSY_3GPP_C3          0.55935729f    /* 1 - C2 / C1 */
+
+#define PSY_SNR_1DB          7.9432821e-1f  /* -1dB */
+#define PSY_SNR_25DB         3.1622776e-3f  /* -25dB */
+
+#define PSY_3GPP_SAVE_SLOPE_L  -0.46666667f
+#define PSY_3GPP_SAVE_SLOPE_S  -0.36363637f
+#define PSY_3GPP_SAVE_ADD_L    -0.84285712f
+#define PSY_3GPP_SAVE_ADD_S    -0.75f
+#define PSY_3GPP_SPEND_SLOPE_L  0.66666669f
+#define PSY_3GPP_SPEND_SLOPE_S  0.81818181f
+#define PSY_3GPP_SPEND_ADD_L   -0.35f
+#define PSY_3GPP_SPEND_ADD_S   -0.26111111f
+#define PSY_3GPP_CLIP_LO_L      0.2f
+#define PSY_3GPP_CLIP_LO_S      0.2f
+#define PSY_3GPP_CLIP_HI_L      0.95f
+#define PSY_3GPP_CLIP_HI_S      0.75f
+
+#define PSY_3GPP_AH_THR_LONG    0.5f
+#define PSY_3GPP_AH_THR_SHORT   0.63f
+
+enum {
+    PSY_3GPP_AH_NONE,
+    PSY_3GPP_AH_INACTIVE,
+    PSY_3GPP_AH_ACTIVE
+};
+
+#define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f)
+
 /* LAME psy model constants */
 #define PSY_LAME_FIR_LEN 21         ///< LAME psy model FIR order
 #define AAC_BLOCK_SIZE_LONG 1024    ///< long block size
@@ -63,6 +103,12 @@ typedef struct AacPsyBand{
     float energy;    ///< band energy
     float thr;       ///< energy threshold
     float thr_quiet; ///< threshold in quiet
+    float nz_lines;     ///< number of non-zero spectral lines
+    float active_lines; ///< number of active spectral lines
+    float pe;           ///< perceptual entropy
+    float pe_const;     ///< constant part of the PE calculation
+    float norm_fac;     ///< normalization factor for linearization
+    int   avoid_holes;  ///< hole avoidance flag
 }AacPsyBand;
 
 /**
@@ -97,6 +143,15 @@ typedef struct AacPsyCoeffs{
  * 3GPP TS26.403-inspired psychoacoustic model specific data
  */
 typedef struct AacPsyContext{
+    int chan_bitrate;     ///< bitrate per channel
+    int frame_bits;       ///< average bits per frame
+    int fill_level;       ///< bit reservoir fill level
+    struct {
+        float min;        ///< minimum allowed PE for bit factor calculation
+        float max;        ///< maximum allowed PE for bit factor calculation
+        float previous;   ///< allowed PE of the previous frame
+        float correction; ///< PE correction factor
+    } pe;
     AacPsyCoeffs psy_coef[2][64];
     AacPsyChannel *ch;
 }AacPsyContext;
@@ -235,16 +290,33 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
     AacPsyContext *pctx;
     float bark;
     int i, j, g, start;
-    float prev, minscale, minath;
+    float prev, minscale, minath, minsnr, pe_min;
+    const int chan_bitrate = ctx->avctx->bit_rate / ctx->avctx->channels;
+    const int bandwidth    = ctx->avctx->cutoff ? ctx->avctx->cutoff : ctx->avctx->sample_rate / 2;
+    const float num_bark   = calc_bark((float)bandwidth);
 
     ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
     pctx = (AacPsyContext*) ctx->model_priv_data;
 
+    pctx->chan_bitrate = chan_bitrate;
+    pctx->frame_bits   = chan_bitrate * AAC_BLOCK_SIZE_LONG / ctx->avctx->sample_rate;
+    pctx->pe.min       =  8.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
+    pctx->pe.max       = 12.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
+    ctx->bitres.size   = 6144 - pctx->frame_bits;
+    ctx->bitres.size  -= ctx->bitres.size % 8;
+    pctx->fill_level   = ctx->bitres.size;
     minath = ath(3410, ATH_ADD);
     for (j = 0; j < 2; j++) {
         AacPsyCoeffs *coeffs = pctx->psy_coef[j];
         const uint8_t *band_sizes = ctx->bands[j];
         float line_to_frequency = ctx->avctx->sample_rate / (j ? 256.f : 2048.0f);
+        float avg_chan_bits = chan_bitrate / ctx->avctx->sample_rate * (j ? 128.0f : 1024.0f);
+        /* reference encoder uses 2.4% here instead of 60% like the spec says */
+        float bark_pe = 0.024f * PSY_3GPP_BITS_TO_PE(avg_chan_bits) / num_bark;
+        float en_spread_low = j ? PSY_3GPP_EN_SPREAD_LOW_S : PSY_3GPP_EN_SPREAD_LOW_L;
+        /* High energy spreading for long blocks <= 22kbps/channel and short blocks are the same. */
+        float en_spread_hi  = (j || (chan_bitrate <= 22.0f)) ? PSY_3GPP_EN_SPREAD_HI_S : PSY_3GPP_EN_SPREAD_HI_L1;
+
         i = 0;
         prev = 0.0;
         for (g = 0; g < ctx->num_bands[j]; g++) {
@@ -258,6 +330,11 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
             float bark_width = coeffs[g+1].barks - coeffs->barks;
             coeff->spread_low[0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_LOW);
             coeff->spread_hi [0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_HI);
+            coeff->spread_low[1] = pow(10.0, -bark_width * en_spread_low);
+            coeff->spread_hi [1] = pow(10.0, -bark_width * en_spread_hi);
+            pe_min = bark_pe * bark_width;
+            minsnr = pow(2.0f, pe_min / band_sizes[g]) - 1.5f;
+            coeff->min_snr = av_clipf(1.0f / minsnr, PSY_SNR_25DB, PSY_SNR_1DB);
         }
         start = 0;
         for (g = 0; g < ctx->num_bands[j]; g++) {
@@ -385,6 +462,97 @@ static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
     return wi;
 }
 
+/* 5.6.1.2 "Calculation of Bit Demand" */
+static int calc_bit_demand(AacPsyContext *ctx, float pe, int bits, int size,
+                           int short_window)
+{
+    const float bitsave_slope  = short_window ? PSY_3GPP_SAVE_SLOPE_S  : PSY_3GPP_SAVE_SLOPE_L;
+    const float bitsave_add    = short_window ? PSY_3GPP_SAVE_ADD_S    : PSY_3GPP_SAVE_ADD_L;
+    const float bitspend_slope = short_window ? PSY_3GPP_SPEND_SLOPE_S : PSY_3GPP_SPEND_SLOPE_L;
+    const float bitspend_add   = short_window ? PSY_3GPP_SPEND_ADD_S   : PSY_3GPP_SPEND_ADD_L;
+    const float clip_low       = short_window ? PSY_3GPP_CLIP_LO_S     : PSY_3GPP_CLIP_LO_L;
+    const float clip_high      = short_window ? PSY_3GPP_CLIP_HI_S     : PSY_3GPP_CLIP_HI_L;
+    float clipped_pe, bit_save, bit_spend, bit_factor, fill_level;
+
+    ctx->fill_level += ctx->frame_bits - bits;
+    ctx->fill_level  = av_clip(ctx->fill_level, 0, size);
+    fill_level = av_clipf((float)ctx->fill_level / size, clip_low, clip_high);
+    clipped_pe = av_clipf(pe, ctx->pe.min, ctx->pe.max);
+    bit_save   = (fill_level + bitsave_add) * bitsave_slope;
+    assert(bit_save <= 0.3f && bit_save >= -0.05000001f);
+    bit_spend  = (fill_level + bitspend_add) * bitspend_slope;
+    assert(bit_spend <= 0.5f && bit_spend >= -0.1f);
+    /* The bit factor graph in the spec is obviously incorrect.
+     *      bit_spend + ((bit_spend - bit_spend))...
+     * The reference encoder subtracts everything from 1, but also seems incorrect.
+     *      1 - bit_save + ((bit_spend + bit_save))...
+     * Hopefully below is correct.
+     */
+    bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (ctx->pe.max - ctx->pe.min)) * (clipped_pe - ctx->pe.min);
+    /* NOTE: The reference encoder attempts to center pe max/min around the current pe. */
+    ctx->pe.max = FFMAX(pe, ctx->pe.max);
+    ctx->pe.min = FFMIN(pe, ctx->pe.min);
+
+    return FFMIN(ctx->frame_bits * bit_factor, ctx->frame_bits + size - bits);
+}
+
+static float calc_pe_3gpp(AacPsyBand *band)
+{
+    float pe, a;
+
+    band->pe           = 0.0f;
+    band->pe_const     = 0.0f;
+    band->active_lines = 0.0f;
+    if (band->energy > band->thr) {
+        a  = log2f(band->energy);
+        pe = a - log2f(band->thr);
+        band->active_lines = band->nz_lines;
+        if (pe < PSY_3GPP_C1) {
+            pe = pe * PSY_3GPP_C3 + PSY_3GPP_C2;
+            a  = a  * PSY_3GPP_C3 + PSY_3GPP_C2;
+            band->active_lines *= PSY_3GPP_C3;
+        }
+        band->pe       = pe * band->nz_lines;
+        band->pe_const = a  * band->nz_lines;
+    }
+
+    return band->pe;
+}
+
+static float calc_reduction_3gpp(float a, float desired_pe, float pe,
+                                 float active_lines)
+{
+    float thr_avg, reduction;
+
+    thr_avg   = powf(2.0f, (a - pe) / (4.0f * active_lines));
+    reduction = powf(2.0f, (a - desired_pe) / (4.0f * active_lines)) - thr_avg;
+
+    return FFMAX(reduction, 0.0f);
+}
+
+static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr,
+                                   float reduction)
+{
+    float thr = band->thr;
+
+    if (band->energy > thr) {
+        thr = powf(thr, 0.25f) + reduction;
+        thr = powf(thr, 4.0f);
+
+        /* This deviates from the 3GPP spec to match the reference encoder.
+         * It performs min(thr_reduced, max(thr, energy/min_snr)) only for bands
+         * that have hole avoidance on (active or inactive). It always reduces the
+         * threshold of bands with hole avoidance off.
+         */
+        if (thr > band->energy * min_snr && band->avoid_holes != PSY_3GPP_AH_NONE) {
+            thr = FFMAX(band->thr, band->energy * min_snr);
+            band->avoid_holes = PSY_3GPP_AH_ACTIVE;
+        }
+    }
+
+    return thr;
+}
+
 /**
  * Calculate band thresholds as suggested in 3GPP TS26.403
  */
@@ -395,18 +563,27 @@ static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
     AacPsyChannel *pch  = &pctx->ch[channel];
     int start = 0;
     int i, w, g;
+    float desired_bits, desired_pe, delta_pe, reduction, spread_en[128] = {0};
+    float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
+    float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
     const int      num_bands  = ctx->num_bands[wi->num_windows == 8];
     const uint8_t *band_sizes = ctx->bands[wi->num_windows == 8];
     AacPsyCoeffs  *coeffs     = pctx->psy_coef[wi->num_windows == 8];
+    const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
 
