Make local functions static in codecs, where possible.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@19612 a1c6a512-1295-4272-9138-f99709370657

12 files changed, 44 insertions, 44 deletions

diff --git a/apps/codecs/a52.c b/apps/codecs/a52.c
index f8eaef26fd..360a5862d7 100644
--- a/apps/codecs/a52.c
+++ b/apps/codecs/a52.c
@@ -43,7 +43,7 @@ static inline void output_audio(sample_t *samples)
     ci->pcmbuf_insert(&samples[0], &samples[256], 256);
 }
 
-void a52_decode_data(uint8_t *start, uint8_t *end)
+static void a52_decode_data(uint8_t *start, uint8_t *end)
 {
     static uint8_t *bufptr = buf;
     static uint8_t *bufpos = buf + 7;
diff --git a/apps/codecs/ape.c b/apps/codecs/ape.c
index 0419a6f6bd..dbe6e0fc9e 100644
--- a/apps/codecs/ape.c
+++ b/apps/codecs/ape.c
@@ -57,11 +57,11 @@ static int32_t decoded1[BLOCKS_PER_LOOP] IBSS_ATTR;
    skip in that frame.
 */
 
-bool ape_calc_seekpos(struct ape_ctx_t* ape_ctx,
-                      uint32_t new_sample,
-                      uint32_t* newframe,
-                      uint32_t* filepos,
-                      uint32_t* samplestoskip)
+static bool ape_calc_seekpos(struct ape_ctx_t* ape_ctx,
+                             uint32_t new_sample,
+                             uint32_t* newframe,
+                             uint32_t* filepos,
+                             uint32_t* samplestoskip)
 {
     uint32_t n;
 
@@ -82,9 +82,9 @@ bool ape_calc_seekpos(struct ape_ctx_t* ape_ctx,
 /* The resume offset is a value in bytes - we need to
    turn it into a frame number and samplestoskip value */
 
-void ape_resume(struct ape_ctx_t* ape_ctx, size_t resume_offset, 
-                uint32_t* currentframe, uint32_t* samplesdone, 
-                uint32_t* samplestoskip, int* firstbyte)
+static void ape_resume(struct ape_ctx_t* ape_ctx, size_t resume_offset, 
+                       uint32_t* currentframe, uint32_t* samplesdone, 
+                       uint32_t* samplestoskip, int* firstbyte)
 {
     off_t newfilepos;
     int64_t framesize;
diff --git a/apps/codecs/demac/libdemac/crc.c b/apps/codecs/demac/libdemac/crc.c
index c23de7d043..816c6594f7 100644
--- a/apps/codecs/demac/libdemac/crc.c
+++ b/apps/codecs/demac/libdemac/crc.c
@@ -2,7 +2,7 @@
 
 libdemac - A Monkey's Audio decoder
 
-$Id:$
+$Id$
 
 Copyright (C) Dave Chapman 2007
 
@@ -24,7 +24,7 @@ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110, USA
 
 #include <inttypes.h>
 
-static uint32_t crctab32[] =
+static const uint32_t crctab32[] =
 {
   0x00000000, 0x77073096, 0xEE0E612C, 0x990951BA,
   0x076DC419, 0x706AF48F, 0xE963A535, 0x9E6495A3,
diff --git a/apps/codecs/flac.c b/apps/codecs/flac.c
index cf3bbca65c..3a23d0b951 100644
--- a/apps/codecs/flac.c
+++ b/apps/codecs/flac.c
@@ -180,7 +180,7 @@ static bool flac_init(FLACContext* fc, int first_frame_offset)
 }
 
 /* Synchronize to next frame in stream - adapted from libFLAC 1.1.3b2 */
-bool frame_sync(FLACContext* fc) {
+static bool frame_sync(FLACContext* fc) {
     unsigned int x = 0;
     bool cached = false;
 
