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authorMichael Sevakis <jethead71@rockbox.org>2006-11-06 18:07:30 +0000
committerMichael Sevakis <jethead71@rockbox.org>2006-11-06 18:07:30 +0000
commit0f5cb94aa4a334366a746fcbb22f3335ca413265 (patch)
tree8f89a96628c1810d51ee9816daf78edb8c76fcd4
parent0b22795e26ee09de14f6ac23219adeda12f2fd5b (diff)
downloadrockbox-0f5cb94aa4a334366a746fcbb22f3335ca413265.tar.gz
rockbox-0f5cb94aa4a334366a746fcbb22f3335ca413265.tar.bz2
rockbox-0f5cb94aa4a334366a746fcbb22f3335ca413265.zip
Big Patch adds primarily: Samplerate and format selection to recording for SWCODEC. Supprort for samplerates changing in playback (just goes with the recording part inseparably). Samplerates to all encoders. Encoders can be configured individually on a menu specific to the encoder in the recording menu. File creation is delayed until flush time to reduce spinups when splitting. Misc: statusbar icons for numbers are individual digits to display any number. Audio buffer was rearranged to maximize memory available to recording and properly reinitialized when trashed. ColdFire PCM stuff moved to target tree to avoid a complicated mess when adding samplerate switching. Some needed API changes and to neaten up growing gap between hardware and software codecs.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11452 a1c6a512-1295-4272-9138-f99709370657
-rw-r--r--apps/SOURCES3
-rw-r--r--apps/codecs.c25
-rw-r--r--apps/codecs.h64
-rw-r--r--apps/codecs/libwavpack/bits.c22
-rw-r--r--apps/codecs/mp3_enc.c1182
-rw-r--r--apps/codecs/wav_enc.c419
-rw-r--r--apps/codecs/wavpack_enc.c539
-rw-r--r--apps/eq_menu.c3
-rw-r--r--apps/gui/statusbar.c196
-rw-r--r--apps/lang/english.lang114
-rw-r--r--apps/main.c3
-rw-r--r--apps/metadata.c48
-rw-r--r--apps/misc.c57
-rw-r--r--apps/misc.h35
-rw-r--r--apps/pcmbuf.c27
-rw-r--r--apps/pcmbuf.h2
-rw-r--r--apps/playback.c262
-rw-r--r--apps/playlist.c12
-rw-r--r--apps/plugin.c8
-rw-r--r--apps/recorder/icons.c90
-rw-r--r--apps/recorder/icons.h60
-rw-r--r--apps/recorder/peakmeter.c6
-rw-r--r--apps/recorder/radio.c25
-rw-r--r--apps/recorder/recording.c250
-rw-r--r--apps/recorder/recording.h13
-rw-r--r--apps/settings.c44
-rw-r--r--apps/settings.h34
-rw-r--r--apps/sound_menu.c288
-rw-r--r--apps/status.c6
-rw-r--r--apps/talk.c27
-rw-r--r--apps/tree.c2
-rw-r--r--bootloader/main.c4
-rw-r--r--firmware/SOURCES26
-rw-r--r--firmware/drivers/tlv320.c45
-rw-r--r--firmware/drivers/uda1380.c84
-rw-r--r--firmware/export/audio.h106
-rw-r--r--firmware/export/config-h100.h6
-rw-r--r--firmware/export/config-h120.h6
-rw-r--r--firmware/export/config-h300.h7
-rw-r--r--firmware/export/config-iaudiox5.h6
-rw-r--r--firmware/export/id3.h91
-rw-r--r--firmware/export/pcm_playback.h16
-rw-r--r--firmware/export/pcm_record.h46
-rw-r--r--firmware/export/system.h81
-rw-r--r--firmware/export/thread.h3
-rw-r--r--firmware/export/tlv320.h10
-rw-r--r--firmware/export/uda1380.h13
-rw-r--r--firmware/id3.c138
-rw-r--r--firmware/mpeg.c20
-rw-r--r--firmware/pcm_playback.c700
-rw-r--r--firmware/pcm_record.c2002
-rw-r--r--firmware/system.c3
-rw-r--r--firmware/target/coldfire/iaudio/x5/system-x5.c7
-rw-r--r--firmware/target/coldfire/iriver/system-iriver.c7
-rw-r--r--firmware/target/coldfire/system-coldfire.c7
-rw-r--r--firmware/target/coldfire/system-target.h29
-rw-r--r--firmware/thread.c8
-rw-r--r--uisimulator/sdl/lcd-charcell.c3
58 files changed, 4664 insertions, 2676 deletions
diff --git a/apps/SOURCES b/apps/SOURCES
index e1d8e7bbdd..ccfc7fa280 100644
--- a/apps/SOURCES
+++ b/apps/SOURCES
@@ -73,6 +73,9 @@ pcmbuf.c
playback.c
codecs.c
dsp.c
+#ifdef HAVE_RECORDING
+enc_config.c
+#endif
eq.c
#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
dsp_cf.S
diff --git a/apps/codecs.c b/apps/codecs.c
index f33957eba2..4491dadf49 100644
--- a/apps/codecs.c
+++ b/apps/codecs.c
@@ -50,6 +50,7 @@
#include "sound.h"
#include "database.h"
#include "splash.h"
+#include "general.h"
#ifdef SIMULATOR
#if CONFIG_CODEC == SWCODEC
@@ -104,6 +105,7 @@ struct codec_api ci = {
PREFIX(remove),
PREFIX(rename),
PREFIX(ftruncate),
+ PREFIX(fsync),
fdprintf,
read_line,
settings_parseline,
@@ -187,6 +189,7 @@ struct codec_api ci = {
get_time,
set_time,
plugin_get_audio_buffer,
+ round_value_to_list32,
#if defined(DEBUG) || defined(SIMULATOR)
debugf,
@@ -213,11 +216,11 @@ struct codec_api ci = {
false,
enc_get_inputs,
enc_set_parameters,
- enc_alloc_chunk,
- enc_free_chunk,
- enc_wavbuf_near_empty,
- enc_get_wav_data,
- &enc_set_header_callback,
+ enc_get_chunk,
+ enc_finish_chunk,
+ enc_pcm_buf_near_empty,
+ enc_get_pcm_data,
+ enc_unget_pcm_data
#endif
/* new stuff at the end, sort into place next time
@@ -225,10 +228,10 @@ struct codec_api ci = {
};
-void codec_get_full_path(char *path, const char *codec_fn)
+void codec_get_full_path(char *path, const char *codec_root_fn)
{
- /* Create full codec path */
- snprintf(path, MAX_PATH-1, CODECS_DIR "/%s", codec_fn);
+ snprintf(path, MAX_PATH-1, CODECS_DIR "/%s." CODEC_EXTENSION,
+ codec_root_fn);
}
int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap,
@@ -254,7 +257,11 @@ int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap,
hdr = (struct codec_header *)codecbuf;
if (size <= (signed)sizeof(struct codec_header)
- || hdr->magic != CODEC_MAGIC
+ || (hdr->magic != CODEC_MAGIC
+#ifdef HAVE_RECORDING
+ && hdr->magic != CODEC_ENC_MAGIC
+#endif
+ )
|| hdr->target_id != TARGET_ID
|| hdr->load_addr != codecbuf
|| hdr->end_addr > codecbuf + CODEC_SIZE)
diff --git a/apps/codecs.h b/apps/codecs.h
index 96804a889b..0b90ef9c19 100644
--- a/apps/codecs.h
+++ b/apps/codecs.h
@@ -46,7 +46,7 @@
#include "profile.h"
#endif
#if (CONFIG_CODEC == SWCODEC)
-#if !defined(SIMULATOR)
+#if !defined(SIMULATOR) && defined(HAVE_RECORDING)
#include "pcm_record.h"
#endif
#include "dsp.h"
@@ -84,15 +84,18 @@
#define PREFIX(_x_) _x_
#endif
+/* magic for normal codecs */
#define CODEC_MAGIC 0x52434F44 /* RCOD */
+/* magic for encoder codecs */
+#define CODEC_ENC_MAGIC 0x52454E43 /* RENC */
/* increase this every time the api struct changes */
-#define CODEC_API_VERSION 9
+#define CODEC_API_VERSION 10
/* update this to latest version if a change to the api struct breaks
backwards compatibility (and please take the opportunity to sort in any
new function which are "waiting" at the end of the function table) */
-#define CODEC_MIN_API_VERSION 8
+#define CODEC_MIN_API_VERSION 10
/* codec return codes */
enum codec_status {
@@ -176,6 +179,7 @@ struct codec_api {
int (*PREFIX(remove))(const char* pathname);
int (*PREFIX(rename))(const char* path, const char* newname);
int (*PREFIX(ftruncate))(int fd, off_t length);
+ int (*PREFIX(fsync))(int fd);
int (*fdprintf)(int fd, const char *fmt, ...);
int (*read_line)(int fd, char* buffer, int buffer_size);
@@ -232,7 +236,8 @@ struct codec_api {
/* sound */
void (*sound_set)(int setting, int value);
#ifndef SIMULATOR
- void (*mp3_play_data)(const unsigned char* start, int size, void (*get_more)(unsigned char** start, int* size));
+ void (*mp3_play_data)(const unsigned char* start,
+ int size, void (*get_more)(unsigned char** start, int* size));
void (*mp3_play_pause)(bool play);
void (*mp3_play_stop)(void);
bool (*mp3_is_playing)(void);
@@ -263,6 +268,10 @@ struct codec_api {
struct tm* (*get_time)(void);
int (*set_time)(const struct tm *tm);
void* (*plugin_get_audio_buffer)(int* buffer_size);
+ int (*round_value_to_list32)(unsigned long value,
+ const unsigned long list[],
+ int count,
+ bool signd);
#if defined(DEBUG) || defined(SIMULATOR)
void (*debugf)(const char *fmt, ...);
@@ -291,18 +300,14 @@ struct codec_api {
#endif
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
- bool enc_codec_loaded;
- void (*enc_get_inputs)(int *buffer_size,
- int *channels, int *quality);
- void (*enc_set_parameters)(int chunk_size, int num_chunks,
- int samp_per_chunk, char *head_ptr, int head_size,
- int enc_id);
- unsigned int* (*enc_alloc_chunk)(void);
- void (*enc_free_chunk)(void);
- int (*enc_wavbuf_near_empty)(void);
- char* (*enc_get_wav_data)(int size);
- void (**enc_set_header_callback)(void *head_buffer,
- int head_size, int num_samples, bool is_file_header);
+ volatile int enc_codec_loaded; /* <0=error, 0=pending, >0=ok */
+ void (*enc_get_inputs)(struct enc_inputs *inputs);
+ void (*enc_set_parameters)(struct enc_parameters *params);
+ struct enc_chunk_hdr * (*enc_get_chunk)(void);
+ void (*enc_finish_chunk)(void);
+ int (*enc_pcm_buf_near_empty)(void);
+ unsigned char * (*enc_get_pcm_data)(size_t size);
+ size_t (*enc_unget_pcm_data)(size_t size);
#endif
/* new stuff at the end, sort into place next time
@@ -312,34 +317,49 @@ struct codec_api {
/* codec header */
struct codec_header {
- unsigned long magic;
+ unsigned long magic; /* RCOD or RENC */
unsigned short target_id;
unsigned short api_version;
unsigned char *load_addr;
unsigned char *end_addr;
enum codec_status(*entry_point)(struct codec_api*);
};
+
#ifdef CODEC
#ifndef SIMULATOR
/* plugin_* is correct, codecs use the plugin linker script */
extern unsigned char plugin_start_addr[];
extern unsigned char plugin_end_addr[];
+/* decoders */
#define CODEC_HEADER \
const struct codec_header __header \
__attribute__ ((section (".header")))= { \
CODEC_MAGIC, TARGET_ID, CODEC_API_VERSION, \
plugin_start_addr, plugin_end_addr, codec_start };
-#else /* SIMULATOR */
+/* encoders */
+#define CODEC_ENC_HEADER \
+ const struct codec_header __header \
+ __attribute__ ((section (".header")))= { \
+ CODEC_ENC_MAGIC, TARGET_ID, CODEC_API_VERSION, \
+ plugin_start_addr, plugin_end_addr, codec_start };
+
+#else /* def SIMULATOR */
+/* decoders */
#define CODEC_HEADER \
const struct codec_header __header = { \
CODEC_MAGIC, TARGET_ID, CODEC_API_VERSION, \
NULL, NULL, codec_start };
-#endif
-#endif
+/* encoders */
+#define CODEC_ENC_HEADER \
+ const struct codec_header __header = { \
+ CODEC_ENC_MAGIC, TARGET_ID, CODEC_API_VERSION, \
+ NULL, NULL, codec_start };
+#endif /* SIMULATOR */
+#endif /* CODEC */
-/* create full codec path from filenames in audio_formats[]
+/* create full codec path from root filenames in audio_formats[]
assumes buffer size is MAX_PATH */
-void codec_get_full_path(char *path, const char *codec_fn);
+void codec_get_full_path(char *path, const char *codec_root_fn);
/* defined by the codec loader (codec.c) */
int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap,
diff --git a/apps/codecs/libwavpack/bits.c b/apps/codecs/libwavpack/bits.c
index 0a148e123f..0f0e79c292 100644
--- a/apps/codecs/libwavpack/bits.c
+++ b/apps/codecs/libwavpack/bits.c
@@ -15,6 +15,7 @@
// the malloc() system is provided.
#include "wavpack.h"
+#include "system.h"
#include <string.h>
@@ -118,19 +119,16 @@ uint32_t bs_close_write (Bitstream *bs)
void little_endian_to_native (void *data, char *format)
{
uchar *cp = (uchar *) data;
- int32_t temp;
while (*format) {
switch (*format) {
case 'L':
- temp = cp [0] + ((int32_t) cp [1] << 8) + ((int32_t) cp [2] << 16) + ((int32_t) cp [3] << 24);
- * (int32_t *) cp = temp;
+ *(long *)cp = letoh32(*(long *)cp);
cp += 4;
break;
case 'S':
- temp = cp [0] + (cp [1] << 8);
- * (short *) cp = (short) temp;
+ *(short *)cp = letoh16(*(short *)cp);
cp += 2;
break;
@@ -148,28 +146,22 @@ void little_endian_to_native (void *data, char *format)
void native_to_little_endian (void *data, char *format)
{
uchar *cp = (uchar *) data;
- int32_t temp;
while (*format) {
switch (*format) {
case 'L':
- temp = * (int32_t *) cp;
- *cp++ = (uchar) temp;
- *cp++ = (uchar) (temp >> 8);
- *cp++ = (uchar) (temp >> 16);
- *cp++ = (uchar) (temp >> 24);
+ *(long *)cp = htole32(*(long *)cp);
+ cp += 4;
break;
case 'S':
- temp = * (short *) cp;
- *cp++ = (uchar) temp;
- *cp++ = (uchar) (temp >> 8);
+ *(short *)cp = htole16(*(short *)cp);
+ cp += 2;
break;
default:
if (*format >= '0' && *format <= '9')
cp += *format - '0';
-
break;
}
diff --git a/apps/codecs/mp3_enc.c b/apps/codecs/mp3_enc.c
index 3caca94f35..cb727ce01e 100644
--- a/apps/codecs/mp3_enc.c
+++ b/apps/codecs/mp3_enc.c
@@ -32,20 +32,19 @@
#ifndef SIMULATOR
+#include <inttypes.h>
#include "codeclib.h"
-CODEC_HEADER
+CODEC_ENC_HEADER
-#ifdef USE_IRAM
-extern char iramcopy[];
-extern char iramstart[];
-extern char iramend[];
-extern char iedata[];
-extern char iend[];
-#endif
-
-
-#define SAMP_PER_FRAME 1152
+#define ENC_PADDING_FRAMES1 2
+#define ENC_PADDING_FRAMES2 4
+#define ENC_DELAY_SAMP 576
+#define ENC_DELAY_SIZE (ENC_DELAY_SAMP*4)
+#define SAMP_PER_FRAME1 1152
+#define SAMP_PER_FRAME2 576
+#define PCM_CHUNK_SIZE1 (SAMP_PER_FRAME1*4)
+#define PCM_CHUNK_SIZE2 (SAMP_PER_FRAME2*4)
#define SAMPL2 576
#define SBLIMIT 32
#define HTN 16
@@ -54,15 +53,16 @@ extern char iend[];
#define putlong(c, s) if(s+sz <= 32) { cc = (cc << s) | c; sz+= s; } \
else { putbits(cc, sz); cc = c; sz = s; }
-enum e_byte_order { order_unknown, order_bigEndian, order_littleEndian };
-
-typedef unsigned long uint32;
-typedef unsigned short uint16;
-typedef unsigned char uint8;
-
+#ifdef USE_IRAM
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+extern char iedata[];
+extern char iend[];
+#endif
typedef struct {
- int type; /* 0=(22.05,24,16kHz) 1=(44.1,48,32kHz) */
+ int type; /* 0=(MPEG2 - 22.05,24,16kHz) 1=(MPEG1 - 44.1,48,32kHz) */
int mode; /* 0=stereo, 1=jstereo, 2=dual, 3=mono */
int bitrate;
int padding;
@@ -73,21 +73,20 @@ typedef struct {
/* Side information */
typedef struct {
- uint32 part2_3_length;
+ uint32_t part2_3_length;
int count1; /* number of 0-1-quadruples */
- uint32 global_gain;
- uint32 table_select[4];
- uint32 region_0_1;
- uint32 address1;
- uint32 address2;
- uint32 address3;
+ uint32_t global_gain;
+ uint32_t table_select[4];
+ uint32_t region_0_1;
+ uint32_t address1;
+ uint32_t address2;
+ uint32_t address3;
long quantStep;
long additStep;
long max_val;
} side_info_t;
typedef struct {
- enum e_byte_order byte_order;
side_info_t cod_info[2][2];
mpeg_t mpg;
long frac_per_frame;
@@ -98,19 +97,18 @@ typedef struct {
int ResvSize;
int channels;
int granules;
- int resample;
long samplerate;
} config_t;
typedef struct {
int bitpos; /* current bitpos for writing */
- uint32 bbuf[263];
+ uint32_t bbuf[263];
} BF_Data;
struct huffcodetab {
int len; /* max. index */
- const uint8 *table; /* pointer to array[len][len] */
- const uint8 *hlen; /* pointer to array[len][len] */
+ const uint8_t *table; /* pointer to array[len][len] */
+ const uint8_t *hlen; /* pointer to array[len][len] */
};
struct huffcodebig {
@@ -127,102 +125,105 @@ struct huffcodebig {
#define shft_n(x,n) ((x) >> n)
#define SQRT 724 /* sqrt(2) * 512 */
-short mfbuf [2*(1152+512)] IBSS_ATTR; /* 3328 Bytes */
-int sb_data [2][2][18][SBLIMIT] IBSS_ATTR; /* 13824 Bytes */
-int mdct_freq [SAMPL2] IBSS_ATTR; /* 9216 Bytes */
-short enc_data [SAMPL2] IBSS_ATTR; /* 4608 Bytes */
-uint32 scalefac [23] IBSS_ATTR; /* 92 Bytes */
-BF_Data CodedData IBSS_ATTR; /* 1056 Bytes */
-int ca [8] IBSS_ATTR; /* 32 Bytes */
-int cs [8] IBSS_ATTR; /* 32 Bytes */
-int cx [9] IBSS_ATTR; /* 36 Bytes */
-int win [18][4] IBSS_ATTR; /* 288 Bytes */
-short enwindow [15*27+24] IBSS_ATTR; /* 862 Bytes */
-short int2idx [4096] IBSS_ATTR; /* 8192 Bytes */
-uint8 ht_count [2][2][16] IBSS_ATTR; /* 64 Bytes */
-uint32 tab01 [ 16] IBSS_ATTR; /* 64 Bytes */
-uint32 tab23 [ 9] IBSS_ATTR; /* 36 Bytes */
-uint32 tab56 [ 16] IBSS_ATTR; /* 64 Bytes */
-uint32 tab1315 [256] IBSS_ATTR; /* 1024 Bytes */
-uint32 tab1624 [256] IBSS_ATTR; /* 1024 Bytes */
-uint32 tab789 [ 36] IBSS_ATTR; /* 144 Bytes */
-uint32 tabABC [ 64] IBSS_ATTR; /* 256 Bytes */
-uint8 t1HB [ 4] IBSS_ATTR;
-uint8 t2HB [ 9] IBSS_ATTR;
-uint8 t3HB [ 9] IBSS_ATTR;
-uint8 t5HB [ 16] IBSS_ATTR;
-uint8 t6HB [ 16] IBSS_ATTR;
-uint8 t7HB [ 36] IBSS_ATTR;
-uint8 t8HB [ 36] IBSS_ATTR;
-uint8 t9HB [ 36] IBSS_ATTR;
-uint8 t10HB [ 64] IBSS_ATTR;
-uint8 t11HB [ 64] IBSS_ATTR;
-uint8 t12HB [ 64] IBSS_ATTR;
-uint8 t13HB [256] IBSS_ATTR;
-uint8 t15HB [256] IBSS_ATTR;
-uint16 t16HB [256] IBSS_ATTR;
-uint16 t24HB [256] IBSS_ATTR;
-uint8 t1l [ 8] IBSS_ATTR;
-uint8 t2l [ 9] IBSS_ATTR;
-uint8 t3l [ 9] IBSS_ATTR;
-uint8 t5l [ 16] IBSS_ATTR;
-uint8 t6l [ 16] IBSS_ATTR;
-uint8 t7l [ 36] IBSS_ATTR;
-uint8 t8l [ 36] IBSS_ATTR;
-uint8 t9l [ 36] IBSS_ATTR;
-uint8 t10l [ 64] IBSS_ATTR;
-uint8 t11l [ 64] IBSS_ATTR;
-uint8 t12l [ 64] IBSS_ATTR;
-uint8 t13l [256] IBSS_ATTR;
-uint8 t15l [256] IBSS_ATTR;
-uint8 t16l [256] IBSS_ATTR;
-uint8 t24l [256] IBSS_ATTR;
-struct huffcodetab ht [HTN] IBSS_ATTR;
-
-static config_t cfg;
+static short mfbuf [2*(1152+512)] IBSS_ATTR; /* 3328 Bytes */
+static int sb_data [2][2][18][SBLIMIT] IBSS_ATTR; /* 13824 Bytes */
+static int mdct_freq [SAMPL2] IBSS_ATTR; /* 9216 Bytes */
+static short enc_data [SAMPL2] IBSS_ATTR; /* 4608 Bytes */
+static uint32_t scalefac [23] IBSS_ATTR; /* 92 Bytes */
+static BF_Data CodedData IBSS_ATTR; /* 1056 Bytes */
+static int ca [8] IBSS_ATTR; /* 32 Bytes */
+static int cs [8] IBSS_ATTR; /* 32 Bytes */
+static int cx [9] IBSS_ATTR; /* 36 Bytes */
+static int win [18][4] IBSS_ATTR; /* 288 Bytes */
+static short enwindow [15*27+24] IBSS_ATTR; /* 862 Bytes */
+static short int2idx [4096] IBSS_ATTR; /* 8192 Bytes */
+static uint8_t ht_count [2][2][16] IBSS_ATTR; /* 64 Bytes */
+static uint32_t tab01 [ 16] IBSS_ATTR; /* 64 Bytes */
+static uint32_t tab23 [ 9] IBSS_ATTR; /* 36 Bytes */
+static uint32_t tab56 [ 16] IBSS_ATTR; /* 64 Bytes */
+static uint32_t tab1315 [256] IBSS_ATTR; /* 1024 Bytes */
+static uint32_t tab1624 [256] IBSS_ATTR; /* 1024 Bytes */
+static uint32_t tab789 [ 36] IBSS_ATTR; /* 144 Bytes */
+static uint32_t tabABC [ 64] IBSS_ATTR; /* 256 Bytes */
+static uint8_t t1HB [ 4] IBSS_ATTR;
+static uint8_t t2HB [ 9] IBSS_ATTR;
+static uint8_t t3HB [ 9] IBSS_ATTR;
+static uint8_t t5HB [ 16] IBSS_ATTR;
+static uint8_t t6HB [ 16] IBSS_ATTR;
+static uint8_t t7HB [ 36] IBSS_ATTR;
+static uint8_t t8HB [ 36] IBSS_ATTR;
+static uint8_t t9HB [ 36] IBSS_ATTR;
+static uint8_t t10HB [ 64] IBSS_ATTR;
+static uint8_t t11HB [ 64] IBSS_ATTR;
+static uint8_t t12HB [ 64] IBSS_ATTR;
+static uint8_t t13HB [256] IBSS_ATTR;
+static uint8_t t15HB [256] IBSS_ATTR;
+static uint16_t t16HB [256] IBSS_ATTR;
+static uint16_t t24HB [256] IBSS_ATTR;
+static uint8_t t1l [ 8] IBSS_ATTR;
+static uint8_t t2l [ 9] IBSS_ATTR;
+static uint8_t t3l [ 9] IBSS_ATTR;
+static uint8_t t5l [ 16] IBSS_ATTR;
+static uint8_t t6l [ 16] IBSS_ATTR;
+static uint8_t t7l [ 36] IBSS_ATTR;
+static uint8_t t8l [ 36] IBSS_ATTR;
+static uint8_t t9l [ 36] IBSS_ATTR;
+static uint8_t t10l [ 64] IBSS_ATTR;
+static uint8_t t11l [ 64] IBSS_ATTR;
+static uint8_t t12l [ 64] IBSS_ATTR;
+static uint8_t t13l [256] IBSS_ATTR;
+static uint8_t t15l [256] IBSS_ATTR;
+static uint8_t t16l [256] IBSS_ATTR;
+static uint8_t t24l [256] IBSS_ATTR;
+static struct huffcodetab ht [HTN] IBSS_ATTR;
+
+static unsigned pcm_chunk_size IBSS_ATTR;
+static unsigned samp_per_frame IBSS_ATTR;
+
+static config_t cfg IBSS_ATTR;
static struct codec_api *ci;
-static int enc_channels;
+static char *res_buffer;
-static const uint8 ht_count_const[2][2][16] =
+static const uint8_t ht_count_const[2][2][16] =
{ { { 1, 5, 4, 5, 6, 5, 4, 4, 7, 3, 6, 0, 7, 2, 3, 1 }, /* table0 */
{ 1, 5, 5, 7, 5, 8, 7, 9, 5, 7, 7, 9, 7, 9, 9,10 } }, /* hleng0 */
{ {15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0 }, /* table1 */
{ 4, 5, 5, 6, 5, 6, 6, 7, 5, 6, 6, 7, 6, 7, 7, 8 } } }; /* hleng1 */
-static const uint8 t1HB_const[4] = {1,1,1,0};
-static const uint8 t2HB_const[9] = {1,2,1,3,1,1,3,2,0};
-static const uint8 t3HB_const[9] = {3,2,1,1,1,1,3,2,0};
-static const uint8 t5HB_const[16] = {1,2,6,5,3,1,4,4,7,5,7,1,6,1,1,0};
-static const uint8 t6HB_const[16] = {7,3,5,1,6,2,3,2,5,4,4,1,3,3,2,0};
+static const uint8_t t1HB_const[4] = {1,1,1,0};
+static const uint8_t t2HB_const[9] = {1,2,1,3,1,1,3,2,0};
+static const uint8_t t3HB_const[9] = {3,2,1,1,1,1,3,2,0};
+static const uint8_t t5HB_const[16] = {1,2,6,5,3,1,4,4,7,5,7,1,6,1,1,0};
+static const uint8_t t6HB_const[16] = {7,3,5,1,6,2,3,2,5,4,4,1,3,3,2,0};
-static const uint8 t7HB_const[36] =
+static const uint8_t t7HB_const[36] =
{ 1, 2,10,19,16,10, 3, 3, 7,10, 5, 3,11, 4,13,17, 8, 4,
12,11,18,15,11, 2, 7, 6, 9,14, 3, 1, 6, 4, 5, 3, 2, 0 };
-static const uint8 t8HB_const[36] =
+static const uint8_t t8HB_const[36] =
{ 3, 4, 6,18,12, 5, 5, 1, 2,16, 9, 3, 7, 3, 5,14, 7, 3,
19,17,15,13,10, 4,13, 5, 8,11, 5, 1,12, 4, 4, 1, 1, 0 };
-static const uint8 t9HB_const[36] =
+static const uint8_t t9HB_const[36] =
{ 7, 5, 9,14,15, 7, 6, 4, 5, 5, 6, 7, 7, 6, 8, 8, 8, 5,
15, 6, 9,10, 5, 1,11, 7, 9, 6, 4, 1,14, 4, 6, 2, 6, 0 };
-static const uint8 t10HB_const[64] =
+static const uint8_t t10HB_const[64] =
{1,2,10,23,35,30,12,17,3,3,8,12,18,21,12,7,11,9,15,21,32,
40,19,6,14,13,22,34,46,23,18,7,20,19,33,47,27,22,9,3,31,22,
41,26,21,20,5,3,14,13,10,11,16,6,5,1,9,8,7,8,4,4,2,0 };
-static const uint8 t11HB_const[64] =
+static const uint8_t t11HB_const[64] =
{3,4,10,24,34,33,21,15,5,3,4,10,32,17,11,10,11,7,13,18,30,
31,20,5,25,11,19,59,27,18,12,5,35,33,31,58,30,16,7,5,28,26,
32,19,17,15,8,14,14,12,9,13,14,9,4,1,11,4,6,6,6,3,2,0 };
-static const uint8 t12HB_const[64] =
+static const uint8_t t12HB_const[64] =
{9,6,16,33,41,39,38,26,7,5,6,9,23,16,26,11,17,7,11,14,21,
30,10,7,17,10,15,12,18,28,14,5,32,13,22,19,18,16,9,5,40,17,
31,29,17,13,4,2,27,12,11,15,10,7,4,1,27,12,8,12,6,3,1,0 };
-static const uint8 t13HB_const[256] =
+static const uint8_t t13HB_const[256] =
{1,5,14,21,34,51,46,71,42,52,68,52,67,44,43,19,3,4,12,19,31,26,44,33,31,24,32,
24,31,35,22,14,15,13,23,36,59,49,77,65,29,40,30,40,27,33,42,16,22,20,37,61,56,
79,73,64,43,76,56,37,26,31,25,14,35,16,60,57,97,75,114,91,54,73,55,41,48,53,
@@ -234,7 +235,7 @@ static const uint8 t13HB_const[256] =
45,21,34,64,56,50,49,45,31,19,12,15,10,7,6,3,48,23,20,39,36,35,53,21,16,23,13,
10,6,1,4,2,16,15,17,27,25,20,29,11,17,12,16,8,1,1,0,1 };
-static const uint8 t15HB_const[256] =
+static const uint8_t t15HB_const[256] =
{7,12,18,53,47,76,124,108,89,123,108,119,107,81,122,63,13,5,16,27,46,36,61,51,
42,70,52,83,65,41,59,36,19,17,15,24,41,34,59,48,40,64,50,78,62,80,56,33,29,28,
25,43,39,63,55,93,76,59,93,72,54,75,50,29,52,22,42,40,67,57,95,79,72,57,89,69,
@@ -246,7 +247,7 @@ static const uint8 t15HB_const[256] =
24,16,22,13,14,7,91,44,39,38,34,63,52,45,31,52,28,19,14,8,9,3,123,60,58,53,47,
43,32,22,37,24,17,12,15,10,2,1,71,37,34,30,28,20,17,26,21,16,10,6,8,6,2,0};
-static const uint16 t16HB_const[256] =
+static const uint16_t t16HB_const[256] =
{1,5,14,44,74,63,110,93,172,149,138,242,225,195,376,17,3,4,12,20,35,62,53,47,
83,75,68,119,201,107,207,9,15,13,23,38,67,58,103,90,161,72,127,117,110,209,
206,16,45,21,39,69,64,114,99,87,158,140,252,212,199,387,365,26,75,36,68,65,
@@ -261,7 +262,7 @@ static const uint16 t16HB_const[256] =
358,711,709,866,1734,871,3458,870,434,0,12,10,7,11,10,17,11,9,13,12,10,7,5,3,
1,3};
-static const uint16 t24HB_const[256] =
+static const uint16_t t24HB_const[256] =
{15,13,46,80,146,262,248,434,426,669,653,649,621,517,1032,88,14,12,21,38,71,
130,122,216,209,198,327,345,319,297,279,42,47,22,41,74,68,128,120,221,207,194,
182,340,315,295,541,18,81,39,75,70,134,125,116,220,204,190,178,325,311,293,
@@ -276,7 +277,7 @@ static const uint16 t24HB_const[256] =
374,369,365,361,357,2,1033,280,278,274,267,264,259,382,378,372,367,363,360,
358,356,0,43,20,19,17,15,13,11,9,7,6,4,7,5,3,1,3};
-static const uint32 tab1315_const[256] =
+static const uint32_t tab1315_const[256] =
{ 0x010003,0x050005,0x070006,0x080008,0x090008,0x0a0009,0x0a000a,0x0b000a,
0x0a000a,0x0b000b,0x0c000b,0x0c000c,0x0d000c,0x0d000c,0x0e000d,0x0e000e,
0x040005,0x060005,0x080007,0x090008,0x0a0009,0x0a0009,0x0b000a,0x0b000a,
@@ -310,18 +311,18 @@ static const uint32 tab1315_const[256] =
0x0d000d,0x0e000d,0x0f000d,0x10000d,0x10000d,0x10000d,0x11000d,0x10000e,
0x11000e,0x11000e,0x12000e,0x12000e,0x15000f,0x14000f,0x15000f,0x12000f };
-static const uint32 tab01_const[16] =
+static const uint32_t tab01_const[16] =
{ 0x10004,0x50005,0x50005,0x70006,0x50005,0x80006,0x70006,0x90007,
0x50005,0x70006,0x70006,0x90007,0x70006,0x90007,0x90007,0xa0008 };
-static const uint32 tab23_const[ 9] =
+static const uint32_t tab23_const[ 9] =
{ 0x10002,0x40003,0x70007,0x40004,0x50004,0x70007,0x60006,0x70007,0x80008 };
-static const uint32 tab56_const[16] =
+static const uint32_t tab56_const[16] =
{ 0x10003,0x40004,0x70006,0x80008,0x40004,0x50004,0x80006,0x90007,
0x70005,0x80006,0x90007,0xa0008,0x80007,0x80007,0x90008,0xa0009 };
-static const uint32 tab789_const[36] =
+static const uint32_t tab789_const[36] =
{0x00100803,0x00401004,0x00701c06,0x00902407,0x00902409,0x00a0280a,0x00401004,
0x00601005,0x00801806,0x00902807,0x00902808,0x00a0280a,0x00701c05,0x00701806,
0x00902007,0x00a02808,0x00a02809,0x00b02c0a,0x00802407,0x00902807,0x00a02808,
@@ -329,7 +330,7 @@ static const uint32 tab789_const[36] =
0x00b0300a,0x00c0300b,0x00902809,0x00a02809,0x00b02c0a,0x00c02c0a,0x00c0340b,
0x00c0340b};
-static const uint32 tabABC_const[64] =
+static const uint32_t tabABC_const[64] =
{0x00100804,0x00401004,0x00701806,0x00902008,0x00a02409,0x00a0280a,0x00a0240a,
0x00b0280a,0x00401004,0x00601405,0x00801806,0x00902007,0x00a02809,0x00b02809,
0x00a0240a,0x00a0280a,0x00701806,0x00801c06,0x00902007,0x00a02408,0x00b02809,
@@ -341,7 +342,7 @@ static const uint32 tabABC_const[64] =
0x00a0240a,0x00a0240a,0x00b0280a,0x00c02c0b,0x00c0300b,0x00d0300b,0x00d0300b,
0x00d0300c};
-static const uint32 tab1624_const[256] =
+static const uint32_t tab1624_const[256] =
{0x00010004,0x00050005,0x00070007,0x00090008,0x000a0009,0x000a000a,0x000b000a,
0x000b000b,0x000c000b,0x000c000c,0x000c000c,0x000d000c,0x000d000c,0x000d000c,
0x000e000d,0x000a000a,0x00040005,0x00060006,0x00080007,0x00090008,0x000a0009,
@@ -380,34 +381,34 @@ static const uint32 tab1624_const[256] =
0x000c0009,0x000c0009,0x000c0009,0x000d0009,0x000d0009,0x000d0009,0x000d000a,
0x000d000a,0x000d000a,0x000d000a,0x000a0006};
-static const uint8 t1l_const[8] = {1,3,2,3,1,4,3,5};
-static const uint8 t2l_const[9] = {1,3,6,3,3,5,5,5,6};
-static const uint8 t3l_const[9] = {2,2,6,3,2,5,5,5,6};
-static const uint8 t5l_const[16] = {1,3,6,7,3,3,6,7,6,6,7,8,7,6,7,8};
-static const uint8 t6l_const[16] = {3,3,5,7,3,2,4,5,4,4,5,6,6,5,6,7};
+static const uint8_t t1l_const[8] = {1,3,2,3,1,4,3,5};
+static const uint8_t t2l_const[9] = {1,3,6,3,3,5,5,5,6};
+static const uint8_t t3l_const[9] = {2,2,6,3,2,5,5,5,6};
+static const uint8_t t5l_const[16] = {1,3,6,7,3,3,6,7,6,6,7,8,7,6,7,8};
+static const uint8_t t6l_const[16] = {3,3,5,7,3,2,4,5,4,4,5,6,6,5,6,7};
-static const uint8 t7l_const[36] =
+static const uint8_t t7l_const[36] =
{1,3,6,8,8,9,3,4,6,7,7,8,6,5,7,8,8,9,7,7,8,9,9,9,7,7,8,9,9,10,8,8,9,10,10,10};
-static const uint8 t8l_const[36] =
+static const uint8_t t8l_const[36] =
{2,3,6,8,8,9,3,2,4,8,8,8,6,4,6,8,8,9,8,8,8,9,9,10,8,7,8,9,10,10,9,8,9,9,11,11};
-static const uint8 t9l_const[36] =
+static const uint8_t t9l_const[36] =
{3,3,5,6,8,9,3,3,4,5,6,8,4,4,5,6,7,8,6,5,6,7,7,8,7,6,7,7,8,9,8,7,8,8,9,9};
-static const uint8 t10l_const[64] =
+static const uint8_t t10l_const[64] =
{1,3,6,8,9,9,9,10,3,4,6,7,8,9,8,8,6,6,7,8,9,10,9,9,7,7,8,9,10,10,9,10,8,8,9,10,
10,10,10,10,9,9,10,10,11,11,10,11,8,8,9,10,10,10,11,11,9,8,9,10,10,11,11,11};
-static const uint8 t11l_const[64] =
+static const uint8_t t11l_const[64] =
{2,3,5,7,8,9,8,9,3,3,4,6,8,8,7,8,5,5,6,7,8,9,8,8,7,6,7,9,8,10,8,9,8,8,8,9,9,10,
9,10,8,8,9,10,10,11,10,11,8,7,7,8,9,10,10,10,8,7,8,9,10,10,10,10};
-static const uint8 t12l_const[64] =
+static const uint8_t t12l_const[64] =
{4,3,5,7,8,9,9,9,3,3,4,5,7,7,8,8,5,4,5,6,7,8,7,8,6,5,6,6,7,8,8,8,7,6,7,7,8,
8,8,9,8,7,8,8,8,9,8,9,8,7,7,8,8,9,9,10,9,8,8,9,9,9,9,10};
-static const uint8 t13l_const[256] =
+static const uint8_t t13l_const[256] =
{1,4,6,7,8,9,9,10,9,10,11,11,12,12,13,13,3,4,6,7,8,8,9,9,9,9,10,10,11,12,12,12,
6,6,7,8,9,9,10,10,9,10,10,11,11,12,13,13,7,7,8,9,9,10,10,10,10,11,11,11,11,12,
13,13,8,7,9,9,10,10,11,11,10,11,11,12,12,13,13,14,9,8,9,10,10,10,11,11,11,11,
@@ -419,7 +420,7 @@ static const uint8 t13l_const[256] =
16,16,13,12,12,13,13,13,15,14,14,17,15,15,15,17,16,16,12,12,13,14,14,14,15,14,
15,15,16,16,19,18,19,16};
-static const uint8 t15l_const[256] =
+static const uint8_t t15l_const[256] =
{3,4,5,7,7,8,9,9,9,10,10,11,11,11,12,13,4,3,5,6,7,7,8,8,8,9,9,10,10,10,11,11,5,
5,5,6,7,7,8,8,8,9,9,10,10,11,11,11,6,6,6,7,7,8,8,9,9,9,10,10,10,11,11,11,7,6,
7,7,8,8,9,9,9,9,10,10,10,11,11,11,8,7,7,8,8,8,9,9,9,9,10,10,11,11,11,12,9,7,8,
@@ -430,7 +431,7 @@ static const uint8 t15l_const[256] =
11,11,11,11,12,12,12,12,12,13,13,12,11,11,11,11,11,11,11,12,12,12,12,13,13,12,
13,12,11,11,11,11,11,11,12,12,12,12,12,13,13,13,13};
-static const uint8 t16l_const[256] =
+static const uint8_t t16l_const[256] =
{1,4,6,8,9,9,10,10,11,11,11,12,12,12,13,9,3,4,6,7,8,9,9,9,10,10,10,11,12,11,12,
8,6,6,7,8,9,9,10,10,11,10,11,11,11,12,12,9,8,7,8,9,9,10,10,10,11,11,12,12,12,
13,13,10,9,8,9,9,10,10,11,11,11,12,12,12,13,13,13,9,9,8,9,9,10,11,11,12,11,12,
@@ -442,7 +443,7 @@ static const uint8 t16l_const[256] =
17,15,11,13,13,11,12,14,14,13,14,14,15,16,15,17,15,14,11,9,8,8,9,9,10,10,10,
11,11,11,11,11,11,11,8};
-static const uint8 t24l_const[256] =
+static const uint8_t t24l_const[256] =
{4,4,6,7,8,9,9,10,10,11,11,11,11,11,12,9,4,4,5,6,7,8,8,9,9,9,10,10,10,10,10,8,
6,5,6,7,7,8,8,9,9,9,9,10,10,10,11,7,7,6,7,7,8,8,8,9,9,9,9,10,10,10,10,7,8,7,7,
8,8,8,8,9,9,9,10,10,10,10,11,7,9,7,8,8,8,8,9,9,9,9,10,10,10,10,10,7,9,8,8,8,8,
@@ -491,8 +492,8 @@ static const struct huffcodebig ht_big[HTN] =
static const struct
{
- uint32 region0_cnt;
- uint32 region1_cnt;
+ uint32_t region0_cnt;
+ uint32_t region1_cnt;
} subdv_table[23] =
{ {0, 0}, /* 0 bands */
{0, 0}, /* 1 bands */
@@ -519,7 +520,7 @@ static const struct
{6, 7}, /* 22 bands */
};
-static const uint32 sfBand[6][23] =
+static const uint32_t sfBand[6][23] =
{
/* Table B.2.b: 22.05 kHz */
{0,6,12,18,24,30,36,44,54,66,80,96,116,140,168,200,238,284,336,396,464,522,576},
@@ -747,9 +748,13 @@ static const int order[32] =
{ 0, 1, 16, 17, 8, 9, 24, 25, 4, 5, 20, 21, 12, 13, 28, 29,
2, 3, 18, 19,10,11, 26, 27, 6, 7, 22, 23, 14, 15, 30, 31 };
-static const int bitr_index[2][15] =
-{ {0, 8,16,24,32,40,48,56, 64, 80, 96,112,128,144,160},