     //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
     for (w = 0; w < wi->num_windows*16; w += 16) {
         for (g = 0; g < num_bands; g++) {
             AacPsyBand *band = &pch->band[w+g];
+            float form_factor = 0.0f;
             band->energy = 0.0f;
-            for (i = 0; i < band_sizes[g]; i++)
+            for (i = 0; i < band_sizes[g]; i++) {
                 band->energy += coefs[start+i] * coefs[start+i];
+                form_factor  += sqrtf(fabs(coefs[start+i]));
+            }
             band->thr     = band->energy * 0.001258925f;
+            band->nz_lines = form_factor / powf(band->energy / band_sizes[g], 0.25f);
+
             start        += band_sizes[g];
         }
     }
@@ -414,10 +591,15 @@ static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
     for (w = 0; w < wi->num_windows*16; w += 16) {
         AacPsyBand *bands = &pch->band[w];
         //5.4.2.3 "Spreading" & 5.4.3 "Spreaded Energy Calculation"
-        for (g = 1; g < num_bands; g++)
+        spread_en[0] = bands[0].energy;
+        for (g = 1; g < num_bands; g++) {
             bands[g].thr = FFMAX(bands[g].thr, bands[g-1].thr * coeffs[g].spread_hi[0]);
-        for (g = num_bands - 2; g >= 0; g--)
+            spread_en[w+g] = FFMAX(bands[g].energy, spread_en[w+g-1] * coeffs[g].spread_hi[1]);
+        }
+        for (g = num_bands - 2; g >= 0; g--) {
             bands[g].thr = FFMAX(bands[g].thr, bands[g+1].thr * coeffs[g].spread_low[0]);
+            spread_en[w+g] = FFMAX(spread_en[w+g], spread_en[w+g+1] * coeffs[g].spread_low[1]);
+        }
         //5.4.2.4 "Threshold in quiet"
         for (g = 0; g < num_bands; g++) {
             AacPsyBand *band = &bands[g];
@@ -426,6 +608,119 @@ static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
             if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (wi->window_type[1] == LONG_START_SEQUENCE && !w)))
                 band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
                                   PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
+
+            /* 5.6.1.3.1 "Prepatory steps of the perceptual entropy calculation" */
+            pe += calc_pe_3gpp(band);
+            a  += band->pe_const;
+            active_lines += band->active_lines;
+
+            /* 5.6.1.3.3 "Selection of the bands for avoidance of holes" */
+            if (spread_en[w+g] * avoid_hole_thr > band->energy || coeffs[g].min_snr > 1.0f)
+                band->avoid_holes = PSY_3GPP_AH_NONE;
+            else
+                band->avoid_holes = PSY_3GPP_AH_INACTIVE;
+        }
+    }
+
+    /* 5.6.1.3.2 "Calculation of the desired perceptual entropy" */
+    ctx->pe[channel] = pe;
+    desired_bits = calc_bit_demand(pctx, pe, ctx->bitres.bits, ctx->bitres.size, wi->num_windows == 8);
+    desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits);
+    /* NOTE: PE correction is kept simple. During initial testing it had very
+     *       little effect on the final bitrate. Probably a good idea to come
+     *       back and do more testing later.
+     */
+    if (ctx->bitres.bits > 0)
+        desired_pe *= av_clipf(pctx->pe.previous / PSY_3GPP_BITS_TO_PE(ctx->bitres.bits),
+                               0.85f, 1.15f);
+    pctx->pe.previous = PSY_3GPP_BITS_TO_PE(desired_bits);
+
+    if (desired_pe < pe) {
+        /* 5.6.1.3.4 "First Estimation of the reduction value" */
+        for (w = 0; w < wi->num_windows*16; w += 16) {
+            reduction = calc_reduction_3gpp(a, desired_pe, pe, active_lines);
+            pe = 0.0f;
+            a  = 0.0f;
+            active_lines = 0.0f;
+            for (g = 0; g < num_bands; g++) {
+                AacPsyBand *band = &pch->band[w+g];
+
+                band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction);
+                /* recalculate PE */
+                pe += calc_pe_3gpp(band);
+                a  += band->pe_const;
+                active_lines += band->active_lines;
+            }
+        }
+
+        /* 5.6.1.3.5 "Second Estimation of the reduction value" */
+        for (i = 0; i < 2; i++) {
+            float pe_no_ah = 0.0f, desired_pe_no_ah;
+            active_lines = a = 0.0f;
+            for (w = 0; w < wi->num_windows*16; w += 16) {
+                for (g = 0; g < num_bands; g++) {
+                    AacPsyBand *band = &pch->band[w+g];
+
+                    if (band->avoid_holes != PSY_3GPP_AH_ACTIVE) {
+                        pe_no_ah += band->pe;
+                        a        += band->pe_const;
+                        active_lines += band->active_lines;
+                    }
+                }
+            }
+            desired_pe_no_ah = FFMAX(desired_pe - (pe - pe_no_ah), 0.0f);
+            if (active_lines > 0.0f)
+                reduction += calc_reduction_3gpp(a, desired_pe_no_ah, pe_no_ah, active_lines);
+
+            pe = 0.0f;
+            for (w = 0; w < wi->num_windows*16; w += 16) {
+                for (g = 0; g < num_bands; g++) {
+                    AacPsyBand *band = &pch->band[w+g];
+
+                    if (active_lines > 0.0f)
+                        band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction);
+                    pe += calc_pe_3gpp(band);
+                    band->norm_fac = band->active_lines / band->thr;
+                    norm_fac += band->norm_fac;
+                }
+            }
+            delta_pe = desired_pe - pe;
+            if (fabs(delta_pe) > 0.05f * desired_pe)
+                break;
+        }
+
+        if (pe < 1.15f * desired_pe) {
+            /* 6.6.1.3.6 "Final threshold modification by linearization" */
+            norm_fac = 1.0f / norm_fac;
+            for (w = 0; w < wi->num_windows*16; w += 16) {
+                for (g = 0; g < num_bands; g++) {
+                    AacPsyBand *band = &pch->band[w+g];
+
+                    if (band->active_lines > 0.5f) {
+                        float delta_sfb_pe = band->norm_fac * norm_fac * delta_pe;
+                        float thr = band->thr;
+
+                        thr *= powf(2.0f, delta_sfb_pe / band->active_lines);
+                        if (thr > coeffs[g].min_snr * band->energy && band->avoid_holes == PSY_3GPP_AH_INACTIVE)
+                            thr = FFMAX(band->thr, coeffs[g].min_snr * band->energy);
+                        band->thr = thr;
+                    }
+                }
+            }
+        } else {
+            /* 5.6.1.3.7 "Further perceptual entropy reduction" */
+            g = num_bands;
+            while (pe > desired_pe && g--) {
+                for (w = 0; w < wi->num_windows*16; w+= 16) {
+                    AacPsyBand *band = &pch->band[w+g];
+                    if (band->avoid_holes != PSY_3GPP_AH_NONE && coeffs[g].min_snr < PSY_SNR_1DB) {
+                        coeffs[g].min_snr = PSY_SNR_1DB;
+                        band->thr = band->energy * PSY_SNR_1DB;
+                        pe += band->active_lines * 1.5f - band->pe;
+                    }
+                }
+            }
+            /* TODO: allow more holes (unused without mid/side) */
         }
     }
 

From 62582a696b4dbe3d644ea19aa6739bbe592482da Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Sat, 23 Apr 2011 02:02:42 +0200
Subject: [PATCH 10/20] cook: Fix small typo in av_log_ask_for_sample message.