@@ -232,7 +232,7 @@ bool frame_sync(FLACContext* fc) {
 }
 
 /* Seek to sample - adapted from libFLAC 1.1.3b2+ */
-bool flac_seek(FLACContext* fc, uint32_t target_sample) {
+static bool flac_seek(FLACContext* fc, uint32_t target_sample) {
     off_t orig_pos = ci->curpos;
     off_t pos = -1;
     unsigned long lower_bound, upper_bound;
@@ -385,7 +385,7 @@ bool flac_seek(FLACContext* fc, uint32_t target_sample) {
 }
 
 /* Seek to file offset */
-bool flac_seek_offset(FLACContext* fc, uint32_t offset) {
+static bool flac_seek_offset(FLACContext* fc, uint32_t offset) {
     unsigned unparseable_count;
     bool got_a_frame = false;
 
diff --git a/apps/codecs/libfaad/pns.c b/apps/codecs/libfaad/pns.c
index 7727b22160..85de391101 100644
--- a/apps/codecs/libfaad/pns.c
+++ b/apps/codecs/libfaad/pns.c
@@ -51,7 +51,7 @@ static void gen_rand_vector(real_t *spec, int16_t scale_factor, uint16_t size,
 
 /* fixed point square root approximation */
 /* !!!! ONLY WORKS FOR EVEN %REAL_BITS% !!!! */
-real_t fp_sqrt(real_t value)
+static real_t fp_sqrt(real_t value)
 {
     real_t root = 0;
 
diff --git a/apps/codecs/libm4a/demux.c b/apps/codecs/libm4a/demux.c
index e9b5c2c482..f0c6922ca6 100644
--- a/apps/codecs/libm4a/demux.c
+++ b/apps/codecs/libm4a/demux.c
@@ -83,7 +83,7 @@ static void read_chunk_ftyp(qtmovie_t *qtmovie, size_t chunk_len)
     }
 }
 
-uint32_t mp4ff_read_mp4_descr_length(stream_t* stream)
+static uint32_t mp4ff_read_mp4_descr_length(stream_t* stream)
 {
     uint8_t b;
     uint8_t numBytes = 0;
diff --git a/apps/codecs/libwavpack/words.c b/apps/codecs/libwavpack/words.c
index c7a8047d03..6da716119c 100644
--- a/apps/codecs/libwavpack/words.c
+++ b/apps/codecs/libwavpack/words.c
@@ -253,7 +253,7 @@ int read_hybrid_profile (WavpackStream *wps, WavpackMetadata *wpmd)
 // currently implemented) this is calculated from the slow_level values and the
 // bitrate accumulators. Note that the bitrate accumulators can be changing.
 
-void update_error_limit (struct words_data *w, uint32_t flags)
+static void update_error_limit (struct words_data *w, uint32_t flags)
 {
     int bitrate_0 = (w->bitrate_acc [0] += w->bitrate_delta [0]) >> 16;
 
diff --git a/apps/codecs/mod.c b/apps/codecs/mod.c
index c8ada66f18..91b5955b40 100644
--- a/apps/codecs/mod.c
+++ b/apps/codecs/mod.c
@@ -264,27 +264,27 @@ void mixer_playsample(int channel, int instrument)
     modplayer.modchannel[channel].instrument = instrument;
 }
 
-inline void mixer_stopsample(int channel)
+static inline void mixer_stopsample(int channel)
 {
     mixer.channel[channel].channelactive = false;
 }
 
-inline void mixer_continuesample(int channel)
+static inline void mixer_continuesample(int channel)
 {
     mixer.channel[channel].channelactive = true;
 }
 
-inline void mixer_setvolume(int channel, int volume)
+static inline void mixer_setvolume(int channel, int volume)
 {
     mixer.channel[channel].volume = volume;
 }
 
-inline void mixer_setpanning(int channel, int panning)
+static inline void mixer_setpanning(int channel, int panning)
 {
     mixer.channel[channel].panning = panning;
 }
 
-inline void mixer_setamigaperiod(int channel, int amigaperiod)
+static inline void mixer_setamigaperiod(int channel, int amigaperiod)
 {
     /* Just to make sure we don't devide by zero
      * amigaperiod shouldn't 0 anyway - if it is the case
@@ -1090,7 +1090,7 @@ void playeffect(int currenttick)
     }
 }
 