- {0,32,40,48,56,64,80,96,112,128,160,192,224,256,320} };
+static const long sampr_index[2][3] =
+{ { 22050, 24000, 16000 }, /* MPEG 2 */
+ { 44100, 48000, 32000 } }; /* MPEG 1 */
+
+static const long bitr_index[2][15] =
+{ {0, 8,16,24,32,40,48,56, 64, 80, 96,112,128,144,160}, /* MPEG 2 */
+ {0,32,40,48,56,64,80,96,112,128,160,192,224,256,320} }; /* MPEG 1 */
static const int num_bands[3][15] =
{ {0,10,10,10,10,12,14,16, 20, 22, 24, 26, 28, 30, 32},
@@ -837,35 +842,55 @@ static const int win_const[18][4] = {
{ 134, -146,-3352,-3072 } };
/* forward declarations */
-int HuffmanCode( short *ix, int *xr, uint32 begin, uint32 end, int table);
-int HuffmanCod1( short *ix, int *xr, uint32 begin, uint32 end, int table);
-void putbits(uint32 val, uint32 nbit);
-int find_best_2( short *ix, uint32 start, uint32 end, const uint32 *table,
- uint32 len, int *bits);
-int find_best_3( short *ix, uint32 start, uint32 end, const uint32 *table,
- uint32 len, int *bits);
-int count_bit1 ( short *ix, uint32 start, uint32 end, int *bits );
-int count_bigv ( short *ix, uint32 start, uint32 end, int table0, int table1,
+static int HuffmanCode( short *ix, int *xr, uint32_t begin, uint32_t end, int table);
+static int HuffmanCod1( short *ix, int *xr, uint32_t begin, uint32_t end, int table);
+static void putbits(uint32_t val, uint32_t nbit);
+static int find_best_2( short *ix, uint32_t start, uint32_t end, const uint32_t *table,
+ uint32_t len, int *bits);
+static int find_best_3( short *ix, uint32_t start, uint32_t end, const uint32_t *table,
+ uint32_t len, int *bits);
+static int count_bit1 ( short *ix, uint32_t start, uint32_t end, int *bits );
+static int count_bigv ( short *ix, uint32_t start, uint32_t end, int table0, int table1,
int *bits);
-void encodeSideInfo( side_info_t si[2][2] )
+static void encodeSideInfo( side_info_t si[2][2] )
{
int gr, ch, header;
- uint32 cc=0, sz=0;
+ uint32_t cc=0, sz=0;
- header = 0xfff00000;
- header |= cfg.mpg.type << 19; /* mp3 type: 1 */
- header |= 1 << 17; /* mp3 layer: 1 */
- header |= 1 << 16; /* mp3 crc: 0 */
+ /*
+ * MPEG header layout:
+ * AAAAAAAA AAABBCCD EEEEFFGH IIJJKLMM
+ * A (31-21) = frame sync
+ * B (20-19) = MPEG type
+ * C (18-17) = MPEG layer
+ * D (16) = protection bit
+ * E (15-12) = bitrate index
+ * F (11-10) = samplerate index
+ * G (9) = padding bit
+ * H (8) = private bit
+ * I (7-6) = channel mode
+ * J (5-4) = mode extension (jstereo only)
+ * K (3) = copyright bit
+ * L (2) = original
+ * M (1-0) = emphasis
+ */
+
+ header = (0xfff00000) | /* frame sync (AAAAAAAAA AAA)
+ mp3 type (upper): 1 (B) */
+ (0x01 << 17) | /* mp3 layer: 01 (CC) */
+ ( 0x1 << 16) | /* mp3 crc: 1 (D) */
+ ( 0x1 << 2); /* mp3 org: 1 (L) */
+ header |= cfg.mpg.type << 19;
header |= cfg.mpg.bitr_id << 12;
header |= cfg.mpg.smpl_id << 10;
header |= cfg.mpg.padding << 9;
header |= cfg.mpg.mode << 6;
- header |= 1 << 2; /* mp3 original: 1 */
+ /* no emphasis (bits 0-1) */
putbits( header, 32 );
- if(cfg.mpg.type)
+ if(cfg.mpg.type == 1)
{ /* MPEG1 */
if(cfg.channels == 2) { putlong( 0, 20); }
else { putlong( 0, 18); }
@@ -910,7 +935,7 @@ void encodeSideInfo( side_info_t si[2][2] )
/* Note the discussion of huffmancodebits() on pages 28 and 29 of the IS,
as well as the definitions of the side information on pages 26 and 27. */
-void Huffmancodebits( short *ix, int *xr, side_info_t *gi )
+static void Huffmancodebits( short *ix, int *xr, side_info_t *gi )
{
int region1 = gi->address1;
int region2 = gi->address2;
@@ -944,10 +969,10 @@ void Huffmancodebits( short *ix, int *xr, side_info_t *gi )
}
}
-int HuffmanCod1( short *ix, int *xr, uint32 begin, uint32 end, int tbl)
+int HuffmanCod1( short *ix, int *xr, uint32_t begin, uint32_t end, int tbl)
{
- uint32 cc=0, sz=0;
- uint32 i, d, p;
+ uint32_t cc=0, sz=0;
+ uint32_t i, d, p;
int sumbit=0, s=0, l=0, v, w, x, y;
#define sgnv (xr[i+0] < 0 ? 1 : 0)
#define sgnw (xr[i+1] < 0 ? 1 : 0)
@@ -995,10 +1020,10 @@ int HuffmanCod1( short *ix, int *xr, uint32 begin, uint32 end, int tbl)
}
/* Implements the pseudocode of page 98 of the IS */
-int HuffmanCode( short *ix, int *xr, uint32 begin, uint32 end, int table)
+int HuffmanCode( short *ix, int *xr, uint32_t begin, uint32_t end, int table)
{
- uint32 cc=0, sz=0, code;
- uint32 i, xl=0, yl=0, idx;
+ uint32_t cc=0, sz=0, code;
+ uint32_t i, xl=0, yl=0, idx;
int x, y, bit, sumbit=0;
#define sign_x (xr[i+0] < 0 ? 1 : 0)
#define sign_y (xr[i+1] < 0 ? 1 : 0)
@@ -1008,9 +1033,9 @@ int HuffmanCode( short *ix, int *xr, uint32 begin, uint32 end, int table)
if( table > 15 )
{ /* ESC-table is used */
- uint32 linbits = ht_big[table-16].linbits;
- uint16 *hffcode = table < 24 ? t16HB : t24HB;
- uint8 *hlen = table < 24 ? t16l : t24l;
+ uint32_t linbits = ht_big[table-16].linbits;
+ uint16_t *hffcode = table < 24 ? t16HB : t24HB;
+ uint8_t *hlen = table < 24 ? t16l : t24l;
for(i=begin; i<end; i+=2)
{
@@ -1088,14 +1113,14 @@ int HuffmanCode( short *ix, int *xr, uint32 begin, uint32 end, int table)
return sumbit;
}
-void putbits(uint32 val, uint32 nbit)
+void putbits(uint32_t val, uint32_t nbit)
{
int new_bitpos = CodedData.bitpos + nbit;
int ptrpos = CodedData.bitpos >> 5;
val = val & (0xffffffff >> (32 - nbit));
- /* data fit in one uint32 */
+ /* data fit in one uint32_t */
if(((new_bitpos - 1) >> 5) == ptrpos)
CodedData.bbuf[ptrpos] |= val << ((32 - new_bitpos) & 31);
else
@@ -1114,9 +1139,9 @@ void putbits(uint32 val, uint32 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 begin, uint32 end, int *bits )
+int choose_table( short *ix, uint32_t begin, uint32_t end, int *bits )
{
- uint32 i;
+ uint32_t i;
int max, table0, table1;
for(i=begin,max=0; i<end; i++)
@@ -1158,10 +1183,10 @@ int choose_table( short *ix, uint32 begin, uint32 end, int *bits )
}
}
-int find_best_2(short *ix, uint32 start, uint32 end, const uint32 *table,
- uint32 len, int *bits)
+int find_best_2(short *ix, uint32_t start, uint32_t end, const uint32_t *table,
+ uint32_t len, int *bits)
{
- uint32 i, sum = 0;
+ uint32_t i, sum = 0;
for(i=start; i<end; i+=2)
sum += table[ix[i] * len + ix[i+1]];
@@ -1178,10 +1203,10 @@ int find_best_2(short *ix, uint32 start, uint32 end, const uint32 *table,
}
}
-int find_best_3(short *ix, uint32 start, uint32 end, const uint32 *table,
- uint32 len, int *bits)
+int find_best_3(short *ix, uint32_t start, uint32_t end, const uint32_t *table,
+ uint32_t len, int *bits)
{
- uint32 i, j, sum = 0;
+ uint32_t i, j, sum = 0;
int sum1 = 0;
int sum2 = 0;
int sum3 = 0;
@@ -1211,9 +1236,9 @@ int find_best_3(short *ix, uint32 start, uint32 end, const uint32 *table,
/*************************************************************************/
/* Function: Count the number of bits necessary to code the subregion. */
/*************************************************************************/
-int count_bit1(short *ix, uint32 start, uint32 end, int *bits )
+int count_bit1(short *ix, uint32_t start, uint32_t end, int *bits )
{
- uint32 i, sum = 0;
+ uint32_t i, sum = 0;
for(i=start; i<end; i+=2)
sum += t1l[4 + ix[i] * 2 + ix[i+1]];
@@ -1223,10 +1248,10 @@ int count_bit1(short *ix, uint32 start, uint32 end, int *bits )
return 1; /* this is table1 */
}
-int count_bigv(short *ix, uint32 start, uint32 end, int table0,
+int count_bigv(short *ix, uint32_t start, uint32_t end, int table0,
int table1, int *bits )
{
- uint32 i, sum0, sum1, sum=0, bigv=0, x, y;
+ uint32_t i, sum0, sum1, sum=0, bigv=0, x, y;
/* ESC-table is used */
for(i=start; i<end; i+=2)
@@ -1264,7 +1289,7 @@ int calc_runlen( short *ix, side_info_t *si )
int p, i, sum = 0;
for(i=SAMPL2; i-=2; )
- if(*(uint32*)&ix[i-2]) /* !!!! short *ix; !!!!! */
+ if(*(uint32_t*)&ix[i-2]) /* !!!! short *ix; !!!!! */
break;
si->count1 = 0;
@@ -1899,276 +1924,275 @@ void mdct_long(int *out, int *in)
out[16] = ct - st;
}
-static int find_bitrate_index(int type, int bitrate)
+static int find_bitrate_index(int type, int bitrate, bool stereo)
{
- int i;
+ if (type == 1 && !stereo && bitrate > 160)
+ bitrate = 160;
- for(i=0;i<14;i++)
- if(bitrate == bitr_index[type][i])
- break;
-
- return i;
+ return ci->round_value_to_list32(bitrate,
+ &bitr_index[type][1], 14, true) + 1;
}
static int find_samplerate_index(long freq, int *mp3_type)
-{ /* MPEG 2 */ /* MPEG 1 */
- static long mpeg[2][3] = { {22050, 24000, 16000}, {44100, 48000, 32000} };
- int mpg, rate;
-
- /* set default values: MPEG1 at 44100/s */
- *mp3_type = 1;
-
- for(mpg=0; mpg<2; mpg++)
- for(rate=0; rate<3; rate++)
- if(freq == mpeg[mpg][rate])
- { *mp3_type = mpg; return rate; }
-
- return 0;
+{
+ int mpeg = freq >= (32000+24000)/2 ? 1 : 0;
+ int i = ci->round_value_to_list32(freq, sampr_index[mpeg], 3, true);
+ *mp3_type = mpeg;
+ return i;
}
-void init_mp3_encoder_engine(bool stereo, int quality, int sample_rate)
+bool init_mp3_encoder_engine(int sample_rate,
+ int num_channels,
+ struct encoder_config *enc_cfg)
{
- /* keep in sync with rec_quality_info_afmt in id3.h/.c */
- static int bitr_s[9] = { 64, 96, 128, 160, 192, 224, 320, 64, 64 };
- static int bitr_m[9] = { 64, 96, 128, 160, 160, 160, 160, 64, 64 };
- uint32 avg_byte_per_frame;
-
- if(quality == 0 && stereo && sample_rate >= 32000)
- { /* use MPEG2 format */
- sample_rate >>= 1;
- cfg.resample = 1;
- cfg.granules = 1;
- }
- else
- { /* use MPEG1 format */
- cfg.resample = 0;
- cfg.granules = 2;
- }
+ const bool stereo = num_channels > 1;
+ uint32_t avg_byte_per_frame;
+
+ cfg.channels = stereo ? 2 : 1;
+ cfg.mpg.mode = stereo ? 0 : 3; /* 0=stereo, 3=mono */
+ cfg.mpg.smpl_id = find_samplerate_index(sample_rate, &cfg.mpg.type);
+ cfg.samplerate = sampr_index[cfg.mpg.type][cfg.mpg.smpl_id];
+ cfg.mpg.bitr_id = find_bitrate_index(cfg.mpg.type,
+ enc_cfg->mp3_enc.bitrate,
+ stereo);
+ cfg.mpg.bitrate = bitr_index[cfg.mpg.type][cfg.mpg.bitr_id];
+ cfg.mpg.num_bands = num_bands[stereo ? cfg.mpg.type : 2][cfg.mpg.bitr_id];
+
+ if (cfg.mpg.type == 1)
+ {
+ cfg.granules = 2;
+ pcm_chunk_size = PCM_CHUNK_SIZE1;
+ samp_per_frame = SAMP_PER_FRAME1;
+ }
+ else
+ {
+ cfg.granules = 1;
+ pcm_chunk_size = PCM_CHUNK_SIZE2;
+ samp_per_frame = SAMP_PER_FRAME2;
+ }
- cfg.byte_order = order_bigEndian;
- cfg.samplerate = sample_rate;
- cfg.channels = stereo ? 2 : 1;
- cfg.mpg.mode = stereo ? 0 : 3; /* 0=stereo, 3=mono */
- cfg.mpg.bitrate = stereo ? bitr_s[quality] : bitr_m[quality];
- cfg.mpg.smpl_id = find_samplerate_index(cfg.samplerate, &cfg.mpg.type);
- cfg.mpg.bitr_id = find_bitrate_index(cfg.mpg.type, cfg.mpg.bitrate);
- cfg.mpg.num_bands = num_bands[stereo ? cfg.mpg.type : 2][cfg.mpg.bitr_id];
-
- memcpy(scalefac, sfBand[cfg.mpg.smpl_id + 3*cfg.mpg.type], sizeof(scalefac));
- memset(mfbuf , 0 , sizeof(mfbuf ));
- memset(mdct_freq , 0 , sizeof(mdct_freq ));
- memset(enc_data , 0 , sizeof(enc_data ));
- memset(sb_data , 0 , sizeof(sb_data ));
- memset(&CodedData, 0 , sizeof(CodedData ));
- memcpy(ca , ca_const , sizeof(ca ));
- memcpy(cs , cs_const , sizeof(cs ));
- memcpy(cx , cx_const , sizeof(cx ));
- memcpy(win , win_const , sizeof(win ));
- memcpy(enwindow , enwindow_const , sizeof(enwindow ));
- memcpy(int2idx , int2idx_const , sizeof(int2idx ));
- memcpy(ht_count , ht_count_const , sizeof(ht_count ));
- memcpy( tab01 , tab01_const , sizeof(tab01 ));
- memcpy( tab23 , tab23_const , sizeof(tab23 ));
- memcpy( tab56 , tab56_const , sizeof(tab56 ));
- memcpy( tab1315 , tab1315_const , sizeof(tab1315 ));
- memcpy( tab1624 , tab1624_const , sizeof(tab1624 ));
- memcpy( tab789 , tab789_const , sizeof(tab789 ));
- memcpy( tabABC , tabABC_const , sizeof(tabABC ));
- memcpy( t1HB , t1HB_const , sizeof(t1HB ));
- memcpy( t2HB , t2HB_const , sizeof(t2HB ));
- memcpy( t3HB , t3HB_const , sizeof(t3HB ));
- memcpy( t5HB , t5HB_const , sizeof(t5HB ));
- memcpy( t6HB , t6HB_const , sizeof(t6HB ));
- memcpy( t7HB , t7HB_const , sizeof(t7HB ));
- memcpy( t8HB , t8HB_const , sizeof(t8HB ));
- memcpy( t9HB , t9HB_const , sizeof(t9HB ));
- memcpy(t10HB , t10HB_const , sizeof(t10HB ));
- memcpy(t11HB , t11HB_const , sizeof(t11HB ));
- memcpy(t12HB , t12HB_const , sizeof(t12HB ));
- memcpy(t13HB , t13HB_const , sizeof(t13HB ));
- memcpy(t15HB , t15HB_const , sizeof(t15HB ));
- memcpy(t16HB , t16HB_const , sizeof(t16HB ));
- memcpy(t24HB , t24HB_const , sizeof(t24HB ));
- memcpy( t1l , t1l_const , sizeof(t1l ));
- memcpy( t2l , t2l_const , sizeof(t2l ));
- memcpy( t3l , t3l_const , sizeof(t3l ));
- memcpy( t5l , t5l_const , sizeof(t5l ));
- memcpy( t6l , t6l_const , sizeof(t6l ));
- memcpy( t7l , t7l_const , sizeof(t7l ));
- memcpy( t8l , t8l_const , sizeof(t8l ));
- memcpy( t9l , t9l_const , sizeof(t9l ));
- memcpy(t10l , t10l_const , sizeof(t10l ));
- memcpy(t11l , t11l_const , sizeof(t11l ));
- memcpy(t12l , t12l_const , sizeof(t12l ));
- memcpy(t13l , t13l_const , sizeof(t13l ));
- memcpy(t15l , t15l_const , sizeof(t15l ));
- memcpy(t16l , t16l_const , sizeof(t16l ));
- memcpy(t24l , t24l_const , sizeof(t24l ));
- memcpy(ht , ht_const , sizeof(ht ));
-
- ht[ 0].table = NULL; ht[ 0].hlen = NULL; /* Apparently not used */
- ht[ 1].table = t1HB; ht[ 1].hlen = t1l;
- ht[ 2].table = t2HB; ht[ 2].hlen = t2l;
- ht[ 3].table = t3HB; ht[ 3].hlen = t3l;
- ht[ 4].table = NULL; ht[ 4].hlen = NULL; /* Apparently not used */
- ht[ 5].table = t5HB; ht[ 5].hlen = t5l;
- ht[ 6].table = t6HB; ht[ 6].hlen = t6l;
- ht[ 7].table = t7HB; ht[ 7].hlen = t7l;
- ht[ 8].table = t8HB; ht[ 8].hlen = t8l;
- ht[ 9].table = t9HB; ht[ 9].hlen = t9l;
- ht[10].table = t10HB; ht[10].hlen = t10l;
- ht[11].table = t11HB; ht[11].hlen = t11l;
- ht[12].table = t12HB; ht[12].hlen = t12l;
- ht[13].table = t13HB; ht[13].hlen = t13l;
- ht[14].table = NULL; ht[14].hlen = NULL; /* Apparently not used */
- ht[15].table = t15HB; ht[15].hlen = t15l;
-
- /* Figure average number of 'bytes' per frame */
- avg_byte_per_frame = SAMPL2 * 16000 * cfg.mpg.bitrate / (2 - cfg.mpg.type);
- avg_byte_per_frame = avg_byte_per_frame / cfg.samplerate;
- cfg.byte_per_frame = avg_byte_per_frame / 64;
- cfg.frac_per_frame = avg_byte_per_frame & 63;
- cfg.slot_lag = 0;
- cfg.sideinfo_len = 32 + (cfg.mpg.type ? (cfg.channels == 1 ? 136 : 256)
- : (cfg.channels == 1 ? 72 : 136));
+ memcpy(scalefac, sfBand[cfg.mpg.smpl_id + 3*cfg.mpg.type], sizeof(scalefac));
+ memset(mfbuf , 0 , sizeof(mfbuf ));
+ memset(mdct_freq , 0 , sizeof(mdct_freq ));
+ memset(enc_data , 0 , sizeof(enc_data ));
+ memset(sb_data , 0 , sizeof(sb_data ));
+ memset(&CodedData, 0 , sizeof(CodedData ));
+ memcpy(ca , ca_const , sizeof(ca ));
+ memcpy(cs , cs_const , sizeof(cs ));
+ memcpy(cx , cx_const , sizeof(cx ));
+ memcpy(win , win_const , sizeof(win ));
+ memcpy(enwindow , enwindow_const , sizeof(enwindow ));
+ memcpy(int2idx , int2idx_const , sizeof(int2idx ));
+ memcpy(ht_count , ht_count_const , sizeof(ht_count ));
+ memcpy( tab01 , tab01_const , sizeof(tab01 ));
+ memcpy( tab23 , tab23_const , sizeof(tab23 ));
+ memcpy( tab56 , tab56_const , sizeof(tab56 ));
+ memcpy( tab1315 , tab1315_const , sizeof(tab1315 ));
+ memcpy( tab1624 , tab1624_const , sizeof(tab1624 ));
+ memcpy( tab789 , tab789_const , sizeof(tab789 ));
+ memcpy( tabABC , tabABC_const , sizeof(tabABC ));
+ memcpy( t1HB , t1HB_const , sizeof(t1HB ));
+ memcpy( t2HB , t2HB_const , sizeof(t2HB ));
+ memcpy( t3HB , t3HB_const , sizeof(t3HB ));
+ memcpy( t5HB , t5HB_const , sizeof(t5HB ));
+ memcpy( t6HB , t6HB_const , sizeof(t6HB ));
+ memcpy( t7HB , t7HB_const , sizeof(t7HB ));
+ memcpy( t8HB , t8HB_const , sizeof(t8HB ));
+ memcpy( t9HB , t9HB_const , sizeof(t9HB ));
+ memcpy(t10HB , t10HB_const , sizeof(t10HB ));
+ memcpy(t11HB , t11HB_const , sizeof(t11HB ));
+ memcpy(t12HB , t12HB_const , sizeof(t12HB ));
+ memcpy(t13HB , t13HB_const , sizeof(t13HB ));
+ memcpy(t15HB , t15HB_const , sizeof(t15HB ));
+ memcpy(t16HB , t16HB_const , sizeof(t16HB ));
+ memcpy(t24HB , t24HB_const , sizeof(t24HB ));
+ memcpy( t1l , t1l_const , sizeof(t1l ));
+ memcpy( t2l , t2l_const , sizeof(t2l ));
+ memcpy( t3l , t3l_const , sizeof(t3l ));
+ memcpy( t5l , t5l_const , sizeof(t5l ));
+ memcpy( t6l , t6l_const , sizeof(t6l ));
+ memcpy( t7l , t7l_const , sizeof(t7l ));
+ memcpy( t8l , t8l_const , sizeof(t8l ));
+ memcpy( t9l , t9l_const , sizeof(t9l ));
+ memcpy(t10l , t10l_const , sizeof(t10l ));
+ memcpy(t11l , t11l_const , sizeof(t11l ));
+ memcpy(t12l , t12l_const , sizeof(t12l ));
+ memcpy(t13l , t13l_const , sizeof(t13l ));
+ memcpy(t15l , t15l_const , sizeof(t15l ));
+ memcpy(t16l , t16l_const , sizeof(t16l ));
+ memcpy(t24l , t24l_const , sizeof(t24l ));
+ memcpy(ht , ht_const , sizeof(ht ));
+
+ ht[ 0].table = NULL; ht[ 0].hlen = NULL; /* Apparently not used */
+ ht[ 1].table = t1HB; ht[ 1].hlen = t1l;
+ ht[ 2].table = t2HB; ht[ 2].hlen = t2l;
+ ht[ 3].table = t3HB; ht[ 3].hlen = t3l;
+ ht[ 4].table = NULL; ht[ 4].hlen = NULL; /* Apparently not used */
+ ht[ 5].table = t5HB; ht[ 5].hlen = t5l;
+ ht[ 6].table = t6HB; ht[ 6].hlen = t6l;
+ ht[ 7].table = t7HB; ht[ 7].hlen = t7l;
+ ht[ 8].table = t8HB; ht[ 8].hlen = t8l;
+ ht[ 9].table = t9HB; ht[ 9].hlen = t9l;
+ ht[10].table = t10HB; ht[10].hlen = t10l;
+ ht[11].table = t11HB; ht[11].hlen = t11l;
+ ht[12].table = t12HB; ht[12].hlen = t12l;
+ ht[13].table = t13HB; ht[13].hlen = t13l;
+ ht[14].table = NULL; ht[14].hlen = NULL; /* Apparently not used */
+ ht[15].table = t15HB; ht[15].hlen = t15l;
+
+ /* Figure average number of 'bytes' per frame */
+ avg_byte_per_frame = SAMPL2 * 16000 * cfg.mpg.bitrate / (2 - cfg.mpg.type);
+ avg_byte_per_frame = avg_byte_per_frame / cfg.samplerate;
+ cfg.byte_per_frame = avg_byte_per_frame / 64;
+ cfg.frac_per_frame = avg_byte_per_frame & 63;
+ cfg.slot_lag = 0;
+ cfg.sideinfo_len = 32 + (cfg.mpg.type ? (cfg.channels == 1 ? 136 : 256)
+ : (cfg.channels == 1 ? 72 : 136));
+
+ return true;
}
-
-enum codec_status codec_start(struct codec_api* api)
+static void to_mono_mm(void) ICODE_ATTR;
+static void to_mono_mm(void)
{
- int i, ii, gr, k, ch, shift, gr_cnt;
- int max, min;
- long *buffer;
- int chunk_size, num_chunks;
- int enc_buffer_size;
- int enc_quality;
- uint32 *mp3_chunk_ptr;
- bool cpu_boosted = true; /* start boosted */
+ /* |llllllllllllllll|rrrrrrrrrrrrrrrr| =>
+ * |mmmmmmmmmmmmmmmm|mmmmmmmmmmmmmmmm|
+ */
+ uint32_t *samp = (uint32_t *)&mfbuf[2*512];
+ uint32_t *samp_end = samp + samp_per_frame;
- /* Generic codec initialisation */
- ci = api;
-
-#ifdef USE_IRAM
- memcpy(iramstart, iramcopy, iramend - iramstart);
- memset(iedata, 0, iend - iedata);
-#endif
-
- if(ci->enc_get_inputs == NULL ||
- ci->enc_set_parameters == NULL ||
- ci->enc_alloc_chunk == NULL ||
- ci->enc_free_chunk == NULL ||
- ci->enc_wavbuf_near_empty == NULL ||
- ci->enc_get_wav_data == NULL ||
- ci->enc_set_header_callback == NULL )
- return CODEC_ERROR;
+ inline void to_mono(uint32_t **samp)
+ {
+ int32_t lr = **samp;
+ int32_t m = ((int16_t)lr + (lr >> 16)) >> 1;
+ *(*samp)++ = (m << 16) | (uint16_t)m;
+ } /* to_mono */
- ci->cpu_boost(true);
+ do
+ {
+ to_mono(&samp);
+ to_mono(&samp);
+ to_mono(&samp);
+ to_mono(&samp);
+ to_mono(&samp);
+ to_mono(&samp);
+ to_mono(&samp);
+ to_mono(&samp);
+ }
+ while (samp < samp_end);
+} /* to_mono_mm */
+
+#ifdef ROCKBOX_LITTLE_ENDIAN
+/* Swaps a frame to big endian */
+static inline void byte_swap_frame32(uint32_t *dst, uint32_t *src,
+ size_t size) ICODE_ATTR;
+static inline void byte_swap_frame32(uint32_t *dst, uint32_t *src,
+ size_t size)
+{
+ uint32_t *src_end = SKIPBYTES(src, size);
- *ci->enc_set_header_callback = NULL;
- ci->enc_get_inputs(&enc_buffer_size, &enc_channels, &enc_quality);
+ do
+ {
+ *dst++ = swap32(*src++);
+ *dst++ = swap32(*src++);
+ *dst++ = swap32(*src++);
+ *dst++ = swap32(*src++);
+ *dst++ = swap32(*src++);
+ *dst++ = swap32(*src++);
+ *dst++ = swap32(*src++);
+ *dst++ = swap32(*src++);
+ }
+ while(src < src_end);
+} /* byte_swap_frame32 */
+#endif /* ROCKBOX_LITTLE_ENDIAN */
- init_mp3_encoder_engine(enc_channels == 2, enc_quality, 44100);
+static void encode_frame(char *buffer, struct enc_chunk_hdr *chunk) ICODE_ATTR;
+static void encode_frame(char *buffer, struct enc_chunk_hdr *chunk)
+{
+ int gr, gr_cnt;
+ int max, min;
- /* must be 4byte aligned */
- chunk_size = (sizeof(long) + cfg.byte_per_frame + 1 + 3) & ~3;
- num_chunks = enc_buffer_size / chunk_size;
+ /* encode one mp3 frame in this loop */
+ CodedData.bitpos = 0;
+ memset(CodedData.bbuf, 0, sizeof(CodedData.bbuf));
- /* inform the main program about the buffer dimensions */
- ci->enc_set_parameters(chunk_size, num_chunks, SAMP_PER_FRAME,
- NULL, 0, AFMT_MPA_L3);
+ if((cfg.slot_lag += cfg.frac_per_frame) >= 64)
+ { /* Padding for this frame */
+ cfg.slot_lag -= 64;
+ cfg.mpg.padding = 1;
+ }
+ else
+ cfg.mpg.padding = 0;
-#ifdef CPU_COLDFIRE
- asm volatile ("move.l #0, %macsr"); /* integer mode */
-#endif
+ cfg.mean_bits = (8 * cfg.byte_per_frame + 8 * cfg.mpg.padding
+ - cfg.sideinfo_len) / cfg.granules / cfg.channels;
- /* main application waits for this flag during encoder loading */
- ci->enc_codec_loaded = true;
+ /* shift out old samples */
+ memcpy(mfbuf, mfbuf + 2*cfg.granules*576, 4*512);
- /* main encoding loop */
- while(!ci->stop_codec)
+ if (chunk->flags & CHUNKF_START_FILE)
{
- while((buffer = (long*)ci->enc_get_wav_data(SAMP_PER_FRAME*4)) != NULL)
- {
- if(ci->stop_codec)
- break;
-
- if(ci->enc_wavbuf_near_empty() == 0)
- {
- if(!cpu_boosted)
- {
- ci->cpu_boost(true);
- cpu_boosted = true;
- }
- }
+ /* prefix silent samples for encoder delay */
+ memset(mfbuf + 2*512, 0, ENC_DELAY_SIZE);
+ /* read new samples to iram for further processing */
+ memcpy(mfbuf + 2*512 + ENC_DELAY_SIZE/2,
+ buffer, pcm_chunk_size - ENC_DELAY_SIZE);
+ chunk->num_pcm = samp_per_frame - ENC_DELAY_SAMP;
+ }
+ else
+ {
+ /* read new samples to iram for further processing */
+ memcpy(mfbuf + 2*512, buffer, pcm_chunk_size);
+ chunk->num_pcm = samp_per_frame;
+ }
- /* encode one mp3 frame in this loop */
- CodedData.bitpos = 0;
- memset(CodedData.bbuf, 0, sizeof(CodedData.bbuf));
-
- if((cfg.slot_lag += cfg.frac_per_frame) >= 64)
- { /* Padding for this frame */
- cfg.slot_lag -= 64;
- cfg.mpg.padding = 1;
- }
- else
- cfg.mpg.padding = 0;
+ if (cfg.channels == 1)
+ to_mono_mm();
- cfg.mean_bits = (8 * cfg.byte_per_frame + 8 * cfg.mpg.padding
- - cfg.sideinfo_len) / cfg.granules / cfg.channels;
+ cfg.ResvSize = 0;
+ gr_cnt = cfg.granules * cfg.channels;
+ CodedData.bitpos = cfg.sideinfo_len; /* leave space for mp3 header */
- /* shift out old samples */
- memcpy(mfbuf, mfbuf + 2*cfg.granules*576, 4*512);
- /* read new samples to iram for further processing */
- memcpy((uint32*)(mfbuf + 2*512), buffer, 4*SAMP_PER_FRAME);
+ for(gr=0; gr<cfg.granules; gr++)
+ {
+ short *wk = mfbuf + 2*286 + gr*1152;
+ int ch;
- if(cfg.resample) /* downsample to half of original */
- for(i=2*512; i<2*512+2*SAMP_PER_FRAME; i+=4)
- {
- mfbuf[i/2+512] = (short)(((int)mfbuf[i+0] + mfbuf[i+2]) >> 1);
- mfbuf[i/2+513] = (short)(((int)mfbuf[i+1] + mfbuf[i+3]) >> 1);
- }
+ /* 16bit packed wav data can be windowed efficiently on coldfire */
+ window_subband1(wk, sb_data[0][1-gr][0], sb_data[1][1-gr][0]);
- if(cfg.channels == 1) /* mix left and right channels to mono */
- for(i=2*512; i<2*512+2*SAMP_PER_FRAME; i+=2)
- mfbuf[i] = mfbuf[i+1] = (short)(((int)mfbuf[i] + mfbuf[i+1]) >> 1);
+ for(ch=0; ch<cfg.channels; ch++)
+ {
+ int ii, k, shift;
- cfg.ResvSize = 0;
- gr_cnt = cfg.granules * cfg.channels;
- CodedData.bitpos = cfg.sideinfo_len; /* leave space for mp3 header */
+ wk = mfbuf + 2*286 + gr*1152 + ch;
- for(gr=0; gr<cfg.granules; gr++)
+ /* 36864=4*18*16*32 */
+ for(k=0; k<18; k++, wk+=64)
{
- short *wk = mfbuf + 2*286 + gr*1152;
-
- /* 16bit packed wav data can be windowed efficiently on coldfire */
- window_subband1(wk, sb_data[0][1-gr][0], sb_data[1][1-gr][0]);
-
- for(ch=0; ch<cfg.channels; ch++)
- {
- int band;
- int *mdct;
-
- wk = mfbuf + 2*286 + gr*1152 + ch;
-
- /* 36864=4*18*16*32 */
- for(k=0; k<18; k++, wk+=64)
+ window_subband2(wk, sb_data[ch][1-gr][k]);
+ /* Compensate for inversion in the analysis filter */
+ if(k & 1)
{
- window_subband2(wk, sb_data[ch][1-gr][k]);
- /* Compensate for inversion in the analysis filter */
- if(k & 1)
+ int band;
for(band=1; band<32; band+=2)
- sb_data[ch][1-gr][k][band] *= -1;
+ sb_data[ch][1-gr][k][band] *= -1;
}
+ }
- /* Perform imdct of 18 previous + 18 current subband samples */
- /* for integer precision do this loop twice (if neccessary) */
- shift = k = 14;
- for(ii=0; ii<2 && k; ii++)
+ /* Perform imdct of 18 previous + 18 current subband samples
+ for integer precision do this loop twice (if neccessary)
+ */
+ shift = k = 14;
+ for(ii=0; ii<2 && k; ii++)
+ {
+ int *mdct = mdct_freq;
+ int band;
+
+ cfg.cod_info[gr][ch].additStep = 4 * (14 - shift);
+
+ for(band=0; band<cfg.mpg.num_bands; band++, mdct+=18)
{
- mdct = mdct_freq;
- cfg.cod_info[gr][ch].additStep = 4 * (14 - shift);
- for(band=0; band<cfg.mpg.num_bands; band++, mdct+=18)
- {
int *band0 = sb_data[ch][ gr][0] + order[band];
int *band1 = sb_data[ch][1-gr][0] + order[band];
int work[18];
@@ -2176,16 +2200,20 @@ enum codec_status codec_start(struct codec_api* api)
/* 9216=4*32*9*8 */
for(k=-9; k<0; k++)
{
- int a = shft_n(band1[(k+9)*32], shift);
- int b = shft_n(band1[(8-k)*32], shift);
- int c = shft_n(band0[(k+9)*32], shift);
- int d = shft_n(band0[(8-k)*32], shift);
-
- work[k+ 9] = shft16(a * win[k+ 9][0] + b * win[k+ 9][1]
- + c * win[k+ 9][2] + d * win[k+ 9][3]);
-
- work[k+18] = shft16(c * win[k+18][0] + d * win[k+18][1]
- + a * win[k+18][2] + b * win[k+18][3]);
+ int a = shft_n(band1[(k+9)*32], shift);
+ int b = shft_n(band1[(8-k)*32], shift);
+ int c = shft_n(band0[(k+9)*32], shift);
+ int d = shft_n(band0[(8-k)*32], shift);
+
+ work[k+ 9] = shft16(a * win[k+ 9][0] +
+ b * win[k+ 9][1] +
+ c * win[k+ 9][2] +
+ d * win[k+ 9][3]);
+
+ work[k+18] = shft16(c * win[k+18][0] +
+ d * win[k+18][1] +
+ a * win[k+18][2] +
+ b * win[k+18][3]);
}
/* 7200=4*18*100 */
@@ -2193,67 +2221,309 @@ enum codec_status codec_start(struct codec_api* api)
/* Perform aliasing reduction butterfly */
if(band != 0)
- for(k=7; k>=0; --k)
- {
- int bu, bd;
- bu = shft15(mdct[k]) * ca[k] + shft15(mdct[-1-k]) * cs[k];
- bd = shft15(mdct[k]) * cs[k] - shft15(mdct[-1-k]) * ca[k];
- mdct[-1-k] = bu;
- mdct[ k ] = bd;
- }
- }
-
- max = min = 0;
- for(k=0; k<576; k++)
- {
+ {
+ for(k=7; k>=0; --k)
+ {
+ int bu, bd;
+ bu = shft15(mdct[k]) * ca[k] +
+ shft15(mdct[-1-k]) * cs[k];
+ bd = shft15(mdct[k]) * cs[k] -
+ shft15(mdct[-1-k]) * ca[k];
+ mdct[-1-k] = bu;
+ mdct[ k ] = bd;
+ }
+ }
+ }
+
+ max = min = 0;
+ for(k=0; k<576; k++)
+ {
mdct_freq[k] = shft13(mdct_freq[k]);
if(max < mdct_freq[k]) max = mdct_freq[k];
if(min > mdct_freq[k]) min = mdct_freq[k];
- }
+ }
- max = (max > -min) ? max : -min;
- cfg.cod_info[gr][ch].max_val = (long)max;
+ max = (max > -min) ? max : -min;
+ cfg.cod_info[gr][ch].max_val = (long)max;
- /* calc new shift for higher integer precision */
- for(k=0; max<(0x3c00>>k); k++);
+ /* calc new shift for higher integer precision */
+ for(k=0; max<(0x3c00>>k); k++);
shift = 12 - k;
- }
+ }
- cfg.cod_info[gr][ch].quantStep += cfg.cod_info[gr][ch].additStep;
+ cfg.cod_info[gr][ch].quantStep +=
+ cfg.cod_info[gr][ch].additStep;
- /* bit and noise allocation */
- iteration_loop(mdct_freq, &cfg.cod_info[gr][ch], gr_cnt--);
- /* write the frame to the bitstream */
- Huffmancodebits(enc_data, mdct_freq, &cfg.cod_info[gr][ch]);
+ /* bit and noise allocation */
+ iteration_loop(mdct_freq, &cfg.cod_info[gr][ch],
+ gr_cnt--);
+ /* write the frame to the bitstream */
+ Huffmancodebits(enc_data, mdct_freq,
+ &cfg.cod_info[gr][ch]);
- cfg.cod_info[gr][ch].quantStep -= cfg.cod_info[gr][ch].additStep;
+ cfg.cod_info[gr][ch].quantStep -=
+ cfg.cod_info[gr][ch].additStep;
- if(cfg.granules == 1)
- memcpy(sb_data[ch][0], sb_data[ch][1], sizeof(sb_data[ch][0]));
- }
+ if(cfg.granules == 1)
+ {
+ memcpy(sb_data[ch][0], sb_data[ch][1],
+ sizeof(sb_data[ch][0]));
}
+ }
+ }
+
+ chunk->enc_size = cfg.byte_per_frame + cfg.mpg.padding;
- mp3_chunk_ptr = (uint32*)ci->enc_alloc_chunk();
- mp3_chunk_ptr[0] = cfg.byte_per_frame + cfg.mpg.padding; //(CodedData.bitpos + 7) >> 3;
/* finish this chunk by adding sideinfo header data */
CodedData.bitpos = 0;
encodeSideInfo( cfg.cod_info );
- /* allocate mp3 chunk, set chunk size, copy chunk to enc_buffer */
- memcpy(&mp3_chunk_ptr[1], CodedData.bbuf, mp3_chunk_ptr[0]);
- ci->enc_free_chunk();
+#ifdef ROCKBOX_BIG_ENDIAN
+ /* copy chunk to enc_buffer */
+ memcpy(chunk->enc_data, CodedData.bbuf, chunk->enc_size);
+#else
+ /* swap frame to big endian */
+ byte_swap_frame32(chunk->enc_data, CodedData.bbuf, chunk->enc_size);
+#endif
+} /* encode_frame */
+
+/* called very often - inline */
+static inline bool is_file_data_ok(struct enc_file_event_data *filed) ICODE_ATTR;
+static inline bool is_file_data_ok(struct enc_file_event_data *filed)
+{
+ return filed->rec_file >= 0 && (long)filed->chunk->flags >= 0;
+} /* is_event_ok */
- ci->yield();
+/* called very often - inline */
+static inline bool on_write_chunk(struct enc_file_event_data *data) ICODE_ATTR;
+static inline bool on_write_chunk(struct enc_file_event_data *data)
+{
+ if (!is_file_data_ok(data))
+ return false;
+
+ if (data->chunk->enc_data == NULL)
+ {
+#ifdef ROCKBOX_HAS_LOGF
+ ci->logf("mp3 enc: NULL data");
+#endif
+ return true;
+ }
+
+ if (ci->write(data->rec_file, data->chunk->enc_data,
+ data->chunk->enc_size) != (ssize_t)data->chunk->enc_size)
+ return false;
+
+ data->num_pcm_samples += data->chunk->num_pcm;
+ return true;
+} /* on_write_chunk */
+
+static bool on_start_file(struct enc_file_event_data *data)
+{
+ if ((data->chunk->flags & CHUNKF_ERROR) || *data->filename == '\0')
+ return false;
+
+ data->rec_file = ci->open(data->filename, O_RDWR|O_CREAT|O_TRUNC);
+
+ if (data->rec_file < 0)
+ return false;
+
+ /* reset sample count */
+ data->num_pcm_samples = 0;
+ return true;
+} /* on_start_file */
+
+static bool on_end_file(struct enc_file_event_data *data)
+{
+ if (!is_file_data_ok(data))
+ return false;
+
+ ci->fsync(data->rec_file);
+ ci->close(data->rec_file);
+ data->rec_file = -1;
+
+ return true;
+} /* on_end_file */
+
+static void on_rec_new_stream(struct enc_buffer_event_data *data)
+{
+ int num_frames = cfg.mpg.type == 1 ?
