[FFmpeg-cvslog] lavc: use avpriv_ prefix for some mpegaudio symbols used in lavf.
Anton Khirnov
git at videolan.org
Fri Oct 21 02:37:11 CEST 2011
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Mon Oct 17 09:28:53 2011 +0200| [82ab61f9015659419e0a2766ee031c367e3f2908] | committer: Anton Khirnov
lavc: use avpriv_ prefix for some mpegaudio symbols used in lavf.
Specifically, ff_mpa_freq_tab, ff_mpa_bitrate_tab, ff_mpa_decode_header,
ff_mpegaudio_decode_header.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=82ab61f9015659419e0a2766ee031c367e3f2908
---
libavcodec/mp3_header_decompress_bsf.c | 4 ++--
libavcodec/mpegaudio_parser.c | 2 +-
libavcodec/mpegaudiodata.c | 4 ++--
libavcodec/mpegaudiodata.h | 4 ++--
libavcodec/mpegaudiodec.c | 6 +++---
libavcodec/mpegaudiodecheader.c | 10 +++++-----
libavcodec/mpegaudiodecheader.h | 4 ++--
libavcodec/mpegaudioenc.c | 6 +++---
libavformat/isom.c | 2 +-
libavformat/mp3dec.c | 4 ++--
libavformat/mp3enc.c | 8 ++++----
libavformat/nutenc.c | 4 ++--
12 files changed, 29 insertions(+), 29 deletions(-)
diff --git a/libavcodec/mp3_header_decompress_bsf.c b/libavcodec/mp3_header_decompress_bsf.c
index 3f30742..78025cc 100644
--- a/libavcodec/mp3_header_decompress_bsf.c
+++ b/libavcodec/mp3_header_decompress_bsf.c
@@ -50,10 +50,10 @@ static int mp3_header_decompress(AVBitStreamFilterContext *bsfc, AVCodecContext
lsf = sample_rate < (24000+32000)/2;
mpeg25 = sample_rate < (12000+16000)/2;
sample_rate_index= (header>>10)&3;
- sample_rate= ff_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
+ sample_rate= avpriv_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
for(bitrate_index=2; bitrate_index<30; bitrate_index++){
- frame_size = ff_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
+ frame_size = avpriv_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
if(frame_size == buf_size + 4)
break;
diff --git a/libavcodec/mpegaudio_parser.c b/libavcodec/mpegaudio_parser.c
index f07d34b..17d329d 100644
--- a/libavcodec/mpegaudio_parser.c
+++ b/libavcodec/mpegaudio_parser.c
@@ -64,7 +64,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
state= (state<<8) + buf[i++];
- ret = ff_mpa_decode_header(avctx, state, &sr, &channels, &frame_size, &bit_rate);
+ ret = avpriv_mpa_decode_header(avctx, state, &sr, &channels, &frame_size, &bit_rate);
if (ret < 4) {
s->header_count= -2;
} else {
diff --git a/libavcodec/mpegaudiodata.c b/libavcodec/mpegaudiodata.c
index b850d22..81a4365 100644
--- a/libavcodec/mpegaudiodata.c
+++ b/libavcodec/mpegaudiodata.c
@@ -27,7 +27,7 @@
#include "mpegaudiodata.h"
-const uint16_t ff_mpa_bitrate_tab[2][3][15] = {
+const uint16_t avpriv_mpa_bitrate_tab[2][3][15] = {
{ {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 } },
@@ -37,7 +37,7 @@ const uint16_t ff_mpa_bitrate_tab[2][3][15] = {
}
};
-const uint16_t ff_mpa_freq_tab[3] = { 44100, 48000, 32000 };
+const uint16_t avpriv_mpa_freq_tab[3] = { 44100, 48000, 32000 };
/*******************************************************/
/* half mpeg encoding window (full precision) */
diff --git a/libavcodec/mpegaudiodata.h b/libavcodec/mpegaudiodata.h
index 8445883..24ea536 100644
--- a/libavcodec/mpegaudiodata.h
+++ b/libavcodec/mpegaudiodata.h
@@ -32,8 +32,8 @@
#define MODE_EXT_MS_STEREO 2
#define MODE_EXT_I_STEREO 1
-extern const uint16_t ff_mpa_bitrate_tab[2][3][15];
-extern const uint16_t ff_mpa_freq_tab[3];
+extern const uint16_t avpriv_mpa_bitrate_tab[2][3][15];
+extern const uint16_t avpriv_mpa_freq_tab[3];
extern const int32_t ff_mpa_enwindow[257];
extern const int ff_mpa_sblimit_table[5];
extern const int ff_mpa_quant_steps[17];
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c
