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>

13 files changed, 99 insertions, 33 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 6b141d3706..63b4952eaf 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -60,7 +60,7 @@ OBJS = ac3_parser.o                                                     \
 # subsystems
 OBJS-$(CONFIG_AANDCTTABLES)            += aandcttab.o
 OBJS-$(CONFIG_AC3DSP)                  += ac3dsp.o ac3.o ac3tab.o
-OBJS-$(CONFIG_ADTS_HEADER)             += adts_header.o mpeg4audio.o
+OBJS-$(CONFIG_ADTS_HEADER)             += adts_header.o mpeg4audio_sample_rates.o
 OBJS-$(CONFIG_AMF)                     += amfenc.o
 OBJS-$(CONFIG_AUDIO_FRAME_QUEUE)       += audio_frame_queue.o
 OBJS-$(CONFIG_ATSC_A53)                += atsc_a53.o
@@ -124,7 +124,7 @@ OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
                                           mpegaudiodsp_fixed.o          \
                                           mpegaudiodsp_float.o
 OBJS-$(CONFIG_MPEGAUDIOHEADER)         += mpegaudiodecheader.o mpegaudiodata.o
-OBJS-$(CONFIG_MPEG4AUDIO)              += mpeg4audio.o
+OBJS-$(CONFIG_MPEG4AUDIO)              += mpeg4audio.o mpeg4audio_sample_rates.o
 OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideodsp.o rl.o \
                                           mpegvideo_motion.o mpegutils.o \
                                           mpegvideodata.o mpegpicture.o
@@ -173,7 +173,8 @@ OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o aacenctab.o    \
                                           aacenc_tns.o \
                                           aacenc_ltp.o \
                                           aacenc_pred.o \
-                                          psymodel.o mpeg4audio.o kbdwin.o
+                                          psymodel.o kbdwin.o \
+                                          mpeg4audio_sample_rates.o
 OBJS-$(CONFIG_AAC_MF_ENCODER)          += mfenc.o mf_utils.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec_float.o ac3dec_data.o ac3.o \
@@ -992,17 +993,18 @@ SHLIBOBJS                              += log2_tab.o reverse.o
 OBJS-$(CONFIG_ISO_MEDIA)               += mpegaudiodata.o
 
 OBJS-$(CONFIG_FITS_DEMUXER)            += fits.o
-OBJS-$(CONFIG_MATROSKA_DEMUXER)        += mpeg4audio.o
 OBJS-$(CONFIG_NUT_MUXER)               += mpegaudiodata.o
-OBJS-$(CONFIG_RTP_MUXER)               += mpeg4audio.o
 OBJS-$(CONFIG_TAK_DEMUXER)             += tak.o
 
 # libavformat dependencies for static builds
+STLIBOBJS-$(CONFIG_FLV_MUXER)          += mpeg4audio_sample_rates.o
 STLIBOBJS-$(CONFIG_HLS_DEMUXER)        += ac3_channel_layout_tab.o
+STLIBOBJS-$(CONFIG_MATROSKA_DEMUXER)   += mpeg4audio_sample_rates.o
 STLIBOBJS-$(CONFIG_MOV_DEMUXER)        += ac3_channel_layout_tab.o
 STLIBOBJS-$(CONFIG_MXF_MUXER)          += golomb.o
 STLIBOBJS-$(CONFIG_RTPDEC)             += jpegtables.o
-STLIBOBJS-$(CONFIG_RTP_MUXER)          += golomb.o jpegtables.o
+STLIBOBJS-$(CONFIG_RTP_MUXER)          += golomb.o jpegtables.o \
+                                          mpeg4audio_sample_rates.o
 STLIBOBJS-$(CONFIG_SPDIF_MUXER)        += dca_sample_rate_tab.o
 
 # libavfilter dependencies
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index e462566078..a1004c3e98 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -998,7 +998,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
 
     /* Samplerate */
     for (i = 0; i < 16; i++)
-        if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
+        if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
             break;
     s->samplerate_index = i;
     ERROR_IF(s->samplerate_index == 16 ||
@@ -1143,7 +1143,7 @@ const AVCodec ff_aac_encoder = {
     .encode2        = aac_encode_frame,
     .close          = aac_encode_end,
     .defaults       = aac_encode_defaults,
-    .supported_samplerates = mpeg4audio_sample_rates,
+    .supported_samplerates = ff_mpeg4audio_sample_rates,
     .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
     .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
diff --git a/libavcodec/aacenctab.h b/libavcodec/aacenctab.h
index f54dd16bed..33cb7ae95b 100644
--- a/libavcodec/aacenctab.h
+++ b/libavcodec/aacenctab.h
@@ -81,13 +81,6 @@ static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
     { 2, 0, 1, 6, 7, 4, 5, 3 },
 };
 