+ ENC_PADDING_FRAMES1 : ENC_PADDING_FRAMES2;
+
+ if (data->flags & CHUNKF_END_FILE)
+ {
+ /* add silent frames to end - encoder will also be flushed for start
+ of next file if any */
+ memset(res_buffer, 0, pcm_chunk_size);
+
+ /* the initial chunk given for the end is at enc_wr_index */
+ while (num_frames-- > 0)
+ {
+ data->chunk->enc_data = ENC_CHUNK_SKIP_HDR(data->chunk->enc_data,
+ data->chunk);
+
+ encode_frame(res_buffer, data->chunk);
+ data->chunk->num_pcm = samp_per_frame;
+
+ ci->enc_finish_chunk();
+ data->chunk = ci->enc_get_chunk();
}
+ }
+ else if (data->flags & CHUNKF_PRERECORD)
+ {
+ /* nothing to add and we cannot change prerecorded data */
+ }
+ else if (data->flags & CHUNKF_START_FILE)
+ {
+ /* starting fresh ... be sure to flush encoder first */
+ struct enc_chunk_hdr *chunk = ENC_CHUNK_HDR(res_buffer);
- if(ci->enc_wavbuf_near_empty())
+ chunk->flags = 0;
+ chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
+
+ while (num_frames-- > 0)
{
- if(cpu_boosted)
+ memset(chunk->enc_data, 0, pcm_chunk_size);
+ encode_frame(chunk->enc_data, chunk);
+ }
+ }
+} /* on_rec_new_stream */
+
+static void enc_events_callback(enum enc_events event, void *data) ICODE_ATTR;
+static void enc_events_callback(enum enc_events event, void *data)
+{
+ if (event == ENC_WRITE_CHUNK)
+ {
+ if (on_write_chunk((struct enc_file_event_data *)data))
+ return;
+ }
+ else if (event == ENC_START_FILE)
+ {
+ if (on_start_file((struct enc_file_event_data *)data))
+ return;
+ }
+ else if (event == ENC_END_FILE)
+ {
+ if (on_end_file((struct enc_file_event_data *)data))
+ return;
+ }
+ else if (event == ENC_REC_NEW_STREAM)
+ {
+ on_rec_new_stream((struct enc_buffer_event_data *)data);
+ return;
+ }
+ else
+ {
+ return;
+ }
+
+ ((struct enc_file_event_data *)data)->chunk->flags |= CHUNKF_ERROR;
+} /* enc_events_callback */
+
+static bool enc_init(void)
+{
+ struct enc_inputs inputs;
+ struct enc_parameters params;
+
+ if (ci->enc_get_inputs == NULL ||
+ ci->enc_set_parameters == NULL ||
+ ci->enc_get_chunk == NULL ||
+ ci->enc_finish_chunk == NULL ||
+ ci->enc_pcm_buf_near_empty == NULL ||
+ ci->enc_get_pcm_data == NULL ||
+ ci->enc_unget_pcm_data == NULL )
+ return false;
+
+ ci->enc_get_inputs(&inputs);
+
+ if (inputs.config->afmt != AFMT_MPA_L3)
+ return false;
+
+ init_mp3_encoder_engine(inputs.sample_rate, inputs.num_channels,
+ inputs.config);
+
+ /* configure the buffer system */
+ params.afmt = AFMT_MPA_L3;
+ params.chunk_size = cfg.byte_per_frame + 1;
+ params.enc_sample_rate = cfg.samplerate;
+ /* need enough reserved bytes to hold one frame of pcm samples + hdr
+ for padding and flushing */
+ params.reserve_bytes = ENC_CHUNK_HDR_SIZE + pcm_chunk_size;
+ params.events_callback = enc_events_callback;
+ ci->enc_set_parameters(&params);
+
+ res_buffer = params.reserve_buffer;
+
+#ifdef CPU_COLDFIRE
+ asm volatile ("move.l #0, %macsr"); /* integer mode */
+#endif
+
+ return true;
+} /* enc_init */
+
+enum codec_status codec_start(struct codec_api* api)
+{
+ bool cpu_boosted;
+
+ /* Generic codec initialisation */
+ ci = api;
+
+#ifdef USE_IRAM
+ memcpy(iramstart, iramcopy, iramend - iramstart);
+ memset(iedata, 0, iend - iedata);
+#endif
+
+ if (!enc_init())
+ {
+ ci->enc_codec_loaded = -1;
+ return CODEC_ERROR;
+ }
+
+ /* main application waits for this flag during encoder loading */
+ ci->enc_codec_loaded = 1;
+
+ ci->cpu_boost(true);
+ cpu_boosted = true;
+
+ /* main encoding loop */
+ while (!ci->stop_codec)
+ {
+ char *buffer;
+
+ while ((buffer = ci->enc_get_pcm_data(pcm_chunk_size)) != NULL)
+ {
+ struct enc_chunk_hdr *chunk;
+
+ if (ci->stop_codec)
+ break;
+
+ if (!cpu_boosted && ci->enc_pcm_buf_near_empty() == 0)
+ {
+ ci->cpu_boost(true);
+ cpu_boosted = true;
+ }
+
+ chunk = ci->enc_get_chunk();
+ chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
+
+ encode_frame(buffer, chunk);
+
+ if (chunk->num_pcm < samp_per_frame)
+ {
+ ci->enc_unget_pcm_data(pcm_chunk_size - chunk->num_pcm*4);
+ chunk->num_pcm = samp_per_frame;
+ }
+
+ ci->enc_finish_chunk();
+
+ ci->yield();
+ }
+
+ if (cpu_boosted && ci->enc_pcm_buf_near_empty())
{
ci->cpu_boost(false);
cpu_boosted = false;
}
- }
+
ci->yield();
}
@@ -2261,12 +2531,12 @@ enum codec_status codec_start(struct codec_api* api)
ci->cpu_boost(false);
/* reset parameters to initial state */
- ci->enc_set_parameters(0, 0, 0, 0, 0, 0);
+ ci->enc_set_parameters(NULL);
/* main application waits for this flag during encoder removing */
- ci->enc_codec_loaded = false;
+ ci->enc_codec_loaded = 0;
return CODEC_OK;
-}
+} /* codec_start */
-#endif
+#endif /* ndef SIMULATOR */
diff --git a/apps/codecs/wav_enc.c b/apps/codecs/wav_enc.c
index 5aabb5d8e8..974a903310 100644
--- a/apps/codecs/wav_enc.c
+++ b/apps/codecs/wav_enc.c
@@ -19,140 +19,364 @@
#ifndef SIMULATOR
+#include <inttypes.h>
#include "codeclib.h"
-CODEC_HEADER
+CODEC_ENC_HEADER
+
+#ifdef USE_IRAM
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+extern char iedata[];
+extern char iend[];
+#endif
+
+struct riff_header
+{
+ uint8_t riff_id[4]; /* 00h - "RIFF" */
+ uint32_t riff_size; /* 04h - sz following headers + data_size */
+ /* format header */
+ uint8_t format[4]; /* 08h - "WAVE" */
+ uint8_t format_id[4]; /* 0Ch - "fmt " */
+ uint32_t format_size; /* 10h - 16 for PCM (sz format data) */
+ /* format data */
+ uint16_t audio_format; /* 14h - 1=PCM */
+ uint16_t num_channels; /* 16h - 1=M, 2=S, etc. */
+ uint32_t sample_rate; /* 18h - HZ */
+ uint32_t byte_rate; /* 1Ch - num_channels*sample_rate*bits_per_sample/8 */
+ uint16_t block_align; /* 20h - num_channels*bits_per_samples/8 */
+ uint16_t bits_per_sample; /* 22h - 8=8 bits, 16=16 bits, etc. */
+ /* Not for audio_format=1 (PCM) */
+/* unsigned short extra_param_size; 24h - size of extra data */
+/* unsigned char *extra_params; */
+ /* data header */
+ uint8_t data_id[4]; /* 24h - "data" */
+ uint32_t data_size; /* 28h - num_samples*num_channels*bits_per_sample/8 */
+/* unsigned char *data; 2ch - actual sound data */
+};
+
+#define RIFF_FMT_HEADER_SIZE 12 /* format -> format_size */
+#define RIFF_FMT_DATA_SIZE 16 /* audio_format -> bits_per_sample */
+#define RIFF_DATA_HEADER_SIZE 8 /* data_id -> data_size */
+
+#define PCM_DEPTH_BYTES 2
+#define PCM_DEPTH_BITS 16
+#define PCM_SAMP_PER_CHUNK 2048
+#define PCM_CHUNK_SIZE (PCM_SAMP_PER_CHUNK*4)
static struct codec_api *ci;
-static int enc_channels;
+static int num_channels;
+uint32_t sample_rate;
+uint32_t enc_size;
-#define CHUNK_SIZE 8192
+static const struct riff_header riff_header =
+{
+ /* "RIFF" header */
+ { 'R', 'I', 'F', 'F' }, /* riff_id */
+ 0, /* riff_size (*) */
+ /* format header */
+ { 'W', 'A', 'V', 'E' }, /* format */
+ { 'f', 'm', 't', ' ' }, /* format_id */
+ H_TO_LE32(16), /* format_size */
+ /* format data */
+ H_TO_LE16(1), /* audio_format */
+ 0, /* num_channels (*) */
+ 0, /* sample_rate (*) */
+ 0, /* byte_rate (*) */
+ 0, /* block_align (*) */
+ H_TO_LE16(PCM_DEPTH_BITS), /* bits_per_sample */
+ /* data header */
+ { 'd', 'a', 't', 'a' }, /* data_id */
+ 0 /* data_size (*) */
+ /* (*) updated during ENC_END_FILE event */
+};
-static unsigned char wav_header[44] =
-{'R','I','F','F',0,0,0,0,'W','A','V','E','f','m','t',' ',16,
- 0,0,0,1,0,2,0,0x44,0xac,0,0,0x10,0xb1,2,0,4,0,16,0,'d','a','t','a',0,0,0,0};
+/* called version often - inline */
+static inline bool is_file_data_ok(struct enc_file_event_data *data) ICODE_ATTR;
+static inline bool is_file_data_ok(struct enc_file_event_data *data)
+{
+ return data->rec_file >= 0 && (long)data->chunk->flags >= 0;
+} /* is_file_data_ok */
-static unsigned char wav_header_mono[44] =
-{'R','I','F','F',0,0,0,0,'W','A','V','E','f','m','t',' ',16,
- 0,0,0,1,0,1,0,0x44,0xac,0,0,0x88,0x58,1,0,2,0,16,0,'d','a','t','a',0,0,0,0};
+/* called version often - inline */
+static inline bool on_write_chunk(struct enc_file_event_data *data) ICODE_ATTR;
+static inline bool on_write_chunk(struct enc_file_event_data *data)
+{
+ if (!is_file_data_ok(data))
+ return false;
-/* update file header info callback function (called by main application) */
-void enc_set_header(void *head_buffer, /* ptr to the file header data */
- int head_size, /* size of this header data */
- int num_pcm_samples, /* amount of processed pcm samples */
- bool is_file_header)
+ if (data->chunk->enc_data == NULL)
+ {
+#ifdef ROCKBOX_HAS_LOGF
+ ci->logf("wav enc: NULL data");
+#endif
+ return true;
+ }
+
+ if (ci->write(data->rec_file, data->chunk->enc_data,
+ data->chunk->enc_size) != (ssize_t)data->chunk->enc_size)
+ return false;
+
+ data->num_pcm_samples += data->chunk->num_pcm;
+ return true;
+} /* on_write_chunk */
+
+static bool on_start_file(struct enc_file_event_data *data)
{
- int num_file_bytes = num_pcm_samples * 2 * enc_channels;
+ if ((data->chunk->flags & CHUNKF_ERROR) || *data->filename == '\0')
+ return false;
+
+ data->rec_file = ci->open(data->filename, O_RDWR|O_CREAT|O_TRUNC);
+
+ if (data->rec_file < 0)
+ return false;
+
+ /* reset sample count */
+ data->num_pcm_samples = 0;
- if(is_file_header)
+ /* write template header */
+ if (ci->write(data->rec_file, &riff_header, sizeof (riff_header))
+ != sizeof (riff_header))
{
- /* update file header before file closing */
- if((int)sizeof(wav_header) < head_size)
+ return false;
+ }
+
+ data->new_enc_size += sizeof (riff_header);
+ return true;
+} /* on_start_file */
+
+static bool on_end_file(struct enc_file_event_data *data)
+{
+ /* update template header */
+ struct riff_header hdr;
+ uint32_t data_size;
+
+ if (!is_file_data_ok(data))
+ return false;
+
+ if (ci->lseek(data->rec_file, 0, SEEK_SET) != 0 ||
+ ci->read(data->rec_file, &hdr, sizeof (hdr)) != sizeof (hdr))
+ {
+ return false;
+ }
+
+ data_size = data->num_pcm_samples*num_channels*PCM_DEPTH_BYTES;
+
+ /* "RIFF" header */
+ hdr.riff_size = htole32(RIFF_FMT_HEADER_SIZE + RIFF_FMT_DATA_SIZE
+ + RIFF_DATA_HEADER_SIZE + data_size);
+
+ /* format data */
+ hdr.num_channels = htole16(num_channels);
+ hdr.sample_rate = htole32(sample_rate);
+ hdr.byte_rate = htole32(sample_rate*num_channels* PCM_DEPTH_BYTES);
+ hdr.block_align = htole16(num_channels*PCM_DEPTH_BYTES);
+
+ /* data header */
+ hdr.data_size = htole32(data_size);
+
+ if (ci->lseek(data->rec_file, 0, SEEK_SET) != 0 ||
+ ci->write(data->rec_file, &hdr, sizeof (hdr)) != sizeof (hdr))
+ {
+ return false;
+ }
+
+ ci->fsync(data->rec_file);
+ ci->close(data->rec_file);
+ data->rec_file = -1;
+
+ return true;
+} /* on_end_file */
+
+static void enc_events_callback(enum enc_events event, void *data) ICODE_ATTR;
+static void enc_events_callback(enum enc_events event, void *data)
+{
+ if (event == ENC_WRITE_CHUNK)
+ {
+ if (on_write_chunk((struct enc_file_event_data *)data))
+ return;
+ }
+ else if (event == ENC_START_FILE)
+ {
+ if (on_start_file((struct enc_file_event_data *)data))
+ return;
+ }
+ else if (event == ENC_END_FILE)
+ {
+ if (on_end_file((struct enc_file_event_data *)data))
+ return;
+ }
+ else
+ {
+ return;
+ }
+
+ ((struct enc_file_event_data *)data)->chunk->flags |= CHUNKF_ERROR;
+} /* enc_events_callback */
+
+/* convert native pcm samples to wav format samples */
+static void chunk_to_wav_format(uint32_t *src, uint32_t *dst) ICODE_ATTR;
+static void chunk_to_wav_format(uint32_t *src, uint32_t *dst)
+{
+ if (num_channels == 1)
+ {
+ /* On big endian:
+ * |LLLLLLLLllllllll|RRRRRRRRrrrrrrrr|
+ * |LLLLLLLLllllllll|RRRRRRRRrrrrrrrr| =>
+ * |mmmmmmmmMMMMMMMM|mmmmmmmmMMMMMMMM|
+ *
+ * On little endian:
+ * |llllllllLLLLLLLL|rrrrrrrrRRRRRRRR|
+ * |llllllllLLLLLLLL|rrrrrrrrRRRRRRRR| =>
+ * |mmmmmmmmMMMMMMMM|mmmmmmmmMMMMMMMM|
+ */
+ uint32_t *src_end = src + PCM_SAMP_PER_CHUNK;
+
+ inline void to_mono(uint32_t **src, uint32_t **dst)
{
- /* update wave header size entries: special to WAV format */
- *(long*)(head_buffer+ 4) = htole32(num_file_bytes + 36);
- *(long*)(head_buffer+40) = htole32(num_file_bytes);
+ int32_t lr1, lr2;
+
+ lr1 = *(*src)++;
+ lr1 = ((int16_t)lr1 + (lr1 >> 16)) >> 1;
+
+ lr2 = *(*src)++;
+ lr2 = ((int16_t)lr2 + (lr2 >> 16)) >> 1;
+ *(*dst)++ = swap_odd_even_be32((lr1 << 16) | (uint16_t)lr2);
+ } /* to_mono */
+
+ do
+ {
+ to_mono(&src, &dst);
+ to_mono(&src, &dst);
+ to_mono(&src, &dst);
+ to_mono(&src, &dst);
+ to_mono(&src, &dst);
+ to_mono(&src, &dst);
+ to_mono(&src, &dst);
+ to_mono(&src, &dst);
}
+ while (src < src_end);
}
-}
+ else
+ {
+#ifdef ROCKBOX_BIG_ENDIAN
+ /* |LLLLLLLLllllllll|RRRRRRRRrrrrrrrr| =>
+ * |llllllllLLLLLLLL|rrrrrrrrRRRRRRRR|
+ */
+ uint32_t *src_end = src + PCM_SAMP_PER_CHUNK;
-/* main codec entry point */
-enum codec_status codec_start(struct codec_api* api)
+ do
+ {
+ *dst++ = swap_odd_even32(*src++);
+ *dst++ = swap_odd_even32(*src++);
+ *dst++ = swap_odd_even32(*src++);
+ *dst++ = swap_odd_even32(*src++);
+ *dst++ = swap_odd_even32(*src++);
+ *dst++ = swap_odd_even32(*src++);
+ *dst++ = swap_odd_even32(*src++);
+ *dst++ = swap_odd_even32(*src++);
+ }
+ while (src < src_end);
+#else
+ /* |llllllllLLLLLLLL|rrrrrrrrRRRRRRRR| =>
+ * |llllllllLLLLLLLL|rrrrrrrrRRRRRRRR|
+ */
+ ci->memcpy(dst, src, PCM_CHUNK_SIZE);
+#endif
+ }
+} /* chunk_to_wav_format */
+
+static bool init_encoder(void)
{
- int i;
- long lr;
- unsigned long t;
- unsigned long *src;
- unsigned long *dst;
- int chunk_size, num_chunks, samp_per_chunk;
- int enc_buffer_size;
- int enc_quality;
- bool cpu_boosted = true; /* start boosted */
+ struct enc_inputs inputs;
+ struct enc_parameters params;
- ci = api; // copy to global api pointer
+ if (ci->enc_get_inputs == NULL ||
+ ci->enc_set_parameters == NULL ||
+ ci->enc_get_chunk == NULL ||
+ ci->enc_finish_chunk == NULL ||
+ ci->enc_pcm_buf_near_empty == NULL ||
+ ci->enc_get_pcm_data == NULL )
+ return false;
- if(ci->enc_get_inputs == NULL ||
- ci->enc_set_parameters == NULL ||
- ci->enc_alloc_chunk == NULL ||
- ci->enc_free_chunk == NULL ||
- ci->enc_wavbuf_near_empty == NULL ||
- ci->enc_get_wav_data == NULL ||
- ci->enc_set_header_callback == NULL )
- return CODEC_ERROR;
+ ci->enc_get_inputs(&inputs);
- ci->cpu_boost(true);
+ if (inputs.config->afmt != AFMT_PCM_WAV)
+ return false;
- *ci->enc_set_header_callback = enc_set_header;
- ci->enc_get_inputs(&enc_buffer_size, &enc_channels, &enc_quality);
+ sample_rate = inputs.sample_rate;
+ num_channels = inputs.num_channels;
/* configure the buffer system */
- chunk_size = sizeof(long) + CHUNK_SIZE * enc_channels / 2;
- num_chunks = enc_buffer_size / chunk_size;
- samp_per_chunk = CHUNK_SIZE / 4;
+ params.afmt = AFMT_PCM_WAV;
+ enc_size = PCM_CHUNK_SIZE*inputs.num_channels / 2;
+ params.chunk_size = enc_size;
+ params.enc_sample_rate = sample_rate;
+ params.reserve_bytes = 0;
+ params.events_callback = enc_events_callback;
+ ci->enc_set_parameters(&params);
+
+ return true;
+} /* init_encoder */
+
+/* main codec entry point */
+enum codec_status codec_start(struct codec_api* api)
+{
+ bool cpu_boosted;
- /* inform the main program about buffer dimensions and other params */
- ci->enc_set_parameters(chunk_size, num_chunks, samp_per_chunk,
- (enc_channels == 2) ? wav_header : wav_header_mono,
- sizeof(wav_header), AFMT_PCM_WAV);
+ ci = api; // copy to global api pointer
+
+#ifdef USE_IRAM
+ ci->memcpy(iramstart, iramcopy, iramend - iramstart);
+ ci->memset(iedata, 0, iend - iedata);
+#endif
+
+ if (!init_encoder())
+ {
+ ci->enc_codec_loaded = -1;
+ return CODEC_ERROR;
+ }
/* main application waits for this flag during encoder loading */
- ci->enc_codec_loaded = true;
+ ci->enc_codec_loaded = 1;
+
+ ci->cpu_boost(true);
+ cpu_boosted = true;
/* main encoding loop */
while(!ci->stop_codec)
{
- while((src = (unsigned long*)ci->enc_get_wav_data(CHUNK_SIZE)) != NULL)
+ uint32_t *src;
+
+ while ((src = (uint32_t *)ci->enc_get_pcm_data(PCM_CHUNK_SIZE)) != NULL)
{
- if(ci->stop_codec)
+ struct enc_chunk_hdr *chunk;
+
+ if (ci->stop_codec)
break;
- if(ci->enc_wavbuf_near_empty() == 0)
+ if (!cpu_boosted && ci->enc_pcm_buf_near_empty() == 0)
{
- if(!cpu_boosted)
- {
- ci->cpu_boost(true);
- cpu_boosted = true;
- }
+ ci->cpu_boost(true);
+ cpu_boosted = true;
}
- dst = (unsigned long*)ci->enc_alloc_chunk();
- *dst++ = CHUNK_SIZE * enc_channels / 2; /* set size info */
+ chunk = ci->enc_get_chunk();
+ chunk->enc_size = enc_size;
+ chunk->num_pcm = PCM_SAMP_PER_CHUNK;
+ chunk->enc_data = ENC_CHUNK_SKIP_HDR(chunk->enc_data, chunk);
- if(enc_channels == 2)
- {
- /* swap byte order & copy to destination */
- for (i=0; i<CHUNK_SIZE/4; i++)
- {
- t = *src++;
- *dst++ = ((t >> 8) & 0xff00ff) | ((t << 8) & 0xff00ff00);
- }
- }
- else
- {
- /* mix left/right, swap byte order & copy to destination */
- for (i=0; i<CHUNK_SIZE/8; i++)
- {
- lr = (long)*src++;
- lr = (((lr<<16)>>16) + (lr>>16)) >> 1; /* left+right */
- t = (lr << 16);
- lr = (long)*src++;
- lr = (((lr<<16)>>16) + (lr>>16)) >> 1; /* left+right */
- t |= lr & 0xffff;
- *dst++ = ((t >> 8) & 0xff00ff) | ((t << 8) & 0xff00ff00);
- }
- }
+ chunk_to_wav_format(src, (uint32_t *)chunk->enc_data);
- ci->enc_free_chunk();
+ ci->enc_finish_chunk();
ci->yield();
}
- if(ci->enc_wavbuf_near_empty())
+ if (cpu_boosted && ci->enc_pcm_buf_near_empty() != 0)
{
- if(cpu_boosted)
- {
- ci->cpu_boost(false);
- cpu_boosted = false;
- }
+ ci->cpu_boost(false);
+ cpu_boosted = false;
}
ci->yield();
@@ -162,11 +386,12 @@ enum codec_status codec_start(struct codec_api* api)
ci->cpu_boost(false);
/* reset parameters to initial state */
- ci->enc_set_parameters(0, 0, 0, 0, 0, 0);
+ ci->enc_set_parameters(NULL);
/* main application waits for this flag during encoder removing */
- ci->enc_codec_loaded = false;
+ ci->enc_codec_loaded = 0;
return CODEC_OK;
-}
-#endif
+} /* codec_start */
+
+#endif /* ndef SIMULATOR */
diff --git a/apps/codecs/wavpack_enc.c b/apps/codecs/wavpack_enc.c
index eced7f1f4e..5318abc8fb 100644
--- a/apps/codecs/wavpack_enc.c
+++ b/apps/codecs/wavpack_enc.c
@@ -22,201 +22,474 @@
#include "codeclib.h"
#include "libwavpack/wavpack.h"
-CODEC_HEADER
+CODEC_ENC_HEADER
+
+#ifdef USE_IRAM
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+extern char iedata[];
+extern char iend[];
+#endif
-typedef unsigned long uint32;
-typedef unsigned short uint16;
-typedef unsigned char uint8;
+/** Types **/
+typedef struct
+{
+ uint8_t type; /* Type of metadata */
+ uint8_t word_size; /* Size of metadata in words */
+} WavpackMetadataHeader;
-static unsigned char wav_header_ster [46] =
-{33,22,'R','I','F','F',0,0,0,0,'W','A','V','E','f','m','t',' ',16,
- 0,0,0,1,0,2,0,0x44,0xac,0,0,0x10,0xb1,2,0,4,0,16,0,'d','a','t','a',0,0,0,0};
+struct riff_header
+{
+ uint8_t riff_id[4]; /* 00h - "RIFF" */
+ uint32_t riff_size; /* 04h - sz following headers + data_size */
+ /* format header */
+ uint8_t format[4]; /* 08h - "WAVE" */
+ uint8_t format_id[4]; /* 0Ch - "fmt " */
+ uint32_t format_size; /* 10h - 16 for PCM (sz format data) */
+ /* format data */
+ uint16_t audio_format; /* 14h - 1=PCM */
+ uint16_t num_channels; /* 16h - 1=M, 2=S, etc. */
+ uint32_t sample_rate; /* 18h - HZ */
+ uint32_t byte_rate; /* 1Ch - num_channels*sample_rate*bits_per_sample/8 */
+ uint16_t block_align; /* 20h - num_channels*bits_per_samples/8 */
+ uint16_t bits_per_sample; /* 22h - 8=8 bits, 16=16 bits, etc. */
+ /* Not for audio_format=1 (PCM) */
+/* unsigned short extra_param_size; 24h - size of extra data */
+/* unsigned char *extra_params; */
+ /* data header */
+ uint8_t data_id[4]; /* 24h - "data" */
+ uint32_t data_size; /* 28h - num_samples*num_channels*bits_per_sample/8 */
+/* unsigned char *data; 2ch - actual sound data */
+};
+
+#define RIFF_FMT_HEADER_SIZE 12 /* format -> format_size */
+#define RIFF_FMT_DATA_SIZE 16 /* audio_format -> bits_per_sample */
+#define RIFF_DATA_HEADER_SIZE 8 /* data_id -> data_size */
+
+#define PCM_DEPTH_BITS 16
+#define PCM_DEPTH_BYTES 2
+#define PCM_SAMP_PER_CHUNK 5000
+#define PCM_CHUNK_SIZE (4*PCM_SAMP_PER_CHUNK)
+
+/** Data **/
+static struct codec_api *ci;
+static int8_t input_buffer[PCM_CHUNK_SIZE*2] IBSS_ATTR;
+static WavpackConfig config IBSS_ATTR;
+static WavpackContext *wpc;
+static int32_t data_size, input_size, input_step IBSS_ATTR;
-static unsigned char wav_header_mono [46] =
-{33,22,'R','I','F','F',0,0,0,0,'W','A','V','E','f','m','t',' ',16,
- 0,0,0,1,0,1,0,0x44,0xac,0,0,0x88,0x58,1,0,2,0,16,0,'d','a','t','a',0,0,0,0};
+static const WavpackMetadataHeader wvpk_mdh =
+{
+ ID_RIFF_HEADER,
+ sizeof (struct riff_header) / sizeof (uint16_t),
+};
-static struct codec_api *ci;
-static int enc_channels;
+static const struct riff_header riff_header =
+{
+ /* "RIFF" header */
+ { 'R', 'I', 'F', 'F' }, /* riff_id */
+ 0, /* riff_size (*) */
+ /* format header */
+ { 'W', 'A', 'V', 'E' }, /* format */
+ { 'f', 'm', 't', ' ' }, /* format_id */
+ H_TO_LE32(16), /* format_size */
+ /* format data */
+ H_TO_LE16(1), /* audio_format */
+ 0, /* num_channels (*) */
+ 0, /* sample_rate (*) */
+ 0, /* byte_rate (*) */
+ 0, /* block_align (*) */
+ H_TO_LE16(PCM_DEPTH_BITS), /* bits_per_sample */
+ /* data header */
+ { 'd', 'a', 't', 'a' }, /* data_id */
+ 0 /* data_size (*) */
+ /* (*) updated during ENC_END_FILE event */
+};
+
+static void chunk_to_int32(int32_t *src) ICODE_ATTR;
+static void chunk_to_int32(int32_t *src)
+{
+ int32_t *dst = (int32_t *)input_buffer + PCM_SAMP_PER_CHUNK;
+ int32_t *src_end = dst + PCM_SAMP_PER_CHUNK;
-#define CHUNK_SIZE 20000
+ /* copy to IRAM before converting data */
+ memcpy(dst, src, PCM_CHUNK_SIZE);
-static long input_buffer[CHUNK_SIZE/2] IBSS_ATTR;
+ src = dst;
+ dst = (int32_t *)input_buffer;
-/* update file header info callback function */
-void enc_set_header(void *head_buffer, /* ptr to the file header data */
- int head_size, /* size of this header data */
- int num_pcm_sampl, /* amount of processed pcm samples */
- bool is_file_header) /* update file/chunk header */
-{
- if(is_file_header)
+ if (config.num_channels == 1)
{
- /* update file header before file closing */
- if(sizeof(WavpackHeader) + sizeof(wav_header_mono) < (unsigned)head_size)
+ /*
+ * |llllllllllllllll|rrrrrrrrrrrrrrrr| =>
+ * |mmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm|
+ */
+ inline void to_int32(int32_t **src, int32_t **dst)
{
- char* riff_header = (char*)head_buffer + sizeof(WavpackHeader);
- char* wv_header = (char*)head_buffer + sizeof(wav_header_mono);
- int num_file_bytes = num_pcm_sampl * 2 * enc_channels;
- unsigned long ckSize;
-
- /* RIFF header and WVPK header have to be swapped */
- /* copy wavpack header to file start position */
- ci->memcpy(head_buffer, wv_header, sizeof(WavpackHeader));
- wv_header = head_buffer; /* recalc wavpack header position */
-
- if(enc_channels == 2)
- ci->memcpy(riff_header, wav_header_ster, sizeof(wav_header_ster));
- else
- ci->memcpy(riff_header, wav_header_mono, sizeof(wav_header_mono));
-
- /* update the Wavpack header first chunk size & total frame count */
- ckSize = htole32(((WavpackHeader*)wv_header)->ckSize)
- + sizeof(wav_header_mono);
- ((WavpackHeader*)wv_header)->total_samples = htole32(num_pcm_sampl);
- ((WavpackHeader*)wv_header)->ckSize = htole32(ckSize);
-
- /* update the RIFF WAV header size entries */
- *(long*)(riff_header+ 6) = htole32(num_file_bytes + 36);
- *(long*)(riff_header+42) = htole32(num_file_bytes);
+ int32_t t = *(*src)++;
+ /* endianness irrelevant */
+ *(*dst)++ = ((int16_t)t + (t >> 16)) >> 1;
+ } /* to_int32 */
+
+ do
+ {
+ /* read 10 longs and write 10 longs */
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
}
+ while(src < src_end);
+
+ return;
}
else
{
- /* update timestamp (block_index) */
- ((WavpackHeader*)head_buffer)->block_index = htole32(num_pcm_sampl);
+ /*
+ * |llllllllllllllll|rrrrrrrrrrrrrrrr| =>
+ * |llllllllllllllllllllllllllllllll|rrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr|
+ */
+ inline void to_int32(int32_t **src, int32_t **dst)
+ {
+ int32_t t = *(*src)++;
+#ifdef ROCKBOX_BIG_ENDIAN
+ *(*dst)++ = t >> 16, *(*dst)++ = (int16_t)t;
+#else
+ *(*dst)++ = (int16_t)t, *(*dst)++ = t >> 16;
+#endif
+ } /* to_int32 */
+
+ do
+ {
+ /* read 10 longs and write 20 longs */
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ to_int32(&src, &dst);
+ }
+ while (src < src_end);
+
+ return;
}
-}
+} /* chunk_to_int32 */
+/* called very often - inline */
+static inline bool is_file_data_ok(struct enc_file_event_data *data) ICODE_ATTR;
+static inline bool is_file_data_ok(struct enc_file_event_data *data)
+{
+ return data->rec_file >= 0 && (long)data->chunk->flags >= 0;
+} /* is_file_data_ok */
-enum codec_status codec_start(struct codec_api* api)
+/* called very often - inline */
+static inline bool on_write_chunk(struct enc_file_event_data *data) ICODE_ATTR;
+static inline bool on_write_chunk(struct enc_file_event_data *data)
+{
+ if (!is_file_data_ok(data))
+ return false;
+
+ if (data->chunk->enc_data == NULL)
+ {
+#ifdef ROCKBOX_HAS_LOGF
+ ci->logf("wvpk enc: NULL data");
+#endif
+ return true;
+ }
+
+ /* update timestamp (block_index) */
+ ((WavpackHeader *)data->chunk->enc_data)->block_index =
+ htole32(data->num_pcm_samples);
+
+ if (ci->write(data->rec_file, data->chunk->enc_data,
+ data->chunk->enc_size) != (ssize_t)data->chunk->enc_size)
+ return false;
+
+ data->num_pcm_samples += data->chunk->num_pcm;
+ return true;
+} /* on_write_chunk */
+
+static bool on_start_file(struct enc_file_event_data *data)
+{
+ if ((data->chunk->flags & CHUNKF_ERROR) || *data->filename == '\0')