---
 libavcodec/cook.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/libavcodec/cook.c b/libavcodec/cook.c
index 282ca1029d516..f1df2405f6a8f 100644
--- a/libavcodec/cook.c
+++ b/libavcodec/cook.c
@@ -1136,7 +1136,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
         switch (q->subpacket[s].cookversion) {
             case MONO:
                 if (q->nb_channels != 1) {
-                    av_log_ask_for_sample(avctx, "Container channels != 1.!\n");
+                    av_log_ask_for_sample(avctx, "Container channels != 1.\n");
                     return -1;
                 }
                 av_log(avctx,AV_LOG_DEBUG,"MONO\n");

From f50d9377252431114e89514cf528825936efcbd9 Mon Sep 17 00:00:00 2001
From: Nathan Caldwell <saintdev@gmail.com>
Date: Tue, 5 Apr 2011 01:05:24 -0600
Subject: [PATCH 11/20] aacenc: Fix whitespace after last commit.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 libavcodec/aacpsy.c | 23 +++++++++++++----------
 1 file changed, 13 insertions(+), 10 deletions(-)

diff --git a/libavcodec/aacpsy.c b/libavcodec/aacpsy.c
index 98be176c0f2bc..5aea1faa40bcc 100644
--- a/libavcodec/aacpsy.c
+++ b/libavcodec/aacpsy.c
@@ -100,9 +100,9 @@ enum {
  * information for single band used by 3GPP TS26.403-inspired psychoacoustic model
  */
 typedef struct AacPsyBand{
-    float energy;    ///< band energy
-    float thr;       ///< energy threshold
-    float thr_quiet; ///< threshold in quiet
+    float energy;       ///< band energy
+    float thr;          ///< energy threshold
+    float thr_quiet;    ///< threshold in quiet
     float nz_lines;     ///< number of non-zero spectral lines
     float active_lines; ///< number of active spectral lines
     float pe;           ///< perceptual entropy
@@ -566,43 +566,46 @@ static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
     float desired_bits, desired_pe, delta_pe, reduction, spread_en[128] = {0};
     float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
     float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
-    const int      num_bands  = ctx->num_bands[wi->num_windows == 8];
-    const uint8_t *band_sizes = ctx->bands[wi->num_windows == 8];
-    AacPsyCoeffs  *coeffs     = pctx->psy_coef[wi->num_windows == 8];
+    const int      num_bands   = ctx->num_bands[wi->num_windows == 8];
+    const uint8_t *band_sizes  = ctx->bands[wi->num_windows == 8];
+    AacPsyCoeffs  *coeffs      = pctx->psy_coef[wi->num_windows == 8];
     const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
 
     //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
     for (w = 0; w < wi->num_windows*16; w += 16) {
         for (g = 0; g < num_bands; g++) {
             AacPsyBand *band = &pch->band[w+g];
+
             float form_factor = 0.0f;
             band->energy = 0.0f;
             for (i = 0; i < band_sizes[g]; i++) {
                 band->energy += coefs[start+i] * coefs[start+i];
                 form_factor  += sqrtf(fabs(coefs[start+i]));
             }
-            band->thr     = band->energy * 0.001258925f;
+            band->thr      = band->energy * 0.001258925f;
             band->nz_lines = form_factor / powf(band->energy / band_sizes[g], 0.25f);
 
-            start        += band_sizes[g];
+            start += band_sizes[g];
         }
     }
     //modify thresholds and energies - spread, threshold in quiet, pre-echo control
     for (w = 0; w < wi->num_windows*16; w += 16) {
         AacPsyBand *bands = &pch->band[w];
+
         //5.4.2.3 "Spreading" & 5.4.3 "Spreaded Energy Calculation"
         spread_en[0] = bands[0].energy;
         for (g = 1; g < num_bands; g++) {
-            bands[g].thr = FFMAX(bands[g].thr, bands[g-1].thr * coeffs[g].spread_hi[0]);
+            bands[g].thr   = FFMAX(bands[g].thr,    bands[g-1].thr * coeffs[g].spread_hi[0]);
             spread_en[w+g] = FFMAX(bands[g].energy, spread_en[w+g-1] * coeffs[g].spread_hi[1]);
         }
         for (g = num_bands - 2; g >= 0; g--) {
-            bands[g].thr = FFMAX(bands[g].thr, bands[g+1].thr * coeffs[g].spread_low[0]);
+            bands[g].thr   = FFMAX(bands[g].thr,   bands[g+1].thr * coeffs[g].spread_low[0]);
             spread_en[w+g] = FFMAX(spread_en[w+g], spread_en[w+g+1] * coeffs[g].spread_low[1]);
         }
         //5.4.2.4 "Threshold in quiet"
         for (g = 0; g < num_bands; g++) {
             AacPsyBand *band = &bands[g];
+
             band->thr_quiet = band->thr = FFMAX(band->thr, coeffs[g].ath);
             //5.4.2.5 "Pre-echo control"
             if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (wi->window_type[1] == LONG_START_SEQUENCE && !w)))

From 046f3cb78903cd04846c49345b473244f095f1a5 Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Sun, 17 Apr 2011 19:44:33 +0200
Subject: [PATCH 12/20] vorbis: Rename decoder/encoder files to follow general
 file naming scheme.

---
 libavcodec/Makefile                      | 4 ++--
 libavcodec/twinvq.c                      | 2 +-
 libavcodec/{vorbis_dec.c => vorbisdec.c} | 0
 libavcodec/{vorbis_enc.c => vorbisenc.c} | 0
 4 files changed, 3 insertions(+), 3 deletions(-)
 rename libavcodec/{vorbis_dec.c => vorbisdec.c} (100%)
 rename libavcodec/{vorbis_enc.c => vorbisenc.c} (100%)

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 784226f7af005..fad435dee39ee 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -386,9 +386,9 @@ OBJS-$(CONFIG_VCR1_ENCODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdav.o
 OBJS-$(CONFIG_VMNC_DECODER)            += vmnc.o
-OBJS-$(CONFIG_VORBIS_DECODER)          += vorbis_dec.o vorbis.o \
+OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbis.o \
                                           vorbis_data.o xiph.o
-OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbis_enc.o vorbis.o \
+OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbisenc.o vorbis.o \
                                           vorbis_data.o
 OBJS-$(CONFIG_VP3_DECODER)             += vp3.o vp3dsp.o
 OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56dsp.o \
diff --git a/libavcodec/twinvq.c b/libavcodec/twinvq.c
index e1a8dc455bb40..985f71c8ca644 100644
--- a/libavcodec/twinvq.c
+++ b/libavcodec/twinvq.c
@@ -234,7 +234,7 @@ static void memset_float(float *buf, float val, int size)
  *        be a multiple of four.
  * @return the LPC value
  *
- * @todo reuse code from vorbis_dec.c: vorbis_floor0_decode
+ * @todo reuse code from Vorbis decoder: vorbis_floor0_decode
  */
 static float eval_lpc_spectrum(const float *lsp, float cos_val, int order)
 {
diff --git a/libavcodec/vorbis_dec.c b/libavcodec/vorbisdec.c
similarity index 100%
rename from libavcodec/vorbis_dec.c
rename to libavcodec/vorbisdec.c
diff --git a/libavcodec/vorbis_enc.c b/libavcodec/vorbisenc.c
similarity index 100%
rename from libavcodec/vorbis_enc.c
rename to libavcodec/vorbisenc.c

From 75e8fba8232108c075a8b24bd7f17b7c5869c7bc Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Sat, 23 Apr 2011 14:14:22 +0200
Subject: [PATCH 13/20] Remove stray libavcore and _g binary references.

---
 .gitignore | 2 --
 Changelog  | 1 -
 2 files changed, 3 deletions(-)

diff --git a/.gitignore b/.gitignore
index eac1758ed9cf0..3d5e38b6bdb11 100644
--- a/.gitignore
+++ b/.gitignore
@@ -6,7 +6,6 @@
 *.ho
 *-example
 *-test
-*_g
 config.*
 doc/*.1
 doc/*.html
@@ -17,7 +16,6 @@ ffplay
 ffprobe
 ffserver
 libavcodec/libavcodec*
-libavcore/libavcore*
 libavdevice/libavdevice*
 libavfilter/libavfilter*
 libavformat/libavformat*
diff --git a/Changelog b/Changelog
index f38ef9eec5d2e..00777e4e6a40a 100644
--- a/Changelog
+++ b/Changelog
@@ -27,7 +27,6 @@ version 0.7_beta1:
 - RTP depacketization of QDM2
 - ANSI/ASCII art playback system
 - Lego Mindstorms RSO de/muxer
-- libavcore added
 - SubRip subtitle file muxer and demuxer
 - Chinese AVS encoding via libxavs
 - ffprobe -show_packets option added

From ff3be572efde575c2b5ee621a84920ac9a5b9b8f Mon Sep 17 00:00:00 2001
From: Brad <brad@comstyle.com>
Date: Sun, 27 Mar 2011 21:05:22 -0400
Subject: [PATCH 14/20] patcheck: Allow overiding grep program(s) through
 environment variables.

patcheck hardcodes the binary names for grep/egrep. This makes overriding the
binary names a pain, e.g. when calling a GNU version of grep on BSD systems.

Signed-off-by: Diego Biurrun <diego@biurrun.de>
---
 tools/patcheck | 50 ++++++++++++++++++++++++++------------------------
 1 file changed, 26 insertions(+), 24 deletions(-)

diff --git a/tools/patcheck b/tools/patcheck
index e21a9f681d4d1..d81a48937fa39 100755
--- a/tools/patcheck
+++ b/tools/patcheck
@@ -8,9 +8,11 @@ if [ $# = 0 ]; then
     exit
 fi
 
+GREP=grep
+EGREP=egrep
 TMP=patcheck.tmp
 OPT="-nH"
-#FILES=`grep '^+++' $* | sed 's/+++ //g'`
+#FILES=`$GREP '^+++' $* | sed 's/+++ //g'`
 
 echo patCHeck 1e10.0
 echo This tool is intended to help a human check/review patches it is very far from
@@ -27,7 +29,7 @@ hiegrep(){
     arg="$1"
     msg="$2"
     shift 2
-    grep $OPT '^+' $* | grep -v ':+++'| egrep --color=always -- "$arg"> $TMP && printf "\n$msg\n"
+    $GREP $OPT '^+' $* | $GREP -v ':+++'| $EGREP --color=always -- "$arg"> $TMP && printf "\n$msg\n"
     cat $TMP
 }
 
@@ -36,7 +38,7 @@ hiegrep2(){
     varg="$2"
     msg="$3"
     shift 3
-    grep $OPT '^+' $* | grep -v ':+++' | egrep -v -- "$varg" | egrep --color=always -- "$arg" > $TMP && printf "\n$msg\n"
+    $GREP $OPT '^+' $* | $GREP -v ':+++' | $EGREP -v -- "$varg" | $EGREP --color=always -- "$arg" > $TMP && printf "\n$msg\n"
     cat $TMP
 }
 
@@ -60,7 +62,7 @@ hiegrep '\+= *1 *;'     'can be simplified to ++' $*
 hiegrep '-= *1 *;'      'can be simplified to --' $*
 hiegrep '((!|=)= *(0|NULL)[^0-9a-z]|[^0-9a-z](0|NULL) *(!|=)=)' 'x==0 / x!=0 can be simplified to !x / x' $*
 