-/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
- * failures */
-static const int mpeg4audio_sample_rates[16] = {
-    96000, 88200, 64000, 48000, 44100, 32000,
-    24000, 22050, 16000, 12000, 11025, 8000, 7350
-};
-
 /** bits needed to code codebook run value for long windows */
 static const uint8_t run_value_bits_long[64] = {
      5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
diff --git a/libavcodec/adts_header.c b/libavcodec/adts_header.c
index e4454529c4..ff4efafbf7 100644
--- a/libavcodec/adts_header.c
+++ b/libavcodec/adts_header.c
@@ -40,7 +40,7 @@ int ff_adts_header_parse(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
     crc_abs = get_bits1(gbc);    /* protection_absent */
     aot     = get_bits(gbc, 2);  /* profile_objecttype */
     sr      = get_bits(gbc, 4);  /* sample_frequency_index */
-    if (!avpriv_mpeg4audio_sample_rates[sr])
+    if (!ff_mpeg4audio_sample_rates[sr])
         return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
     skip_bits1(gbc);             /* private_bit */
     ch = get_bits(gbc, 3);       /* channel_configuration */
@@ -63,7 +63,7 @@ int ff_adts_header_parse(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
     hdr->crc_absent     = crc_abs;
     hdr->num_aac_frames = rdb + 1;
     hdr->sampling_index = sr;
-    hdr->sample_rate    = avpriv_mpeg4audio_sample_rates[sr];
+    hdr->sample_rate    = ff_mpeg4audio_sample_rates[sr];
     hdr->samples        = (rdb + 1) * 1024;
     hdr->bit_rate       = size * 8 * hdr->sample_rate / hdr->samples;
     hdr->frame_length   = size;
diff --git a/libavcodec/mpeg4audio.c b/libavcodec/mpeg4audio.c
index be50de9052..ed72a80f6d 100644
--- a/libavcodec/mpeg4audio.c
+++ b/libavcodec/mpeg4audio.c
@@ -57,13 +57,6 @@ static int parse_config_ALS(GetBitContext *gb, MPEG4AudioConfig *c, void *logctx
     return 0;
 }
 
-/* XXX: make sure to update the copies in the different encoders if you change
- * this table */
-const int avpriv_mpeg4audio_sample_rates[16] = {
-    96000, 88200, 64000, 48000, 44100, 32000,
-    24000, 22050, 16000, 12000, 11025, 8000, 7350
-};
-
 const uint8_t ff_mpeg4audio_channels[14] = {
     0,
     1, // mono (1/0)
@@ -93,7 +86,7 @@ static inline int get_sample_rate(GetBitContext *gb, int *index)
 {
     *index = get_bits(gb, 4);
     return *index == 0x0f ? get_bits(gb, 24) :
-        avpriv_mpeg4audio_sample_rates[*index];
+        ff_mpeg4audio_sample_rates[*index];
 }
 
 int ff_mpeg4audio_get_config_gb(MPEG4AudioConfig *c, GetBitContext *gb,
diff --git a/libavcodec/mpeg4audio.h b/libavcodec/mpeg4audio.h
index 3187df68d2..c486a3ddef 100644
--- a/libavcodec/mpeg4audio.h
+++ b/libavcodec/mpeg4audio.h
@@ -27,7 +27,6 @@
 #include "libavutil/attributes.h"
 
 #include "get_bits.h"
-#include "internal.h"
 #include "put_bits.h"
 
 typedef struct MPEG4AudioConfig {
@@ -45,7 +44,7 @@ typedef struct MPEG4AudioConfig {
     int frame_length_short;
 } MPEG4AudioConfig;
 
-extern av_export_avcodec const int avpriv_mpeg4audio_sample_rates[16];
+extern const int     ff_mpeg4audio_sample_rates[16];
 extern const uint8_t ff_mpeg4audio_channels[14];
 
 /**
diff --git a/libavcodec/mpeg4audio_sample_rates.c b/libavcodec/mpeg4audio_sample_rates.c
new file mode 100644
index 0000000000..b5ceb59c6e
--- /dev/null
+++ b/libavcodec/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 "mpeg4audio_sample_rates.h"
diff --git a/libavcodec/mpeg4audio_sample_rates.h b/libavcodec/mpeg4audio_sample_rates.h
new file mode 100644
index 0000000000..0b8caa6d76
--- /dev/null
+++ b/libavcodec/mpeg4audio_sample_rates.h
@@ -0,0 +1,30 @@
+/*
+ * 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
+ */
+
+#ifndef AVCODEC_MPEG4AUDIO_SAMPLE_RATES_H
+#define AVCODEC_MPEG4AUDIO_SAMPLE_RATES_H
+
+const int ff_mpeg4audio_sample_rates[16] = {
+    96000, 88200, 64000, 48000, 44100, 32000,
+    24000, 22050, 16000, 12000, 11025, 8000, 7350
+};
+#endif
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");