+ return false;
+
+ data->rec_file = ci->open(data->filename, O_RDWR|O_CREAT|O_TRUNC);
+
+ if (data->rec_file < 0)
+ return false;
+
+ /* reset sample count */
+ data->num_pcm_samples = 0;
+
+ /* write template headers */
+ if (ci->write(data->rec_file, &wvpk_mdh, sizeof (wvpk_mdh))
+ != sizeof (wvpk_mdh) ||
+ ci->write(data->rec_file, &riff_header, sizeof (riff_header))
+ != sizeof (riff_header))
+ {
+ return false;
+ }
+
+ data->new_enc_size += sizeof(wvpk_mdh) + sizeof(riff_header);
+ return true;
+} /* on_start_file */
+
+static bool on_end_file(struct enc_file_event_data *data)
+{
+ struct
+ {
+ WavpackMetadataHeader wpmdh;
+ struct riff_header rhdr;
+ WavpackHeader wph;
+ } __attribute__ ((packed)) h;
+
+ uint32_t data_size;
+
+ if (!is_file_data_ok(data))
+ return false;
+
+ /* read template headers at start */
+ if (ci->lseek(data->rec_file, 0, SEEK_SET) != 0 ||
+ ci->read(data->rec_file, &h, sizeof (h)) != sizeof (h))
+ return false;
+
+ data_size = data->num_pcm_samples*config.num_channels*PCM_DEPTH_BYTES;
+
+ /** "RIFF" header **/
+ h.rhdr.riff_size = htole32(RIFF_FMT_HEADER_SIZE +
+ RIFF_FMT_DATA_SIZE + RIFF_DATA_HEADER_SIZE + data_size);
+
+ /* format data */
+ h.rhdr.num_channels = htole16(config.num_channels);
+ h.rhdr.sample_rate = htole32(config.sample_rate);
+ h.rhdr.byte_rate = htole32(config.sample_rate*config.num_channels*
+ PCM_DEPTH_BYTES);
+ h.rhdr.block_align = htole16(config.num_channels*PCM_DEPTH_BYTES);
+
+ /* data header */
+ h.rhdr.data_size = htole32(data_size);
+
+ /** Wavpack header **/
+ h.wph.ckSize = htole32(letoh32(h.wph.ckSize) + sizeof (h.wpmdh)
+ + sizeof (h.rhdr));
+ h.wph.total_samples = htole32(data->num_pcm_samples);
+
+ /* MDH|RIFF|WVPK => WVPK|MDH|RIFF */
+ if (ci->lseek(data->rec_file, 0, SEEK_SET)
+ != 0 ||
+ ci->write(data->rec_file, &h.wph, sizeof (h.wph))
+ != sizeof (h.wph) ||
+ ci->write(data->rec_file, &h.wpmdh, sizeof (h.wpmdh))
+ != sizeof (h.wpmdh) ||
+ ci->write(data->rec_file, &h.rhdr, sizeof (h.rhdr))
+ != sizeof (h.rhdr))
+ {
+ return false;
+ }
+
+ ci->fsync(data->rec_file);
+ ci->close(data->rec_file);
+ data->rec_file = -1;
+
+ return true;
+} /* on_end_file */
+
+static void enc_events_callback(enum enc_events event, void *data) ICODE_ATTR;
+static void enc_events_callback(enum enc_events event, void *data)
{
- int i;
- long t;
- uint32 *src;
- uint32 *dst;
- int chunk_size, num_chunks, samp_per_chunk;
- int enc_buffer_size;
- int enc_quality;
- WavpackConfig config;
- WavpackContext *wpc;
- bool cpu_boosted = true; /* start boosted */
-
- ci = api; // copy to global api pointer
+ if (event == ENC_WRITE_CHUNK)
+ {
+ if (on_write_chunk((struct enc_file_event_data *)data))
+ return;
+ }
+ else if (event == ENC_START_FILE)
+ {
+ /* write metadata header and RIFF header */
+ if (on_start_file((struct enc_file_event_data *)data))
+ return;
+ }
+ else if (event == ENC_END_FILE)
+ {
+ if (on_end_file((struct enc_file_event_data *)data))
+ return;
+ }
+ else
+ {
+ return;
+ }
+
+ ((struct enc_file_event_data *)data)->chunk->flags |= CHUNKF_ERROR;
+} /* enc_events_callback */
+
+static bool init_encoder(void)
+{
+ struct enc_inputs inputs;
+ struct enc_parameters params;
codec_init(ci);
- if(ci->enc_get_inputs == NULL ||
- ci->enc_set_parameters == NULL ||
- ci->enc_alloc_chunk == NULL ||
- ci->enc_free_chunk == NULL ||
- ci->enc_wavbuf_near_empty == NULL ||
- ci->enc_get_wav_data == NULL ||
- ci->enc_set_header_callback == NULL )
- return CODEC_ERROR;
+ if (ci->enc_get_inputs == NULL ||
+ ci->enc_set_parameters == NULL ||
+ ci->enc_get_chunk == NULL ||
+ ci->enc_finish_chunk == NULL ||
+ ci->enc_pcm_buf_near_empty == NULL ||
+ ci->enc_get_pcm_data == NULL ||
+ ci->enc_unget_pcm_data == NULL )
+ return false;
- ci->cpu_boost(true);
+ ci->enc_get_inputs(&inputs);
+
+ if (inputs.config->afmt != AFMT_WAVPACK)
+ return false;
- *ci->enc_set_header_callback = enc_set_header;
- ci->enc_get_inputs(&enc_buffer_size, &enc_channels, &enc_quality);
+ memset(&config, 0, sizeof (config));
+ config.bits_per_sample = PCM_DEPTH_BITS;
+ config.bytes_per_sample = PCM_DEPTH_BYTES;
+ config.sample_rate = inputs.sample_rate;
+ config.num_channels = inputs.num_channels;
+
+ wpc = WavpackOpenFileOutput ();
+
+ if (!WavpackSetConfiguration(wpc, &config, -1))
+ return false;
/* configure the buffer system */
- chunk_size = sizeof(long) + CHUNK_SIZE * enc_channels / 2;
- num_chunks = enc_buffer_size / chunk_size;
- samp_per_chunk = CHUNK_SIZE / 4;
+ params.afmt = AFMT_WAVPACK;
+ input_size = PCM_CHUNK_SIZE*inputs.num_channels / 2;
+ data_size = 105*input_size / 100;
+ input_size *= 2;
+ input_step = input_size / 4;
+ params.chunk_size = data_size;
+ params.enc_sample_rate = inputs.sample_rate;
+ params.reserve_bytes = 0;
+ params.events_callback = enc_events_callback;
- /* inform the main program about buffer dimensions and other params */
- /* add wav_header_mono as place holder to file start position */
- /* wav header and wvpk header have to be reordered later */
- ci->enc_set_parameters(chunk_size, num_chunks, samp_per_chunk,
- wav_header_mono, sizeof(wav_header_mono),
- AFMT_WAVPACK);
+ ci->enc_set_parameters(&params);
- wpc = WavpackOpenFileOutput ();
+ return true;
+} /* init_encoder */
+
+enum codec_status codec_start(struct codec_api* api)
+{
+ bool cpu_boosted;
+
+ ci = api; /* copy to global api pointer */
- memset (&config, 0, sizeof (config));
- config.bits_per_sample = 16;
- config.bytes_per_sample = 2;
- config.sample_rate = 44100;
- config.num_channels = enc_channels;
+#ifdef USE_IRAM
+ ci->memcpy(iramstart, iramcopy, iramend - iramstart);
+ ci->memset(iedata, 0, iend - iedata);
+#endif
- if (!WavpackSetConfiguration (wpc, &config, 1))
+ /* initialize params and config */
+ if (!init_encoder())
+ {
+ ci->enc_codec_loaded = -1;
return CODEC_ERROR;
+ }
/* main application waits for this flag during encoder loading */
- ci->enc_codec_loaded = true;
+ ci->enc_codec_loaded = 1;
+
+ ci->cpu_boost(true);
+ cpu_boosted = true;
/* main encoding loop */
while(!ci->stop_codec)
{
- while((src = (uint32*)ci->enc_get_wav_data(CHUNK_SIZE)) != NULL)
+ uint8_t *src;
+
+ while ((src = ci->enc_get_pcm_data(PCM_CHUNK_SIZE)) != NULL)
{
+ struct enc_chunk_hdr *chunk;
+ bool abort_chunk;
+ uint8_t *dst;
+ uint8_t *src_end;
+
if(ci->stop_codec)
break;
- if(ci->enc_wavbuf_near_empty() == 0)
+ abort_chunk = true;
+
+ if (!cpu_boosted && ci->enc_pcm_buf_near_empty() == 0)
{
- if(!cpu_boosted)
- {
- ci->cpu_boost(true);
- cpu_boosted = true;
- }
+ ci->cpu_boost(true);
+ cpu_boosted = true;
}
- dst = (uint32*)ci->enc_alloc_chunk() + 1;
+ chunk = ci->enc_get_chunk();
- WavpackStartBlock (wpc, (uint8*)dst, (uint8*)dst + CHUNK_SIZE);
+ /* reset counts and pointer */
+ chunk->enc_size = 0;
+ chunk->num_pcm = 0;
+ chunk->enc_data = NULL;
- if(enc_channels == 2)
- {
- for (i=0; i<CHUNK_SIZE/4; i++)
- {
- t = (long)*src++;
+ dst = ENC_CHUNK_SKIP_HDR(dst, chunk);
- input_buffer[2*i + 0] = t >> 16;
- input_buffer[2*i + 1] = (short)t;
- }
- }
- else
+ WavpackStartBlock(wpc, dst, dst + data_size);
+
+ chunk_to_int32((uint32_t*)src);
+ src = input_buffer;
+ src_end = src + input_size;
+
+ /* encode chunk in four steps yielding between each */
+ do
{
- for (i=0; i<CHUNK_SIZE/4; i++)
+ if (WavpackPackSamples(wpc, (int32_t *)src,
+ PCM_SAMP_PER_CHUNK/4))
{
- t = (long)*src++;
- t = (((t<<16)>>16) + (t>>16)) >> 1; /* left+right */
-
- input_buffer[i] = t;
+ chunk->num_pcm += PCM_SAMP_PER_CHUNK/4;
+ ci->yield();
+ /* could've been stopped in some way */
+ abort_chunk = ci->stop_codec ||
+ (chunk->flags & CHUNKF_ABORT);
}
- }
- if (!WavpackPackSamples (wpc, input_buffer, CHUNK_SIZE/4))
- return CODEC_ERROR;
+ src += input_step;
+ }
+ while (!abort_chunk && src < src_end);
+ if (!abort_chunk)
+ {
+ chunk->enc_data = dst;
+ if (chunk->num_pcm < PCM_SAMP_PER_CHUNK)
+ ci->enc_unget_pcm_data(PCM_CHUNK_SIZE - chunk->num_pcm*4);
/* finish the chunk and store chunk size info */
- dst[-1] = WavpackFinishBlock (wpc);
-
- ci->enc_free_chunk();
- ci->yield();
+ chunk->enc_size = WavpackFinishBlock(wpc);
+ ci->enc_finish_chunk();
+ }
}
- if(ci->enc_wavbuf_near_empty())
+ if (cpu_boosted && ci->enc_pcm_buf_near_empty() != 0)
{
- if(cpu_boosted)
- {
- ci->cpu_boost(false);
- cpu_boosted = false;
- }
+ ci->cpu_boost(false);
+ cpu_boosted = false;
}
+
ci->yield();
}
- if(cpu_boosted) /* set initial boost state */
+ if (cpu_boosted) /* set initial boost state */
ci->cpu_boost(false);
/* reset parameters to initial state */
- ci->enc_set_parameters(0, 0, 0, 0, 0, 0);
+ ci->enc_set_parameters(NULL);
/* main application waits for this flag during encoder removing */
- ci->enc_codec_loaded = false;
+ ci->enc_codec_loaded = 0;
return CODEC_OK;
-}
-#endif
+} /* codec_start */
+
+#endif /* ndef SIMULATOR */
diff --git a/apps/eq_menu.c b/apps/eq_menu.c
index 6c4dde4a78..9939ee77fe 100644
--- a/apps/eq_menu.c
+++ b/apps/eq_menu.c
@@ -710,7 +710,8 @@ static bool eq_save_preset(void)
char filename[MAX_PATH];
int *setting;
- create_numbered_filename(filename, EQS_DIR, "eq", ".cfg", 2);
+ create_numbered_filename(filename, EQS_DIR, "eq", ".cfg", 2
+ IF_CNFN_NUM_(, NULL));
/* allow user to modify filename */
while (true) {
diff --git a/apps/gui/statusbar.c b/apps/gui/statusbar.c
index a8b8c5061b..ae8bba0538 100644
--- a/apps/gui/statusbar.c
+++ b/apps/gui/statusbar.c
@@ -124,19 +124,6 @@
#endif
#define STATUSBAR_TIME_X_END(statusbar_width) statusbar_width - 1 - \
STATUSBAR_DISK_WIDTH
-#if defined(HAVE_RECORDING)
-/* analogue frequency numbers taken from the order of frequencies in sample_rate */
-#define FREQ_44 7
-#define FREQ_48 8
-#define FREQ_32 6
-#define FREQ_22 4
-#define FREQ_24 5
-#define FREQ_16 3
-#ifdef HAVE_SPDIF_IN
-#define SOURCE_SPDIF 2
-#endif
-#endif
-
struct gui_syncstatusbar statusbars;
void gui_statusbar_init(struct gui_statusbar * bar)
@@ -600,41 +587,113 @@ void gui_statusbar_time(struct screen * display, int hour, int minute)
#endif
#ifdef HAVE_RECORDING
-void gui_statusbar_icon_recording_info(struct screen * display)
+#if CONFIG_CODEC == SWCODEC
+/**
+ * Write a number to the display using bitmaps and return new position
+ */
+static int write_bitmap_number(struct screen * display, int value,
+ int x, int y)
{
-#if (CONFIG_CODEC != SWCODEC) || (defined(SIMULATOR) && defined(HAVE_SPDIF_IN))
- char buffer[3];
+ char buf[12], *ptr;
+ snprintf(buf, sizeof(buf), "%d", value);
+
+ for (ptr = buf; *ptr != '\0'; ptr++, x += BM_GLYPH_WIDTH)
+ display->mono_bitmap(bitmap_glyphs_4x8[*ptr - '0'], x, y,
+ BM_GLYPH_WIDTH, STATUSBAR_HEIGHT);
+ return x;
+}
+
+/**
+ * Write format info bitmaps - right justified
+ */
+static void gui_statusbar_write_format_info(struct screen * display)
+{
+ /* Can't fit info for sw codec targets in statusbar using FONT_SYSFIXED
+ so must use icons */
+ int rec_format = global_settings.rec_format;
+ unsigned bitrk = 0; /* compiler warns about unitialized use !! */
+ int xpos = STATUSBAR_ENCODER_X_POS;
+ int width = STATUSBAR_ENCODER_WIDTH;
+ const unsigned char *bm = bitmap_formats_18x8[rec_format];
+
+ if (rec_format == REC_FORMAT_MPA_L3)
+ {
+ /* Special handling for mp3 */
+ bitrk = global_settings.mp3_enc_config.bitrate;
+ bitrk = mp3_enc_bitr[bitrk];
+
+ width = BM_MPA_L3_M_WIDTH;
+
+ /* Slide 'M' to right if fewer than three digits used */
+ if (bitrk > 999)
+ bitrk = 999; /* neurotic safety check if corrupted */
+ else
+ {
+ if (bitrk < 100)
+ xpos += BM_GLYPH_WIDTH;
+ if (bitrk < 10)
+ xpos += BM_GLYPH_WIDTH;
+ }
+ }
+
+
+ /* Show bitmap - clipping right edge if needed */
+ display->mono_bitmap_part(bm, 0, 0, STATUSBAR_ENCODER_WIDTH,
+ xpos, STATUSBAR_Y_POS, width, STATUSBAR_HEIGHT);
+
+ if (rec_format == REC_FORMAT_MPA_L3)
+ {
+ xpos += BM_MPA_L3_M_WIDTH; /* to right of 'M' */
+ write_bitmap_number(display, bitrk, xpos, STATUSBAR_Y_POS);
+ }
+}
+
+/**
+ * Write sample rate using bitmaps - left justified
+ */
+static void gui_statusbar_write_samplerate_info(struct screen * display)
+{
+ unsigned long samprk;
+ int xpos;
+
+#ifdef SIMULATOR
+ samprk = 44100;
+#else
+#ifdef HAVE_SPDIF_IN
+ if (global_settings.rec_source == AUDIO_SRC_SPDIF)
+ /* Use rate in use, not current measured rate if it changed */
+ samprk = pcm_rec_sample_rate();
+ else
#endif
+ samprk = rec_freq_sampr[global_settings.rec_frequency];
+#endif /* SIMULATOR */
+
+ samprk /= 1000;
+ if (samprk > 99)
+ samprk = 99; /* Limit to 3 glyphs */
+
+ xpos = write_bitmap_number(display, (unsigned)samprk,
+ STATUSBAR_RECFREQ_X_POS, STATUSBAR_Y_POS);
+
+ /* write the 'k' */
+ display->mono_bitmap(bitmap_glyphs_4x8[Glyph_4x8_k], xpos,
+ STATUSBAR_Y_POS, BM_GLYPH_WIDTH,
+ STATUSBAR_HEIGHT);
+}
+#endif /* CONFIG_CODEC == SWCODEC */
+
+void gui_statusbar_icon_recording_info(struct screen * display)
+{
#if CONFIG_CODEC != SWCODEC
+ char buffer[3];
int width, height;
- static char* const sample_rate[12] =
- {
- "8",
- "11",
- "12",
- "16",
- "22",
- "24",
- "32",
- "44",
- "48",
- "64",
- "88",
- "96"
- };
-
display->setfont(FONT_SYSFIXED);
-#endif
+#endif /* CONFIG_CODEC != SWCODEC */
/* Display Codec info in statusbar */
#if CONFIG_CODEC == SWCODEC
- /* Can't fit info for sw codec targets in statusbar using FONT_SYSFIXED
- so must use icons */
- display->mono_bitmap(bitmap_icons_18x8[global_settings.rec_quality],
- STATUSBAR_ENCODER_X_POS, STATUSBAR_Y_POS,
- STATUSBAR_ENCODER_WIDTH, STATUSBAR_HEIGHT);
-#else
-
+ gui_statusbar_write_format_info(display);
+#else /* !SWCODEC */
display->mono_bitmap(bitmap_icons_5x8[Icon_q],
STATUSBAR_ENCODER_X_POS + 8, STATUSBAR_Y_POS,
5, STATUSBAR_HEIGHT);
@@ -643,56 +702,37 @@ void gui_statusbar_icon_recording_info(struct screen * display)
display->getstringsize(buffer, &width, &height);
if (height <= STATUSBAR_HEIGHT)
display->putsxy(STATUSBAR_ENCODER_X_POS + 13, STATUSBAR_Y_POS, buffer);
-#endif
+#endif /* CONFIG_CODEC == SWCODEC */
/* Display Samplerate info in statusbar */
-#if defined(HAVE_SPDIF_IN)
- if (global_settings.rec_source == SOURCE_SPDIF)
+#if CONFIG_CODEC == SWCODEC
+ /* SWCODEC targets use bitmaps for glyphs */
+ gui_statusbar_write_samplerate_info(display);
+#else /* !SWCODEC */
+ /* hwcodec targets have sysfont characters */
+#ifdef HAVE_SPDIF_IN
+ if (global_settings.rec_source == AUDIO_SRC_SPDIF)
{
-#if (CONFIG_CODEC != MAS3587F) && !defined(SIMULATOR)
- display->mono_bitmap(bitmap_icons_12x8[audio_get_spdif_sample_rate()],
- STATUSBAR_RECFREQ_X_POS, STATUSBAR_Y_POS,
- STATUSBAR_RECFREQ_WIDTH, STATUSBAR_HEIGHT);
-#else
/* Can't measure S/PDIF sample rate on Archos/Sim yet */
snprintf(buffer, sizeof(buffer), "--");
-#endif
}
else
#endif /* HAVE_SPDIF_IN */
{
- /* Analogue frequency in wrong order so remap settings numbers */
- int freq = global_settings.rec_frequency;
- if (freq == 0)
- freq = FREQ_44;
- else if (freq == 1)
- freq = FREQ_48;
- else if (freq == 2)
- freq = FREQ_32;
- else if (freq == 3)
- freq = FREQ_22;
- else if (freq == 4)
- freq = FREQ_24;
- else if (freq == 5)
- freq = FREQ_16;
-
-#if CONFIG_CODEC == SWCODEC
- /* samplerate icons for swcodec targets*/
- display->mono_bitmap(bitmap_icons_12x8[freq],
- STATUSBAR_RECFREQ_X_POS, STATUSBAR_Y_POS,
- STATUSBAR_RECFREQ_WIDTH, STATUSBAR_HEIGHT);
- }
-#else
- /* hwcodec targets have sysfont characters */
- snprintf(buffer, sizeof(buffer), "%s", sample_rate[freq]);
- display->getstringsize(buffer, &width, &height);
+ static char const * const freq_strings[12] =
+ { "44", "48", "32", "22", "24", "16" };
+ snprintf(buffer, sizeof(buffer), "%s",
+ freq_strings[global_settings.rec_frequency]);
}
- if (height <= STATUSBAR_HEIGHT)
- display->putsxy(STATUSBAR_RECFREQ_X_POS, STATUSBAR_Y_POS, buffer);
+ display->getstringsize(buffer, &width, &height);
+
+ if (height <= STATUSBAR_HEIGHT)
+ display->putsxy(STATUSBAR_RECFREQ_X_POS, STATUSBAR_Y_POS, buffer);
+
+ display->setfont(FONT_UI);
+#endif /* CONFIG_CODEC == SWCODEC */
- display->setfont(FONT_UI);
-#endif
/* Display Channel status in status bar */
if(global_settings.rec_channels)
{
diff --git a/apps/lang/english.lang b/apps/lang/english.lang
index 40d7bb7b19..8f7deb78a9 100644
--- a/apps/lang/english.lang
+++ b/apps/lang/english.lang
@@ -9802,7 +9802,7 @@
</phrase>
<phrase>
id: VOICE_KBIT_PER_SEC
- desc: spoken only, for file extension
+ desc: spoken only, a unit postfix
user:
<source>
*: ""
@@ -10032,3 +10032,115 @@
*: ""
</voice>
</phrase>
+<phrase>
+ id: LANG_RECORDING_FORMAT
+ desc: audio format item in recording menu
+ user:
+ <source>
+ *: "Format"
+ </source>
+ <dest>
+ *: "Format"
+ </dest>
+ <voice>
+ *: "Format"
+ </voice>
+</phrase>
+<phrase>
+ id: LANG_AFMT_MPA_L3
+ desc: audio format description
+ user:
+ <source>
+ *: "MPEG Layer 3"
+ </source>
+ <dest>
+ *: "MPEG Layer 3"
+ </dest>
+ <voice>
+ *: "MPEG Layer 3"
+ </voice>
+</phrase>
+<phrase>
+ id: LANG_AFMT_PCM_WAV
+ desc: audio format description
+ user:
+ <source>
+ *: "PCM Wave"
+ </source>
+ <dest>
+ *: "PCM Wave"
+ </dest>
+ <voice>
+ *: "PCM Wave"
+ </voice>
+</phrase>
+<phrase>
+ id: LANG_AFMT_WAVPACK
+ desc: audio format description
+ user:
+ <source>
+ *: "WavPack"
+ </source>
+ <dest>
+ *: "WavPack"
+ </dest>
+ <voice>
+ *: "WavPack"
+ </voice>
+</phrase>
+<phrase>
+ id: LANG_ENCODER_SETTINGS
+ desc: encoder settings
+ user:
+ <source>
+ *: "Encoder Settings"
+ </source>
+ <dest>
+ *: "Encoder Settings"
+ </dest>
+ <voice>
+ *: "Encoder Settings"
+ </voice>
+</phrase>
+<phrase>
+ id: LANG_NO_SETTINGS
+ desc: when something has settings in a certain context
+ user:
+ <source>
+ *: "(No Settings)"
+ </source>
+ <dest>
+ *: "(No Settings)"
+ </dest>
+ <voice>
+ *: "No settings available"
+ </voice>
+</phrase>
+<phrase>
+ id: LANG_SOURCE_FREQUENCY
+ desc: when recording source frequency setting must follow source
+ user:
+ <source>
+ *: "(Same As Source)"
+ </source>
+ <dest>
+ *: "(Same As Source)"
+ </dest>
+ <voice>
+ *: "Same As Source"
+ </voice>
+</phrase>
+<phrase>
+ id: LANG_BITRATE
+ desc: bits-kilobits per unit time
+ user:
+ <source>
+ *: "Bitrate"
+ </source>
+ <dest>
+ *: "Bitrate"
+ </dest>
+ <voice>
+ *: "Bitrate"
+ </voice>
+</phrase>
diff --git a/apps/main.c b/apps/main.c
index c4ee45cb89..05b4ab54a3 100644
--- a/apps/main.c
+++ b/apps/main.c
@@ -286,6 +286,9 @@ void init(void)
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
set_cpu_frequency(CPUFREQ_NORMAL);
+#ifdef CPU_COLDFIRE
+ coldfire_set_pllcr_audio_bits(DEFAULT_PLLCR_AUDIO_BITS);
+#endif
cpu_boost_id(true, CPUBOOSTID_MAININIT);
#endif
diff --git a/apps/metadata.c b/apps/metadata.c
index ee0100ecf7..845536877c 100644
--- a/apps/metadata.c
+++ b/apps/metadata.c
@@ -88,36 +88,6 @@ struct apetag_item_header
long flags;
};
-struct format_list
-{
- char format;
- char extension[5];
-};
-
-static const struct format_list formats[] =
-{
- { AFMT_MPA_L1, "mp1" },
- { AFMT_MPA_L2, "mp2" },
- { AFMT_MPA_L2, "mpa" },
- { AFMT_MPA_L3, "mp3" },
-#if CONFIG_CODEC == SWCODEC
- { AFMT_OGG_VORBIS, "ogg" },
- { AFMT_PCM_WAV, "wav" },
- { AFMT_FLAC, "flac" },
- { AFMT_MPC, "mpc" },
- { AFMT_A52, "a52" },
- { AFMT_A52, "ac3" },
- { AFMT_WAVPACK, "wv" },
- { AFMT_ALAC, "m4a" },
- { AFMT_AAC, "mp4" },
- { AFMT_SHN, "shn" },
- { AFMT_AIFF, "aif" },
- { AFMT_AIFF, "aiff" },
- { AFMT_SID, "sid" },
- { AFMT_ADX, "adx" },
-#endif
-};
-
#if CONFIG_CODEC == SWCODEC
static const unsigned short a52_bitrates[] =
{
@@ -1691,14 +1661,24 @@ unsigned int probe_file_format(const char *filename)
return AFMT_UNKNOWN;
}
- suffix += 1;
+ /* skip '.' */
+ suffix++;
+
+ for (i = 1; i < AFMT_NUM_CODECS; i++)
+ {
+ /* search extension list for type */
+ const char *ext = audio_formats[i].ext_list;
- for (i = 0; i < sizeof(formats) / sizeof(formats[0]); i++)
+ do
{
- if (strcasecmp(suffix, formats[i].extension) == 0)
+ if (strcasecmp(suffix, ext) == 0)
{
- return formats[i].format;
+ return i;
+ }
+
+ ext += strlen(ext) + 1;
}
+ while (*ext != '\0');
}
return AFMT_UNKNOWN;
diff --git a/apps/misc.c b/apps/misc.c
index c36d61914b..01463851be 100644
--- a/apps/misc.c
+++ b/apps/misc.c
@@ -58,6 +58,8 @@
#endif /* End HAVE_LCD_BITMAP */
#include "gui/gwps-common.h"
+#include "misc.h"
+
/* Format a large-range value for output, using the appropriate unit so that
* the displayed value is in the range 1 <= display < 1000 (1024 for "binary"
* units) if possible, and 3 significant digits are shown. If a buffer is
@@ -114,16 +116,20 @@ char *output_dyn_value(char *buf, int buf_size, int value,
}
/* Create a filename with a number part in a way that the number is 1
- higher than the highest numbered file matching the same pattern.
- It is allowed that buffer and path point to the same memory location,
- saving a strcpy(). Path must always be given without trailing slash,. */
+ * higher than the highest numbered file matching the same pattern.
+ * It is allowed that buffer and path point to the same memory location,
+ * saving a strcpy(). Path must always be given without trailing slash.
+ * "num" can point to an int specifying the number to use or NULL or a value
+ * less than zero to number automatically. The final number used will also
+ * be returned in *num. If *num is >= 0 then *num will be incremented by
+ * one. */
char *create_numbered_filename(char *buffer, const char *path,
const char *prefix, const char *suffix,
- int numberlen)
+ int numberlen IF_CNFN_NUM_(, int *num))
{
DIR *dir;
struct dirent *entry;
- int max_num = 0;
+ int max_num;
int pathlen;
int prefixlen = strlen(prefix);
char fmtstring[12];
@@ -133,6 +139,18 @@ char *create_numbered_filename(char *buffer, const char *path,
pathlen = strlen(buffer);
+#ifdef IF_CNFN_NUM
+ if (num && *num >= 0)
+ {
+ /* number specified */
+ max_num = *num;
+ }
+ else
+#endif
+ {
+ /* automatic numbering */
+ max_num = 0;
+
dir = opendir(pathlen ? buffer : "/");
if (!dir)
return NULL;
@@ -149,11 +167,20 @@ char *create_numbered_filename(char *buffer, const char *path,
if (curr_num > max_num)
max_num = curr_num;
}
+
closedir(dir);
+ }
+
+ max_num++;
snprintf(fmtstring, sizeof(fmtstring), "/%%s%%0%dd%%s", numberlen);
snprintf(buffer + pathlen, MAX_PATH - pathlen, fmtstring, prefix,
- max_num + 1, suffix);
+ max_num, suffix);
+
+#ifdef IF_CNFN_NUM
+ if (num)
+ *num = max_num;
+#endif
return buffer;
}
@@ -161,13 +188,22 @@ char *create_numbered_filename(char *buffer, const char *path,
#ifdef CONFIG_RTC
/* Create a filename with a date+time part.
It is allowed that buffer and path point to the same memory location,
- saving a strcpy(). Path must always be given without trailing slash. */
+ saving a strcpy(). Path must always be given without trailing slash.
+ unique_time as true makes the function wait until the current time has
+ changed. */
char *create_datetime_filename(char *buffer, const char *path,
- const char *prefix, const char *suffix)
+ const char *prefix, const char *suffix,
+ bool unique_time)
{
struct tm *tm = get_time();
+ static struct tm last_tm;
int pathlen;
+ while (unique_time && !memcmp(get_time(), &last_tm, sizeof (struct tm)))
+ sleep(HZ/10);
+
+ last_tm = *tm;
+
if (buffer != path)
strncpy(buffer, path, MAX_PATH);
@@ -356,9 +392,10 @@ void screen_dump(void)
#endif
#ifdef CONFIG_RTC
- create_datetime_filename(filename, "", "dump ", ".bmp");
+ create_datetime_filename(filename, "", "dump ", ".bmp", false);
#else
- create_numbered_filename(filename, "", "dump_", ".bmp", 4);
+ create_numbered_filename(filename, "", "dump_", ".bmp", 4
+ IF_CNFN_NUM_(, NULL));
#endif
fh = creat(filename, O_WRONLY);
diff --git a/apps/misc.h b/apps/misc.h
index 1bc9a23447..6c660e0a5e 100644
--- a/apps/misc.h
+++ b/apps/misc.h
@@ -19,21 +19,46 @@
#ifndef MISC_H
#define MISC_H
+#include <stdbool.h>
+
/* Format a large-range value for output, using the appropriate unit so that
* the displayed value is in the range 1 <= display < 1000 (1024 for "binary"
* units) if possible, and 3 significant digits are shown. If a buffer is
* given, the result is snprintf()'d into that buffer, otherwise the result is
* voiced.*/
-void output_dyn_value(char *buf, int buf_size, int value,
+char *output_dyn_value(char *buf, int buf_size, int value,
const unsigned char **units, bool bin_scale);
+/* Create a filename with a number part in a way that the number is 1
+ * higher than the highest numbered file matching the same pattern.
+ * It is allowed that buffer and path point to the same memory location,
+ * saving a strcpy(). Path must always be given without trailing slash.
+ *
+ * "num" can point to an int specifying the number to use or NULL or a value
+ * less than zero to number automatically. The final number used will also
+ * be returned in *num. If *num is >= 0 then *num will be incremented by
+ * one. */
+#if CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING) && !defined(CONFIG_RTC)
+/* this feature is needed by SWCODEC recording without a RTC to prevent
+ disk access when changing files */
+#define IF_CNFN_NUM_(...) __VA_ARGS__
+#define IF_CNFN_NUM
+#else
+#define IF_CNFN_NUM_(...)
+#endif
char *create_numbered_filename(char *buffer, const char *path,
const char *prefix, const char *suffix,
- int numberlen);
+ int numberlen IF_CNFN_NUM_(, int *num));
#ifdef CONFIG_RTC
+/* Create a filename with a date+time part.
+ It is allowed that buffer and path point to the same memory location,
+ saving a strcpy(). Path must always be given without trailing slash.