-egrep $OPT '^\+ *(const *|)static' $*| egrep --color=always '[^=]= *(0|NULL)[^0-9a-zA-Z]'> $TMP && printf '\nuseless 0 init\n'
+$EGREP $OPT '^\+ *(const *|)static' $*| $EGREP --color=always '[^=]= *(0|NULL)[^0-9a-zA-Z]'> $TMP && printf '\nuseless 0 init\n'
 cat $TMP
 hiegrep '# *ifdef * (HAVE|CONFIG)_' 'ifdefs that should be #if' $*
 
@@ -89,7 +91,7 @@ hiegrep2 '\.sample_fmts *= *\(' 'const' 'missing const for sample_fmts array' $*
 hiegrep2 '\.supported_framerates *= *\(' 'const' 'missing const for supported_framerates array' $*
 hiegrep2 '\.channel_layouts *= *\(' 'const' 'missing const for channel_layouts array' $*
 
-#egrep $OPT '^\+.*const ' $*| grep -v 'static'> $TMP && printf '\nnon static const\n'
+#$EGREP $OPT '^\+.*const ' $*| $GREP -v 'static'> $TMP && printf '\nnon static const\n'
 #cat $TMP
 
 hiegrep2 "$ERE_TYPES" '(static|av_|ff_|typedef|:\+[^a-zA-Z_])' 'Non static with no ff_/av_ prefix' $*
@@ -104,58 +106,58 @@ hiegrep ':\+  *{ *$' '{ should be on the same line as the related previous state
 
 
 rm $TMP
-for i in `grep -H '^+.*@param' $*| sed 's/^\([^:]*\):.*@param\(\[.*\]\|\) *\([a-zA-Z0-9_]*\) .*$/\1:\3/'` ; do
+for i in `$GREP -H '^+.*@param' $*| sed 's/^\([^:]*\):.*@param\(\[.*\]\|\) *\([a-zA-Z0-9_]*\) .*$/\1:\3/'` ; do
     doxpar=`echo $i | sed 's/^.*:\(.*\)$/\1/'`
     file=`echo $i | sed 's/^\([^:]*\):.*$/\1/'`
-    grep " *$doxpar *[),]" $file | grep -v '@param' >/dev/null || grep --color=always "@param *$doxpar" $file >>$TMP
+    $GREP " *$doxpar *[),]" $file | $GREP -v '@param' >/dev/null || $GREP --color=always "@param *$doxpar" $file >>$TMP
 done
 if test -e $TMP ; then
     printf '\nmismatching doxy params\n'
     cat $TMP
 fi
 
-egrep -B2 $OPT '^(\+|) *('"$ERE_TYPES"'|# *define)' $* | egrep -A2 --color=always '(:|-)\+[^/]*/(\*([^*]|$)|/([^/]|$))' > $TMP && printf "\n Non doxy comments\n"
+$EGREP -B2 $OPT '^(\+|) *('"$ERE_TYPES"'|# *define)' $* | $EGREP -A2 --color=always '(:|-)\+[^/]*/(\*([^*]|$)|/([^/]|$))' > $TMP && printf "\n Non doxy comments\n"
 cat $TMP
 
 rm $TMP
 for i in \
-    `egrep -H '^\+ *'"$ERE_TYPES" $* |\
-    grep -v '(' | egrep -v '\Wgoto\W' |\
+    `$EGREP -H '^\+ *'"$ERE_TYPES" $* |\
+    $GREP -v '(' | $EGREP -v '\Wgoto\W' |\
     xargs -d '\n' -n 1 |\
-    grep -o '[* ][* ]*[a-zA-Z][0-9a-zA-Z_]* *[,;=]' |\
+    $GREP -o '[* ][* ]*[a-zA-Z][0-9a-zA-Z_]* *[,;=]' |\
     sed 's/.[* ]*\([a-zA-Z][0-9a-zA-Z_]*\) *[,;=]/\1/'` \
     ; do
-    echo $i | grep '^NULL$' && continue
-    egrep $i' *(\+|-|\*|/|\||&|%|)=[^=]' $* >/dev/null || echo "possibly never written:"$i >> $TMP
-    egrep '(=|\(|return).*'$i'(==|[^=])*$'    $* >/dev/null || echo "possibly never read   :"$i >> $TMP
-    egrep -o $i' *((\+|-|\*|/|\||&|%|)=[^=]|\+\+|--) *(0x|)[0-9]*(;|)'   $* |\
-           egrep -v $i' *= *(0x|)[0-9]{1,};'>/dev/null || echo "possibly constant     :"$i >> $TMP
+    echo $i | $GREP '^NULL$' && continue
+    $EGREP $i' *(\+|-|\*|/|\||&|%|)=[^=]' $* >/dev/null || echo "possibly never written:"$i >> $TMP
+    $EGREP '(=|\(|return).*'$i'(==|[^=])*$'    $* >/dev/null || echo "possibly never read   :"$i >> $TMP
+    $EGREP -o $i' *((\+|-|\*|/|\||&|%|)=[^=]|\+\+|--) *(0x|)[0-9]*(;|)'   $* |\
+           $EGREP -v $i' *= *(0x|)[0-9]{1,};'>/dev/null || echo "possibly constant     :"$i >> $TMP
 done
 if test -e $TMP ; then
     printf '\npossibly unused variables\n'
     cat $TMP
 fi
 
-grep '^+++ .*Changelog' $* >/dev/null || printf "\nMissing changelog entry (ignore if minor change)\n"
+$GREP '^+++ .*Changelog' $* >/dev/null || printf "\nMissing changelog entry (ignore if minor change)\n"
 
-cat $* | tr '\n' '@' | egrep --color=always -o '(fprintf|av_log|printf)\([^)]*\)[+ ;@]*\1'  >$TMP && printf "\nMergeable calls\n"
+cat $* | tr '\n' '@' | $EGREP --color=always -o '(fprintf|av_log|printf)\([^)]*\)[+ ;@]*\1'  >$TMP && printf "\nMergeable calls\n"
 cat $TMP | tr '@' '\n'
 
-cat $* | tr '\n' '@' | egrep --color=always -o '\+ *if *\( *([A-Za-z0-9_]*) *[<>]=? *[0-9]* *\) * \1 *= *[0-9]* *;[ @\\+]*else *if *\( *\1 *[<>]=? *[0-9]* *\) *\1 *= *[0-9]* *;'  >$TMP && printf "\nav_clip / av_clip_uint8 / av_clip_int16 / ...\n"
+cat $* | tr '\n' '@' | $EGREP --color=always -o '\+ *if *\( *([A-Za-z0-9_]*) *[<>]=? *[0-9]* *\) * \1 *= *[0-9]* *;[ @\\+]*else *if *\( *\1 *[<>]=? *[0-9]* *\) *\1 *= *[0-9]* *;'  >$TMP && printf "\nav_clip / av_clip_uint8 / av_clip_int16 / ...\n"
 cat $TMP | tr '@' '\n'
 
-cat $* | tr '\n' '@' | egrep --color=always -o '\+ *if *\( *([A-Za-z0-9_]*) *[<>]=? *([A-Za-z0-9_]*) *\)[ @\\+]*(\1|\2) *= *(\1|\2) *;'  >$TMP && printf "\nFFMIN/FFMAX\n"
+cat $* | tr '\n' '@' | $EGREP --color=always -o '\+ *if *\( *([A-Za-z0-9_]*) *[<>]=? *([A-Za-z0-9_]*) *\)[ @\\+]*(\1|\2) *= *(\1|\2) *;'  >$TMP && printf "\nFFMIN/FFMAX\n"
 cat $TMP | tr '@' '\n'
 
-cat $* | tr '\n' '@' | egrep --color=always -o '\+ *if *\( *([A-Za-z0-9_]*) *\)[ @\\+]*av_free(p|) *\( *(&|) *\1[^-.]'  >$TMP && printf "\nav_free(NULL) is safe\n"
+cat $* | tr '\n' '@' | $EGREP --color=always -o '\+ *if *\( *([A-Za-z0-9_]*) *\)[ @\\+]*av_free(p|) *\( *(&|) *\1[^-.]'  >$TMP && printf "\nav_free(NULL) is safe\n"
 cat $TMP | tr '@' '\n'
 
-cat $* | tr '\n' '@' | egrep --color=always -o '[^a-zA-Z0-9_]([a-zA-Z0-9_]*) *= *av_malloc *\([^)]*\)[ @;\\+]*memset *\( *\1'  >$TMP && printf "\nav_mallocz()\n"
+cat $* | tr '\n' '@' | $EGREP --color=always -o '[^a-zA-Z0-9_]([a-zA-Z0-9_]*) *= *av_malloc *\([^)]*\)[ @;\\+]*memset *\( *\1'  >$TMP && printf "\nav_mallocz()\n"
 cat $TMP | tr '@' '\n'
 
 
 # doesnt work
-#cat $* | tr '\n' '@' | egrep -o '[^a-zA-Z_0-9]([a-zA-Z][a-zA-Z_0-9]*) *=[^=].*\1' | egrep -o '[^a-zA-Z_0-9]([a-zA-Z][a-zA-Z_0-9]*) *=[^=].*\1 *=[^=]'  >$TMP && printf "\nPossibly written 2x before read\n"
+#cat $* | tr '\n' '@' | $EGREP -o '[^a-zA-Z_0-9]([a-zA-Z][a-zA-Z_0-9]*) *=[^=].*\1' | $EGREP -o '[^a-zA-Z_0-9]([a-zA-Z][a-zA-Z_0-9]*) *=[^=].*\1 *=[^=]'  >$TMP && printf "\nPossibly written 2x before read\n"
 #cat $TMP | tr '@' '\n'
 
 exit
@@ -163,7 +165,7 @@ exit
 TODO/idea list:
 
 for all demuxers & muxers
-    grep for "avctx->priv_data"
+    $EGREP for "avctx->priv_data"
 
 vertical align =
 /* and * align

From 6252040e773c30efa21c743124d5e813320a0ec2 Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Tue, 19 Apr 2011 00:16:24 +0200
Subject: [PATCH 15/20] Replace `` by $() syntax in shell scripts.