index 6f841e8..1b36937 100644
--- a/libavcodec/mpegaudiodec.c
+++ b/libavcodec/mpegaudiodec.c
@@ -1790,7 +1790,7 @@ static int decode_frame(AVCodecContext * avctx,
return -1;
}
- if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
+ if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
/* free format: prepare to compute frame size */
s->frame_size = -1;
return -1;
@@ -1863,7 +1863,7 @@ static int decode_frame_adu(AVCodecContext * avctx,
return buf_size;
}
- ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
+ avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
/* update codec info */
avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
@@ -2016,7 +2016,7 @@ static int decode_frame_mp3on4(AVCodecContext * avctx,
if (ff_mpa_check_header(header) < 0) // Bad header, discard block
break;
- ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
+ avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
out_size += mp_decode_frame(m, outptr, buf, fsize);
buf += fsize;
len -= fsize;
diff --git a/libavcodec/mpegaudiodecheader.c b/libavcodec/mpegaudiodecheader.c
index be7abc6..dbd67ff 100644
--- a/libavcodec/mpegaudiodecheader.c
+++ b/libavcodec/mpegaudiodecheader.c
@@ -31,7 +31,7 @@
#include "mpegaudiodecheader.h"
-int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
+int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
{
int sample_rate, frame_size, mpeg25, padding;
int sample_rate_index, bitrate_index;
@@ -46,7 +46,7 @@ int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
s->layer = 4 - ((header >> 17) & 3);
/* extract frequency */
sample_rate_index = (header >> 10) & 3;
- sample_rate = ff_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
+ sample_rate = avpriv_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
sample_rate_index += 3 * (s->lsf + mpeg25);
s->sample_rate_index = sample_rate_index;
s->error_protection = ((header >> 16) & 1) ^ 1;
@@ -67,7 +67,7 @@ int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
s->nb_channels = 2;
if (bitrate_index != 0) {
- frame_size = ff_mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index];
+ frame_size = avpriv_mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index];
s->bit_rate = frame_size * 1000;
switch(s->layer) {
case 1:
@@ -109,14 +109,14 @@ int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
return 0;
}
-int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate)
+int avpriv_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate)
{
MPADecodeHeader s1, *s = &s1;
if (ff_mpa_check_header(head) != 0)
return -1;
- if (ff_mpegaudio_decode_header(s, head) != 0) {
+ if (avpriv_mpegaudio_decode_header(s, head) != 0) {
return -1;
}
diff --git a/libavcodec/mpegaudiodecheader.h b/libavcodec/mpegaudiodecheader.h
index 2991595..764e8ab 100644
--- a/libavcodec/mpegaudiodecheader.h
+++ b/libavcodec/mpegaudiodecheader.h
@@ -50,11 +50,11 @@ typedef struct MPADecodeHeader {
/* header decoding. MUST check the header before because no
consistency check is done there. Return 1 if free format found and
that the frame size must be computed externally */
-int ff_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header);
+int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header);
/* useful helper to get mpeg audio stream infos. Return -1 if error in
header, otherwise the coded frame size in bytes */
-int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate);
+int avpriv_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate);
/* fast header check for resync */
static inline int ff_mpa_check_header(uint32_t header){
diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c
index 1cadef7..71ea393 100644