+ unique_time as true makes the function wait until the current time has
+ changed. */
char *create_datetime_filename(char *buffer, const char *path,
- const char *prefix, const char *suffix);
-#endif
+ const char *prefix, const char *suffix,
+ bool unique_time);
+#endif /* CONFIG_RTC */
/* Read (up to) a line of text from fd into buffer and return number of bytes
* read (which may be larger than the number of bytes stored in buffer). If
@@ -57,4 +82,4 @@ long default_event_handler(long event);
void car_adapter_mode_init(void);
extern int show_logo(void);
-#endif
+#endif /* MISC_H */
diff --git a/apps/pcmbuf.c b/apps/pcmbuf.c
index 44f175c67d..5119d20ebd 100644
--- a/apps/pcmbuf.c
+++ b/apps/pcmbuf.c
@@ -51,9 +51,11 @@ struct pcmbufdesc
void (*callback)(void);
};
+#define PCMBUF_DESCS(bufsize) ((bufsize) / PCMBUF_MINAVG_CHUNK)
+
/* Size of the PCM buffer. */
static size_t pcmbuf_size IDATA_ATTR = 0;
-
+static char *pcmbuf_bufend IDATA_ATTR;
static char *audiobuffer IDATA_ATTR;
/* Current audio buffer write index. */
static size_t audiobuffer_pos IDATA_ATTR;
@@ -360,7 +362,7 @@ int pcmbuf_used_descs(void) {
}
int pcmbuf_descs(void) {
- return pcmbuf_size / PCMBUF_MINAVG_CHUNK;
+ return PCMBUF_DESCS(pcmbuf_size);
}
size_t get_pcmbuf_descsize(void) {
@@ -371,28 +373,37 @@ static void pcmbuf_init_pcmbuffers(void) {
struct pcmbufdesc *next = pcmbuf_write;
next++;
pcmbuf_write_end = pcmbuf_write;
- while ((void *)next < (void *)audiobufend) {
+ while ((void *)next < (void *)pcmbuf_bufend) {
pcmbuf_write_end->link=next;
pcmbuf_write_end=next;
next++;
}
}
+bool pcmbuf_is_same_size(size_t bufsize)
+{
+ /* keep calculations synced with pcmbuf_init */
+ bufsize += PCMBUF_MIX_CHUNK * 2 + PCMBUF_DESCS(bufsize);
+ return bufsize == (size_t)(pcmbuf_bufend - audiobuffer);
+}
+
/* Initialize the pcmbuffer the structure looks like this:
- * ...CODECBUFFER|---------PCMBUF---------|GUARDBUF|DESCS| */
-void pcmbuf_init(size_t bufsize)
+ * ...|---------PCMBUF---------|FADEBUF|VOICEBUF|DESCS|... */
+size_t pcmbuf_init(size_t bufsize, char *bufend)
{
pcmbuf_size = bufsize;
+ pcmbuf_bufend = bufend;
pcmbuf_descsize = pcmbuf_descs()*sizeof(struct pcmbufdesc);
- audiobuffer = (char *)&audiobuf[(audiobufend - audiobuf) -
- (pcmbuf_size + PCMBUF_MIX_CHUNK * 2 + pcmbuf_descsize)];
+ audiobuffer = pcmbuf_bufend - (pcmbuf_size + PCMBUF_MIX_CHUNK * 2
+ + pcmbuf_descsize);
fadebuf = &audiobuffer[pcmbuf_size];
voicebuf = &fadebuf[PCMBUF_MIX_CHUNK];
- pcmbuf_write = (struct pcmbufdesc *)(&voicebuf[PCMBUF_MIX_CHUNK]);
+ pcmbuf_write = (struct pcmbufdesc *)&voicebuf[PCMBUF_MIX_CHUNK];
pcmbuf_init_pcmbuffers();
position_callback = NULL;
pcmbuf_event_handler = NULL;
pcmbuf_play_stop();
+ return pcmbuf_bufend - audiobuffer;
}
size_t pcmbuf_get_bufsize(void)
diff --git a/apps/pcmbuf.h b/apps/pcmbuf.h
index b5035f4405..a408cdae42 100644
--- a/apps/pcmbuf.h
+++ b/apps/pcmbuf.h
@@ -38,7 +38,7 @@
/* Returns true if the buffer needs to change size */
bool pcmbuf_is_same_size(size_t bufsize);
-void pcmbuf_init(size_t bufsize);
+size_t pcmbuf_init(size_t bufsize, char *bufend);
/* Size in bytes used by the pcmbuffer */
size_t pcmbuf_get_bufsize(void);
size_t get_pcmbuf_descsize(void);
diff --git a/apps/playback.c b/apps/playback.c
index f8372665a4..af6b573f1d 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -54,8 +54,6 @@
#include "playlist.h"
#include "playback.h"
#include "pcmbuf.h"
-#include "pcm_playback.h"
-#include "pcm_record.h"
#include "buffer.h"
#include "dsp.h"
#include "abrepeat.h"
@@ -78,6 +76,7 @@
#ifdef HAVE_RECORDING
#include "recording.h"
+#include "talk.h"
#endif
#define PLAYBACK_VOICE
@@ -93,9 +92,13 @@
* for their correct seeek target, 32k seems a good size */
#define AUDIO_REBUFFER_GUESS_SIZE (1024*32)
-/* macros to enable logf for queues */
+/* macros to enable logf for queues
+ logging on SYS_TIMEOUT can be disabled */
#ifdef SIMULATOR
-#define PLAYBACK_LOGQUEUES /* Define this for logf output of all queuing */
+/* Define this for logf output of all queuing except SYS_TIMEOUT */
+#define PLAYBACK_LOGQUEUES
+/* Define this to logf SYS_TIMEOUT messages */
+#define PLAYBACK_LOGQUEUES_SYS_TIMEOUT
#endif
#ifdef PLAYBACK_LOGQUEUES
@@ -104,6 +107,18 @@
#define LOGFQUEUE(s)
#endif
+#ifdef PLAYBACK_LOGQUEUES_SYS_TIMEOUT
+#define LOGFQUEUE_SYS_TIMEOUT(s) logf("%s", s)
+#else
+#define LOGFQUEUE_SYS_TIMEOUT(s)
+#endif
+
+
+/* Define one constant that includes recording related functionality */
+#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
+#define AUDIO_HAVE_RECORDING
+#endif
+
enum {
Q_AUDIO_PLAY = 1,
Q_AUDIO_STOP,
@@ -122,6 +137,9 @@ enum {
#if MEM > 8
Q_AUDIO_FILL_BUFFER_IF_ACTIVE_ATA,
#endif
+#ifdef AUDIO_HAVE_RECORDING
+ Q_AUDIO_LOAD_ENCODER,
+#endif
Q_CODEC_REQUEST_PENDING,
Q_CODEC_REQUEST_COMPLETE,
@@ -133,7 +151,7 @@ enum {
Q_CODEC_LOAD,
Q_CODEC_LOAD_DISK,
-#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
+#ifdef AUDIO_HAVE_RECORDING
Q_ENCODER_LOAD_DISK,
Q_ENCODER_RECORD,
#endif
@@ -178,11 +196,16 @@ static volatile bool paused; /* Is audio paused? (A/C-) */
static volatile bool filling IDATA_ATTR; /* Is file buffer currently being refilled? (A/C-) */
/* Ring buffer where tracks and codecs are loaded */
-static char *filebuf; /* Pointer to start of ring buffer (A/C-) */
+static unsigned char *filebuf; /* Pointer to start of ring buffer (A/C-) */
size_t filebuflen; /* Total size of the ring buffer FIXME: make static (A/C-)*/
static volatile size_t buf_ridx IDATA_ATTR; /* Ring buffer read position (A/C) FIXME? should be (C/A-) */
static volatile size_t buf_widx IDATA_ATTR; /* Ring buffer read position (A/C-) */
+#define BUFFER_STATE_TRASHED -1 /* Buffer is in a trashed state and must be reset */
+#define BUFFER_STATE_NORMAL 0 /* Buffer is arranged for voice and audio */
+#define BUFFER_STATE_VOICED_ONLY 1 /* Buffer is arranged for voice-only use */
+static int buffer_state = BUFFER_STATE_TRASHED; /* Buffer state */
+
#define RINGBUF_ADD(p,v) ((p+v)<filebuflen ? p+v : p+v-filebuflen)
#define RINGBUF_SUB(p,v) ((p>=v) ? p-v : p+filebuflen-v)
#define RINGBUF_ADD_CROSS(p1,v,p2) ((p1<p2)?(int)(p1+v)-(int)p2:(int)(p1+v-p2)-(int)filebuflen)
@@ -235,7 +258,7 @@ static const char audio_thread_name[] = "audio";
static void audio_thread(void);
static void audio_initiate_track_change(long direction);
static bool audio_have_tracks(void);
-static void audio_reset_buffer(void);
+static void audio_reset_buffer(size_t pcmbufsize);
/* Codec thread */
extern struct codec_api ci;
@@ -294,6 +317,10 @@ static void voice_thread(void);
void mp3_play_data(const unsigned char* start, int size,
void (*get_more)(unsigned char** start, int* size))
{
+ /* must reset the buffer before any playback begins if needed */
+ if (buffer_state == BUFFER_STATE_TRASHED)
+ audio_reset_buffer(pcmbuf_get_bufsize());
+
#ifdef PLAYBACK_VOICE
static struct voice_info voice_clip;
voice_clip.callback = get_more;
@@ -330,38 +357,95 @@ void mpeg_id3_options(bool _v1first)
v1first = _v1first;
}
-void audio_load_encoder(int enc_id)
+unsigned char *audio_get_buffer(bool talk_buf, size_t *buffer_size)
{
-#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
- const char *enc_fn = get_codec_filename(enc_id | CODEC_TYPE_ENCODER);
+ unsigned char *buf = audiobuf;
+ unsigned char *end = audiobufend;
+
+ audio_stop();
+
+ if (talk_buf || !talk_voice_required()
+ || buffer_state == BUFFER_STATE_TRASHED)
+ {
+ logf("get buffer: talk_buf");
+ /* ok to use everything from audiobuf to audiobufend */
+ if (buffer_state != BUFFER_STATE_TRASHED)
+ talk_buffer_steal();
+ buffer_state = BUFFER_STATE_TRASHED;
+ }
+ else
+ {
+ /* skip talk buffer and move pcm buffer to end */
+ logf("get buffer: voice");
+ mp3_play_stop();
+ buf += talk_get_bufsize();
+ end -= pcmbuf_init(pcmbuf_get_bufsize(), audiobufend);
+ buffer_state = BUFFER_STATE_VOICED_ONLY;
+ }
+
+ *buffer_size = end - buf;
+
+ return buf;
+}
+
+#ifdef HAVE_RECORDING
+unsigned char *audio_get_recording_buffer(size_t *buffer_size)
+{
+ /* don't allow overwrite of voice swap area or we'll trash the
+ swapped-out voice codec but can use whole thing if none */
+ unsigned char *end = iram_buf[CODEC_IDX_VOICE] ?
+ iram_buf[CODEC_IDX_VOICE] : audiobufend;
+
+ audio_stop();
+ talk_buffer_steal();
+
+ buffer_state = BUFFER_STATE_TRASHED;
+
+ *buffer_size = end - audiobuf;
+
+ return (unsigned char *)audiobuf;
+}
+
+bool audio_load_encoder(int afmt)
+{
+#ifndef SIMULATOR
+ const char *enc_fn = get_codec_filename(afmt | CODEC_TYPE_ENCODER);
if (!enc_fn)
- return;
+ return false;
audio_remove_encoder();
+ ci.enc_codec_loaded = 0; /* clear any previous error condition */
- LOGFQUEUE("audio > codec Q_ENCODER_LOAD_DISK");
- queue_post(&codec_queue, Q_ENCODER_LOAD_DISK, (void *)enc_fn);
+ LOGFQUEUE("audio > Q_AUDIO_LOAD_ENCODER");
+ queue_post(&audio_queue, Q_AUDIO_LOAD_ENCODER, (void *)enc_fn);
- while (!ci.enc_codec_loaded)
+ while (ci.enc_codec_loaded == 0)
yield();
+
+ logf("codec loaded: %d", ci.enc_codec_loaded);
+
+ return ci.enc_codec_loaded > 0;
+#else
+ (void)afmt;
+ return true;
#endif
- return;
- (void)enc_id;
} /* audio_load_encoder */
void audio_remove_encoder(void)
{
-#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
- /* force encoder codec unload (if previously loaded) */
- if (!ci.enc_codec_loaded)
+#ifndef SIMULATOR
+ /* force encoder codec unload (if currently loaded) */
+ if (ci.enc_codec_loaded <= 0)
return;
ci.stop_codec = true;
- while (ci.enc_codec_loaded)
+ while (ci.enc_codec_loaded > 0)
yield();
#endif
} /* audio_remove_encoder */
+#endif /* HAVE_RECORDING */
+
struct mp3entry* audio_current_track(void)
{
const char *filename;
@@ -553,6 +637,9 @@ void audio_flush_and_reload_tracks(void)
void audio_error_clear(void)
{
+#ifdef AUDIO_HAVE_RECORDING
+ pcm_rec_error_clear();
+#endif
}
int audio_status(void)
@@ -573,11 +660,6 @@ int audio_status(void)
return ret;
}
-bool audio_query_poweroff(void)
-{
- return !(playing && paused);
-}
-
int audio_get_file_pos(void)
{
return 0;
@@ -617,7 +699,7 @@ void audio_set_crossfade(int enable)
enable = 0;
size = NATIVE_FREQUENCY*2;
#endif
- if (pcmbuf_get_bufsize() == size)
+ if (buffer_state == BUFFER_STATE_NORMAL && pcmbuf_is_same_size(size))
return ;
if (was_playing)
@@ -633,9 +715,8 @@ void audio_set_crossfade(int enable)
voice_stop();
/* Re-initialize audio system. */
- pcmbuf_init(size);
+ audio_reset_buffer(size);
pcmbuf_crossfade_enable(enable);
- audio_reset_buffer();
logf("abuf:%dB", pcmbuf_get_bufsize());
logf("fbuf:%dB", filebuflen);
@@ -714,8 +795,7 @@ void voice_stop(void)
{
#ifdef PLAYBACK_VOICE
/* Messages should not be posted to voice codec queue unless it is the
- current codec or deadlocks happen.
- -- jhMikeS */
+ current codec or deadlocks happen. */
if (current_codec != CODEC_IDX_VOICE)
return;
@@ -784,21 +864,32 @@ static void set_filebuf_watermark(int seconds)
conf_watermark = bytes;
}
-static const char * get_codec_filename(int enc_spec)
+static const char * get_codec_filename(int cod_spec)
{
const char *fname;
- int type = enc_spec & CODEC_TYPE_MASK;
- int afmt = enc_spec & CODEC_AFMT_MASK;
+
+#ifdef HAVE_RECORDING
+ /* Can choose decoder or encoder if one available */
+ int type = cod_spec & CODEC_TYPE_MASK;
+ int afmt = cod_spec & CODEC_AFMT_MASK;
if ((unsigned)afmt >= AFMT_NUM_CODECS)
type = AFMT_UNKNOWN | (type & CODEC_TYPE_MASK);
- fname = (type == CODEC_TYPE_DECODER) ?
- audio_formats[afmt].codec_fn : audio_formats[afmt].codec_enc_fn;
+ fname = (type == CODEC_TYPE_ENCODER) ?
+ audio_formats[afmt].codec_enc_root_fn :
+ audio_formats[afmt].codec_root_fn;
logf("%s: %d - %s",
(type == CODEC_TYPE_ENCODER) ? "Encoder" : "Decoder",
afmt, fname ? fname : "<unknown>");
+#else /* !HAVE_RECORDING */
+ /* Always decoder */
+ if ((unsigned)cod_spec >= AFMT_NUM_CODECS)
+ cod_spec = AFMT_UNKNOWN;
+ fname = audio_formats[cod_spec].codec_root_fn;
+ logf("Codec: %d - %s", cod_spec, fname ? fname : "<unknown>");
+#endif /* HAVE_RECORDING */
return fname;
} /* get_codec_filename */
@@ -940,7 +1031,7 @@ static void* voice_request_buffer_callback(size_t *realsize, size_t reqsize)
}
break;
-#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
+#ifdef AUDIO_HAVE_RECORDING
case Q_ENCODER_RECORD:
LOGFQUEUE("voice < Q_ENCODER_RECORD");
swap_codec();
@@ -995,7 +1086,7 @@ static void* voice_request_buffer_callback(size_t *realsize, size_t reqsize)
goto voice_play_clip;
case SYS_TIMEOUT:
- LOGFQUEUE("voice < SYS_TIMEOUT");
+ LOGFQUEUE_SYS_TIMEOUT("voice < SYS_TIMEOUT");
goto voice_play_clip;
default:
@@ -1773,7 +1864,7 @@ static void codec_thread(void)
#endif
break ;
-#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
+#ifdef AUDIO_HAVE_RECORDING
case Q_ENCODER_LOAD_DISK:
LOGFQUEUE("codec < Q_ENCODER_LOAD_DISK");
audio_codec_loaded = false; /* Not audio codec! */
@@ -1785,12 +1876,14 @@ static void codec_thread(void)
}
#endif
mutex_lock(&mutex_codecthread);
+ logf("loading encoder");
current_codec = CODEC_IDX_AUDIO;
ci.stop_codec = false;
status = codec_load_file((const char *)ev.data, &ci);
mutex_unlock(&mutex_codecthread);
+ logf("encoder stopped");
break;
-#endif
+#endif /* AUDIO_HAVE_RECORDING */
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
@@ -1872,6 +1965,24 @@ static void codec_thread(void)
}
break;
+#ifdef AUDIO_HAVE_RECORDING
+ case Q_ENCODER_LOAD_DISK:
+ LOGFQUEUE("codec < Q_ENCODER_LOAD_DISK");
+
+ if (status == CODEC_OK)
+ break;
+
+ logf("Encoder failure");
+ gui_syncsplash(HZ*2, true, "Encoder failure");
+
+ if (ci.enc_codec_loaded < 0)
+ break;
+
+ logf("Encoder failed to load");
+ ci.enc_codec_loaded = -1;
+ break;
+#endif /* AUDIO_HAVE_RECORDING */
+
default:
LOGFQUEUE("codec < default");
@@ -2992,6 +3103,10 @@ static void audio_play_start(size_t offset)
/* Wait for any previously playing audio to flush - TODO: Not necessary? */
audio_stop_codec_flush();
+ /* must reset the buffer before any playback begins if needed */
+ if (buffer_state != BUFFER_STATE_NORMAL)
+ audio_reset_buffer(pcmbuf_get_bufsize());
+
track_changed = true;
playlist_end = false;
@@ -3084,51 +3199,60 @@ static void audio_initiate_dir_change(long direction)
ci.new_track = direction;
}
-static void audio_reset_buffer(void)
+/*
+ * Layout audio buffer as follows:
+ * [|TALK]|MALLOC|FILE|GUARD|PCM|AUDIOCODEC|[VOICECODEC|]
+ */
+static void audio_reset_buffer(size_t pcmbufsize)
{
+ /* see audio_get_recording_buffer if this is modified */
size_t offset;
- /* Set up file buffer as all space available */
- filebuf = (char *)&audiobuf[talk_get_bufsize()+MALLOC_BUFSIZE];
- filebuflen = audiobufend - (unsigned char *) filebuf - GUARD_BUFSIZE -
- (pcmbuf_get_bufsize() + get_pcmbuf_descsize() + PCMBUF_MIX_CHUNK * 2);
+ logf("audio_reset_buffer");
+ logf(" size:%08X", pcmbufsize);
+
+ /* Initially set up file buffer as all space available */
+ filebuf = audiobuf + MALLOC_BUFSIZE + talk_get_bufsize();
+ filebuflen = audiobufend - filebuf;
- /* Allow for codec(s) at end of file buffer */
+ /* Allow for codec(s) at end of audio buffer */
if (talk_voice_required())
{
- /* Allow 2 codecs at end of file buffer */
+#ifdef PLAYBACK_VOICE
+ /* Allow 2 codecs at end of audio buffer */
filebuflen -= 2 * (CODEC_IRAM_SIZE + CODEC_SIZE);
-#ifdef PLAYBACK_VOICE
- iram_buf[0] = &filebuf[filebuflen];
- iram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE];
- dram_buf[0] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE*2];
- dram_buf[1] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE*2+CODEC_SIZE];
+ iram_buf[CODEC_IDX_AUDIO] = filebuf + filebuflen;
+ dram_buf[CODEC_IDX_AUDIO] = iram_buf[CODEC_IDX_AUDIO] + CODEC_IRAM_SIZE;
+ iram_buf[CODEC_IDX_VOICE] = dram_buf[CODEC_IDX_AUDIO] + CODEC_SIZE;
+ dram_buf[CODEC_IDX_VOICE] = iram_buf[CODEC_IDX_VOICE] + CODEC_IRAM_SIZE;
#endif
}
else
{
- /* Allow for 1 codec at end of file buffer */
+#ifdef PLAYBACK_VOICE
+ /* Allow for 1 codec at end of audio buffer */
filebuflen -= CODEC_IRAM_SIZE + CODEC_SIZE;
-#ifdef PLAYBACK_VOICE
- iram_buf[0] = &filebuf[filebuflen];
- iram_buf[1] = NULL;
- dram_buf[0] = (unsigned char *)&filebuf[filebuflen+CODEC_IRAM_SIZE];
- dram_buf[1] = NULL;
+ iram_buf[CODEC_IDX_AUDIO] = filebuf + filebuflen;
+ dram_buf[CODEC_IDX_AUDIO] = iram_buf[CODEC_IDX_AUDIO] + CODEC_IRAM_SIZE;
+ iram_buf[CODEC_IDX_VOICE] = NULL;
+ dram_buf[CODEC_IDX_VOICE] = NULL;
#endif
}
+ filebuflen -= pcmbuf_init(pcmbufsize, filebuf + filebuflen) + GUARD_BUFSIZE;
+
/* Ensure that file buffer is aligned */
- offset = (-(size_t)filebuf) & 3;
+ offset = -(size_t)filebuf & 3;
filebuf += offset;
filebuflen -= offset;
filebuflen &= ~3;
/* Clear any references to the file buffer */
+ buffer_state = BUFFER_STATE_NORMAL;
}
-
#ifdef ROCKBOX_HAS_LOGF
static void audio_test_track_changed_event(struct mp3entry *id3)
{
@@ -3149,9 +3273,8 @@ static void audio_playback_init(void)
logf("playback api init");
pcm_init();
-#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
- /* Set the input multiplexer to Line In */
- pcm_rec_mux(0);
+#ifdef AUDIO_HAVE_RECORDING
+ rec_set_source(AUDIO_SRC_PLAYBACK, SRCF_PLAYBACK);
#endif
#ifdef ROCKBOX_HAS_LOGF
@@ -3219,8 +3342,9 @@ static void audio_playback_init(void)
#endif
}
- filebuf = (char *)&audiobuf[MALLOC_BUFSIZE]; /* Will be reset by reset_buffer */
-
+ /* initialize the buffer */
+ filebuf = audiobuf; /* must be non-NULL for audio_set_crossfade */
+ buffer_state = BUFFER_STATE_TRASHED; /* force it */
audio_set_crossfade(global_settings.crossfade);
audio_is_initialized = true;
@@ -3358,6 +3482,14 @@ static void audio_thread(void)
playlist_update_resume_info(audio_current_track());
break ;
+#ifdef AUDIO_HAVE_RECORDING
+ case Q_AUDIO_LOAD_ENCODER:
+ LOGFQUEUE("audio < Q_AUDIO_LOAD_ENCODER");
+ LOGFQUEUE("audio > codec Q_ENCODER_LOAD_DISK");
+ queue_post(&codec_queue, Q_ENCODER_LOAD_DISK, ev.data);
+ break;
+#endif
+
#ifndef SIMULATOR
case SYS_USB_CONNECTED:
LOGFQUEUE("audio < SYS_USB_CONNECTED");
@@ -3368,7 +3500,7 @@ static void audio_thread(void)
#endif
case SYS_TIMEOUT:
- LOGFQUEUE("audio < SYS_TIMEOUT");
+ LOGFQUEUE_SYS_TIMEOUT("audio < SYS_TIMEOUT");
break;
default:
diff --git a/apps/playlist.c b/apps/playlist.c
index 5a5313b736..134b52ea8b 100644
--- a/apps/playlist.c
+++ b/apps/playlist.c
@@ -155,7 +155,7 @@ static int recreate_control(struct playlist_info* playlist);
static void update_playlist_filename(struct playlist_info* playlist,
const char *dir, const char *file);
static int add_indices_to_playlist(struct playlist_info* playlist,
- char* buffer, int buflen);
+ char* buffer, size_t buflen);
static int add_track_to_playlist(struct playlist_info* playlist,
const char *filename, int position,
bool queue, int seek_pos);
@@ -457,7 +457,7 @@ static void update_playlist_filename(struct playlist_info* playlist,
* calculate track offsets within a playlist file
*/
static int add_indices_to_playlist(struct playlist_info* playlist,
- char* buffer, int buflen)
+ char* buffer, size_t buflen)
{
unsigned int nread;
unsigned int i = 0;
@@ -489,8 +489,7 @@ static int add_indices_to_playlist(struct playlist_info* playlist,
buflen = (audiobufend - audiobuf);
buffer = (char *)audiobuf;
#else
- buflen = (audiobufend - audiobuf - talk_get_bufsize());
- buffer = (char *)&audiobuf[talk_get_bufsize()];
+ buffer = (char *)audio_get_buffer(false, &buflen);
#endif
}
@@ -1853,7 +1852,7 @@ int playlist_resume(void)
{
struct playlist_info* playlist = &current_playlist;
char *buffer;
- int buflen;
+ size_t buflen;
int nread;
int total_read = 0;
int control_file_size = 0;
@@ -1866,8 +1865,7 @@ int playlist_resume(void)
buflen = (audiobufend - audiobuf);
buffer = (char *)audiobuf;
#else
- buflen = (audiobufend - audiobuf - talk_get_bufsize());
- buffer = (char *)&audiobuf[talk_get_bufsize()];
+ buffer = (char *)audio_get_buffer(false, &buflen);
#endif
empty_playlist(playlist, true);
diff --git a/apps/plugin.c b/apps/plugin.c
index 25f1865c9e..3a893fc537 100644
--- a/apps/plugin.c
+++ b/apps/plugin.c
@@ -674,12 +674,18 @@ void* plugin_get_buffer(int* buffer_size)
}
/* Returns a pointer to the mp3 buffer.
- Playback gets stopped, to avoid conflicts. */
+ Playback gets stopped, to avoid conflicts.
+ Talk buffer is stolen as well.
+ */
void* plugin_get_audio_buffer(int* buffer_size)
{
+#if CONFIG_CODEC == SWCODEC
+ return audio_get_buffer(true, (size_t *)buffer_size);
+#else
audio_stop();
talk_buffer_steal(); /* we use the mp3 buffer, need to tell */
*buffer_size = audiobufend - audiobuf;
+#endif
return audiobuf;
}
diff --git a/apps/recorder/icons.c b/apps/recorder/icons.c
index 46d628e780..ba22bb5a2c 100644
--- a/apps/recorder/icons.c
+++ b/apps/recorder/icons.c
@@ -30,12 +30,17 @@
const unsigned char bitmap_icons_5x8[][5] =
{
- [Icon_Lock_Main] ={0x78,0x7f,0x49,0x7f,0x78}, /* Lock Main */
- [Icon_Lock_Remote]={0x78,0x7f,0x49,0x7f,0x78}, /* Lock Remote */
- [Icon_Stereo]={0x7f, 0x1c, 0x00, 0x1c, 0x7f}, /* Stereo recording */
- [Icon_Mono]={0x00, 0x1c, 0x7f, 0x00, 0x00}, /* Mono recording */
+ [Icon_Lock_Main] =
+ {0x78, 0x7f, 0x49, 0x7f, 0x78}, /* Lock Main */
+ [Icon_Lock_Remote] =
+ {0x78, 0x7f, 0x49, 0x7f, 0x78}, /* Lock Remote */
+ [Icon_Stereo] =
+ {0x7f, 0x22, 0x1c, 0x22, 0x7f}, /* Stereo recording */
+ [Icon_Mono] =
+ {0x00, 0x1c, 0x22, 0x7f, 0x00}, /* Mono recording */
#if CONFIG_CODEC != SWCODEC
- [Icon_q]={0x1e, 0x21, 0x31, 0x21, 0x5e} /* Q icon */
+ [Icon_q] =
+ {0x1e, 0x21, 0x31, 0x21, 0x5e} /* Q icon */
#endif
};
@@ -81,45 +86,52 @@ const unsigned char bitmap_icons_7x8[][7] =
{0x7f,0x04,0x4e,0x5f,0x44,0x38,0x7f} /* Repeat-AB playmode */
};
-#if CONFIG_CODEC == SWCODEC
-const unsigned char bitmap_icons_18x8[][18] =
+#if CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING)
+const unsigned char bitmap_glyphs_4x8[][4] =
{
- {0x00, 0x00, 0x00, 0x00,0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x3e, 0x2a,
- 0x3a, 0x00, 0x0e, 0x08, 0x3e, 0x00}, /* mp3 64kbps */
- {0x00, 0x00, 0x00, 0x00,0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x0e, 0x0a,
- 0x3e, 0x00, 0x3e, 0x2a, 0x3a, 0x00}, /* mp3 96kbps */
- {0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x24, 0x3e, 0x20, 0x00, 0x3a, 0x2a,
- 0x2e, 0x00, 0x3e, 0x2a, 0x3e, 0x00}, /* mp3 128kbps */
- {0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x24, 0x3e, 0x20, 0x00, 0x3e, 0x2a,
- 0x3a, 0x00, 0x3e, 0x22, 0x3e, 0x00}, /* mp3 160kbps */
- {0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x24, 0x3e, 0x20, 0x00, 0x0e, 0x0a,
- 0x3e, 0x00, 0x3a, 0x2a, 0x2e, 0x00}, /* mp3 192kbps */
- {0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x3a, 0x2a, 0x2e, 0x00, 0x3a, 0x2a,
- 0x2e, 0x00, 0x0e, 0x08, 0x3e, 0x00}, /* mp3 224kbps */
- {0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x22, 0x2a, 0x36, 0x00, 0x3a, 0x2a,
- 0x2e, 0x00, 0x3e, 0x22, 0x3e, 0x00}, /* mp3 320kbps */
- {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,0x1e, 0x20, 0x18, 0x20, 0x1e,
- 0x00, 0x1e, 0x20, 0x18, 0x06, 0x00}, /* wv */
- {0x00, 0x00, 0x1e, 0x20, 0x18, 0x20, 0x1e, 0x00, 0x3c, 0x12, 0x12, 0x3c,
- 0x00, 0x1e, 0x20, 0x18, 0x06, 0x00} /* wav */
+ /* Keep digits together and first! */
+ [0] =
+ {0x00, 0x3e, 0x22, 0x3e}, /* 0 */
+ [1] =
+ {0x00, 0x24, 0x3e, 0x20}, /* 1 */
+ [2] =
+ {0x00, 0x3a, 0x2a, 0x2e}, /* 2 */
+ [3] =
+ {0x00, 0x22, 0x2a, 0x36}, /* 3 */
+ [4] =
+ {0x00, 0x0e, 0x08, 0x3e}, /* 4 */
+ [5] =
+ {0x00, 0x2e, 0x2a, 0x3a}, /* 5 */
+ [6] =
+ {0x00, 0x3e, 0x2a, 0x3a}, /* 6 */
+ [7] =
+ {0x00, 0x02, 0x02, 0x3e}, /* 7 */
+ [8] =
+ {0x00, 0x3e, 0x2a, 0x3e}, /* 8 */
+ [9] =
+ {0x00, 0x0e, 0x0a, 0x3e}, /* 9 */
+ [10 ... Glyph_4x8Last-1] =
+ {0x00, 0x00, 0x00, 0x00}, /* auto-blank */
+ [Glyph_4x8_k] =
+ {0x00, 0x3e, 0x10, 0x28}, /* k */
};
-const unsigned char bitmap_icons_12x8[][12] =
+const unsigned char bitmap_formats_18x8[Format_18x8Last][18]=
{
- {0x00, 0x00, 0x00, 0x00, 0x00, 0x3e, 0x2a, 0x3e, 0x00, 0x3e, 0x10, 0x28}, /* 8khz */
- {0x00, 0x24, 0x3e, 0x20, 0x00, 0x24, 0x3e, 0x20, 0x00, 0x3e, 0x10, 0x28}, /* 11khz */
- {0x00, 0x24, 0x3e, 0x20, 0x00, 0x3a, 0x2a, 0x2e, 0x00, 0x3e, 0x10, 0x28}, /* 12khz */
- {0x00, 0x24, 0x3e, 0x20, 0x00, 0x3e, 0x2a, 0x3a, 0x00, 0x3e, 0x10, 0x28}, /* 16khz */
- {0x00, 0x3a, 0x2a, 0x2e, 0x00, 0x3a, 0x2a, 0x2e, 0x00, 0x3e, 0x10, 0x28}, /* 22khz */
- {0x00, 0x3a, 0x2a, 0x2e, 0x00, 0x0e, 0x08, 0x3e, 0x00, 0x3e, 0x10, 0x28}, /* 24khz */
- {0x00, 0x22, 0x2a, 0x36, 0x00, 0x3a, 0x2a, 0x2e, 0x00, 0x3e, 0x10, 0x28}, /* 32khz */
- {0x00, 0x0e, 0x08, 0x3e, 0x00, 0x0e, 0x08, 0x3e, 0x00, 0x3e, 0x10, 0x28}, /* 44.1khz */
- {0x00, 0x0e, 0x08, 0x3e, 0x00, 0x3e, 0x2a, 0x3e, 0x00, 0x3e, 0x10, 0x28}, /* 48khz */
- {0x00, 0x3e, 0x2a, 0x3a, 0x00, 0x0e, 0x08, 0x3e, 0x00, 0x3e, 0x10, 0x28}, /* 64khz */
- {0x00, 0x3e, 0x2a, 0x3e, 0x00, 0x3e, 0x2a, 0x3e, 0x00, 0x3e, 0x10, 0x28}, /* 88.2khz */
- {0x00, 0x0e, 0x0a, 0x3e, 0x00, 0x3e, 0x2a, 0x3a, 0x00, 0x3e, 0x10, 0x28} /* 96khz */
+ [0 ... Format_18x8Last-1] = /* auto-blank */
+ {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}, /* ___ */
+ [Format_18x8_MPA_L3] =
+ {0x00, 0x3e, 0x04, 0x08, 0x04, 0x3e, 0x00, 0x00, 0x00,
+ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}, /* M__ */
+ [Format_18x8_WAVPACK] =
+ {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x1e, 0x20,
+ 0x18, 0x20, 0x1e, 0x00, 0x0e, 0x10, 0x20, 0x10, 0x0e}, /* _WV */
+ [Format_18x8_PCM_WAV] =
+ {0x00, 0x1e, 0x20, 0x18, 0x20, 0x1e, 0x00, 0x3c, 0x0a,
+ 0x0a, 0x0a, 0x3c, 0x00, 0x0e, 0x10, 0x20, 0x10, 0x0e}, /* WAV */
};
-#endif
+#endif /* CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING) */
/* Disk/MMC activity */
const unsigned char bitmap_icon_disk[12] =
diff --git a/apps/recorder/icons.h b/apps/recorder/icons.h
index 75401f6f0b..1e7b8dba1e 100644
--- a/apps/recorder/icons.h
+++ b/apps/recorder/icons.h
@@ -89,44 +89,40 @@ enum icons_7x8 {
Icon7x8Last
};
-#if CONFIG_CODEC == SWCODEC
-enum icons_12x8 {
- Icon_8000,
- Icon_11025,
- Icon_12000,
- Icon_16000,
- Icon_22050,
- Icon_24000,
- Icon_32000,
- Icon_44100,
- Icon_48000,
- Icon_64000,
- Icon_88200,
- Icon_96000,
- Icon12x8Last
+#if CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING)
+#define BM_GLYPH_WIDTH 4
+enum Glyphs_4x8 {
+ Glyph_4x8_0 = 0,
+ Glyph_4x8_1,
+ Glyph_4x8_2,
+ Glyph_4x8_3,
+ Glyph_4x8_4,
+ Glyph_4x8_5,
+ Glyph_4x8_6,
+ Glyph_4x8_7,
+ Glyph_4x8_8,
+ Glyph_4x8_9,
+ Glyph_4x8_k,
+ Glyph_4x8Last
};
-
-enum icons_18x8 {
- Icon_mp364,
- Icon_mp396,
- Icon_mp3128,
- Icon_mp3160,
- Icon_mp3192,
- Icon_mp3224,
- Icon_mp3320,
- Icon_wv,
- Icon_wav,
- Icon18x8Last
+extern const unsigned char bitmap_glyphs_4x8[Glyph_4x8Last][4];
+
+#define BM_MPA_L3_M_WIDTH 6
+#ifdef ID3_H
+/* This enum is redundant but sort of in keeping with the style */
+enum rec_format_18x8 {
+ Format_18x8_MPA_L3 = REC_FORMAT_MPA_L3,
+ Format_18x8_WAVPACK = REC_FORMAT_WAVPACK,
+ Format_18x8_PCM_WAV = REC_FORMAT_PCM_WAV,
+ Format_18x8Last = REC_NUM_FORMATS
};
-#endif
+extern const unsigned char bitmap_formats_18x8[Format_18x8Last][18];
+#endif /* ID3_H */
+#endif /* CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING) */
extern const unsigned char bitmap_icons_5x8[Icon5x8Last][5];
extern const unsigned char bitmap_icons_6x8[Icon6x8Last][6];
extern const unsigned char bitmap_icons_7x8[Icon7x8Last][7];
-#if CONFIG_CODEC == SWCODEC
-extern const unsigned char bitmap_icons_12x8[Icon12x8Last][12];
-extern const unsigned char bitmap_icons_18x8[Icon18x8Last][18];
-#endif
extern const unsigned char bitmap_icon_disk[];
#define STATUSBAR_X_POS 0
diff --git a/apps/recorder/peakmeter.c b/apps/recorder/peakmeter.c
index 0370f4deea..44be43124a 100644
--- a/apps/recorder/peakmeter.c
+++ b/apps/recorder/peakmeter.c
@@ -540,10 +540,8 @@ void peak_meter_peek(void)
if (pm_playback)
pcm_calculate_peaks(&pm_cur_left, &pm_cur_right);
#ifdef HAVE_RECORDING
- if (!pm_playback)
- {
- pcm_rec_get_peaks(&pm_cur_left, &pm_cur_right);
- }
+ else
+ pcm_calculate_rec_peaks(&pm_cur_left, &pm_cur_right);
#endif
left = pm_cur_left;
right = pm_cur_right;
diff --git a/apps/recorder/radio.c b/apps/recorder/radio.c
index d74437a8c9..7a0cc6543e 100644
--- a/apps/recorder/radio.c
+++ b/apps/recorder/radio.c
@@ -386,6 +386,7 @@ bool radio_screen(void)
unsigned int last_seconds = 0;
#if CONFIG_CODEC != SWCODEC
int hours, minutes;
+ struct audio_recording_options rec_options;
#endif
bool keep_playing = false;
bool statusbar = global_settings.statusbar;
@@ -436,12 +437,9 @@ bool radio_screen(void)
peak_meter_enabled = true;
- rec_set_recording_options(global_settings.rec_frequency,
- global_settings.rec_quality,
- AUDIO_SRC_LINEIN, 0,
- global_settings.rec_channels,
- global_settings.rec_editable,
- global_settings.rec_prerecord_time);
+ rec_init_recording_options(&rec_options);
+ rec_options.rec_source = AUDIO_SRC_LINEIN;
+ rec_set_recording_options(&rec_options);
audio_set_recording_gain(sound_default(SOUND_LEFT_GAIN),
sound_default(SOUND_RIGHT_GAIN), AUDIO_GAIN_LINEIN);
@@ -881,7 +879,7 @@ bool radio_screen(void)
}
else
{
- if(global_settings.rec_prerecord_time)
+ if(rec_options.rec_prerecord_time)
{
snprintf(buf, 32, "%s %02d",
str(LANG_RECORD_PRERECORD), seconds%60);
@@ -1173,7 +1171,8 @@ bool save_preset_list(void)
if(!opendir(FMPRESET_PATH)) /* Check if there is preset folder */
mkdir(FMPRESET_PATH, 0);
- create_numbered_filename(filepreset,FMPRESET_PATH,"preset",".fmr",2);
+ create_numbered_filename(filepreset, FMPRESET_PATH, "preset",
+ ".fmr", 2 IF_CNFN_NUM_(, NULL));
while(bad_file_name)
{
@@ -1534,12 +1533,10 @@ static bool fm_recording_settings(void)
#if CONFIG_CODEC != SWCODEC
if (!ret)
{
- rec_set_recording_options(global_settings.rec_frequency,
- global_settings.rec_quality,
- AUDIO_SRC_LINEIN, 0,
- global_settings.rec_channels,
- global_settings.rec_editable,
- global_settings.rec_prerecord_time);
+ struct audio_recording_options rec_options;
+ rec_init_recording_options(&rec_options);
+ rec_options.rec_source = AUDIO_SRC_LINEIN;
+ rec_set_recording_options(&rec_options);
}
#endif
diff --git a/apps/recorder/recording.c b/apps/recorder/recording.c
index 0a21d96566..6a053cd12e 100644
--- a/apps/recorder/recording.c
+++ b/apps/recorder/recording.c
@@ -30,8 +30,10 @@
#include "mpeg.h"
#include "audio.h"
#if CONFIG_CODEC == SWCODEC
-#include "pcm_record.h"
+#include "thread.h"
+#include "pcm_playback.h"
#include "playback.h"
+#include "enc_config.h"
#endif
#ifdef HAVE_UDA1380
#include "uda1380.h"
@@ -73,36 +75,40 @@
#define PM_HEIGHT ((LCD_HEIGHT >= 72) ? 2 : 1)
+#if CONFIG_KEYPAD == RECORDER_PAD
bool f2_rec_screen(void);
bool f3_rec_screen(void);
+#endif
#define MAX_FILE_SIZE 0x7F800000 /* 2 GB - 4 MB */
int screen_update = NB_SCREENS;
bool remote_display_on = true;
-const char* const freq_str[6] =
-{
- "44.1kHz",
- "48kHz",
- "32kHz",
- "22.05kHz",
- "24kHz",
- "16kHz"
-};
+/** File name creation **/
#if CONFIG_CODEC == SWCODEC
-#define REC_ENCODER_ID(q) \
- rec_quality_info_afmt[q]
-#define REC_QUALITY_LABEL(q) \
- (audio_formats[REC_ENCODER_ID(q)].label)
-#define REC_FILE_ENDING(q) \
- (audio_formats[REC_ENCODER_ID(q)].ext)
-#else
+
+#ifdef IF_CNFN_NUM
+/* current file number to assist in creating unique numbered filenames
+ without actually having to create the file on disk */
+static int file_number = -1;
+#endif /* IF_CNFN_NUM */
+
+#define REC_FILE_ENDING(rec_format) \
+ (audio_formats[rec_format_afmt[rec_format]].ext_list)
+
+#else /* CONFIG_CODEC != SWCODEC */
+
/* default record file extension for HWCODEC */
-#define REC_QUALITY_LABEL(q) "MP3"
-#define REC_FILE_ENDING(q) ".mp3"
-#endif
+#define REC_FILE_ENDING(rec_format) \
+ (audio_formats[AFMT_MPA_L3].ext_list)
+#endif /* CONFIG_CODEC == SWCODEC */
+
+/* path for current file */
+static char path_buffer[MAX_PATH];
+
+/** Automatic Gain Control (AGC) **/
#ifdef HAVE_AGC
/* Timing counters:
* peak_time is incremented every 0.2s, every 2nd run of record screen loop.