$() is easier to nest and POSIX, which we require in other places.
---
 tools/clean-diff |  2 +-
 tools/patcheck   | 14 +++++++-------
 2 files changed, 8 insertions(+), 8 deletions(-)

diff --git a/tools/clean-diff b/tools/clean-diff
index 98e26a79f9a42..4600702b10fb6 100755
--- a/tools/clean-diff
+++ b/tools/clean-diff
@@ -1,6 +1,6 @@
 #!/bin/sh
 sed '/^+[^+]/!s/	/TaBBaT/g' |\
- expand -t `seq -s , 9 8 200` |\
+ expand -t $(seq -s , 9 8 200) |\
  sed 's/TaBBaT/	/g' |\
  sed '/^+[^+]/s/ * $//' |\
  tr -d '\015' |\
diff --git a/tools/patcheck b/tools/patcheck
index d81a48937fa39..a8c0cc8cf7be7 100755
--- a/tools/patcheck
+++ b/tools/patcheck
@@ -12,7 +12,7 @@ GREP=grep
 EGREP=egrep
 TMP=patcheck.tmp
 OPT="-nH"
-#FILES=`$GREP '^+++' $* | sed 's/+++ //g'`
+#FILES=$($GREP '^+++' $* | sed 's/+++ //g')
 
 echo patCHeck 1e10.0
 echo This tool is intended to help a human check/review patches it is very far from
@@ -43,7 +43,7 @@ hiegrep2(){
 }
 
 hiegrep '[[:space:]]$'    'trailing whitespace' $*
-hiegrep "`echo x | tr 'x' '\t'`"         'tabs' $*
+hiegrep "$(echo x | tr 'x' '\t')"         'tabs' $*
 #hiegrep ':\+$'          'Empty lines' $*
 hiegrep ';;'              'double ;' $*
 hiegrep2 '\b_[a-zA-Z0-9_]{1,}' '__(asm|attribute)([^a-zA-Z0-9]|$)' 'reserved identifer' $*
@@ -106,9 +106,9 @@ hiegrep ':\+  *{ *$' '{ should be on the same line as the related previous state
 
 
 rm $TMP
-for i in `$GREP -H '^+.*@param' $*| sed 's/^\([^:]*\):.*@param\(\[.*\]\|\) *\([a-zA-Z0-9_]*\) .*$/\1:\3/'` ; do
-    doxpar=`echo $i | sed 's/^.*:\(.*\)$/\1/'`
-    file=`echo $i | sed 's/^\([^:]*\):.*$/\1/'`
+for i in $($GREP -H '^+.*@param' $*| sed 's/^\([^:]*\):.*@param\(\[.*\]\|\) *\([a-zA-Z0-9_]*\) .*$/\1:\3/') ; do
+    doxpar=$(echo $i | sed 's/^.*:\(.*\)$/\1/')
+    file=$(echo $i | sed 's/^\([^:]*\):.*$/\1/')
     $GREP " *$doxpar *[),]" $file | $GREP -v '@param' >/dev/null || $GREP --color=always "@param *$doxpar" $file >>$TMP
 done
 if test -e $TMP ; then
@@ -121,11 +121,11 @@ cat $TMP
 
 rm $TMP
 for i in \
-    `$EGREP -H '^\+ *'"$ERE_TYPES" $* |\
+    $($EGREP -H '^\+ *'"$ERE_TYPES" $*  |\
     $GREP -v '(' | $EGREP -v '\Wgoto\W' |\
     xargs -d '\n' -n 1 |\
     $GREP -o '[* ][* ]*[a-zA-Z][0-9a-zA-Z_]* *[,;=]' |\
-    sed 's/.[* ]*\([a-zA-Z][0-9a-zA-Z_]*\) *[,;=]/\1/'` \
+    sed 's/.[* ]*\([a-zA-Z][0-9a-zA-Z_]*\) *[,;=]/\1/') \
     ; do
     echo $i | $GREP '^NULL$' && continue
     $EGREP $i' *(\+|-|\*|/|\||&|%|)=[^=]' $* >/dev/null || echo "possibly never written:"$i >> $TMP

From 43fb279f5643225799841f512d2dad4690186165 Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Sat, 23 Apr 2011 16:39:03 +0200
Subject: [PATCH 16/20] Replace more FFmpeg instances by Libav or ffmpeg.

---
 doc/general.texi      | 2 +-
 doc/issue_tracker.txt | 2 +-
 ffserver.c            | 2 +-
 libavcodec/avcodec.h  | 6 +++---
 libavcodec/dca.c      | 2 +-
 5 files changed, 7 insertions(+), 7 deletions(-)

diff --git a/doc/general.texi b/doc/general.texi
index 303159eba7d3f..e68e83e5f230e 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -268,7 +268,7 @@ library:
 
 @section Image Formats
 
-FFmpeg can read and write images for each frame of a video sequence. The
+Libav can read and write images for each frame of a video sequence. The
 following image formats are supported:
 
 @multitable @columnfractions .4 .1 .1 .4
diff --git a/doc/issue_tracker.txt b/doc/issue_tracker.txt
index 5a15a9b665963..e5733ec9d5ca7 100644
--- a/doc/issue_tracker.txt
+++ b/doc/issue_tracker.txt
@@ -39,7 +39,7 @@ NOTE: issue = (bug report || patch || feature request)
 Type:
 -----
 bug
-    An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
+    An error, flaw, mistake, failure, or fault in ffmpeg or libav* that
     prevents it from behaving as intended.
 
 feature request
diff --git a/ffserver.c b/ffserver.c
index d341182a4f30a..9cb65da0179a2 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -2265,7 +2265,7 @@ static int http_prepare_data(HTTPContext *c)
 
         /*
          * HACK to avoid mpeg ps muxer to spit many underflow errors
-         * Default value from FFmpeg
+         * Default value from Libav
          * Try to set it use configuration option
          */
         c->fmt_ctx.preload   = (int)(0.5*AV_TIME_BASE);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 14f00bb5d0432..ec31f666d196f 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -960,7 +960,7 @@ typedef struct AVPanScan{
     int64_t reordered_opaque;\
 \
     /**\
-     * hardware accelerator private data (FFmpeg allocated)\
+     * hardware accelerator private data (Libav-allocated)\
      * - encoding: unused\
      * - decoding: Set by libavcodec\
      */\
@@ -2633,8 +2633,8 @@ typedef struct AVCodecContext {
      * Hardware accelerator context.
      * For some hardware accelerators, a global context needs to be
      * provided by the user. In that case, this holds display-dependent
-     * data FFmpeg cannot instantiate itself. Please refer to the
-     * FFmpeg HW accelerator documentation to know how to fill this
+     * data Libav cannot instantiate itself. Please refer to the
+     * Libav HW accelerator documentation to know how to fill this
      * is. e.g. for VA API, this is a struct vaapi_context.
      * - encoding: unused
      * - decoding: Set by user
diff --git a/libavcodec/dca.c b/libavcodec/dca.c
index e3c64660dbd45..525832f802f50 100644
--- a/libavcodec/dca.c
+++ b/libavcodec/dca.c
@@ -1798,7 +1798,7 @@ static int dca_decode_frame(AVCodecContext * avctx,
 
 
     /* There is nothing that prevents a dts frame to change channel configuration
-       but FFmpeg doesn't support that so only set the channels if it is previously
+       but Libav doesn't support that so only set the channels if it is previously
        unset. Ideally during the first probe for channels the crc should be checked
        and only set avctx->channels when the crc is ok. Right now the decoder could
        set the channels based on a broken first frame.*/

From 3a50894eaaad3d5a7ec9bad99c08ecb8305c51c1 Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Sat, 23 Apr 2011 16:40:01 +0200
Subject: [PATCH 17/20] Refer to ff* tools by their lowercase names.

---
 doc/faq.texi      |  4 ++--
 doc/ffmpeg.texi   | 26 ++++++++++++--------------
 doc/ffprobe.texi  |  6 +++---
 doc/ffserver.texi | 12 ++++++------
 4 files changed, 23 insertions(+), 25 deletions(-)

diff --git a/doc/faq.texi b/doc/faq.texi
index a29af1e1adf59..3d7a275f89827 100644
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -280,7 +280,7 @@ Just create an "input.avs" text file with this single line ...
 @example
   DirectShowSource("C:\path to your file\yourfile.asf")
 @end example
-... and then feed that text file to FFmpeg:
+... and then feed that text file to ffmpeg:
 @example
   ffmpeg -i input.avs
 @end example
@@ -348,7 +348,7 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
 @end example
 
-@section FFmpeg does not adhere to the -maxrate setting, some frames are bigger than maxrate/fps.
+@section The ffmpeg program does not respect the -maxrate setting, some frames are bigger than maxrate/fps.
 
 Read the MPEG spec about video buffer verifier.
 
diff --git a/doc/ffmpeg.texi b/doc/ffmpeg.texi
index 38da9186e62da..62282b2dd0eb6 100644
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -22,17 +22,15 @@ ffmpeg [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{
 @chapter Description
 @c man begin DESCRIPTION
 
-FFmpeg is a very fast video and audio converter. It can also grab from
-a live audio/video source.
+ffmpeg is a very fast video and audio converter that can also grab from
+a live audio/video source. It can also convert between arbitrary sample
+rates and resize video on the fly with a high quality polyphase filter.
 
 The command line interface is designed to be intuitive, in the sense
-that FFmpeg tries to figure out all parameters that can possibly be
+that ffmpeg tries to figure out all parameters that can possibly be
 derived automatically. You usually only have to specify the target
 bitrate you want.
 
-FFmpeg can also convert from any sample rate to any other, and resize
-video on the fly with a high quality polyphase filter.
-
 As a general rule, options are applied to the next specified
 file. Therefore, order is important, and you can have the same
 option on the command line multiple times. Each occurrence is
@@ -61,7 +59,7 @@ ffmpeg -r 1 -i input.m2v -r 24 output.avi
 
 The format option may be needed for raw input files.
 
-By default, FFmpeg tries to convert as losslessly as possible: It
+By default ffmpeg tries to convert as losslessly as possible: It
 uses the same audio and video parameters for the outputs as the one
 specified for the inputs.
 