-inline int clip(int i)
+static inline int clip(int i)
 {
     if (i > 32767) return(32767);
     else if (i < -32768) return(-32768);
diff --git a/apps/codecs/mp3_enc.c b/apps/codecs/mp3_enc.c
index 18aa1bfe1a..94d4c2a5f3 100644
--- a/apps/codecs/mp3_enc.c
+++ b/apps/codecs/mp3_enc.c
@@ -1156,7 +1156,7 @@ void putbits(uint32_t val, uint32_t nbit)
 /*  of the Huffman tables as defined in the IS (Table B.7), and will not   */
 /*  work with any arbitrary tables.                                        */
 /***************************************************************************/
-int choose_table( short *ix, uint32_t begin, uint32_t end, int *bits )
+static int choose_table( short *ix, uint32_t begin, uint32_t end, int *bits )
 {
   uint32_t i;
   int    max, table0, table1;
@@ -1301,7 +1301,7 @@ int count_bigv(short *ix, uint32_t start, uint32_t end, int table0,
 /* Function: Calculation of rzero, count1, address3                      */
 /* (Partitions ix into big values, quadruples and zeros).                */
 /*************************************************************************/
-int calc_runlen( short *ix, side_info_t *si )
+static int calc_runlen( short *ix, side_info_t *si )
 {
   int  p, i, sum = 0;
 
@@ -1347,7 +1347,7 @@ int calc_runlen( short *ix, side_info_t *si )
 /*************************************************************************/
 /*   Function: Quantization of the vector xr ( -> ix)                    */
 /*************************************************************************/
-int quantize_int(int *xr, short *ix, side_info_t *si)
+static int quantize_int(int *xr, short *ix, side_info_t *si)
 {
   unsigned int i, idx, s, frac_pow[] = { 0x10000, 0xd745, 0xb505, 0x9838 };
 
@@ -1379,7 +1379,7 @@ int quantize_int(int *xr, short *ix, side_info_t *si)
 /*************************************************************************/
 /* subdivides the bigvalue region which will use separate Huffman tables */
 /*************************************************************************/
-void subdivide(side_info_t *si)
+static void subdivide(side_info_t *si)
 {
   int scfb, count0, count1;
   
@@ -1407,7 +1407,7 @@ void subdivide(side_info_t *si)
 /*******************************************************************/
 /* Count the number of bits necessary to code the bigvalues region */
 /*******************************************************************/
-int bigv_bitcount(short *ix, side_info_t *gi)
+static int bigv_bitcount(short *ix, side_info_t *gi)
 {
   int b1=0, b2=0, b3=0;
 
@@ -1428,7 +1428,7 @@ int bigv_bitcount(short *ix, side_info_t *gi)
   return b1+b2+b3;
 }
 
-int quantize_and_count_bits(int *xr, short *ix, side_info_t *si)
+static int quantize_and_count_bits(int *xr, short *ix, side_info_t *si)
 {
   int bits = 10000;
 
@@ -1445,7 +1445,7 @@ int quantize_and_count_bits(int *xr, short *ix, side_info_t *si)
 /************************************************************************/
 /* The code selects the best quantStep for a particular set of scalefacs*/
 /************************************************************************/ 
-int inner_loop(int *xr, int max_bits, side_info_t *si)
+static int inner_loop(int *xr, int max_bits, side_info_t *si)
 {
   int bits;
 
@@ -1469,7 +1469,7 @@ int inner_loop(int *xr, int max_bits, side_info_t *si)
   return bits;
 }
 
-void iteration_loop(int *xr, side_info_t *si, int gr_cnt)
+static void iteration_loop(int *xr, side_info_t *si, int gr_cnt)
 {
   int remain, tar_bits, max_bits = cfg.mean_bits;
 
@@ -1971,10 +1971,10 @@ static int find_samplerate_index(long freq, int *mp3_type)
     return i;    
 }
 