@@ -496,20 +502,24 @@ void adjust_cursor(void)
char *rec_create_filename(char *buffer)
{
+ char ext[16];
+
if(global_settings.rec_directory)
getcwd(buffer, MAX_PATH);
else
strncpy(buffer, rec_base_directory, MAX_PATH);
+ snprintf(ext, sizeof(ext), ".%s",
+ REC_FILE_ENDING(global_settings.rec_format));
#ifdef CONFIG_RTC
- create_datetime_filename(buffer, buffer, "R",
- REC_FILE_ENDING(global_settings.rec_quality));
+ /* We'll wait at least up to the start of the next second so no duplicate
+ names are created */
+ return create_datetime_filename(buffer, buffer, "R", ext, true);
#else
- create_numbered_filename(buffer, buffer, "rec_",
- REC_FILE_ENDING(global_settings.rec_quality), 4);
+ return create_numbered_filename(buffer, buffer, "rec_", ext, 4
+ IF_CNFN_NUM_(, &file_number));
#endif
- return buffer;
}
int rec_create_directory(void)
@@ -557,9 +567,15 @@ static void rec_boost(bool state)
/**
* Selects an audio source for recording or playback
- * powers/unpowers related devices.
+ * powers/unpowers related devices and sets up monitoring.
* Here because it calls app code and used only for HAVE_RECORDING atm.
* Would like it in pcm_record.c.
+ *
+ * Behaves like a firmware function in that it does not use global settings
+ * to determine the state.
+ *
+ * The order of setting monitoring may need tweaking dependent upon the
+ * selected source to get the smoothest transition.
*/
#if defined(HAVE_UDA1380)
#define ac_disable_recording uda1380_disable_recording
@@ -571,7 +587,13 @@ static void rec_boost(bool state)
#define ac_set_monitor tlv320_set_monitor
#endif
-void rec_set_source(int source, int flags)
+#ifdef HAVE_SPDIF_IN
+#define rec_spdif_set_monitor(m) audio_spdif_set_monitor(m)
+#else
+#define rec_spdif_set_monitor(m)
+#endif
+
+void rec_set_source(int source, unsigned flags)
{
/* Prevent pops from unneeded switching */
static int last_source = AUDIO_SRC_PLAYBACK;
@@ -586,7 +608,9 @@ void rec_set_source(int source, int flags)
/** Do power up/down of associated device(s) **/
+ /** SPDIF **/
#ifdef HAVE_SPDIF_IN
+ /* Always boost for SPDIF */
if ((source == AUDIO_SRC_SPDIF) != (source == last_source))
rec_boost(source == AUDIO_SRC_SPDIF);
@@ -595,10 +619,11 @@ void rec_set_source(int source, int flags)
both optical in and out is controlled by the same power source, which is
the case on H1x0. */
spdif_power_enable((source == AUDIO_SRC_SPDIF) ||
- global_settings.spdif_enable);
+ audio_get_spdif_power_setting());
#endif
#endif
+ /** Tuner **/
#ifdef CONFIG_TUNER
/* Switch radio off or on per source and flags. */
if (source != AUDIO_SRC_FMRADIO)
@@ -612,12 +637,15 @@ void rec_set_source(int source, int flags)
switch (source)
{
default: /* playback - no recording */
+ source = AUDIO_SRC_PLAYBACK;
+ case AUDIO_SRC_PLAYBACK:
pm_playback = true;
if (source == last_source)
break;
ac_disable_recording();
ac_set_monitor(false);
pcm_rec_mux(0); /* line in */
+ rec_spdif_set_monitor(-1); /* silence it */
break;
case AUDIO_SRC_MIC: /* recording only */
@@ -625,6 +653,7 @@ void rec_set_source(int source, int flags)
break;
ac_enable_recording(true); /* source mic */
pcm_rec_mux(0); /* line in */
+ rec_spdif_set_monitor(0);
break;
case AUDIO_SRC_LINEIN: /* recording only */
@@ -632,29 +661,20 @@ void rec_set_source(int source, int flags)
break;
pcm_rec_mux(0); /* line in */
ac_enable_recording(false); /* source line */
+ rec_spdif_set_monitor(0);
break;
#ifdef HAVE_SPDIF_IN
case AUDIO_SRC_SPDIF: /* recording only */
- if (recording)
- {
- /* This was originally done in audio_set_recording_options only */
-#ifdef HAVE_SPDIF_POWER
- EBU1CONFIG = global_settings.spdif_enable ? (1 << 2) : 0;
- /* Input source is EBUin1, Feed-through monitoring if desired */
-#else
- EBU1CONFIG = (1 << 2);
- /* Input source is EBUin1, Feed-through monitoring */
-#endif
- }
-
- if (source != last_source)
- uda1380_disable_recording();
+ if (source == last_source)
+ break;
+ ac_disable_recording();
+ audio_spdif_set_monitor(1);
break;
#endif /* HAVE_SPDIF_IN */
#ifdef HAVE_FMRADIO_IN
- case AUDIO_SRC_FMRADIO:
+ case AUDIO_SRC_FMRADIO: /* recording and playback */
if (!recording)
{
audio_set_recording_gain(sound_default(SOUND_LEFT_GAIN),
@@ -687,6 +707,8 @@ void rec_set_source(int source, int flags)
tlv320_set_monitor(true); /* analog bypass */
}
#endif
+
+ rec_spdif_set_monitor(0);
break;
/* #elif defined(CONFIG_TUNER) */
/* Have radio but cannot record it */
@@ -702,33 +724,50 @@ void rec_set_source(int source, int flags)
} /* rec_set_source */
#endif /* CONFIG_CODEC == SWCODEC && !defined(SIMULATOR) */
-/* steal the mp3 buffer then actually set options */
-void rec_set_recording_options(int frequency, int quality,
- int source, int source_flags,
- int channel_mode, bool editable,
- int prerecord_time)
+void rec_init_recording_options(struct audio_recording_options *options)
+{
+ options->rec_source = global_settings.rec_source;
+ options->rec_frequency = global_settings.rec_frequency;
+ options->rec_channels = global_settings.rec_channels;
+ options->rec_prerecord_time = global_settings.rec_prerecord_time;
+#if CONFIG_CODEC == SWCODEC
+ options->rec_source_flags = 0;
+ options->enc_config.rec_format = global_settings.rec_format;
+ global_to_encoder_config(&options->enc_config);
+#else
+ options->rec_quality = global_settings.rec_quality;
+ options->rec_editable = global_settings.rec_editable;
+#endif
+}
+
+void rec_set_recording_options(struct audio_recording_options *options)
{
#if CONFIG_CODEC != SWCODEC
if (global_settings.rec_prerecord_time)
-#endif
talk_buffer_steal(); /* will use the mp3 buffer */
+#endif
+
+#ifdef HAVE_SPDIF_IN
+#ifdef HAVE_SPDIF_POWER
+ audio_set_spdif_power_setting(global_settings.spdif_enable);
+#endif
+#endif
#if CONFIG_CODEC == SWCODEC
- rec_set_source(source, source_flags | SRCF_RECORDING);
-#else
- (void)source_flags;
+ rec_set_source(options->rec_source,
+ options->rec_source_flags | SRCF_RECORDING);
#endif
- audio_set_recording_options(frequency, quality, source,
- channel_mode, editable, prerecord_time);
+ audio_set_recording_options(options);
}
-static char path_buffer[MAX_PATH];
-
/* steals mp3 buffer, creates unique filename and starts recording */
void rec_record(void)
{
+#if CONFIG_CODEC != SWCODEC
talk_buffer_steal(); /* we use the mp3 buffer */
+#endif
+ IF_CNFN_NUM_(file_number = -1;) /* Hit disk for number */
audio_record(rec_create_filename(path_buffer));
}
@@ -753,7 +792,6 @@ static void trigger_listener(int trigger_status)
case TRIG_GO:
if((audio_status() & AUDIO_STATUS_RECORD) != AUDIO_STATUS_RECORD)
{
- talk_buffer_steal(); /* we use the mp3 buffer */
rec_record();
/* give control to mpeg thread so that it can start
recording */
@@ -831,6 +869,8 @@ bool recording_screen(bool no_source)
ID2P(LANG_GIGABYTE)
};
+ struct audio_recording_options rec_options;
+
global_settings.recscreen_on = true;
cursor = 0;
#if (CONFIG_LED == LED_REAL) && !defined(SIMULATOR)
@@ -838,35 +878,26 @@ bool recording_screen(bool no_source)
#endif
#if CONFIG_CODEC == SWCODEC
- audio_stop();
- voice_stop();
/* recording_menu gets messed up: so reset talk_menu */
talk_menu = global_settings.talk_menu;
global_settings.talk_menu = 0;
+ /* audio_init_recording stops anything playing when it takes the audio
+ buffer */
#else
/* Yes, we use the D/A for monitoring */
peak_meter_enabled = true;
peak_meter_playback(true);
#endif
-#if CONFIG_CODEC == SWCODEC
- audio_init_recording(talk_get_bufsize());
-#else
audio_init_recording(0);
-#endif
sound_set_volume(global_settings.volume);
#ifdef HAVE_AGC
peak_meter_get_peakhold(&peak_l, &peak_r);
#endif
- rec_set_recording_options(global_settings.rec_frequency,
- global_settings.rec_quality,
- global_settings.rec_source,
- 0,
- global_settings.rec_channels,
- global_settings.rec_editable,
- global_settings.rec_prerecord_time);
+ rec_init_recording_options(&rec_options);
+ rec_set_recording_options(&rec_options);
set_gain();
settings_apply_trigger();
@@ -1025,7 +1056,6 @@ bool recording_screen(bool no_source)
{
/* manual recording */
have_recorded = true;
- talk_buffer_steal(); /* we use the mp3 buffer */
rec_record();
last_seconds = 0;
if (talk_menu)
@@ -1253,16 +1283,10 @@ bool recording_screen(bool no_source)
#if CONFIG_CODEC == SWCODEC
/* reinit after submenu exit */
audio_close_recording();
- audio_init_recording(talk_get_bufsize());
+ audio_init_recording(0);
#endif
- rec_set_recording_options(
- global_settings.rec_frequency,
- global_settings.rec_quality,
- global_settings.rec_source,
- 0,
- global_settings.rec_channels,
- global_settings.rec_editable,
- global_settings.rec_prerecord_time);
+ rec_init_recording_options(&rec_options);
+ rec_set_recording_options(&rec_options);
if(rec_create_directory() > 0)
have_recorded = true;
@@ -1739,11 +1763,7 @@ bool recording_screen(bool no_source)
}
} /* end while(!done) */
-#if CONFIG_CODEC == SWCODEC
- audio_stat = pcm_rec_status();
-#else
audio_stat = audio_status();
-#endif
if (audio_stat & AUDIO_STATUS_ERROR)
{
gui_syncsplash(0, true, str(LANG_SYSFONT_DISK_FULL));
@@ -1804,11 +1824,22 @@ bool recording_screen(bool no_source)
#if CONFIG_KEYPAD == RECORDER_PAD
bool f2_rec_screen(void)
{
+ static const char* const freq_str[6] =
+ {
+ "44.1kHz",
+ "48kHz",
+ "32kHz",
+ "22.05kHz",
+ "24kHz",
+ "16kHz"
+ };
+
bool exit = false;
bool used = false;
int w, h, i;
char buf[32];
int button;
+ struct audio_recording_options rec_options;
FOR_NB_SCREENS(i)
{
@@ -1919,13 +1950,8 @@ bool f2_rec_screen(void)
}
}
- rec_set_recording_options(global_settings.rec_frequency,
- global_settings.rec_quality,
- global_settings.rec_source,
- 0,
- global_settings.rec_channels,
- global_settings.rec_editable,
- global_settings.rec_prerecord_time);
+ rec_init_recording_options(&rec_options);
+ rec_set_recording_options(&rec_options);
set_gain();
@@ -1948,6 +1974,8 @@ bool f3_rec_screen(void)
str(LANG_SYSFONT_RECORDING_SRC_LINE),
str(LANG_SYSFONT_RECORDING_SRC_DIGITAL)
};
+ struct audio_recording_options rec_options;
+
FOR_NB_SCREENS(i)
{
screens[i].setfont(FONT_SYSFIXED);
@@ -2019,13 +2047,8 @@ bool f3_rec_screen(void)
}
}
- rec_set_recording_options(global_settings.rec_frequency,
- global_settings.rec_quality,
- global_settings.rec_source,
- 0,
- global_settings.rec_channels,
- global_settings.rec_editable,
- global_settings.rec_prerecord_time);
+ rec_init_recording_options(&rec_options);
+ rec_set_recording_options(&rec_options);
set_gain();
@@ -2066,23 +2089,30 @@ unsigned long audio_num_recorded_bytes(void)
}
#if CONFIG_CODEC == SWCODEC
-void rec_set_source(int source, int flags)
+void rec_set_source(int source, unsigned flags)
{
source = source;
flags = flags;
}
-#endif
-void audio_set_recording_options(int frequency, int quality,
- int source, int channel_mode,
- bool editable, int prerecord_time)
+#ifdef HAVE_SPDIF_IN
+#ifdef HAVE_SPDIF_POWER
+void audio_set_spdif_power_setting(bool on)
{
- frequency = frequency;
- quality = quality;
- source = source;
- channel_mode = channel_mode;
- editable = editable;
- prerecord_time = prerecord_time;
+ on = on;
+}
+
+bool audio_get_spdif_power_setting(void)
+{
+ return true;
+}
+#endif /* HAVE_SPDIF_POWER */
+#endif /* HAVE_SPDIF_IN */
+#endif /* CONFIG_CODEC == SWCODEC */
+
+void audio_set_recording_options(struct audio_recording_options *options)
+{
+ options = options;
}
void audio_set_recording_gain(int left, int right, int type)
@@ -2104,7 +2134,7 @@ void audio_resume_recording(void)
{
}
-void pcm_rec_get_peaks(int *left, int *right)
+void pcm_calculate_rec_peaks(int *left, int *right)
{
if (left)
*left = 0;
diff --git a/apps/recorder/recording.h b/apps/recorder/recording.h
index aa216e757f..a977efa749 100644
--- a/apps/recorder/recording.h
+++ b/apps/recorder/recording.h
@@ -32,15 +32,16 @@ int rec_create_directory(void);
#define SRCF_FMRADIO_PLAYING 0x0000 /* default */
#define SRCF_FMRADIO_PAUSED 0x2000
#endif
-void rec_set_source(int source, int flags);
+void rec_set_source(int source, unsigned flags);
#endif /* CONFIG_CODEC == SW_CODEC */
+/* Initializes a recording_options structure with global settings.
+ pass returned data to audio_set_recording_options or
+ rec_set_recording_options */
+void rec_init_recording_options(struct audio_recording_options *options);
/* steals mp3 buffer, sets source and then options */
-/* SRCF_RECORDING is implied */
-void rec_set_recording_options(int frequency, int quality,
- int source, int source_flags,
- int channel_mode, bool editable,
- int prerecord_time);
+/* SRCF_RECORDING is implied for SWCODEC */
+void rec_set_recording_options(struct audio_recording_options *options);
/* steals mp3 buffer, creates unique filename and starts recording */
void rec_record(void);
diff --git a/apps/settings.c b/apps/settings.c
index a4320eda7b..ec96cc760b 100644
--- a/apps/settings.c
+++ b/apps/settings.c
@@ -90,13 +90,16 @@ const char rec_base_directory[] = REC_BASE_DIR;
#include "pcmbuf.h"
#include "pcm_playback.h"
#include "dsp.h"
+#ifdef HAVE_RECORDING
+#include "enc_config.h"
#endif
+#endif /* CONFIG_CODEC == SWCODEC */
#ifdef HAVE_WM8758
#include "eq_menu.h"
#endif
-#define CONFIG_BLOCK_VERSION 55
+#define CONFIG_BLOCK_VERSION 56
#define CONFIG_BLOCK_SIZE 512
#define RTC_BLOCK_SIZE 44
@@ -514,7 +517,7 @@ static const struct bit_entry hd_bits[] =
{1, S_O(rec_editable), false, "editable recordings", off_on },
#endif /* CONFIG_CODEC == MAS3587F */
-#if CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING)
+#if CONFIG_CODEC == SWCODEC
#ifdef HAVE_UDA1380
{8|SIGNED, S_O(rec_mic_gain), 16 /* 8 dB */, "rec mic gain", NULL }, /* -128...+108 */
#endif
@@ -524,16 +527,20 @@ static const struct bit_entry hd_bits[] =
#endif
{8|SIGNED, S_O(rec_left_gain), 0, "rec left gain", NULL }, /* -128...+96 */
{8|SIGNED, S_O(rec_right_gain), 0, "rec right gain", NULL }, /* -128...+96 */
-#if 0 /* Till samplerates are added for SWCODEC */
- {3, S_O(rec_frequency), 0, /* 0=44.1kHz */
- "rec frequency", "44,48,32,22,24,16" },
-#else
- {3, S_O(rec_frequency), 0, /* 0=44.1kHz */
- "rec frequency", "44" },
-#endif
-
- {4, S_O(rec_quality), 4 /* MP3 L3 192 kBit/s */, "rec quality", NULL },
-#endif /* CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING) */
+ {REC_FREQ_CFG_NUM_BITS, S_O(rec_frequency), REC_FREQ_DEFAULT,
+ "rec frequency", REC_FREQ_CFG_VAL_LIST },
+ {REC_FORMAT_CFG_NUM_BITS ,S_O(rec_format), REC_FORMAT_DEFAULT,
+ "rec format", REC_FORMAT_CFG_VAL_LIST },
+ /** Encoder settings start - keep these together **/
+ /* mp3_enc */
+ {5,S_O(mp3_enc_config.bitrate), MP3_ENC_BITRATE_CFG_DEFAULT,
+ "mp3_enc bitrate", MP3_ENC_BITRATE_CFG_VALUE_LIST },
+ /* wav_enc */
+ /* (no settings yet) */
+ /* wavpack_enc */
+ /* (no settings yet) */
+ /** Encoder settings end **/
+#endif /* CONFIG_CODEC == SWCODEC */
/* values for the trigger */
{8 | SIGNED, S_O(rec_start_thres), -35, "trigger start threshold", NULL},
@@ -1301,6 +1308,11 @@ void settings_apply(void)
lcd_set_sleep_after_backlight_off(global_settings.lcd_sleep_after_backlight_off);
#endif
#endif /* CONFIG_BACKLIGHT */
+
+ /* This should stay last */
+#if defined(HAVE_RECORDING) && CONFIG_CODEC == SWCODEC
+ enc_global_settings_apply();
+#endif
}
@@ -1727,13 +1739,13 @@ static void save_cfg_table(const struct bit_entry* p_table, int count, int fd)
}
}
-
bool settings_save_config(void)
{
int fd;
char filename[MAX_PATH];
- create_numbered_filename(filename, ROCKBOX_DIR, "config", ".cfg", 2);
+ create_numbered_filename(filename, ROCKBOX_DIR, "config", ".cfg", 2
+ IF_CNFN_NUM_(, NULL));
/* allow user to modify filename */
while (true) {
@@ -1887,6 +1899,10 @@ void settings_reset(void) {
global_settings.kbd_file[0] = '\0';
#endif
global_settings.hold_lr_for_scroll_in_list = true;
+
+#if defined (HAVE_RECORDING) && CONFIG_CODEC == SWCODEC
+ enc_global_settings_reset();
+#endif
}
bool set_bool(const char* string, bool* variable )
diff --git a/apps/settings.h b/apps/settings.h
index 435d8e9ce2..09d4974eee 100644
--- a/apps/settings.h
+++ b/apps/settings.h
@@ -29,6 +29,10 @@
#include "tagcache.h"
#include "button.h"
+#if CONFIG_CODEC == SWCODEC
+#include "audio.h"
+#endif
+
#ifdef HAVE_BACKLIGHT_BRIGHTNESS
#include "backlight.h" /* for [MIN|MAX]_BRIGHTNESS_SETTING */
#endif
@@ -142,18 +146,22 @@ struct user_settings
int crossfade_fade_out_mixmode; /* Fade out mode (0=crossfade,1=mix) */
#endif
+#if CONFIG_CODEC == SWCODEC
+ int rec_format; /* record format index */
+#else
int rec_quality; /* 0-7 */
- int rec_source; /* 0=mic, 1=line, 2=S/PDIF */
- int rec_frequency; /* 0 = 44.1kHz
+#endif /* CONFIG_CODEC == SWCODEC */
+ int rec_source; /* 0=mic, 1=line, 2=S/PDIF, 2 or 3=FM Radio */
+ int rec_frequency; /* 0 = 44.1kHz (depends on target)
1 = 48kHz
2 = 32kHz
3 = 22.05kHz
4 = 24kHz
5 = 16kHz */
int rec_channels; /* 0=Stereo, 1=Mono */
- int rec_mic_gain; /* 0-15 */
- int rec_left_gain; /* 0-15 */
- int rec_right_gain; /* 0-15 */
+ int rec_mic_gain; /* depends on target */
+ int rec_left_gain; /* depends on target */
+ int rec_right_gain; /* depands on target */
bool rec_editable; /* true means that the bit reservoir is off */
bool recscreen_on; /* true if using the recording screen */
@@ -504,6 +512,20 @@ struct user_settings
#endif
bool audioscrobbler; /* Audioscrobbler logging */
+ /* If values are just added to the end, no need to bump plugin API
+ version. */
+ /* new stuff to be added at the end */
+
+#if defined(HAVE_RECORDING) && CONFIG_CODEC == SWCODEC
+ /* Encoder Settings Start - keep these together */
+ struct mp3_enc_config mp3_enc_config;
+#if 0 /* These currently contain no members but their places in line
+ should be held */
+ struct wav_enc_config wav_enc_config;
+ struct wavpack_enc_config wavpack_enc_config;
+#endif
+ /* Encoder Settings End */
+#endif /* CONFIG_CODEC == SWCODEC */
};
enum optiontype { INT, BOOL };
@@ -584,7 +606,7 @@ extern const char rec_base_directory[];
/* argument bits for settings_load() */
#define SETTINGS_RTC 1 /* only the settings from the RTC nonvolatile RAM */
-#define SETTINGS_HD 2 /* only the settings fron the disk sector */
+#define SETTINGS_HD 2 /* only the settings from the disk sector */
#define SETTINGS_ALL 3 /* both */
/* repeat mode options */
diff --git a/apps/sound_menu.c b/apps/sound_menu.c
index c10ba9417e..fb766d604c 100644
--- a/apps/sound_menu.c
+++ b/apps/sound_menu.c
@@ -54,6 +54,10 @@
#include "dsp.h"
#include "eq_menu.h"
#include "pcmbuf.h"
+#ifdef HAVE_RECORDING
+#include "enc_config.h"
+#endif
+#include "general.h"
#endif
#include "action.h"
@@ -308,22 +312,20 @@ static bool recsource(void)
{
int n_opts = AUDIO_NUM_SOURCES;
- struct opt_items names[AUDIO_NUM_SOURCES] = {
- { STR(LANG_RECORDING_SRC_MIC) },
- { STR(LANG_RECORDING_SRC_LINE) },
+ static const struct opt_items names[AUDIO_NUM_SOURCES] = {
+ [AUDIO_SRC_MIC] = { STR(LANG_RECORDING_SRC_MIC) },
+ [AUDIO_SRC_LINEIN] = { STR(LANG_RECORDING_SRC_LINE) },
#ifdef HAVE_SPDIF_IN
- { STR(LANG_RECORDING_SRC_DIGITAL) },
+ [AUDIO_SRC_SPDIF] = { STR(LANG_RECORDING_SRC_DIGITAL) },
+#endif
+#ifdef HAVE_FMRADIO_IN
+ [AUDIO_SRC_FMRADIO] = { STR(LANG_FM_RADIO) }
#endif
};
/* caveat: assumes it's the last item! */
#ifdef HAVE_FMRADIO_IN
- if (radio_hardware_present())
- {
- names[AUDIO_SRC_FMRADIO].string = ID2P(LANG_FM_RADIO);
- names[AUDIO_SRC_FMRADIO].voice_id = LANG_FM_RADIO;
- }
- else
+ if (!radio_hardware_present())
n_opts--;
#endif
@@ -332,28 +334,7 @@ static bool recsource(void)
n_opts, NULL );
}
-/* To be removed when we add support for sample rates and channel settings */
-#if CONFIG_CODEC == SWCODEC
-static bool recquality(void)
-{
- static const struct opt_items names[] = {
- { "MP3 64 kBit/s", TALK_ID( 64, UNIT_KBIT) },
- { "MP3 96 kBit/s", TALK_ID( 96, UNIT_KBIT) },
- { "MP3 128 kBit/s", TALK_ID( 128, UNIT_KBIT) },
- { "MP3 160 kBit/s", TALK_ID( 160, UNIT_KBIT) },
- { "MP3 192 kBit/s", TALK_ID( 192, UNIT_KBIT) },
- { "MP3 224 kBit/s", TALK_ID( 224, UNIT_KBIT) },
- { "MP3 320 kBit/s", TALK_ID( 320, UNIT_KBIT) },
- { "WV 900 kBit/s", TALK_ID( 900, UNIT_KBIT) },
- { "WAV 1411 kBit/s", TALK_ID(1411, UNIT_KBIT) }
- };
-
- return set_option(str(LANG_RECORDING_QUALITY),
- &global_settings.rec_quality, INT,
- names, sizeof (names)/sizeof(struct opt_items),
- NULL );
-}
-#elif CONFIG_CODEC == MAS3587F
+#if CONFIG_CODEC == MAS3587F
static bool recquality(void)
{
return set_int(str(LANG_RECORDING_QUALITY), "", UNIT_INT,
@@ -368,32 +349,182 @@ static bool receditable(void)
}
#endif /* CONFIG_CODEC == MAS3587F */
+#if CONFIG_CODEC == SWCODEC
+/* Makes an options list from a source list of options and indexes */
+void make_options_from_indexes(const struct opt_items *src_names,
+ const long *src_indexes,
+ int n_indexes,
+ struct opt_items *dst_names)
+{
+ while (--n_indexes >= 0)
+ dst_names[n_indexes] = src_names[src_indexes[n_indexes]];
+} /* make_options_from_indexes */
+
+static bool recformat(void)
+{
+ static const struct opt_items names[REC_NUM_FORMATS] = {
+ [REC_FORMAT_MPA_L3] = { STR(LANG_AFMT_MPA_L3) },
+ [REC_FORMAT_WAVPACK] = { STR(LANG_AFMT_WAVPACK) },
+ [REC_FORMAT_PCM_WAV] = { STR(LANG_AFMT_PCM_WAV) },
+ };
+
+ int rec_format = global_settings.rec_format;
+ bool res = set_option(str(LANG_RECORDING_FORMAT), &rec_format, INT,
+ names, REC_NUM_FORMATS, NULL );
+
+ if (rec_format != global_settings.rec_format)
+ {
+ global_settings.rec_format = rec_format;
+ enc_global_settings_apply();
+ }
+
+ return res;
+} /* recformat */
+
+#endif /* CONFIG_CODEC == SWCODEC */
+
static bool recfrequency(void)
{
- static const struct opt_items names[] = {
+#if CONFIG_CODEC == MAS3587F
+ static const struct opt_items names[6] = {
{ "44.1kHz", TALK_ID(44, UNIT_KHZ) },
-#if CONFIG_CODEC != SWCODEC /* This is temporary */
{ "48kHz", TALK_ID(48, UNIT_KHZ) },
{ "32kHz", TALK_ID(32, UNIT_KHZ) },
{ "22.05kHz", TALK_ID(22, UNIT_KHZ) },
{ "24kHz", TALK_ID(24, UNIT_KHZ) },
{ "16kHz", TALK_ID(16, UNIT_KHZ) }
-#endif
};
return set_option(str(LANG_RECORDING_FREQUENCY),
&global_settings.rec_frequency, INT,
- names, sizeof(names)/sizeof(*names), NULL );
+ names, 6, NULL );
+#endif /* CONFIG_CODEC == MAS3587F */
+
+#if CONFIG_CODEC == SWCODEC
+ static const struct opt_items names[REC_NUM_FREQ] = {
+ REC_HAVE_96_([REC_FREQ_96] = { "96kHz", TALK_ID(96, UNIT_KHZ) },)
+ REC_HAVE_88_([REC_FREQ_88] = { "88.2kHz", TALK_ID(88, UNIT_KHZ) },)
+ REC_HAVE_64_([REC_FREQ_64] = { "64kHz", TALK_ID(64, UNIT_KHZ) },)
+ REC_HAVE_48_([REC_FREQ_48] = { "48kHz", TALK_ID(48, UNIT_KHZ) },)
+ REC_HAVE_44_([REC_FREQ_44] = { "44.1kHz", TALK_ID(44, UNIT_KHZ) },)
+ REC_HAVE_32_([REC_FREQ_32] = { "32kHz", TALK_ID(32, UNIT_KHZ) },)
+ REC_HAVE_24_([REC_FREQ_24] = { "24kHz", TALK_ID(24, UNIT_KHZ) },)
+ REC_HAVE_22_([REC_FREQ_22] = { "22.05kHz", TALK_ID(22, UNIT_KHZ) },)
+ REC_HAVE_16_([REC_FREQ_16] = { "16kHz", TALK_ID(16, UNIT_KHZ) },)
+ REC_HAVE_12_([REC_FREQ_12] = { "12kHz", TALK_ID(12, UNIT_KHZ) },)
+ REC_HAVE_11_([REC_FREQ_11] = { "11.025kHz", TALK_ID(11, UNIT_KHZ) },)
+ REC_HAVE_8_( [REC_FREQ_8 ] = { "8kHz", TALK_ID( 8, UNIT_KHZ) },)
+ };
+
+ struct opt_items opts[REC_NUM_FREQ];
+ unsigned long table[REC_NUM_FREQ];
+ int n_opts;
+ int rec_frequency;
+ bool ret;
+
+#ifdef HAVE_SPDIF_IN
+ if (global_settings.rec_source == AUDIO_SRC_SPDIF)
+ {
+ /* Inform user that frequency follows the source's frequency */
+ opts[0].string = ID2P(LANG_SOURCE_FREQUENCY);
+ opts[0].voice_id = LANG_SOURCE_FREQUENCY;
+ n_opts = 1;
+ rec_frequency = 0;
}
+ else
+#endif
+ {
+ struct encoder_caps caps;
+ struct encoder_config cfg;
+
+ cfg.rec_format = global_settings.rec_format;
+ global_to_encoder_config(&cfg);
+
+ if (!enc_get_caps(&cfg, &caps, true))
+ return false;
+
+ /* Construct samplerate menu based upon encoder settings */
+ n_opts = make_list_from_caps32(REC_SAMPR_CAPS, NULL,
+ caps.samplerate_caps, table);
+
+ if (n_opts == 0)
+ return false; /* No common flags...?? */
+
+ make_options_from_indexes(names, table, n_opts, opts);
+
+ /* Find closest rate that the potentially restricted list
+ comes to */
+ make_list_from_caps32(REC_SAMPR_CAPS, rec_freq_sampr,
+ caps.samplerate_caps, table);
+
+ rec_frequency = round_value_to_list32(
+ rec_freq_sampr[global_settings.rec_frequency],
+ table, n_opts, false);
+ }
+
+ ret = set_option(str(LANG_RECORDING_FREQUENCY),
+ &rec_frequency, INT, opts, n_opts, NULL );
+
+ if (!ret
+#ifdef HAVE_SPDIF_IN
+ && global_settings.rec_source != AUDIO_SRC_SPDIF
+#endif
+ )
+ {
+ /* Translate back to full index */
+ global_settings.rec_frequency =
+ round_value_to_list32(table[rec_frequency],
+ rec_freq_sampr,
+ REC_NUM_FREQ,
+ false);
+ }
+
+ return ret;
+#endif /* CONFIG_CODEC == SWCODEC */
+} /* recfrequency */
static bool recchannels(void)
{
- static const struct opt_items names[] = {
- { STR(LANG_CHANNEL_STEREO) },
- { STR(LANG_CHANNEL_MONO) }
+ static const struct opt_items names[CHN_NUM_MODES] = {
+ [CHN_MODE_STEREO] = { STR(LANG_CHANNEL_STEREO) },
+ [CHN_MODE_MONO] = { STR(LANG_CHANNEL_MONO) }
};
+#if CONFIG_CODEC == MAS3587F
return set_option(str(LANG_RECORDING_CHANNELS),
&global_settings.rec_channels, INT,
- names, 2, NULL );
+ names, CHN_NUM_MODES, NULL );
+#endif /* CONFIG_CODEC == MAS3587F */
+
+#if CONFIG_CODEC == SWCODEC
+ struct opt_items opts[CHN_NUM_MODES];
+ long table[CHN_NUM_MODES];
+ struct encoder_caps caps;
+ struct encoder_config cfg;
+ int n_opts;
+ int rec_channels;
+ bool ret;
+
+ cfg.rec_format = global_settings.rec_format;
+ global_to_encoder_config(&cfg);
+
+ if (!enc_get_caps(&cfg, &caps, true))
+ return false;
+
+ n_opts = make_list_from_caps32(CHN_CAP_ALL, NULL,
+ caps.channel_caps, table);
+
+ rec_channels = round_value_to_list32(global_settings.rec_channels,
+ table, n_opts, false);
+
+ make_options_from_indexes(names, table, n_opts, opts);
+
+ ret = set_option(str(LANG_RECORDING_CHANNELS), &rec_channels,
+ INT, opts, n_opts, NULL );
+
+ if (!ret)
+ global_settings.rec_channels = table[rec_channels];
+
+ return ret;
+#endif /* CONFIG_CODEC == SWCODEC */
}
static bool rectimesplit(void)
@@ -1049,58 +1180,59 @@ bool rectrigger(void)
action_signalscreenchange();
return retval;
}
-#endif
+#endif /* !defined(SIMULATOR) && CONFIG_CODEC == MAS3587F */
bool recording_menu(bool no_source)
{
- int m;
- int i = 0;
- struct menu_item items[13];
- bool result;
-
-#if CONFIG_CODEC == MAS3587F || CONFIG_CODEC == SWCODEC
- items[i].desc = ID2P(LANG_RECORDING_QUALITY);
- items[i++].function = recquality;
+ static const struct menu_item static_items[] = {
+#if CONFIG_CODEC == MAS3587F
+ { ID2P(LANG_RECORDING_QUALITY), recquality },
#endif
- items[i].desc = ID2P(LANG_RECORDING_FREQUENCY);
- items[i++].function = recfrequency;
- if(!no_source) {
- items[i].desc = ID2P(LANG_RECORDING_SOURCE);
- items[i++].function = recsource;
- }
- items[i].desc = ID2P(LANG_RECORDING_CHANNELS);
- items[i++].function = recchannels;
+#if CONFIG_CODEC == SWCODEC
+ { ID2P(LANG_RECORDING_FORMAT), recformat },
+ { ID2P(LANG_ENCODER_SETTINGS), enc_global_config_menu },
+#endif
+ { ID2P(LANG_RECORDING_FREQUENCY), recfrequency },
+ { ID2P(LANG_RECORDING_SOURCE), recsource }, /* not shown if no_source */
+ { ID2P(LANG_RECORDING_CHANNELS), recchannels },
#if CONFIG_CODEC == MAS3587F
- items[i].desc = ID2P(LANG_RECORDING_EDITABLE);