@@ -486,7 +484,7 @@ Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).
 macroblock decision
 @table @samp
 @item 0
-FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).
+FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in ffmpeg).
 @item 1
 FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.
 @item 2
@@ -868,22 +866,22 @@ It allows almost lossless encoding.
 
 @section Video and Audio grabbing
 
-FFmpeg can grab video and audio from devices given that you specify the input
-format and device.
+If you specify the input format and device then ffmpeg can grab video
+and audio directly.
 
 @example
 ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg
 @end example
 
 Note that you must activate the right video source and channel before
-launching FFmpeg with any TV viewer such as xawtv
+launching ffmpeg with any TV viewer such as xawtv
 (@url{http://linux.bytesex.org/xawtv/}) by Gerd Knorr. You also
 have to set the audio recording levels correctly with a
 standard mixer.
 
 @section X11 grabbing
 
-FFmpeg can grab the X11 display.
+Grab the X11 display with ffmpeg via
 
 @example
 ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg
@@ -901,7 +899,7 @@ variable. 10 is the x-offset and 20 the y-offset for the grabbing.
 
 @section Video and Audio file format conversion
 
-FFmpeg can use any supported file format and protocol as input:
+Any supported file format and protocol can serve as input to ffmpeg:
 
 Examples:
 @itemize
@@ -921,7 +919,7 @@ It will use the files:
 The Y files use twice the resolution of the U and V files. They are
 raw files, without header. They can be generated by all decent video
 decoders. You must specify the size of the image with the @option{-s} option
-if FFmpeg cannot guess it.
+if ffmpeg cannot guess it.
 
 @item
 You can input from a raw YUV420P file:
diff --git a/doc/ffprobe.texi b/doc/ffprobe.texi
index b775f558ddb68..15090220f3151 100644
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -22,7 +22,7 @@ ffprobe [options] [@file{input_file}]
 @chapter Description
 @c man begin DESCRIPTION
 
-FFprobe gathers information from multimedia streams and prints it in
+ffprobe gathers information from multimedia streams and prints it in
 human- and machine-readable fashion.
 
 For example it can be used to check the format of the container used
@@ -33,7 +33,7 @@ If a filename is specified in input, ffprobe will try to open and
 probe the file content. If the file cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.
 
-FFprobe may be employed both as a standalone application or in
+ffprobe may be employed both as a standalone application or in
 combination with a textual filter, which may perform more
 sophisticated processing, e.g. statistical processing or plotting.
 
@@ -41,7 +41,7 @@ Options are used to list some of the formats supported by ffprobe or
 for specifying which information to display, and for setting how
 ffprobe will show it.
 
-FFprobe output is designed to be easily parsable by a textual filter,
+ffprobe output is designed to be easily parsable by a textual filter,
 and consists of one or more sections of the form:
 @example
 [SECTION]
diff --git a/doc/ffserver.texi b/doc/ffserver.texi
index 021b237de588a..e5c9d40235913 100644
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -22,12 +22,12 @@ ffserver [options]
 @chapter Description
 @c man begin DESCRIPTION
 
-FFserver is a streaming server for both audio and video. It supports
+ffserver is a streaming server for both audio and video. It supports
 several live feeds, streaming from files and time shifting on live feeds
 (you can seek to positions in the past on each live feed, provided you
 specify a big enough feed storage in ffserver.conf).
 
-FFserver runs in daemon mode by default; that is, it puts itself in
+ffserver runs in daemon mode by default; that is, it puts itself in
 the background and detaches from its TTY, unless it is launched in
 debug mode or a NoDaemon option is specified in the configuration
 file.
@@ -39,7 +39,7 @@ information.
 
 @section How does it work?
 
-FFserver receives prerecorded files or FFM streams from some ffmpeg
+ffserver receives prerecorded files or FFM streams from some ffmpeg
 instance as input, then streams them over RTP/RTSP/HTTP.
 
 An ffserver instance will listen on some port as specified in the
@@ -57,7 +57,7 @@ file.
 
 @section Status stream
 
-FFserver supports an HTTP interface which exposes the current status
+ffserver supports an HTTP interface which exposes the current status
 of the server.
 
 Simply point your browser to the address of the special status stream
@@ -249,8 +249,8 @@ For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
 Use @file{configfile} instead of @file{/etc/ffserver.conf}.
 @item -n
 Enable no-launch mode. This option disables all the Launch directives
-within the various <Stream> sections. FFserver will not launch any
-ffmpeg instance, so you will have to launch them manually.
+within the various <Stream> sections. Since ffserver will not launch
+any ffmpeg instances, you will have to launch them manually.
 @item -d
 Enable debug mode. This option increases log verbosity, directs log
 messages to stdout and causes ffserver to run in the foreground

From 89b503b55f2b2713f1c3cc8981102c1a7b663281 Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Sat, 23 Apr 2011 15:19:17 +0200
Subject: [PATCH 18/20] Lowercase all ff* program names.

---
 doc/ffmpeg.texi   | 6 +++---
 doc/ffplay.texi   | 4 ++--
 doc/ffprobe.texi  | 6 +++---
 doc/ffserver.texi | 6 +++---
 ffmpeg.c          | 4 ++--
 ffplay.c          | 4 ++--
 ffprobe.c         | 4 ++--
 ffserver.c        | 2 +-
 8 files changed, 18 insertions(+), 18 deletions(-)

diff --git a/doc/ffmpeg.texi b/doc/ffmpeg.texi
index 62282b2dd0eb6..0d12f65f02aea 100644
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle FFmpeg Documentation
+@settitle ffmpeg Documentation
 @titlepage
-@center @titlefont{FFmpeg Documentation}
+@center @titlefont{ffmpeg Documentation}
 @end titlepage
 
 @top
@@ -1046,7 +1046,7 @@ file to which you want to add them.
 @ignore
 
 @setfilename ffmpeg
-@settitle FFmpeg video converter
+@settitle ffmpeg video converter
 
 @c man begin SEEALSO
 ffplay(1), ffprobe(1), ffserver(1) and the Libav HTML documentation
diff --git a/doc/ffplay.texi b/doc/ffplay.texi
index 6199e6d2693ee..e5de3d2d6de33 100644
--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle FFplay Documentation
+@settitle ffplay Documentation
 @titlepage
-@center @titlefont{FFplay Documentation}
+@center @titlefont{ffplay Documentation}
 @end titlepage
 
 @top
diff --git a/doc/ffprobe.texi b/doc/ffprobe.texi
index 15090220f3151..5e856e66458e9 100644
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle FFprobe Documentation
+@settitle ffprobe Documentation
 @titlepage
-@center @titlefont{FFprobe Documentation}
+@center @titlefont{ffprobe Documentation}
 @end titlepage
 
 @top
@@ -119,7 +119,7 @@ with name "STREAM".
 @ignore
 
 @setfilename ffprobe
-@settitle FFprobe media prober
+@settitle ffprobe media prober
 
 @c man begin SEEALSO
 ffmpeg(1), ffplay(1), ffserver(1) and the Libav HTML documentation
diff --git a/doc/ffserver.texi b/doc/ffserver.texi
index e5c9d40235913..d247016bbcd65 100644
--- a/doc/ffserver.texi
+++ b/doc/ffserver.texi
@@ -1,8 +1,8 @@
 \input texinfo @c -*- texinfo -*-
 
-@settitle FFserver Documentation
+@settitle ffserver Documentation
 @titlepage
-@center @titlefont{FFserver Documentation}
+@center @titlefont{ffserver Documentation}
 @end titlepage
 
 @top
@@ -261,7 +261,7 @@ rather than as a daemon.
 @ignore
 
 @setfilename ffserver
-@settitle FFserver video server
+@settitle ffserver video server
 
 @c man begin SEEALSO
 
diff --git a/ffmpeg.c b/ffmpeg.c
index 886d5da7de27a..d54785d5978af 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -1,5 +1,5 @@
 /*
- * FFmpeg main
+ * ffmpeg main
  * Copyright (c) 2000-2003 Fabrice Bellard
  *
  * This file is part of Libav.
@@ -78,7 +78,7 @@
 
 #include "libavutil/avassert.h"
 
-const char program_name[] = "FFmpeg";
+const char program_name[] = "ffmpeg";
 const int program_birth_year = 2000;
 
 /* select an input stream for an output stream */
diff --git a/ffplay.c b/ffplay.c
index 766d5caf9c562..18010ef07600a 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -1,5 +1,5 @@
 /*
- * FFplay : Simple Media Player based on the Libav libraries
+ * ffplay : Simple Media Player based on the Libav libraries
  * Copyright (c) 2003 Fabrice Bellard
  *
  * This file is part of Libav.
@@ -55,7 +55,7 @@
 #include <unistd.h>
 #include <assert.h>
 
-const char program_name[] = "FFplay";
+const char program_name[] = "ffplay";
 const int program_birth_year = 2003;
 
 //#define DEBUG
diff --git a/ffprobe.c b/ffprobe.c
index cc3699bf9b887..6e26c115a7b30 100644
--- a/ffprobe.c
+++ b/ffprobe.c
@@ -1,5 +1,5 @@
 /*
- * FFprobe : Simple Media Prober based on the Libav libraries
+ * ffprobe : Simple Media Prober based on the Libav libraries
  * Copyright (c) 2007-2010 Stefano Sabatini
  *
  * This file is part of Libav.
@@ -28,7 +28,7 @@
 #include "libavdevice/avdevice.h"
 #include "cmdutils.h"
 
-const char program_name[] = "FFprobe";
+const char program_name[] = "ffprobe";
 const int program_birth_year = 2007;
 
 static int do_show_format  = 0;
diff --git a/ffserver.c b/ffserver.c
index 9cb65da0179a2..829eb2ef7efa6 100644
--- a/ffserver.c
+++ b/ffserver.c
@@ -59,7 +59,7 @@
 
 #include "cmdutils.h"
 
-const char program_name[] = "FFserver";
+const char program_name[] = "ffserver";
 const int program_birth_year = 2000;
 
 static const OptionDef options[];

From 9261e6cf3fe579fa02a96761c8e81a77bb3d8b2e Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Martin=20Storsj=C3=B6?= <martin@martin.st>
Date: Wed, 20 Apr 2011 15:36:37 +0300
Subject: [PATCH 19/20] rtp: Rename the open/close functions to alloc/free
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

This avoids clashes if we internally want to override the global
open function.