--- a/libavcodec/mpegaudioenc.c
+++ b/libavcodec/mpegaudioenc.c
@@ -84,9 +84,9 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
/* encoding freq */
s->lsf = 0;
for(i=0;i<3;i++) {
- if (ff_mpa_freq_tab[i] == freq)
+ if (avpriv_mpa_freq_tab[i] == freq)
break;
- if ((ff_mpa_freq_tab[i] / 2) == freq) {
+ if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
s->lsf = 1;
break;
}
@@ -99,7 +99,7 @@ static av_cold int MPA_encode_init(AVCodecContext *avctx)
/* encoding bitrate & frequency */
for(i=0;i<15;i++) {
- if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
+ if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
break;
}
if (i == 15){
diff --git a/libavformat/isom.c b/libavformat/isom.c
index d7c0af1..e5fd859 100644
--- a/libavformat/isom.c
+++ b/libavformat/isom.c
@@ -437,7 +437,7 @@ int ff_mp4_read_dec_config_descr(AVFormatContext *fc, AVStream *st, AVIOContext
st->codec->extradata_size);
st->codec->channels = cfg.channels;
if (cfg.object_type == 29 && cfg.sampling_index < 3) // old mp3on4
- st->codec->sample_rate = ff_mpa_freq_tab[cfg.sampling_index];
+ st->codec->sample_rate = avpriv_mpa_freq_tab[cfg.sampling_index];
else if (cfg.ext_sample_rate)
st->codec->sample_rate = cfg.ext_sample_rate;
else
diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 09494de..9011f9f 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -51,7 +51,7 @@ static int mp3_read_probe(AVProbeData *p)
for(frames = 0; buf2 < end; frames++) {
header = AV_RB32(buf2);
- fsize = ff_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate);
+ fsize = avpriv_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate);
if(fsize < 0)
break;
buf2 += fsize;
@@ -86,7 +86,7 @@ static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
if(ff_mpa_check_header(v) < 0)
return -1;
- if (ff_mpegaudio_decode_header(&c, v) == 0)
+ if (avpriv_mpegaudio_decode_header(&c, v) == 0)
vbrtag_size = c.frame_size;
if(c.layer != 3)
return -1;
diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index 07c28e1..64f8957 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -217,12 +217,12 @@ static void mp3_write_xing(AVFormatContext *s)
MPADecodeHeader mpah;
int srate_idx, i, channels;
- for (i = 0; i < FF_ARRAY_ELEMS(ff_mpa_freq_tab); i++)
- if (ff_mpa_freq_tab[i] == codec->sample_rate) {
+ for (i = 0; i < FF_ARRAY_ELEMS(avpriv_mpa_freq_tab); i++)
+ if (avpriv_mpa_freq_tab[i] == codec->sample_rate) {
srate_idx = i;
break;
}
- if (i == FF_ARRAY_ELEMS(ff_mpa_freq_tab)) {
+ if (i == FF_ARRAY_ELEMS(avpriv_mpa_freq_tab)) {
av_log(s, AV_LOG_ERROR, "Unsupported sample rate.\n");
return;
}
@@ -240,7 +240,7 @@ static void mp3_write_xing(AVFormatContext *s)
header |= channels << 6;
avio_wb32(s->pb, header);
- ff_mpegaudio_decode_header(&mpah, header);
+ avpriv_mpegaudio_decode_header(&mpah, header);
ffio_fill(s->pb, 0, xing_offset);
ffio_wfourcc(s->pb, "Xing");
diff --git a/libavformat/nutenc.c b/libavformat/nutenc.c
index f8a078c..bde6ba2 100644
--- a/libavformat/nutenc.c
+++ b/libavformat/nutenc.c
@@ -59,10 +59,10 @@ static int find_expected_header(AVCodecContext *c, int size, int key_frame, uint
else if(sample_rate < (44100 + 48000)/2) sample_rate_index=0;
else sample_rate_index=1;
- sample_rate= ff_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25);
+ sample_rate= avpriv_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25);
for(bitrate_index=2; bitrate_index<30; bitrate_index++){
- frame_size = ff_mpa_bitrate_tab[lsf][layer-1][bitrate_index>>1];
+ frame_size = avpriv_mpa_bitrate_tab[lsf][layer-1][bitrate_index>>1];
frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
if(frame_size == size)
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