- items[i++].function = receditable;
+ { ID2P(LANG_RECORDING_EDITABLE), receditable },
#endif
- items[i].desc = ID2P(LANG_RECORD_TIMESPLIT);
- items[i++].function = filesplitoptionsmenu;
- items[i].desc = ID2P(LANG_RECORD_PRERECORD_TIME);
- items[i++].function = recprerecord;
- items[i].desc = ID2P(LANG_RECORD_DIRECTORY);
- items[i++].function = recdirectory;
- items[i].desc = ID2P(LANG_RECORD_STARTUP);
- items[i++].function = reconstartup;
+ { ID2P(LANG_RECORD_TIMESPLIT), filesplitoptionsmenu },
+ { ID2P(LANG_RECORD_PRERECORD_TIME), recprerecord },
+ { ID2P(LANG_RECORD_DIRECTORY), recdirectory },
+ { ID2P(LANG_RECORD_STARTUP), reconstartup },
#ifdef CONFIG_BACKLIGHT
- items[i].desc = ID2P(LANG_CLIP_LIGHT);
- items[i++].function = cliplight;
+ { ID2P(LANG_CLIP_LIGHT), cliplight },
#endif
#if !defined(SIMULATOR) && CONFIG_CODEC == MAS3587F
- items[i].desc = ID2P(LANG_RECORD_TRIGGER);
- items[i++].function = rectrigger;
+ { ID2P(LANG_RECORD_TRIGGER), rectrigger },
#endif
#ifdef HAVE_AGC
- items[i].desc = ID2P(LANG_RECORD_AGC_PRESET);
- items[i++].function = agc_preset;
- items[i].desc = ID2P(LANG_RECORD_AGC_CLIPTIME);
- items[i++].function = agc_cliptime;
+ { ID2P(LANG_RECORD_AGC_PRESET), agc_preset },
+ { ID2P(LANG_RECORD_AGC_CLIPTIME), agc_cliptime },
#endif
+ };
- m=menu_init( items, i, NULL, NULL, NULL, NULL);
+ struct menu_item items[ARRAYLEN(static_items)];
+ int i, n_items;
+ int m;
+
+ bool result;
+
+ for (i = 0, n_items = 0; i < (int)ARRAYLEN(items); i++)
+ {
+ const struct menu_item *mi = &static_items[i];
+ if (no_source && mi->function == recsource)
+ continue;
+ items[n_items++] = *mi;
+ }
+
+ m = menu_init(items, n_items, NULL, NULL, NULL, NULL);
result = menu_run(m);
menu_exit(m);
return result;
-}
-#endif
+} /* recording_menu */
+
+#endif /* HAVE_RECORDING */
diff --git a/apps/status.c b/apps/status.c
index 2a57db0f89..75219d604c 100644
--- a/apps/status.c
+++ b/apps/status.c
@@ -46,7 +46,7 @@
#ifdef CONFIG_TUNER
#include "radio.h"
#endif
-#if CONFIG_CODEC == SWCODEC
+#if defined(HAVE_RECORDING) && CONFIG_CODEC == SWCODEC
#include "pcm_record.h"
#endif
@@ -87,10 +87,6 @@ int current_playmode(void)
}
#ifdef HAVE_RECORDING
-#if CONFIG_CODEC == SWCODEC
- audio_stat = pcm_rec_status();
-#endif
-
if(audio_stat & AUDIO_STATUS_RECORD)
{
if(audio_stat & AUDIO_STATUS_PAUSE)
diff --git a/apps/talk.c b/apps/talk.c
index d81aa082c9..018f6ed5ab 100644
--- a/apps/talk.c
+++ b/apps/talk.c
@@ -46,13 +46,17 @@
MASCODEC | MASCODEC | SWCODEC
(playing) | (stopped) |
- audiobuf-----------+-----------+-----------
+ audiobuf-----------+-----------+------------
audio | voice | thumbnail
- |-----------|----------- filebuf
+ |-----------|------------
| thumbnail | voice
- | |-----------
+ | |------------
+ | | filebuf
+ | |------------
| | audio
- audiobufend----------+-----------+-----------
+ | |------------
+ | | codec swap
+ audiobufend----------+-----------+------------
SWCODEC allocates dedicated buffers, MASCODEC reuses audiobuf. */
@@ -102,7 +106,7 @@ struct queue_entry /* one entry of the internal queue */
/***************** Globals *****************/
-static unsigned char* p_thumbnail; /* buffer for thumbnail */
+static unsigned char* p_thumbnail = NULL; /* buffer for thumbnail */
static long size_for_thumbnail; /* leftover buffer size for it */
static struct voicefile* p_voicefile; /* loaded voicefile */
static bool has_voicefile; /* a voicefile file is present */
@@ -479,11 +483,14 @@ static void reset_state(void)
queue_write = queue_read = 0; /* reset the queue */
p_voicefile = NULL; /* indicate no voicefile (trashed) */
#if CONFIG_CODEC == SWCODEC
- /* Allocate a dedicated thumbnail buffer */
- size_for_thumbnail = audiobufend - audiobuf;
- if (size_for_thumbnail > MAX_THUMBNAIL_BUFSIZE)
- size_for_thumbnail = MAX_THUMBNAIL_BUFSIZE;
- p_thumbnail = buffer_alloc(size_for_thumbnail);
+ /* Allocate a dedicated thumbnail buffer - once */
+ if (p_thumbnail == NULL)
+ {
+ size_for_thumbnail = audiobufend - audiobuf;
+ if (size_for_thumbnail > MAX_THUMBNAIL_BUFSIZE)
+ size_for_thumbnail = MAX_THUMBNAIL_BUFSIZE;
+ p_thumbnail = buffer_alloc(size_for_thumbnail);
+ }
#else
/* Just use the audiobuf, without allocating anything */
p_thumbnail = audiobuf;
diff --git a/apps/tree.c b/apps/tree.c
index 653da791a8..6465b50e6f 100644
--- a/apps/tree.c
+++ b/apps/tree.c
@@ -58,7 +58,9 @@
#include "misc.h"
#include "filetree.h"
#include "tagtree.h"
+#ifdef HAVE_RECORDING
#include "recorder/recording.h"
+#endif
#include "rtc.h"
#include "dircache.h"
#ifdef HAVE_TAGCACHE
diff --git a/bootloader/main.c b/bootloader/main.c
index 99eb449151..0f3d706d7b 100644
--- a/bootloader/main.c
+++ b/bootloader/main.c
@@ -204,6 +204,7 @@ void main(void)
kernel_init();
set_cpu_frequency(CPUFREQ_NORMAL);
+ coldfire_set_pllcr_audio_bits(DEFAULT_PLLCR_AUDIO_BITS);
set_irq_level(0);
lcd_init();
@@ -311,6 +312,9 @@ void main(void)
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
/* Set up waitstates for the peripherals */
set_cpu_frequency(0); /* PLL off */
+#ifdef CPU_COLDFIRE
+ coldfire_set_pllcr_audio_bits(DEFAULT_PLLCR_AUDIO_BITS);
+#endif
#endif
#ifdef HAVE_UDA1380
diff --git a/firmware/SOURCES b/firmware/SOURCES
index 1ec3c82616..df38169be3 100644
--- a/firmware/SOURCES
+++ b/firmware/SOURCES
@@ -4,6 +4,7 @@ logf.c
#endif
backlight.c
buffer.c
+general.c
common/atoi.c
common/crc32.c
common/ctype.c
@@ -45,7 +46,12 @@ target/coldfire/memcpy-coldfire.S
target/coldfire/memmove-coldfire.S
target/coldfire/memset-coldfire.S
target/coldfire/memset16-coldfire.S
+#ifndef SIMULATOR
+#ifndef BOOTLOADER
+target/coldfire/pcm-coldfire.c
+#endif
target/coldfire/system-coldfire.c
+#endif
#elif (CONFIG_CPU == SH7034)
target/sh/memcpy-sh.S
target/sh/memmove-sh.S
@@ -207,15 +213,21 @@ drivers/wm8731l.c
#elif defined(HAVE_TLV320) && !defined(SIMULATOR)
drivers/tlv320.c
#endif
-#if (CONFIG_CODEC == SWCODEC) && !defined(SIMULATOR)
-pcm_playback.c
-#endif
-#if CONFIG_CODEC == SWCODEC
+#if (CONFIG_CODEC == SWCODEC) && !defined(BOOTLOADER)
+pcm_sampr.c
replaygain.c
-#endif
-#if defined(CPU_COLDFIRE) && !defined(SIMULATOR)
+#ifndef SIMULATOR
+pcm_playback.c
+#endif /* SIMULATOR */
+#ifdef HAVE_RECORDING
+enc_base.c
+#if defined(CPU_COLDFIRE)
+#ifndef SIMULATOR
pcm_record.c
-#endif
+#endif /* SIMULATOR */
+#endif /* CPU_COLDFIRE */
+#endif /* HAVE_RECORDING */
+#endif /* SWCODEC && !BOOTLOADER */
sound.c
#if defined(IRIVER_IFP7XX_SERIES) && defined(STUB)
common/sscanf.c
diff --git a/firmware/drivers/tlv320.c b/firmware/drivers/tlv320.c
index abce31ef81..7c4bbbd1ee 100644
--- a/firmware/drivers/tlv320.c
+++ b/firmware/drivers/tlv320.c
@@ -82,7 +82,7 @@ void tlv320_init(void)
tlv320_write_reg(REG_DAP, 0x00); /* No deemphasis */
tlv320_write_reg(REG_DAIF, DAIF_IWL_16 | DAIF_FOR_I2S);
tlv320_write_reg(REG_DIA, DIA_ACT);
- tlv320_write_reg(REG_SRC, (1 << 5)); /* 44.1kHz */
+ tlv320_set_frequency(-1); /* default */
/* All ON except ADC, MIC and LINE */
tlv320_write_reg(REG_PC, PC_ADC | PC_MIC | PC_LINE);
}
@@ -96,6 +96,32 @@ void tlv320_reset(void)
}
/**
+ * Sets internal sample rate for DAC and ADC relative to MCLK
+ * Selection for frequency:
+ * Fs: tlv: with:
+ * 11025: 0 = MCLK/2 MCLK/2 SCLK, LRCK: Audio Clk / 16
+ * 22050: 0 = MCLK/2 MCLK SCLK, LRCK: Audio Clk / 8
+ * 44100: 1 = MCLK MCLK SCLK, LRCK: Audio Clk / 4 (default)
+ * 88200: 2 = MCLK*2 MCLK SCLK, LRCK: Audio Clk / 2
+ */
+void tlv320_set_frequency(unsigned fsel)
+{
+ /* All rates available for 11.2896MHz besides 8.021 */
+ unsigned char values_src[3] =
+ {
+ /* Fs: */
+ (0x8 << 2) | SRC_CLKIN, /* 11025, 22050 */
+ (0x8 << 2), /* 44100 */
+ (0xf << 2), /* 88200 */
+ };
+
+ if (fsel >= ARRAYLEN(values_src))
+ fsel = 1;
+
+ tlv320_write_reg(REG_SRC, values_src[fsel]);
+}
+
+/**
* Sets left and right headphone volume
*
* Left & Right: 48 .. 121 .. 127 => Volume -73dB (mute) .. +0 dB .. +6 dB
@@ -142,7 +168,6 @@ void tlv320_set_recvol(int left, int right, int type)
value_aap &= ~AAP_MICB;
tlv320_write_reg(REG_AAP, value_aap);
-
}
else if (type == AUDIO_GAIN_LINEIN)
{
@@ -180,15 +205,17 @@ void tlv320_mute(bool mute)
}
/* Nice shutdown of TLV320 codec */
-void tlv320_close()
+void tlv320_close(void)
{
+ tlv320_mute(true);
+ sleep(HZ/8);
+
tlv320_write_reg(REG_PC, PC_OFF | PC_CLK | PC_OSC | PC_OUT |
PC_DAC | PC_ADC | PC_MIC | PC_LINE); /* All OFF */
}
void tlv320_enable_recording(bool source_mic)
{
- unsigned value_daif = tlv320_regs[REG_DAIF];
unsigned value_aap, value_pc;
if (source_mic)
@@ -205,20 +232,12 @@ void tlv320_enable_recording(bool source_mic)
tlv320_write_reg(REG_PC, value_pc);
tlv320_write_reg(REG_AAP, value_aap);
-
- /* Enable MASTER mode (start sending I2S data to the CPU) */
- value_daif |= DAIF_MS;
- tlv320_write_reg(REG_DAIF, value_daif);
}
-void tlv320_disable_recording()
+void tlv320_disable_recording(void)
{
unsigned value_pc = tlv320_regs[REG_PC];
unsigned value_aap = tlv320_regs[REG_AAP];
- unsigned value_daif = tlv320_regs[REG_DAIF];
-
- value_daif &= ~DAIF_MS; /* disable MASTER mode */
- tlv320_write_reg(REG_DAIF, value_daif);
value_aap |= AAP_MICM; /* mute MIC */
tlv320_write_reg(REG_PC, value_aap);
diff --git a/firmware/drivers/uda1380.c b/firmware/drivers/uda1380.c
index 241a117385..d6dfe6623b 100644
--- a/firmware/drivers/uda1380.c
+++ b/firmware/drivers/uda1380.c
@@ -49,9 +49,10 @@ short recgain_line;
#define NUM_DEFAULT_REGS 13
unsigned short uda1380_defaults[2*NUM_DEFAULT_REGS] =
{
- REG_0, EN_DAC | EN_INT | EN_DEC | SYSCLK_256FS | WSPLL_25_50,
+ REG_0, EN_DAC | EN_INT | EN_DEC | ADC_CLK | DAC_CLK |
+ SYSCLK_256FS | WSPLL_25_50,
REG_I2S, I2S_IFMT_IIS,
- REG_PWR, PON_BIAS,
+ REG_PWR, PON_PLL | PON_BIAS,
/* PON_HP & PON_DAC is enabled later */
REG_AMIX, AMIX_RIGHT(0x3f) | AMIX_LEFT(0x3f),
/* 00=max, 3f=mute */
@@ -60,7 +61,7 @@ unsigned short uda1380_defaults[2*NUM_DEFAULT_REGS] =
REG_MIX_VOL, MIX_VOL_CH_1(0) | MIX_VOL_CH_2(0xff),
/* 00=max, ff=mute */
REG_EQ, EQ_MODE_MAX,
- /* Bass and tremble = 0 dB */
+ /* Bass and treble = 0 dB */
REG_MUTE, MUTE_MASTER | MUTE_CH2,
/* Mute everything to start with */
REG_MIX_CTL, MIX_CTL_MIX,
@@ -192,6 +193,43 @@ void uda1380_reset(void)
#endif
}
+/**
+ * Sets frequency settings for DAC and ADC relative to MCLK
+ *
+ * Selection for frequency ranges:
+ * Fs: range: with:
+ * 11025: 0 = 6.25 to 12.5 MCLK/2 SCLK, LRCK: Audio Clk / 16
+ * 22050: 1 = 12.5 to 25 MCLK/2 SCLK, LRCK: Audio Clk / 8
+ * 44100: 2 = 25 to 50 MCLK SCLK, LRCK: Audio Clk / 4 (default)
+ * 88200: 3 = 50 to 100 MCLK SCLK, LRCK: Audio Clk / 2 <= TODO: Needs WSPLL
+ */
+void uda1380_set_frequency(unsigned fsel)
+{
+ static const unsigned short values_reg[4][2] =
+ {
+ /* Fs: */
+ { 0, WSPLL_625_125 | SYSCLK_512FS }, /* 11025 */
+ { 0, WSPLL_125_25 | SYSCLK_256FS }, /* 22050 */
+ { MIX_CTL_SEL_NS, WSPLL_25_50 | SYSCLK_256FS }, /* 44100 */
+ { MIX_CTL_SEL_NS, WSPLL_50_100 | SYSCLK_256FS }, /* 88200 */
+ };
+
+ const unsigned short *ent;
+
+ if (fsel >= ARRAYLEN(values_reg))
+ fsel = 2;
+
+ ent = values_reg[fsel];
+
+ /* Set WSPLL input frequency range or SYSCLK divider */
+ uda1380_regs[REG_0] &= ~0xf;
+ uda1380_write_reg(REG_0, uda1380_regs[REG_0] | ent[1]);
+
+ /* Choose 3rd order or 5th order noise shaper */
+ uda1380_regs[REG_MIX_CTL] &= ~MIX_CTL_SEL_NS;
+ uda1380_write_reg(REG_MIX_CTL, uda1380_regs[REG_MIX_CTL] | ent[0]);
+}
+
/* Initialize UDA1380 codec with default register values (uda1380_defaults) */
int uda1380_init(void)
{
@@ -227,30 +265,34 @@ void uda1380_close(void)
*/
void uda1380_enable_recording(bool source_mic)
{
+ uda1380_regs[REG_0] &= ~(ADC_CLK | DAC_CLK);
uda1380_write_reg(REG_0, uda1380_regs[REG_0] | EN_ADC);
if (source_mic)
{
/* VGA_GAIN: 0=0 dB, F=30dB */
+ /* Output of left ADC is fed into right bitstream */
+ uda1380_regs[REG_PWR] &= ~(PON_PLL | PON_PGAR | PON_ADCR);
uda1380_write_reg(REG_PWR, uda1380_regs[REG_PWR] | PON_LNA | PON_ADCL);
+ uda1380_regs[REG_ADC] &= ~SKIP_DCFIL;
uda1380_write_reg(REG_ADC, (uda1380_regs[REG_ADC] & VGA_GAIN_MASK)
| SEL_LNA | SEL_MIC | EN_DCFIL);
uda1380_write_reg(REG_PGA, 0);
- } else
+ }
+ else
{
/* PGA_GAIN: 0=0 dB, F=24dB */
+ uda1380_regs[REG_PWR] &= ~(PON_PLL | PON_LNA);
uda1380_write_reg(REG_PWR, uda1380_regs[REG_PWR] | PON_PGAL | PON_ADCL
| PON_PGAR | PON_ADCR);
uda1380_write_reg(REG_ADC, EN_DCFIL);
- uda1380_write_reg(REG_PGA, (uda1380_regs[REG_PGA] & PGA_GAIN_MASK)
- | PGA_GAINL(0) | PGA_GAINR(0));
+ uda1380_write_reg(REG_PGA, uda1380_regs[REG_PGA] & PGA_GAIN_MASK);
}
sleep(HZ/8);
uda1380_write_reg(REG_I2S, uda1380_regs[REG_I2S] | I2S_MODE_MASTER);
uda1380_write_reg(REG_MIX_CTL, MIX_MODE(1));
-
}
/**
@@ -262,10 +304,13 @@ void uda1380_disable_recording(void)
sleep(HZ/8);
uda1380_write_reg(REG_I2S, I2S_IFMT_IIS);
- uda1380_write_reg(REG_PWR, uda1380_regs[REG_PWR] & ~(PON_LNA | PON_ADCL
- | PON_ADCR | PON_PGAL
- | PON_PGAR));
- uda1380_write_reg(REG_0, uda1380_regs[REG_0] & ~EN_ADC);
+
+ uda1380_regs[REG_PWR] &= ~(PON_LNA | PON_ADCL | PON_ADCR | PON_PGAL | PON_PGAR);
+ uda1380_write_reg(REG_PWR, uda1380_regs[REG_PWR] | PON_PLL);
+
+ uda1380_regs[REG_0] &= ~EN_ADC;
+ uda1380_write_reg(REG_0, uda1380_regs[REG_0] | ADC_CLK | DAC_CLK);
+
uda1380_write_reg(REG_ADC, SKIP_DCFIL);
}
@@ -373,20 +418,3 @@ void uda1380_set_monitor(int enable)
else /* mute channel 2 */
uda1380_write_reg(REG_MUTE, uda1380_regs[REG_MUTE] | MUTE_CH2);
}
-
-/* Change the order of the noise chaper,
- 5th order is recommended above 32kHz */
-void uda1380_set_nsorder(int order)
-{
- switch(order)
- {
- case 5:
- uda1380_write_reg(REG_MIX_CTL, uda1380_regs[REG_MIX_CTL]
- | MIX_CTL_SEL_NS);
- break;
- case 3:
- default:
- uda1380_write_reg(REG_MIX_CTL, uda1380_regs[REG_MIX_CTL]
- & ~MIX_CTL_SEL_NS);
- }
-}
diff --git a/firmware/export/audio.h b/firmware/export/audio.h
index 9099cb3765..d3f544de94 100644
--- a/firmware/export/audio.h
+++ b/firmware/export/audio.h
@@ -20,6 +20,20 @@
#define AUDIO_H
#include <stdbool.h>
+#include <sys/types.h>
+/* These must always be included with audio.h for this to compile under
+ cetain conditions. Do it here or else spread the complication around to
+ many files. */
+#if CONFIG_CODEC == SWCODEC
+#include "pcm_sampr.h"
+#include "pcm_playback.h"
+#ifdef HAVE_RECORDING
+#include "pcm_record.h"
+#include "id3.h"
+#include "enc_base.h"
+#endif /* HAVE_RECORDING */
+#endif /* CONFIG_CODEC == SWCODEC */
+
#ifdef SIMULATOR
#define audio_play(x) sim_audio_play(x)
@@ -31,9 +45,6 @@
#define AUDIO_STATUS_PRERECORD 8
#define AUDIO_STATUS_ERROR 16
-#define AUDIO_STATUS_STAYON_FLAGS \
- (AUDIO_STATUS_PLAY | AUDIO_STATUS_PAUSE | AUDIO_STATUS_RECORD | AUDIO_)
-
#define AUDIOERR_DISK_FULL 1
#define AUDIO_GAIN_LINEIN 0
@@ -72,10 +83,11 @@ void audio_resume(void);
void audio_next(void);
void audio_prev(void);
int audio_status(void);
-bool audio_query_poweroff(void);
+#if CONFIG_CODEC == SWCODEC
int audio_track_count(void); /* SWCODEC only */
long audio_filebufused(void); /* SWCODEC only */
void audio_pre_ff_rewind(void); /* SWCODEC only */
+#endif /* CONFIG_CODEC == SWCODEC */
void audio_ff_rewind(long newtime);
void audio_flush_and_reload_tracks(void);
struct mp3entry* audio_current_track(void);
@@ -89,18 +101,28 @@ void audio_error_clear(void);
int audio_get_file_pos(void);
void audio_beep(int duration);
void audio_init_playback(void);
+unsigned char *audio_get_buffer(bool talk_buf, size_t *buffer_size);
-/* audio recording functions */
-void audio_init_recording(unsigned int buffer_offset);
-void audio_close_recording(void);
-void audio_record(const char *filename);
-void audio_stop_recording(void);
-void audio_pause_recording(void);
-void audio_resume_recording(void);
-void audio_new_file(const char *filename);
+/* channel modes */
+enum rec_channel_modes
+{
+ __CHN_MODE_START_INDEX = -1,
+
+ CHN_MODE_STEREO,
+ CHN_MODE_MONO,
+
+ CHN_NUM_MODES
+};
+
+#if CONFIG_CODEC == SWCODEC
+/* channel mode capability bits */
+#define CHN_CAP_STEREO (1 << CHN_MODE_STEREO)
+#define CHN_CAP_MONO (1 << CHN_MODE_MONO)
+#define CHN_CAP_ALL (CHN_CAP_STEREO | CHN_CAP_MONO)
+#endif /* CONFIG_CODEC == SWCODEC */
/* audio sources */
-enum
+enum audio_sources
{
AUDIO_SRC_PLAYBACK = -1, /* for audio playback (default) */
AUDIO_SRC_MIC, /* monitor mic */
@@ -123,33 +145,57 @@ enum
AUDIO_SRC_MAX = AUDIO_NUM_SOURCES-1
};
-/* channel modes */
-enum
+#ifdef HAVE_RECORDING
+/* parameters for audio_set_recording_options */
+struct audio_recording_options
{
- CHN_MODE_MONO = 1,
- CHN_MODE_STEREO,
+ int rec_source;
+ int rec_frequency;
+ int rec_channels;
+ int rec_prerecord_time;
+#if CONFIG_CODEC == SWCODEC
+ int rec_source_flags; /* for rec_set_source */
+ struct encoder_config enc_config;
+#else
+ int rec_quality;
+ bool rec_editable;
+#endif
};
-void audio_set_recording_options(int frequency, int quality,
- int source, int channel_mode,
- bool editable, int prerecord_time);
+
+/* audio recording functions */
+void audio_init_recording(unsigned int buffer_offset);
+void audio_close_recording(void);
+void audio_record(const char *filename);
+void audio_stop_recording(void);
+void audio_pause_recording(void);
+void audio_resume_recording(void);
+void audio_new_file(const char *filename);
+void audio_set_recording_options(struct audio_recording_options *options);
void audio_set_recording_gain(int left, int right, int type);
unsigned long audio_recorded_time(void);
unsigned long audio_num_recorded_bytes(void);
-#if 0
-#ifdef HAVE_SPDIF_POWER
-void audio_set_spdif_power_setting(bool on);
-#endif
-#endif
-unsigned long audio_get_spdif_sample_rate(void);
-unsigned long audio_prev_elapsed(void);
+
#if CONFIG_CODEC == SWCODEC
-/* audio encoder functions (defined in playback.c) */
-int audio_get_encoder_id(void);
-void audio_load_encoder(int enc_id);
+/* SWCODEC recoring functions */
+/* playback.c */
+bool audio_load_encoder(int afmt);
void audio_remove_encoder(void);
+unsigned char *audio_get_recording_buffer(size_t *buffer_size);
#endif /* CONFIG_CODEC == SWCODEC */
+#endif /* HAVE_RECORDING */
+#ifdef HAVE_SPDIF_IN
+#ifdef HAVE_SPDIF_POWER
+void audio_set_spdif_power_setting(bool on);
+bool audio_get_spdif_power_setting(void);
+#endif
+/* returns index into rec_master_sampr_list */
+int audio_get_spdif_sample_rate(void);
+/* > 0: monitor EBUin, 0: Monitor IISrecv, <0: reset only */
+void audio_spdif_set_monitor(int monitor_spdif);
+#endif /* HAVE_SPDIF_IN */
+unsigned long audio_prev_elapsed(void);
/***********************************************************************/
diff --git a/firmware/export/config-h100.h b/firmware/export/config-h100.h
index 6f74078e1e..285ab88930 100644
--- a/firmware/export/config-h100.h
+++ b/firmware/export/config-h100.h
@@ -84,6 +84,12 @@
/* define this if you have recording possibility */
#define HAVE_RECORDING 1
+/* define hardware samples rate caps mask */
+#define HW_SAMPR_CAPS (SAMPR_CAP_88 | SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
+/* define the bitmask of recording sample rates */
+#define REC_SAMPR_CAPS (SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
#define HAVE_AGC
#ifndef SIMULATOR
diff --git a/firmware/export/config-h120.h b/firmware/export/config-h120.h
index 1476102100..b22ff0eb22 100644
--- a/firmware/export/config-h120.h
+++ b/firmware/export/config-h120.h
@@ -77,6 +77,12 @@
/* define this if you have recording possibility */
#define HAVE_RECORDING 1
+/* define hardware samples rate caps mask */
+#define HW_SAMPR_CAPS (SAMPR_CAP_88 | SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
+/* define the bitmask of recording sample rates */
+#define REC_SAMPR_CAPS (SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
#define HAVE_AGC
#define BATTERY_CAPACITY_DEFAULT 1300 /* default battery capacity */
diff --git a/firmware/export/config-h300.h b/firmware/export/config-h300.h
index 31f0f6729f..748635dcb4 100644
--- a/firmware/export/config-h300.h
+++ b/firmware/export/config-h300.h
@@ -72,6 +72,12 @@
/* define this if you have recording possibility */
#define HAVE_RECORDING 1
+/* define hardware samples rate caps mask */
+#define HW_SAMPR_CAPS (SAMPR_CAP_88 | SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
+/* define the bitmask of recording sample rates */
+#define REC_SAMPR_CAPS (SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
#define HAVE_AGC
#define BATTERY_CAPACITY_DEFAULT 1300 /* default battery capacity */
@@ -157,4 +163,3 @@
/* Define this for FM radio input available */
#define HAVE_FMRADIO_IN
-
diff --git a/firmware/export/config-iaudiox5.h b/firmware/export/config-iaudiox5.h
index 80b010a6b0..d4c904ed23 100644
--- a/firmware/export/config-iaudiox5.h
+++ b/firmware/export/config-iaudiox5.h
@@ -9,6 +9,12 @@
/* define this if you have recording possibility */
#define HAVE_RECORDING 1
+/* define the bitmask of hardware sample rates */
+#define HW_SAMPR_CAPS (SAMPR_CAP_88 | SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
+/* define the bitmask of recording sample rates */
+#define REC_SAMPR_CAPS (SAMPR_CAP_88 | SAMPR_CAP_44 | SAMPR_CAP_22 | SAMPR_CAP_11)
+
/* define this if you have a bitmap LCD display */
#define HAVE_LCD_BITMAP 1
diff --git a/firmware/export/id3.h b/firmware/export/id3.h
index 1d07affbfa..dd099e0204 100644
--- a/firmware/export/id3.h
+++ b/firmware/export/id3.h
@@ -24,13 +24,19 @@
#include "file.h"
/* Audio file types. */
-enum {
+enum
+{
AFMT_UNKNOWN = 0, /* Unknown file format */
+ /* start formats */
+
AFMT_MPA_L1, /* MPEG Audio layer 1 */
AFMT_MPA_L2, /* MPEG Audio layer 2 */
AFMT_MPA_L3, /* MPEG Audio layer 3 */
+ AFMT_AIFF, /* Audio Interchange File Format */
+
+#if CONFIG_CODEC == SWCODEC
AFMT_PCM_WAV, /* Uncompressed PCM in a WAV file */
AFMT_OGG_VORBIS, /* Ogg Vorbis */
AFMT_FLAC, /* FLAC */
@@ -40,54 +46,91 @@ enum {
AFMT_ALAC, /* Apple Lossless Audio Codec */
AFMT_AAC, /* Advanced Audio Coding (AAC) in M4A container */
AFMT_SHN, /* Shorten */
- AFMT_AIFF, /* Audio Interchange File Format */
AFMT_SID, /* SID File Format */
- AFMT_ADX, /* ADX */
+ AFMT_ADX, /* ADX File Format */
+#endif
- /* New formats must be added to the end of this list */
+ /* add new formats at any index above this line to have a sensible order -
+ specified array index inits are used */
+ /* format arrays defined in id3.c */
AFMT_NUM_CODECS,
-#if CONFIG_CODEC == SWCODEC
+#if CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING)
/* masks to decompose parts */
CODEC_AFMT_MASK = 0x0fff,
CODEC_TYPE_MASK = 0x7000,
/* switch for specifying codec type when requesting a filename */
CODEC_TYPE_DECODER = (0 << 12), /* default */
- CODEC_TYPE_ENCODER = (1 << 12)
-#endif
+ CODEC_TYPE_ENCODER = (1 << 12),
+#endif /* CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING) */
};
#if CONFIG_CODEC == SWCODEC
-#define AFMT_ENTRY(label, codec_fname, codec_enc_fname, enc_ext) \
- { label, codec_fname, codec_enc_fname, enc_ext }
-#else
-#define AFMT_ENTRY(label, codec_fname, codec_enc_fname, enc_ext) \
- { label }
-#endif
+#define CODEC_EXTENSION "codec"
+
+#ifdef HAVE_RECORDING
+#define ENCODER_SUFFIX "_enc"
+enum rec_format_indexes
+{
+ __REC_FORMAT_START_INDEX = -1,
+
+ /* start formats */
+
+ REC_FORMAT_PCM_WAV,
+ REC_FORMAT_WAVPACK,
+ REC_FORMAT_MPA_L3,
+
+ /* add new formats at any index above this line to have a sensible order -
+ specified array index inits are used
+ REC_FORMAT_CFG_NUM_BITS should allocate enough bits to hold the range
+ REC_FORMAT_CFG_VALUE_LIST should be in same order as indexes
+ */
+
+ REC_NUM_FORMATS,
+
+ REC_FORMAT_DEFAULT = REC_FORMAT_PCM_WAV,
+ REC_FORMAT_CFG_NUM_BITS = 2
+};
+
+#define REC_FORMAT_CFG_VAL_LIST "wave,wvpk,mpa3"
+
+/* get REC_FORMAT_* corresponding AFMT_* */
+extern const int rec_format_afmt[REC_NUM_FORMATS];
+/* get AFMT_* corresponding REC_FORMAT_* */
+extern const int afmt_rec_format[AFMT_NUM_CODECS];
+
+#define AFMT_ENTRY(label, root_fname, enc_root_fname, ext_list) \
+ { label, root_fname, enc_root_fname, ext_list }
+#else /* !HAVE_RECORDING */
+#define AFMT_ENTRY(label, root_fname, enc_root_fname, ext_list) \
+ { label, root_fname, ext_list }
+#endif /* HAVE_RECORDING */
+#else /* !SWCODEC */
+
+#define AFMT_ENTRY(label, root_fname, enc_root_fname, ext_list) \
+ { label, ext_list }
+#endif /* CONFIG_CODEC == SWCODEC */
/* record describing the audio format */
struct afmt_entry
{
-#if CONFIG_CODEC == SWCODEC
char label[8]; /* format label */
- char *codec_fn; /* filename of decoder codec */
- char *codec_enc_fn; /* filename of encoder codec */
- char *ext; /* default extension for file (enc only for now) */
-#else
- char label[4];
+#if CONFIG_CODEC == SWCODEC
+ char *codec_root_fn; /* root codec filename (sans _enc and .codec) */
+#ifdef HAVE_RECORDING
+ char *codec_enc_root_fn; /* filename of encoder codec */
+#endif
#endif
+ char *ext_list; /* double NULL terminated extension
+ list for type with the first as
+ the default for recording */
};
/* database of labels and codecs. add formats per above enum */
extern const struct afmt_entry audio_formats[AFMT_NUM_CODECS];
-#if CONFIG_CODEC == SWCODEC
-/* recording quality to AFMT_* */
-extern const int rec_quality_info_afmt[9];
-#endif
-
struct mp3entry {
char path[MAX_PATH];
char* title;
diff --git a/firmware/export/pcm_playback.h b/firmware/export/pcm_playback.h
index a4cd93969b..9c3e96ba63 100644
--- a/firmware/export/pcm_playback.h
+++ b/firmware/export/pcm_playback.h
@@ -19,11 +19,23 @@
#ifndef PCM_PLAYBACK_H
#define PCM_PLAYBACK_H
+#include <sys/types.h>
+
+/* Typedef for registered callback (play and record) */
+typedef void (*pcm_more_callback_type)(unsigned char **start,
+ size_t *size);
+
void pcm_init(void);
+
+/* set the pcm frequency - use values in hw_sampr_list
+ * use -1 for the default frequency
+ */
void pcm_set_frequency(unsigned int frequency);
+/* apply settings to hardware immediately */
+void pcm_apply_settings(bool reset);
/* This is for playing "raw" PCM data */
-void pcm_play_data(void (*get_more)(unsigned char** start, size_t* size),
+void pcm_play_data(pcm_more_callback_type get_more,
unsigned char* start, size_t size);
void pcm_calculate_peaks(int *left, int *right);
@@ -35,4 +47,4 @@ void pcm_play_pause(bool play);
bool pcm_is_paused(void);
bool pcm_is_playing(void);
-#endif
+#endif /* PCM_PLAYBACK_H */
diff --git a/firmware/export/pcm_record.h b/firmware/export/pcm_record.h
index b217335340..c1187a4c6c 100644
--- a/firmware/export/pcm_record.h
+++ b/firmware/export/pcm_record.h
@@ -20,24 +20,44 @@
#ifndef PCM_RECORD_H
#define PCM_RECORD_H
-void enc_set_parameters(int chunk_size, int num_chunks,
- int samp_per_chunk, char *head_ptr, int head_size,
- int enc_id);
-void enc_get_inputs(int *buffer_size, int *channels, int *quality);
-unsigned int* enc_alloc_chunk(void);
-void enc_free_chunk(void);
-int enc_wavbuf_near_empty(void);
-char* enc_get_wav_data(int size);
-extern void (*enc_set_header_callback)(void *head_buffer, int head_size,
- int num_pcm_samples, bool is_file_header);
+#define DMA_REC_ERROR_DMA ((size_t)-1)
+#ifdef HAVE_SPDIF_IN
+#define DMA_REC_ERROR_SPDIF ((size_t)-2)
+#endif
+/* Use AUDIO_SRC_* enumeration values */
+void pcm_set_monitor(int monitor);
+void pcm_set_rec_source(int source);
+
+/**
+ * RAW pcm data recording
+ * These calls are nescessary only when using the raw pcm apis directly.