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 libavformat/rdt.c          | 4 ++--
 libavformat/rtpdec.h       | 4 ++--
 libavformat/rtpdec_amr.c   | 8 ++++----
 libavformat/rtpdec_asf.c   | 4 ++--
 libavformat/rtpdec_h264.c  | 4 ++--
 libavformat/rtpdec_latm.c  | 4 ++--
 libavformat/rtpdec_mpeg4.c | 8 ++++----
 libavformat/rtpdec_qcelp.c | 4 ++--
 libavformat/rtpdec_qdm2.c  | 4 ++--
 libavformat/rtpdec_qt.c    | 4 ++--
 libavformat/rtpdec_svq3.c  | 4 ++--
 libavformat/rtpdec_vp8.c   | 4 ++--
 libavformat/rtpdec_xiph.c  | 8 ++++----
 libavformat/rtsp.c         | 6 +++---
 14 files changed, 35 insertions(+), 35 deletions(-)

diff --git a/libavformat/rdt.c b/libavformat/rdt.c
index dfb31d15f0723..bc3c17bd0d234 100644
--- a/libavformat/rdt.c
+++ b/libavformat/rdt.c
@@ -551,8 +551,8 @@ static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
     .codec_type       = t, \
     .codec_id         = CODEC_ID_NONE, \
     .parse_sdp_a_line = rdt_parse_sdp_line, \
-    .open             = rdt_new_context, \
-    .close            = rdt_free_context, \
+    .alloc            = rdt_new_context, \
+    .free             = rdt_free_context, \
     .parse_packet     = rdt_parse_packet \
 }
 
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
index da53efc1f446a..a4d21aa688d06 100644
--- a/libavformat/rtpdec.h
+++ b/libavformat/rtpdec.h
@@ -126,8 +126,8 @@ struct RTPDynamicProtocolHandler_s {
                              int st_index,
                              PayloadContext *priv_data,
                              const char *line); ///< Parse the a= line from the sdp field
-    PayloadContext *(*open) (void); ///< allocate any data needed by the rtp parsing for this dynamic data.
-    void (*close)(PayloadContext *protocol_data); ///< free any data needed by the rtp parsing for this dynamic data.
+    PayloadContext *(*alloc) (void); ///< allocate any data needed by the rtp parsing for this dynamic data.
+    void (*free)(PayloadContext *protocol_data); ///< free any data needed by the rtp parsing for this dynamic data.
     DynamicPayloadPacketHandlerProc parse_packet; ///< parse handler for this dynamic packet.
 
     struct RTPDynamicProtocolHandler_s *next;
diff --git a/libavformat/rtpdec_amr.c b/libavformat/rtpdec_amr.c
index 802e7c1ce8cc2..b7ff3aa05ea44 100644
--- a/libavformat/rtpdec_amr.c
+++ b/libavformat/rtpdec_amr.c
@@ -191,8 +191,8 @@ RTPDynamicProtocolHandler ff_amr_nb_dynamic_handler = {
     .codec_type       = AVMEDIA_TYPE_AUDIO,
     .codec_id         = CODEC_ID_AMR_NB,
     .parse_sdp_a_line = amr_parse_sdp_line,
-    .open             = amr_new_context,
-    .close            = amr_free_context,
+    .alloc            = amr_new_context,
+    .free             = amr_free_context,
     .parse_packet     = amr_handle_packet,
 };
 
@@ -201,8 +201,8 @@ RTPDynamicProtocolHandler ff_amr_wb_dynamic_handler = {
     .codec_type       = AVMEDIA_TYPE_AUDIO,
     .codec_id         = CODEC_ID_AMR_WB,
     .parse_sdp_a_line = amr_parse_sdp_line,
-    .open             = amr_new_context,
-    .close            = amr_free_context,
+    .alloc            = amr_new_context,
+    .free             = amr_free_context,
     .parse_packet     = amr_handle_packet,
 };
 
diff --git a/libavformat/rtpdec_asf.c b/libavformat/rtpdec_asf.c
index a8326cf123ae6..ecacc0eb4e5c8 100644
--- a/libavformat/rtpdec_asf.c
+++ b/libavformat/rtpdec_asf.c
@@ -286,8 +286,8 @@ RTPDynamicProtocolHandler ff_ms_rtp_ ## n ## _handler = { \
     .codec_type       = t, \
     .codec_id         = CODEC_ID_NONE, \
     .parse_sdp_a_line = asfrtp_parse_sdp_line, \
-    .open             = asfrtp_new_context, \
-    .close            = asfrtp_free_context, \
+    .alloc            = asfrtp_new_context, \
+    .free             = asfrtp_free_context, \
     .parse_packet     = asfrtp_parse_packet,   \
 }
 
diff --git a/libavformat/rtpdec_h264.c b/libavformat/rtpdec_h264.c
index 4c9b8ba0bdc8e..effdc1fe66c49 100644
--- a/libavformat/rtpdec_h264.c
+++ b/libavformat/rtpdec_h264.c
@@ -398,7 +398,7 @@ RTPDynamicProtocolHandler ff_h264_dynamic_handler = {
     .codec_type       = AVMEDIA_TYPE_VIDEO,
     .codec_id         = CODEC_ID_H264,
     .parse_sdp_a_line = parse_h264_sdp_line,
-    .open             = h264_new_context,
-    .close            = h264_free_context,
+    .alloc            = h264_new_context,
+    .free             = h264_free_context,
     .parse_packet     = h264_handle_packet
 };
diff --git a/libavformat/rtpdec_latm.c b/libavformat/rtpdec_latm.c
index 5bf4c1964e2f1..bde34b7ab9633 100644
--- a/libavformat/rtpdec_latm.c
+++ b/libavformat/rtpdec_latm.c
@@ -181,7 +181,7 @@ RTPDynamicProtocolHandler ff_mp4a_latm_dynamic_handler = {
     .codec_type         = AVMEDIA_TYPE_AUDIO,
     .codec_id           = CODEC_ID_AAC,
     .parse_sdp_a_line   = latm_parse_sdp_line,
-    .open               = latm_new_context,
-    .close              = latm_free_context,
+    .alloc              = latm_new_context,
+    .free               = latm_free_context,
     .parse_packet       = latm_parse_packet
 };
diff --git a/libavformat/rtpdec_mpeg4.c b/libavformat/rtpdec_mpeg4.c
index 9f2fcb3dbca56..7a63cc315b33c 100644
--- a/libavformat/rtpdec_mpeg4.c
+++ b/libavformat/rtpdec_mpeg4.c
@@ -235,8 +235,8 @@ RTPDynamicProtocolHandler ff_mp4v_es_dynamic_handler = {
     .codec_type         = AVMEDIA_TYPE_VIDEO,
     .codec_id           = CODEC_ID_MPEG4,
     .parse_sdp_a_line   = parse_sdp_line,
-    .open               = NULL,
-    .close              = NULL,
+    .alloc              = NULL,
+    .free               = NULL,
     .parse_packet       = NULL
 };
 
@@ -245,7 +245,7 @@ RTPDynamicProtocolHandler ff_mpeg4_generic_dynamic_handler = {
     .codec_type         = AVMEDIA_TYPE_AUDIO,
     .codec_id           = CODEC_ID_AAC,
     .parse_sdp_a_line   = parse_sdp_line,
-    .open               = new_context,
-    .close              = free_context,
+    .alloc              = new_context,
+    .free               = free_context,
     .parse_packet       = aac_parse_packet
 };
diff --git a/libavformat/rtpdec_qcelp.c b/libavformat/rtpdec_qcelp.c
index cc16ec1f7f3d5..325683c3961f3 100644
--- a/libavformat/rtpdec_qcelp.c
+++ b/libavformat/rtpdec_qcelp.c
@@ -223,7 +223,7 @@ RTPDynamicProtocolHandler ff_qcelp_dynamic_handler = {
     .codec_type         = AVMEDIA_TYPE_AUDIO,
     .codec_id           = CODEC_ID_QCELP,
     .static_payload_id  = 12,
-    .open               = qcelp_new_context,
-    .close              = qcelp_free_context,
+    .alloc              = qcelp_new_context,
+    .free               = qcelp_free_context,
     .parse_packet       = qcelp_parse_packet
 };
diff --git a/libavformat/rtpdec_qdm2.c b/libavformat/rtpdec_qdm2.c
index 0d744be8200d4..7f5f0771bcfd7 100644
--- a/libavformat/rtpdec_qdm2.c
+++ b/libavformat/rtpdec_qdm2.c
@@ -309,7 +309,7 @@ RTPDynamicProtocolHandler ff_qdm2_dynamic_handler = {
     .enc_name         = "X-QDM",
     .codec_type       = AVMEDIA_TYPE_AUDIO,
     .codec_id         = CODEC_ID_NONE,
-    .open             = qdm2_extradata_new,
-    .close            = qdm2_extradata_free,
+    .alloc            = qdm2_extradata_new,
+    .free             = qdm2_extradata_free,
     .parse_packet     = qdm2_parse_packet,
 };
diff --git a/libavformat/rtpdec_qt.c b/libavformat/rtpdec_qt.c
index a295ba7381c5f..8dd2968398398 100644
--- a/libavformat/rtpdec_qt.c
+++ b/libavformat/rtpdec_qt.c
@@ -244,8 +244,8 @@ RTPDynamicProtocolHandler ff_ ## m ## _rtp_ ## n ## _handler = { \
     .enc_name         = s, \
     .codec_type       = t, \
     .codec_id         = CODEC_ID_NONE, \
-    .open             = qt_rtp_new,    \
-    .close            = qt_rtp_free,   \
+    .alloc            = qt_rtp_new,    \
+    .free             = qt_rtp_free,   \
     .parse_packet     = qt_rtp_parse_packet, \
 }
 