-bool init_mp3_encoder_engine(int sample_rate,
-                             int num_channels,
-                             int rec_mono_mode,
-                             struct encoder_config *enc_cfg)
+static bool init_mp3_encoder_engine(int sample_rate,
+                                    int num_channels,
+                                    int rec_mono_mode,
+                                    struct encoder_config *enc_cfg)
 {
     const bool stereo = num_channels > 1;
     uint32_t avg_byte_per_frame;
@@ -2157,7 +2157,7 @@ static inline void byte_swap_frame32(uint32_t *dst, uint32_t *src,
 } /* byte_swap_frame32 */
 #endif /* ROCKBOX_LITTLE_ENDIAN */
 
-void set_scale_facs(int *mdct_freq)
+static void set_scale_facs(int *mdct_freq)
 {
   unsigned int i, is, ie, k, s;
   int max_freq_val, avrg_freq_val;
diff --git a/apps/codecs/mpa.c b/apps/codecs/mpa.c
index 37a1afadfa..1a0b03c272 100644
--- a/apps/codecs/mpa.c
+++ b/apps/codecs/mpa.c
@@ -56,7 +56,7 @@ unsigned char mad_main_data[MAD_BUFFER_MDLEN] IBSS_ATTR;
 int mpeg_latency[3] = { 0, 481, 529 };
 int mpeg_framesize[3] = {384, 1152, 1152};
 
-void init_mad(void)
+static void init_mad(void)
 {
     ci->memset(&stream, 0, sizeof(struct mad_stream));
     ci->memset(&frame, 0, sizeof(struct mad_frame));
@@ -85,7 +85,7 @@ void init_mad(void)
     
 }
 
-int get_file_pos(int newtime)
+static int get_file_pos(int newtime)
 {
     int pos = -1;
     struct mp3entry *id3 = ci->id3;
diff --git a/apps/codecs/mpc.c b/apps/codecs/mpc.c
index 36a7469088..a227fb81cf 100644
--- a/apps/codecs/mpc.c
+++ b/apps/codecs/mpc.c
@@ -27,14 +27,14 @@ CODEC_HEADER
 mpc_decoder decoder IBSS_ATTR;
 
 /* Our implementations of the mpc_reader callback functions. */
-mpc_int32_t read_impl(void *data, void *ptr, mpc_int32_t size)
+static mpc_int32_t read_impl(void *data, void *ptr, mpc_int32_t size)
 {
     struct codec_api *ci = (struct codec_api *)data;
 
     return ((mpc_int32_t)(ci->read_filebuf(ptr, size)));
 }
 
-mpc_bool_t seek_impl(void *data, mpc_int32_t offset)
+static mpc_bool_t seek_impl(void *data, mpc_int32_t offset)
 {  
     struct codec_api *ci = (struct codec_api *)data;
 
@@ -43,21 +43,21 @@ mpc_bool_t seek_impl(void *data, mpc_int32_t offset)
     return ci->seek_buffer(offset);
 }
 
-mpc_int32_t tell_impl(void *data)
+static mpc_int32_t tell_impl(void *data)
 {
     struct codec_api *ci = (struct codec_api *)data;
 
     return ci->curpos;
 }
 
-mpc_int32_t get_size_impl(void *data)
+static mpc_int32_t get_size_impl(void *data)
 {
     struct codec_api *ci = (struct codec_api *)data;
     
     return ci->filesize;
 }
 
-mpc_bool_t canseek_impl(void *data)
+static mpc_bool_t canseek_impl(void *data)
 {
     (void)data;
     
diff --git a/apps/codecs/nsf.c b/apps/codecs/nsf.c
index c7239837eb..6beb8fe3e6 100644
--- a/apps/codecs/nsf.c
+++ b/apps/codecs/nsf.c
@@ -4294,7 +4294,7 @@ jammed:
 
 /****************** rockbox interface ******************/
 
-void set_codec_track(int t, int d) {
+static void set_codec_track(int t, int d) {
     int track,fade,def=0;
     SetTrack(t);