avcodec/mpeg4audio: Unavpriv and deduplicate mpeg4audio_sample_rates

avpriv_mpeg4audio_sample_rates has a size of 64B and it is currently
avpriv; a clone of it exists in aacenctab.h and from there it is inlined
in aacenc.c (which also uses the avpriv version) and in the FLV muxer.
This means that despite it being avpriv both libavformat as well as
libavcodec have copies already.

This situation is clearly suboptimal. Given the overhead of exporting
symbols (for x64 Elf/Linux/GNU: 2x2B version, 2x24B .dynsym, 24B .rela.dyn,
8B .got, 4B hash + twice the size of the name (here 31B)) the object is
unavprived, i.e. duplicated into libavformat when creating a shared
build; but the duplicates in the AAC encoder and FLV muxer are removed.

This involves splitting of the sample rate table into a file of its own;
this allowed to break some spurious dependencies (e.g. both the AAC
encoder as well as the Matroska demuxer actually don't need the
mpeg4audio_get_config stuff).

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>

5 files changed, 32 insertions, 6 deletions

diff --git a/libavformat/Makefile b/libavformat/Makefile
index f5d6b8fecf..6482f247b6 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -681,11 +681,14 @@ OBJS-$(CONFIG_LIBZMQ_PROTOCOL)           += libzmq.o
 
 # Objects duplicated from other libraries for shared builds
 SHLIBOBJS                                += log2_tab.o
+SHLIBOBJS-$(CONFIG_FLV_MUXER)            += mpeg4audio_sample_rates.o
 SHLIBOBJS-$(CONFIG_HLS_DEMUXER)          += ac3_channel_layout_tab.o
+SHLIBOBJS-$(CONFIG_MATROSKA_DEMUXER)     += mpeg4audio_sample_rates.o
 SHLIBOBJS-$(CONFIG_MOV_DEMUXER)          += ac3_channel_layout_tab.o
 SHLIBOBJS-$(CONFIG_MXF_MUXER)            += golomb_tab.o
 SHLIBOBJS-$(CONFIG_RTPDEC)               += jpegtables.o
-SHLIBOBJS-$(CONFIG_RTP_MUXER)            += golomb_tab.o jpegtables.o
+SHLIBOBJS-$(CONFIG_RTP_MUXER)            += golomb_tab.o jpegtables.o \
+                                            mpeg4audio_sample_rates.o
 SHLIBOBJS-$(CONFIG_SPDIF_MUXER)          += dca_sample_rate_tab.o
 
 # libavdevice dependencies
diff --git a/libavformat/flvenc.c b/libavformat/flvenc.c
index 31d1d93c23..66c530a2ff 100644
--- a/libavformat/flvenc.c
+++ b/libavformat/flvenc.c
@@ -24,6 +24,7 @@
 #include "libavutil/intfloat.h"
 #include "libavutil/avassert.h"
 #include "libavutil/mathematics.h"
+#include "libavcodec/mpeg4audio.h"
 #include "avio_internal.h"
 #include "avio.h"
 #include "avc.h"
@@ -33,7 +34,6 @@
 #include "metadata.h"
 #include "libavutil/opt.h"
 #include "libavcodec/put_bits.h"
-#include "libavcodec/aacenctab.h"
 
 
 static const AVCodecTag flv_video_codec_ids[] = {
@@ -514,7 +514,7 @@ static void flv_write_codec_header(AVFormatContext* s, AVCodecParameters* par, i
                 for (samplerate_index = 0; samplerate_index < 16;
                         samplerate_index++)
                     if (flv->audio_par->sample_rate
-                            == mpeg4audio_sample_rates[samplerate_index])
+                            == ff_mpeg4audio_sample_rates[samplerate_index])
                         break;
 
                 init_put_bits(&pbc, data, sizeof(data));
diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
index 6ce553205d..e271916bf1 100644
--- a/libavformat/matroskadec.c
+++ b/libavformat/matroskadec.c
@@ -2029,8 +2029,8 @@ static int matroska_aac_sri(int samplerate)
 {
     int sri;
 
-    for (sri = 0; sri < FF_ARRAY_ELEMS(avpriv_mpeg4audio_sample_rates); sri++)
-        if (avpriv_mpeg4audio_sample_rates[sri] == samplerate)
+    for (sri = 0; sri < FF_ARRAY_ELEMS(ff_mpeg4audio_sample_rates); sri++)
+        if (ff_mpeg4audio_sample_rates[sri] == samplerate)
             break;
     return sri;
 }
diff --git a/libavformat/mpeg4audio_sample_rates.c b/libavformat/mpeg4audio_sample_rates.c
new file mode 100644
index 0000000000..212385f348
--- /dev/null
+++ b/libavformat/mpeg4audio_sample_rates.c
@@ -0,0 +1,23 @@
+/*
+ * MPEG-4 Audio sample rates
+ * Copyright (c) 2008 Baptiste Coudurier <baptiste.coudurier@free.fr>
+ * Copyright (c) 2009 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavcodec/mpeg4audio_sample_rates.h"
diff --git a/libavformat/sdp.c b/libavformat/sdp.c
index 111a25b126..2230d74742 100644
--- a/libavformat/sdp.c
+++ b/libavformat/sdp.c
@@ -481,7 +481,7 @@ static int latm_context2config(AVFormatContext *s, const AVCodecParameters *par,
     *out = NULL;
 
     for (rate_index = 0; rate_index < 16; rate_index++)
-        if (avpriv_mpeg4audio_sample_rates[rate_index] == par->sample_rate)
+        if (ff_mpeg4audio_sample_rates[rate_index] == par->sample_rate)
             break;
     if (rate_index == 16) {
         av_log(s, AV_LOG_ERROR, "Unsupported sample rate\n");