diff --git a/libavformat/rtpdec_svq3.c b/libavformat/rtpdec_svq3.c
index 8c0926212cc29..331434245ebe6 100644
--- a/libavformat/rtpdec_svq3.c
+++ b/libavformat/rtpdec_svq3.c
@@ -128,7 +128,7 @@ RTPDynamicProtocolHandler ff_svq3_dynamic_handler = {
     .enc_name         = "X-SV3V-ES",
     .codec_type       = AVMEDIA_TYPE_VIDEO,
     .codec_id         = CODEC_ID_NONE,      // see if (config_packet) above
-    .open             = svq3_extradata_new,
-    .close            = svq3_extradata_free,
+    .alloc            = svq3_extradata_new,
+    .free             = svq3_extradata_free,
     .parse_packet     = svq3_parse_packet,
 };
diff --git a/libavformat/rtpdec_vp8.c b/libavformat/rtpdec_vp8.c
index 9e50cc44e1728..026728ec9dee5 100644
--- a/libavformat/rtpdec_vp8.c
+++ b/libavformat/rtpdec_vp8.c
@@ -148,7 +148,7 @@ RTPDynamicProtocolHandler ff_vp8_dynamic_handler = {
     .enc_name       = "VP8",
     .codec_type     = AVMEDIA_TYPE_VIDEO,
     .codec_id       = CODEC_ID_VP8,
-    .open           = vp8_new_context,
-    .close          = vp8_free_context,
+    .alloc          = vp8_new_context,
+    .free           = vp8_free_context,
     .parse_packet   = vp8_handle_packet,
 };
diff --git a/libavformat/rtpdec_xiph.c b/libavformat/rtpdec_xiph.c
index 4d8e83481346a..a7f36ef59a5e7 100644
--- a/libavformat/rtpdec_xiph.c
+++ b/libavformat/rtpdec_xiph.c
@@ -389,8 +389,8 @@ RTPDynamicProtocolHandler ff_theora_dynamic_handler = {
     .codec_type       = AVMEDIA_TYPE_VIDEO,
     .codec_id         = CODEC_ID_THEORA,
     .parse_sdp_a_line = xiph_parse_sdp_line,
-    .open             = xiph_new_context,
-    .close            = xiph_free_context,
+    .alloc            = xiph_new_context,
+    .free             = xiph_free_context,
     .parse_packet     = xiph_handle_packet
 };
 
@@ -399,7 +399,7 @@ RTPDynamicProtocolHandler ff_vorbis_dynamic_handler = {
     .codec_type       = AVMEDIA_TYPE_AUDIO,
     .codec_id         = CODEC_ID_VORBIS,
     .parse_sdp_a_line = xiph_parse_sdp_line,
-    .open             = xiph_new_context,
-    .close            = xiph_free_context,
+    .alloc            = xiph_new_context,
+    .free             = xiph_free_context,
     .parse_packet     = xiph_handle_packet
 };
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 58f7ddc1a66c3..14111e69fd774 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -132,8 +132,8 @@ static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
         return;
     codec->codec_id          = handler->codec_id;
     rtsp_st->dynamic_handler = handler;
-    if (handler->open)
-        rtsp_st->dynamic_protocol_context = handler->open();
+    if (handler->alloc)
+        rtsp_st->dynamic_protocol_context = handler->alloc();
 }
 
 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
@@ -526,7 +526,7 @@ void ff_rtsp_close_streams(AVFormatContext *s)
         rtsp_st = rt->rtsp_streams[i];
         if (rtsp_st) {
             if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
-                rtsp_st->dynamic_handler->close(
+                rtsp_st->dynamic_handler->free(
                     rtsp_st->dynamic_protocol_context);
             av_free(rtsp_st);
         }

From b1ac139d89b9fc55b70ad3411af2f75fe8b17805 Mon Sep 17 00:00:00 2001
From: Kirill Gavrilov <gavr.mail@gmail.com>
Date: Wed, 20 Apr 2011 14:36:44 +0300
Subject: [PATCH 20/20] Handle unicode file names on windows
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

All file names should be in UTF-8 within libavformat.

This is handled by mapping the open() function to an internal one
in os_support.h for windows.

fopen() could be overridden in the same way, but if that would be
used from ffmpeg.c, it would add a dependency on an ff prefixed
internal lavf function.

Signed-off-by: Martin Storsjö <martin@martin.st>
---
 cmdutils.c               | 63 ++++++++++++++++++++++++++++++++++++++++
 libavformat/os_support.c | 28 ++++++++++++++++++
 libavformat/os_support.h |  5 ++++
 libavformat/version.h    |  2 +-
 4 files changed, 97 insertions(+), 1 deletion(-)

diff --git a/cmdutils.c b/cmdutils.c
index f1cbd55373b5c..f25f61d4e5173 100644
--- a/cmdutils.c
+++ b/cmdutils.c
@@ -155,6 +155,66 @@ static const OptionDef* find_option(const OptionDef *po, const char *name){
     return po;
 }
 
+#if defined(_WIN32) && !defined(__MINGW32CE__)
+/* Will be leaked on exit */
+static char** win32_argv_utf8 = NULL;
+static int win32_argc = 0;
+
+/**
+ * Prepare command line arguments for executable.
+ * For Windows - perform wide-char to UTF-8 conversion.
+ * Input arguments should be main() function arguments.
+ * @param argc_ptr Arguments number (including executable)
+ * @param argv_ptr Arguments list.
+ */
+static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
+{
+    char *argstr_flat;
+    wchar_t **argv_w;
+    int i, buffsize = 0, offset = 0;
+
+    if (win32_argv_utf8) {
+        *argc_ptr = win32_argc;
+        *argv_ptr = win32_argv_utf8;
+        return;
+    }
+
+    win32_argc = 0;
+    argv_w = CommandLineToArgvW(GetCommandLineW(), &win32_argc);
+    if (win32_argc <= 0 || !argv_w)
+        return;
+
+    /* determine the UTF-8 buffer size (including NULL-termination symbols) */
+    for (i = 0; i < win32_argc; i++)
+        buffsize += WideCharToMultiByte(CP_UTF8, 0, argv_w[i], -1,
+                                        NULL, 0, NULL, NULL);
+
+    win32_argv_utf8 = av_mallocz(sizeof(char*) * (win32_argc + 1) + buffsize);
+    argstr_flat     = (char*)win32_argv_utf8 + sizeof(char*) * (win32_argc + 1);
+    if (win32_argv_utf8 == NULL) {
+        LocalFree(argv_w);
+        return;
+    }
+
+    for (i = 0; i < win32_argc; i++) {
+        win32_argv_utf8[i] = &argstr_flat[offset];
+        offset += WideCharToMultiByte(CP_UTF8, 0, argv_w[i], -1,
+                                      &argstr_flat[offset],
+                                      buffsize - offset, NULL, NULL);
+    }
+    win32_argv_utf8[i] = NULL;
+    LocalFree(argv_w);
+
+    *argc_ptr = win32_argc;
+    *argv_ptr = win32_argv_utf8;
+}
+#else
+static inline void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
+{
+    /* nothing to do */
+}
+#endif /* WIN32 && !__MINGW32CE__ */
+
 void parse_options(int argc, char **argv, const OptionDef *options,
                    void (* parse_arg_function)(const char*))
 {
@@ -162,6 +222,9 @@ void parse_options(int argc, char **argv, const OptionDef *options,
     int optindex, handleoptions=1;
     const OptionDef *po;
 
+    /* perform system-dependent conversions for arguments list */
+    prepare_app_arguments(&argc, &argv);
+
     /* parse options */
     optindex = 1;
     while (optindex < argc) {
diff --git a/libavformat/os_support.c b/libavformat/os_support.c
index 5a3a1bbfe014a..05577b7553f7d 100644
--- a/libavformat/os_support.c
+++ b/libavformat/os_support.c
@@ -28,6 +28,34 @@
 #include "avformat.h"
 #include "os_support.h"
 
+#if defined(_WIN32) && !defined(__MINGW32CE__)
+#include <windows.h>
+
+#undef open
+int ff_win32_open(const char *filename_utf8, int oflag, int pmode)
+{
+    int fd;
+    int num_chars;
+    wchar_t *filename_w;
+
+    /* convert UTF-8 to wide chars */
+    num_chars = MultiByteToWideChar(CP_UTF8, 0, filename_utf8, -1, NULL, 0);
+    if (num_chars <= 0)
+        return -1;
+    filename_w = av_mallocz(sizeof(wchar_t) * num_chars);
+    MultiByteToWideChar(CP_UTF8, 0, filename_utf8, -1, filename_w, num_chars);
+
+    fd = _wopen(filename_w, oflag, pmode);
+    av_freep(&filename_w);
+
+    /* filename maybe be in CP_ACP */
+    if (fd == -1 && !(oflag & O_CREAT))
+        return open(filename_utf8, oflag, pmode);
+
+    return fd;
+}
+#endif
+
 #if CONFIG_NETWORK
 #include <fcntl.h>
 #include <unistd.h>
diff --git a/libavformat/os_support.h b/libavformat/os_support.h
index dc01e6413c8c0..521e9978a2415 100644
--- a/libavformat/os_support.h
+++ b/libavformat/os_support.h
@@ -45,6 +45,11 @@ static inline int is_dos_path(const char *path)
     return 0;
 }
 
+#if defined(_WIN32) && !defined(__MINGW32CE__)
+int ff_win32_open(const char *filename, int oflag, int pmode);
+#define open ff_win32_open
+#endif
+
 #if CONFIG_NETWORK
 #if !HAVE_SOCKLEN_T
 typedef int socklen_t;
diff --git a/libavformat/version.h b/libavformat/version.h
index 9041f92626f1e..22b5dc9791ea6 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -25,7 +25,7 @@
 
 #define LIBAVFORMAT_VERSION_MAJOR 53
 #define LIBAVFORMAT_VERSION_MINOR  0
-#define LIBAVFORMAT_VERSION_MICRO  2
+#define LIBAVFORMAT_VERSION_MICRO  3
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
                                                LIBAVFORMAT_VERSION_MINOR, \