+ */
+
+/* Initialize pcm recording interface */
+void pcm_init_recording(void);
+/* Uninitialze pcm recording interface */
+void pcm_close_recording(void);
+
+/* Start recording "raw" PCM data */
+void pcm_record_data(pcm_more_callback_type more_ready,
+ unsigned char *start, size_t size);
+/* Stop tranferring data into supplied buffer */
+void pcm_stop_recording(void);
+
+void pcm_calculate_rec_peaks(int *left, int *right);
+
+/** General functions for high level codec recording **/
+void pcm_rec_error_clear(void);
unsigned long pcm_rec_status(void);
void pcm_rec_init(void);
void pcm_rec_mux(int source);
int pcm_rec_current_bitrate(void);
+int pcm_rec_encoder_afmt(void); /* AFMT_* value, AFMT_UNKNOWN if none */
+int pcm_rec_rec_format(void); /* Format index or -1 otherwise */
+unsigned long pcm_rec_sample_rate(void);
int pcm_get_num_unprocessed(void);
-void pcm_rec_get_peaks(int *left, int *right);
-/* audio.h contains audio recording functions */
+/* audio.h contains audio_* recording functions */
-#endif
+#endif /* PCM_RECORD_H */
diff --git a/firmware/export/system.h b/firmware/export/system.h
index 4a33d80466..9b90a6e80c 100644
--- a/firmware/export/system.h
+++ b/firmware/export/system.h
@@ -21,7 +21,6 @@
#define __SYSTEM_H__
#include "cpu.h"
-#include "config.h"
#include "stdbool.h"
extern void system_reboot (void);
@@ -111,6 +110,23 @@ const char *get_cpu_boost_tracker(void);
#define MAX(a, b) (((a)>(b))?(a):(b))
#endif
+/* return number of elements in array a */
+#define ARRAYLEN(a) (sizeof(a)/sizeof((a)[0]))
+
+/* return p incremented by specified number of bytes */
+#define SKIPBYTES(p, count) ((typeof (p))((char *)(p) + (count)))
+
+#define P2_M1(p2) ((1 << (p2))-1)
+
+/* align up or down to nearest 2^p2 */
+#define ALIGN_DOWN_P2(n, p2) ((n) & ~P2_M1(p2))
+#define ALIGN_UP_P2(n, p2) ALIGN_DOWN_P2((n) + P2_M1(p2),p2)
+
+/* align up or down to nearest integer multiple of a */
+#define ALIGN_DOWN(n, a) ((n)/(a)*(a))
+#define ALIGN_UP(n, a) ALIGN_DOWN((n)+((a)-1),a)
+
+/* live endianness conversion */
#ifdef ROCKBOX_LITTLE_ENDIAN
#define letoh16(x) (x)
#define letoh32(x) (x)
@@ -120,6 +136,8 @@ const char *get_cpu_boost_tracker(void);
#define betoh32(x) swap32(x)
#define htobe16(x) swap16(x)
#define htobe32(x) swap32(x)
+#define swap_odd_even_be32(x) (x)
+#define swap_odd_even_le32(x) swap_odd_even32(x)
#else
#define letoh16(x) swap16(x)
#define letoh32(x) swap32(x)
@@ -129,6 +147,37 @@ const char *get_cpu_boost_tracker(void);
#define betoh32(x) (x)
#define htobe16(x) (x)
#define htobe32(x) (x)
+#define swap_odd_even_be32(x) swap_odd_even32(x)
+#define swap_odd_even_le32(x) (x)
+#endif
+
+/* static endianness conversion */
+#define SWAP_16(x) ((typeof(x))(unsigned short)(((unsigned short)(x) >> 8) | \
+ ((unsigned short)(x) << 8)))
+
+#define SWAP_32(x) ((typeof(x))(unsigned long)( ((unsigned long)(x) >> 24) | \
+ (((unsigned long)(x) & 0xff0000ul) >> 8) | \
+ (((unsigned long)(x) & 0xff00ul) << 8) | \
+ ((unsigned long)(x) << 24)))
+
+#ifdef ROCKBOX_LITTLE_ENDIAN
+#define LE_TO_H16(x) (x)
+#define LE_TO_H32(x) (x)
+#define H_TO_LE16(x) (x)
+#define H_TO_LE32(x) (x)
+#define BE_TO_H16(x) SWAP_16(x)
+#define BE_TO_H32(x) SWAP_32(x)
+#define H_TO_BE16(x) SWAP_16(x)
+#define H_TO_BE32(x) SWAP_32(x)
+#else
+#define LE_TO_H16(x) SWAP_16(x)
+#define LE_TO_H32(x) SWAP_32(x)
+#define H_TO_LE16(x) SWAP_16(x)
+#define H_TO_LE32(x) SWAP_32(x)
+#define BE_TO_H16(x) (x)
+#define BE_TO_H32(x) (x)
+#define H_TO_BE16(x) (x)
+#define H_TO_BE32(x) (x)
#endif
@@ -181,6 +230,7 @@ enum {
: /* %0 */ I_CONSTRAINT((char)(mask)), \
/* %1 */ "z"(address-GBR))
+
#endif /* CONFIG_CPU == SH7034 */
#ifndef SIMULATOR
@@ -388,7 +438,20 @@ static inline unsigned long swap32(unsigned long value)
#define invalidate_icache()
#endif
-#else
+
+#ifndef CPU_COLDFIRE
+static inline unsigned long swap_odd_even32(unsigned long value)
+{
+ /*
+ result[31..24],[15.. 8] = value[23..16],[ 7.. 0]
+ result[23..16],[ 7.. 0] = value[31..24],[15.. 8]
+ */
+ unsigned long t = value & 0xff00ff00;
+ return (t >> 8) | ((t ^ value) << 8);
+}
+#endif
+
+#else /* SIMULATOR */
static inline unsigned short swap16(unsigned short value)
/*
@@ -412,8 +475,18 @@ static inline unsigned long swap32(unsigned long value)
return (lo << 16) | hi;
}
+static inline unsigned long swap_odd_even32(unsigned long value)
+{
+ /*
+ result[31..24],[15.. 8] = value[23..16],[ 7.. 0]
+ result[23..16],[ 7.. 0] = value[31..24],[15.. 8]
+ */
+ unsigned long t = value & 0xff00ff00;
+ return (t >> 8) | ((t ^ value) << 8);
+}
+
#define invalidate_icache()
-#endif
+#endif /* !SIMULATOR */
-#endif
+#endif /* __SYSTEM_H__ */
diff --git a/firmware/export/thread.h b/firmware/export/thread.h
index 17e6e3aa88..72c692ec3b 100644
--- a/firmware/export/thread.h
+++ b/firmware/export/thread.h
@@ -142,7 +142,10 @@ void switch_thread(bool save_context, struct thread_entry **blocked_list);
void sleep_thread(int ticks);
void block_thread(struct thread_entry **thread, int timeout);
void wakeup_thread(struct thread_entry **thread);
+#ifdef HAVE_PRIORITY_SCHEDULING
int thread_set_priority(struct thread_entry *thread, int priority);
+int thread_get_priority(struct thread_entry *thread);
+#endif
void init_threads(void);
int thread_stack_usage(const struct thread_entry *thread);
int thread_get_status(const struct thread_entry *thread);
diff --git a/firmware/export/tlv320.h b/firmware/export/tlv320.h
index dfcbec4373..023ec9d874 100644
--- a/firmware/export/tlv320.h
+++ b/firmware/export/tlv320.h
@@ -24,6 +24,16 @@
extern void tlv320_init(void);
extern void tlv320_reset(void);
+/**
+ * Sets internal sample rate for DAC and ADC relative to MCLK
+ * Selection for frequency:
+ * Fs: tlv: with:
+ * 11025: 0 = MCLK/2 MCLK/2 SCLK, LRCK: Audio Clk / 16
+ * 22050: 0 = MCLK/2 MCLK SCLK, LRCK: Audio Clk / 8
+ * 44100: 1 = MCLK MCLK SCLK, LRCK: Audio Clk / 4 (default)
+ * 88200: 2 = MCLK*2 MCLK SCLK, LRCK: Audio Clk / 2
+ */
+extern void tlv320_set_frequency(unsigned fsel);
extern void tlv320_enable_output(bool enable);
extern void tlv320_set_headphone_vol(int vol_l, int vol_r);
extern void tlv320_set_recvol(int left, int right, int type);
diff --git a/firmware/export/uda1380.h b/firmware/export/uda1380.h
index 9c761c6a7d..639ca8aa5c 100644
--- a/firmware/export/uda1380.h
+++ b/firmware/export/uda1380.h
@@ -28,8 +28,17 @@ extern void uda1380_set_bass(int value);
extern void uda1380_set_treble(int value);
extern int uda1380_mute(int mute);
extern void uda1380_close(void);
-extern void uda1380_set_nsorder(int order);
-
+/**
+ * Sets frequency settings for DAC and ADC relative to MCLK
+ *
+ * Selection for frequency ranges:
+ * Fs: range: with:
+ * 11025: 0 = 6.25 to 12.5 SCLK, LRCK: Audio Clk / 16
+ * 22050: 1 = 12.5 to 25 SCLK, LRCK: Audio Clk / 8
+ * 44100: 2 = 25 to 50 SCLK, LRCK: Audio Clk / 4 (default)
+ * 88200: 3 = 50 to 100 SCLK, LRCK: Audio Clk / 2
+ */
+extern void uda1380_set_frequency(unsigned fsel);
extern void uda1380_enable_recording(bool source_mic);
extern void uda1380_disable_recording(void);
extern void uda1380_set_recvol(int left, int right, int type);
diff --git a/firmware/id3.c b/firmware/id3.c
index 92f60a2095..7d03c75331 100644
--- a/firmware/id3.c
+++ b/firmware/id3.c
@@ -44,6 +44,89 @@
#include "replaygain.h"
#include "rbunicode.h"
+/** Database of audio formats **/
+const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
+{
+ /* Unknown file format */
+ [AFMT_UNKNOWN] =
+ AFMT_ENTRY("???", NULL, NULL, NULL ),
+
+ /* MPEG Audio layer 1 */
+ [AFMT_MPA_L1] =
+ AFMT_ENTRY("MP1", "mpa", NULL, "mp1\0" ),
+ /* MPEG Audio layer 2 */
+ [AFMT_MPA_L2] =
+ AFMT_ENTRY("MP2", "mpa", NULL, "mpa\0mp2\0" ),
+ /* MPEG Audio layer 3 */
+ [AFMT_MPA_L3] =
+ AFMT_ENTRY("MP3", "mpa", "mp3_enc", "mp3\0" ),
+
+ /* Audio Interchange File Format */
+ [AFMT_AIFF] =
+ AFMT_ENTRY("AIFF", "aiff", NULL, "aiff\0aif\0"),
+
+#if CONFIG_CODEC == SWCODEC
+ /* Uncompressed PCM in a WAV file */
+ [AFMT_PCM_WAV] =
+ AFMT_ENTRY("WAV", "wav", "wav_enc", "wav\0" ),
+ /* Ogg Vorbis */
+ [AFMT_OGG_VORBIS] =
+ AFMT_ENTRY("Ogg", "vorbis", NULL, "ogg\0" ),
+ /* FLAC */
+ [AFMT_FLAC] =
+ AFMT_ENTRY("FLAC", "flac", NULL, "flac\0" ),
+ /* Musepack */
+ [AFMT_MPC] =
+ AFMT_ENTRY("MPC", "mpc", NULL, "mpc\0" ),
+ /* A/52 (aka AC3) audio */
+ [AFMT_A52] =
+ AFMT_ENTRY("AC3", "a52", NULL, "a52\0ac3\0" ),
+ /* WavPack */
+ [AFMT_WAVPACK] =
+ AFMT_ENTRY("WV", "wavpack", "wavpack_enc", "wv\0" ),
+ /* Apple Lossless Audio Codec */
+ [AFMT_ALAC] =
+ AFMT_ENTRY("ALAC", "alac", NULL, "m4a\0" ),
+ /* Advanced Audio Coding in M4A container */
+ [AFMT_AAC] =
+ AFMT_ENTRY("AAC", "aac", NULL, "mp4\0" ),
+ /* Shorten */
+ [AFMT_SHN] =
+ AFMT_ENTRY("SHN", "shorten", NULL, "shn\0" ),
+ /* SID File Format */
+ [AFMT_SID] =
+ AFMT_ENTRY("SID", "sid", NULL, "sid\0" ),
+ /* ADX File Format */
+ [AFMT_ADX] =
+ AFMT_ENTRY("ADX", "adx", NULL, "adx\0" ),
+#endif
+};
+
+#if CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING)
+/* get REC_FORMAT_* corresponding AFMT_* */
+const int rec_format_afmt[REC_NUM_FORMATS] =
+{
+ /* give AFMT_UNKNOWN by default */
+ [0 ... REC_NUM_FORMATS-1] = AFMT_UNKNOWN,
+ /* add new entries below this line */
+ [REC_FORMAT_MPA_L3] = AFMT_MPA_L3,
+ [REC_FORMAT_WAVPACK] = AFMT_WAVPACK,
+ [REC_FORMAT_PCM_WAV] = AFMT_PCM_WAV,
+};
+
+/* get AFMT_* corresponding REC_FORMAT_* */
+const int afmt_rec_format[AFMT_NUM_CODECS] =
+{
+ /* give -1 by default */
+ [0 ... AFMT_NUM_CODECS-1] = -1,
+ /* add new entries below this line */
+ [AFMT_MPA_L3] = REC_FORMAT_MPA_L3,
+ [AFMT_WAVPACK] = REC_FORMAT_WAVPACK,
+ [AFMT_PCM_WAV] = REC_FORMAT_PCM_WAV,
+};
+#endif /* CONFIG_CODEC == SWCODEC && defined (HAVE_RECORDING) */
+/****/
+
#define UNSYNC(b0,b1,b2,b3) (((long)(b0 & 0x7F) << (3*7)) | \
((long)(b1 & 0x7F) << (2*7)) | \
((long)(b2 & 0x7F) << (1*7)) | \
@@ -85,61 +168,6 @@ static const char* const genres[] = {
"Synthpop"
};
-/* database of audio formats */
-const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
-{
- /* Unknown file format */
- AFMT_ENTRY("???", NULL, NULL, NULL ),
- /* MPEG Audio layer 1 */
- AFMT_ENTRY("MP1", "mpa.codec", NULL, NULL ),
- /* MPEG Audio layer 2 */
- AFMT_ENTRY("MP2", "mpa.codec", NULL, NULL ),
- /* MPEG Audio layer 3 */
- AFMT_ENTRY("MP3", "mpa.codec", "mp3_enc.codec", ".mp3"),
-#if CONFIG_CODEC == SWCODEC
- /* Uncompressed PCM in a WAV file */
- AFMT_ENTRY("WAV", "wav.codec", "wav_enc.codec", ".wav"),
- /* Ogg Vorbis */
- AFMT_ENTRY("Ogg", "vorbis.codec", NULL, NULL ),
- /* FLAC */
- AFMT_ENTRY("FLAC", "flac.codec", NULL, NULL ),
- /* Musepack */
- AFMT_ENTRY("MPC", "mpc.codec", NULL, NULL ),
- /* A/52 (aka AC3) audio */
- AFMT_ENTRY("AC3", "a52.codec", NULL, NULL ),
- /* WavPack */
- AFMT_ENTRY("WV", "wavpack.codec", "wavpack_enc.codec", ".wv" ),
- /* Apple Lossless Audio Codec */
- AFMT_ENTRY("ALAC", "alac.codec", NULL, NULL ),
- /* Advanced Audio Coding in M4A container */
- AFMT_ENTRY("AAC", "aac.codec", NULL, NULL ),
- /* Shorten */
- AFMT_ENTRY("SHN", "shorten.codec", NULL, NULL ),
- /* Audio Interchange File Format */
- AFMT_ENTRY("AIFF", "aiff.codec", NULL, NULL ),
- /* SID File Format */
- AFMT_ENTRY("SID", "sid.codec", NULL, NULL ),
- /* ADX File Format */
- AFMT_ENTRY("ADX", "adx.codec", NULL, NULL ),
-#endif
-};
-
-#if CONFIG_CODEC == SWCODEC
-/* recording quality to AFMT_* */
-const int rec_quality_info_afmt[9] =
-{
- AFMT_MPA_L3, /* MPEG L3 64 kBit/s */
- AFMT_MPA_L3, /* MPEG L3 96 kBit/s */
- AFMT_MPA_L3, /* MPEG L3 128 kBit/s */
- AFMT_MPA_L3, /* MPEG L3 160 kBit/s */
- AFMT_MPA_L3, /* MPEG L3 192 kBit/s */
- AFMT_MPA_L3, /* MPEG L3 224 kBit/s */
- AFMT_MPA_L3, /* MPEG L3 320 kBit/s */
- AFMT_WAVPACK, /* WavPack 909 kBit/s */
- AFMT_PCM_WAV, /* PCM Wav 1411 kBit/s */
-};
-#endif /* SWCODEC */
-
char* id3_get_genre(const struct mp3entry* id3)
{
if( id3->genre_string )
diff --git a/firmware/mpeg.c b/firmware/mpeg.c
index ce1d995461..bb438a3ab4 100644
--- a/firmware/mpeg.c
+++ b/firmware/mpeg.c
@@ -2453,34 +2453,32 @@ static void stop_recording(void)
resume_recording();
}
-void audio_set_recording_options(int frequency, int quality,
- int source, int channel_mode,
- bool editable, int prerecord_time)
+void audio_set_recording_options(struct audio_recording_options *options)
{
bool is_mpeg1;
- is_mpeg1 = (frequency < 3)?true:false;
+ is_mpeg1 = (options->rec_frequency < 3)?true:false;
rec_version_index = is_mpeg1?3:2;
- rec_frequency_index = frequency % 3;
+ rec_frequency_index = options->rec_frequency % 3;
- shadow_encoder_control = (quality << 17) |
+ shadow_encoder_control = (options->rec_quality << 17) |
(rec_frequency_index << 10) |
((is_mpeg1?1:0) << 9) |
- (((channel_mode * 2 + 1) & 3) << 6) |
+ (((options->rec_channels * 2 + 1) & 3) << 6) |
(1 << 5) /* MS-stereo */ |
(1 << 2) /* Is an original */;
mas_writemem(MAS_BANK_D0, MAS_D0_ENCODER_CONTROL, &shadow_encoder_control,1);
DEBUGF("mas_writemem(MAS_BANK_D0, ENCODER_CONTROL, %x)\n", shadow_encoder_control);
- shadow_soft_mute = editable?4:0;
+ shadow_soft_mute = options->rec_editable?4:0;
mas_writemem(MAS_BANK_D0, MAS_D0_SOFT_MUTE, &shadow_soft_mute,1);
DEBUGF("mas_writemem(MAS_BANK_D0, SOFT_MUTE, %x)\n", shadow_soft_mute);
shadow_io_control_main = ((1 << 10) | /* Monitoring ON */
- ((source < 2)?1:2) << 8) | /* Input select */
+ ((options->rec_source < 2)?1:2) << 8) | /* Input select */
(1 << 5) | /* SDO strobe invert */
((is_mpeg1?0:1) << 3) |
(1 << 2) | /* Inverted SIBC clock signal */
@@ -2489,7 +2487,7 @@ void audio_set_recording_options(int frequency, int quality,
DEBUGF("mas_writemem(MAS_BANK_D0, IO_CONTROL_MAIN, %x)\n", shadow_io_control_main);
- if(source == AUDIO_SRC_MIC)
+ if(options->rec_source == AUDIO_SRC_MIC)
{
/* Copy left channel to right (mono mode) */
mas_codec_writereg(8, 0x8000);
@@ -2500,7 +2498,7 @@ void audio_set_recording_options(int frequency, int quality,
mas_codec_writereg(8, 0);
}
- prerecording_max_seconds = prerecord_time;
+ prerecording_max_seconds = options->rec_prerecord_time;
if(prerecording_max_seconds)
{
prerecording = true;
diff --git a/firmware/pcm_playback.c b/firmware/pcm_playback.c
index cd14f123d1..b7ea96f3d3 100644
--- a/firmware/pcm_playback.c
+++ b/firmware/pcm_playback.c
@@ -16,259 +16,86 @@
* KIND, either express or implied.
*
****************************************************************************/
-#include <stdbool.h>
-#include "config.h"
-#include "debug.h"
-#include "panic.h"
-#include <kernel.h>
-#include "cpu.h"
-#include "i2c.h"
-#if defined(HAVE_UDA1380)
-#include "uda1380.h"
-#elif defined(HAVE_WM8975)
+#include "system.h"
+#include "kernel.h"
+#include "logf.h"
+#include "audio.h"
+#if defined(HAVE_WM8975)
#include "wm8975.h"
#elif defined(HAVE_WM8758)
#include "wm8758.h"
-#elif defined(HAVE_TLV320)
-#include "tlv320.h"
#elif defined(HAVE_WM8731) || defined(HAVE_WM8721)
#include "wm8731l.h"
#elif CONFIG_CPU == PNX0101
+#include "string.h"
#include "pnx0101.h"
#endif
-#include "system.h"
-#include "logf.h"
-#include <stdio.h>
-#include <string.h>
-#include <stdarg.h>
-#include "pcm_playback.h"
-#include "lcd.h"
-#include "button.h"
-#include "file.h"
-#include "buffer.h"
-#include "sprintf.h"
-#include "button.h"
-#include <string.h>
-
-static bool pcm_playing;
-static bool pcm_paused;
+/**
+ * APIs implemented in the target-specific portion:
+ * Public -
+ * pcm_init
+ * pcm_get_bytes_waiting
+ * pcm_calculate_peaks
+ * Semi-private -
+ * pcm_play_dma_start
+ * pcm_play_dma_stop
+ * pcm_play_pause_pause
+ * pcm_play_pause_unpause
+ */
+
+/** These items may be implemented target specifically or need to
+ be shared semi-privately **/
/* the registered callback function to ask for more mp3 data */
-static void (*callback_for_more)(unsigned char**, size_t*) IDATA_ATTR = NULL;
+pcm_more_callback_type pcm_callback_for_more = NULL;
+bool pcm_playing = false;
+bool pcm_paused = false;
+
+void pcm_play_dma_start(const void *addr, size_t size);
+void pcm_play_dma_stop(void);
+void pcm_play_pause_pause(void);
+void pcm_play_pause_unpause(void);
+
+/** Functions that require targeted implementation **/
+
+#ifndef CPU_COLDFIRE
#if (CONFIG_CPU == S3C2440)
/* TODO: Implement for Gigabeat
For now, just implement some dummy functions.
*/
-
void pcm_init(void)
{
-
}
-static void dma_start(const void *addr, size_t size)
+void pcm_play_dma_start(const void *addr, size_t size)
{
(void)addr;
(void)size;
}
-void pcm_set_frequency(unsigned int frequency)
-{
- (void)frequency;
-}
-
-void pcm_play_stop(void)
+void pcm_play_dma_stop(void)
{
}
-size_t pcm_get_bytes_waiting(void)
+void pcm_play_pause_pause(void)
{
- return 0;
}
-#else
-#ifdef CPU_COLDFIRE
-#ifdef HAVE_SPDIF_OUT
-#define EBU_DEFPARM ((7 << 12) | (3 << 8) | (1 << 5) | (5 << 2))
-#endif
-#define IIS_DEFPARM(freq) ((freq << 12) | 0x300 | 4 << 2)
-#define IIS_RESET 0x800
-
-#ifdef IAUDIO_X5
-#define SET_IIS_CONFIG(x) IIS1CONFIG = (x);
-#else
-#define SET_IIS_CONFIG(x) IIS2CONFIG = (x);
-#endif
-
-static int pcm_freq = 0x6; /* 44.1 is default */
-
-int peak_left = 0, peak_right = 0;
-
-/* Set up the DMA transfer that kicks in when the audio FIFO gets empty */
-static void dma_start(const void *addr, size_t size)
+void pcm_play_pause_unpause(void)
{
- pcm_playing = true;
-
- addr = (void *)((unsigned long)addr & ~3); /* Align data */
- size &= ~3; /* Size must be multiple of 4 */
-
- /* Reset the audio FIFO */
-#ifdef HAVE_SPDIF_OUT
- EBU1CONFIG = IIS_RESET | EBU_DEFPARM;
-#endif
-
- /* Set up DMA transfer */
- SAR0 = (unsigned long)addr; /* Source address */
- DAR0 = (unsigned long)&PDOR3; /* Destination address */
- BCR0 = size; /* Bytes to transfer */
-
- /* Enable the FIFO and force one write to it */
- SET_IIS_CONFIG(IIS_DEFPARM(pcm_freq));
- /* Also send the audio to S/PDIF */
-#ifdef HAVE_SPDIF_OUT
- EBU1CONFIG = EBU_DEFPARM;
-#endif
- DCR0 = DMA_INT | DMA_EEXT | DMA_CS | DMA_AA | DMA_SINC | (3 << 20) | DMA_START;
}
-/* Stops the DMA transfer and interrupt */
-static void dma_stop(void)
-{
- pcm_playing = false;
-
- DCR0 = 0;
- DSR0 = 1;
- /* Reset the FIFO */
- SET_IIS_CONFIG(IIS_RESET | IIS_DEFPARM(pcm_freq));
-#ifdef HAVE_SPDIF_OUT
- EBU1CONFIG = IIS_RESET | EBU_DEFPARM;
-#endif
-}
-
-/* sets frequency of input to DAC */
void pcm_set_frequency(unsigned int frequency)
{
- switch(frequency)
- {
- case 11025:
- pcm_freq = 0x2;
-#ifdef HAVE_UDA1380
- uda1380_set_nsorder(3);
-#endif
- break;
- case 22050:
- pcm_freq = 0x4;
-#ifdef HAVE_UDA1380
- uda1380_set_nsorder(3);
-#endif
- break;
- case 44100:
- default:
- pcm_freq = 0x6;
-#ifdef HAVE_UDA1380
- uda1380_set_nsorder(5);
-#endif
- break;
- }
+ (void)frequency;
}
size_t pcm_get_bytes_waiting(void)
{
- return (BCR0 & 0xffffff);
-}
-
-/* DMA0 Interrupt is called when the DMA has finished transfering a chunk */
-void DMA0(void) __attribute__ ((interrupt_handler, section(".icode")));
-void DMA0(void)
-{
- int res = DSR0;
-
- DSR0 = 1; /* Clear interrupt */
- DCR0 &= ~DMA_EEXT;
-
- /* Stop on error */
- if(res & 0x70)
- {
- dma_stop();
- logf("DMA Error:0x%04x", res);
- }
- else
- {
- size_t next_size;
- unsigned char *next_start;
- {
- void (*get_more)(unsigned char**, size_t*) = callback_for_more;
- if (get_more)
- get_more(&next_start, &next_size);
- else
- {
- next_size = 0;
- next_start = NULL;
- }
- }
- if(next_size)
- {
- SAR0 = (unsigned long)next_start; /* Source address */
- BCR0 = next_size; /* Bytes to transfer */
- DCR0 |= DMA_EEXT;
- }
- else
- {
- /* Finished playing */
- dma_stop();
- logf("DMA No Data:0x%04x", res);
- }
- }
-
- IPR |= (1<<14); /* Clear pending interrupt request */
-}
-
-void pcm_init(void)
-{
- pcm_playing = false;
- pcm_paused = false;
-
- MPARK = 0x81; /* PARK[1,0]=10 + BCR24BIT */
- DIVR0 = 54; /* DMA0 is mapped into vector 54 in system.c */
- DMAROUTE = (DMAROUTE & 0xffffff00) | DMA0_REQ_AUDIO_1;
- DMACONFIG = 1; /* DMA0Req = PDOR3 */
-
- /* Reset the audio FIFO */
- SET_IIS_CONFIG(IIS_RESET);
-
- /* Enable interrupt at level 7, priority 0 */
- ICR6 = 0x1c;
- IMR &= ~(1<<14); /* bit 14 is DMA0 */
-
- pcm_set_frequency(44100);
-
- /* Prevent pops (resets DAC to zero point) */
- SET_IIS_CONFIG(IIS_DEFPARM(pcm_freq) | IIS_RESET);
-
-#if defined(HAVE_UDA1380)
- /* Initialize default register values. */
- uda1380_init();
-
- /* Sleep a while so the power can stabilize (especially a long
- delay is needed for the line out connector). */
- sleep(HZ);
-
- /* Power on FSDAC and HP amp. */
- uda1380_enable_output(true);
-
- /* Unmute the master channel (DAC should be at zero point now). */
- uda1380_mute(false);
-
-#elif defined(HAVE_TLV320)
- tlv320_init();
- sleep(HZ/4);
- tlv320_mute(false);
-#endif
-
- /* Call dma_stop to initialize everything. */
- dma_stop();
+ return 0;
}
#elif defined(HAVE_WM8975) || defined(HAVE_WM8758) \
@@ -286,14 +113,14 @@ void pcm_init(void)
#define FIFO_FREE_COUNT 4 /* TODO: make this sensible */
#endif
-static int pcm_freq = 44100; /* 44.1 is default */
+static int pcm_freq = HW_SAMPR_DEFAULT; /* 44.1 is default */
/* NOTE: The order of these two variables is important if you use the iPod
assembler optimised fiq handler, so don't change it. */
unsigned short* p IBSS_ATTR;
size_t p_size IBSS_ATTR;
-static void dma_start(const void *addr, size_t size)
+void pcm_play_dma_start(const void *addr, size_t size)
{
p=(unsigned short*)addr;
p_size=size;
@@ -341,7 +168,7 @@ static void dma_start(const void *addr, size_t size)
}
/* Stops the DMA transfer and interrupt */
-static void dma_stop(void)
+void pcm_play_dma_stop(void)
{
pcm_playing = false;
@@ -365,9 +192,58 @@ static void dma_stop(void)
disable_fiq();
}
+void pcm_play_pause_pause(void)
+{
+#if CONFIG_CPU == PP5020
+ /* Disable the interrupt */
+ IISCONFIG &= ~0x2;
+ /* Disable playback FIFO */
+ IISCONFIG &= ~0x20000000;
+#elif CONFIG_CPU == PP5002
+ /* Disable the interrupt */
+ IISFIFO_CFG &= ~(1<<9);
+ /* Disable playback FIFO */
+ IISCONFIG &= ~0x4;
+#endif
+ disable_fiq();
+}
+
+void pcm_play_pause_unpause(void)
+{
+ /* Enable the FIFO and fill it */
+
+ enable_fiq();
+
+ /* Enable playback FIFO */
+#if CONFIG_CPU == PP5020
+ IISCONFIG |= 0x20000000;
+#elif CONFIG_CPU == PP5002
+ IISCONFIG |= 0x4;
+#endif
+
+ /* Fill the FIFO - we assume there are enough bytes in the
+ pcm buffer to fill the 32-byte FIFO. */
+ while (p_size > 0) {
+ if (FIFO_FREE_COUNT < 2) {
+ /* Enable interrupt */
+#if CONFIG_CPU == PP5020
+ IISCONFIG |= 0x2;
+#elif CONFIG_CPU == PP5002
+ IISFIFO_CFG |= (1<<9);
+#endif
+ return;
+ }
+
+ IISFIFO_WR = (*(p++))<<16;
+ IISFIFO_WR = (*(p++))<<16;
+ p_size-=4;
+ }
+}
+
void pcm_set_frequency(unsigned int frequency)
{
- pcm_freq=frequency;
+ (void)frequency;
+ pcm_freq = HW_SAMPR_DEFAULT;
}
size_t pcm_get_bytes_waiting(void)
@@ -378,8 +254,8 @@ size_t pcm_get_bytes_waiting(void)
/* ASM optimised FIQ handler. GCC fails to make use of the fact that FIQ mode
has registers r8-r14 banked, and so does not need to be saved. This routine
uses only these registers, and so will never touch the stack unless it
- actually needs to do so when calling callback_for_more. C version is still
- included below for reference.
+ actually needs to do so when calling pcm_callback_for_more. C version is
+ still included below for reference.
*/
#if CONFIG_CPU == PP5020 || CONFIG_CPU == PP5002
void fiq(void) ICODE_ATTR __attribute__((naked));
@@ -433,10 +309,10 @@ void fiq(void)
"add r1, r11, #4 \n\t" /* r1 = &p_size */
"str r9, [r0] \n\t" /* save internal copies of variables back */
"str r8, [r1] \n\t"
- "ldr r2, =callback_for_more\n\t"
+ "ldr r2, =pcm_callback_for_more\n\t"
"ldr r2, [r2] \n\t" /* get callback address */
"cmp r2, #0 \n\t" /* check for null pointer */
- "movne lr, pc \n\t" /* call callback_for_more */
+ "movne lr, pc \n\t" /* call pcm_callback_for_more */
"bxne r2 \n\t"
"ldmia sp!, { r0-r3, r12, lr}\n\t"
"ldr r8, [r11, #4] \n\t" /* reload p_size and p */
@@ -477,7 +353,7 @@ void fiq(void)
"b .exit \n\t"
);
}
-#else
+#else /* !(CONFIG_CPU == PP5020 || CONFIG_CPU == PP5002) */
void fiq(void) ICODE_ATTR __attribute__ ((interrupt ("FIQ")));
void fiq(void)
{
@@ -507,20 +383,21 @@ void fiq(void)
}
/* p is empty, get some more data */
- if (callback_for_more) {
- callback_for_more((unsigned char**)&p,&p_size);
+ if (pcm_callback_for_more) {
+ pcm_callback_for_more((unsigned char**)&p,&p_size);
}
} while (p_size);
/* No more data, so disable the FIFO/FIQ */
- dma_stop();
+ pcm_play_dma_stop();
}
-#endif
+#endif /* CONFIG_CPU == PP5020 || CONFIG_CPU == PP5002 */
void pcm_init(void)
{
pcm_playing = false;
pcm_paused = false;
+ pcm_callback_for_more = NULL;
/* Initialize default register values. */
wmcodec_init();
@@ -531,8 +408,8 @@ void pcm_init(void)
/* Unmute the master channel (DAC should be at zero point now). */
wmcodec_mute(false);
- /* Call dma_stop to initialize everything. */
- dma_stop();
+ /* Call pcm_play_dma_stop to initialize everything. */
+ pcm_play_dma_stop();
}
#elif (CONFIG_CPU == PNX0101)
@@ -542,12 +419,16 @@ void pcm_init(void)
short __attribute__((section(".dmabuf"))) dma_buf_left[DMA_BUF_SAMPLES];
short __attribute__((section(".dmabuf"))) dma_buf_right[DMA_BUF_SAMPLES];
-static int pcm_freq = 44100; /* 44.1 is default */
+static int pcm_freq = HW_SAMPR_DEFAULT; /* 44.1 is default */
unsigned short* p IBSS_ATTR;
size_t p_size IBSS_ATTR;
-static void dma_start(const void *addr, size_t size)
+void pcm_init(void)
+{
+}
+
+void pcm_play_dma_start(const void *addr, size_t size)
{
p = (unsigned short*)addr;
p_size = size;
@@ -555,11 +436,19 @@ static void dma_start(const void *addr, size_t size)
pcm_playing = true;
}
-static void dma_stop(void)
+void pcm_play_dma_stop(void)
{
pcm_playing = false;
}
+void pcm_play_pause_pause(void)
+{
+}
+
+void pcm_play_pause_unpause(void)
+{
+}
+
static inline void fill_dma_buf(int offset)
{
short *l, *r, *lend;
@@ -611,8 +500,8 @@ static inline void fill_dma_buf(int offset)
p = tmp_p;
if (l >= lend)
return;
- else if (callback_for_more)
- callback_for_more((unsigned char**)&p,
+ else if (pcm_callback_for_more)
+ pcm_callback_for_more((unsigned char**)&p,
&p_size);
}
while (p_size);
@@ -647,9 +536,10 @@ unsigned long physical_address(void *p)
void pcm_init(void)
{
int i;
- callback_for_more = NULL;
+
pcm_playing = false;
pcm_paused = false;
+ pcm_callback_for_more = NULL;
memset(dma_buf_left, 0, sizeof(dma_buf_left));
memset(dma_buf_right, 0, sizeof(dma_buf_right));
@@ -691,271 +581,37 @@ void pcm_init(void)
void pcm_set_frequency(unsigned int frequency)
{
- pcm_freq=frequency;
+ (void)frequency;
+ pcm_freq = HW_SAMPR_DEFAULT;
}
size_t pcm_get_bytes_waiting(void)
{
return p_size;
}
-#endif
+#endif /* CONFIG_CPU == */
-void pcm_play_stop(void)
+/* dummy functions for those not actually supporting all this yet */
+void pcm_apply_settings(bool reset)
{
- if (pcm_playing) {
- dma_stop();
- }
+ (void)reset;
}
-#endif
-
-void pcm_play_data(void (*get_more)(unsigned char** start, size_t* size),
- unsigned char* start, size_t size)
+void pcm_set_monitor(int monitor)
{
- callback_for_more = get_more;
-
- if (!(start && size))
- {
- if (get_more)
- get_more(&start, &size);
- else
- return;
- }
- if (start && size)
- {
- dma_start(start, size);
- if (pcm_paused) {
- pcm_paused = false;
- pcm_play_pause(false);
- }
- }
+ (void)monitor;
}
+/** **/
void pcm_mute(bool mute)
{
-#ifdef HAVE_UDA1380
- uda1380_mute(mute);
-#elif defined(HAVE_WM8975) || defined(HAVE_WM8758) \
+#if defined(HAVE_WM8975) || defined(HAVE_WM8758) \
|| defined(HAVE_WM8731) || defined(HAVE_WM8721)
wmcodec_mute(mute);
-#elif defined(HAVE_TLV320)
- tlv320_mute(mute);
#endif
if (mute)
sleep(HZ/16);
}
-void pcm_play_pause(bool play)
-{
- bool needs_change = pcm_paused == play;
-
- /* This needs to be done ahead of the rest to prevent infinite
- * recursion from dma_start */
- pcm_paused = !play;
- if (pcm_playing && needs_change) {
- if(play) {
- if (pcm_get_bytes_waiting()) {
- logf("unpause");
-
-#ifdef CPU_COLDFIRE
- /* Enable the FIFO and force one write to it */
- SET_IIS_CONFIG(IIS_DEFPARM(pcm_freq));
-#ifdef HAVE_SPDIF_OUT
- EBU1CONFIG = EBU_DEFPARM;
-#endif
- DCR0 |= DMA_EEXT | DMA_START;
-#elif defined(HAVE_WM8975) || defined(HAVE_WM8758) \
- || defined(HAVE_WM8731) || defined(HAVE_WM8721)
- /* Enable the FIFO and fill it */
-
- enable_fiq();
-
- /* Enable playback FIFO */
-#if CONFIG_CPU == PP5020
- IISCONFIG |= 0x20000000;
-#elif CONFIG_CPU == PP5002
- IISCONFIG |= 0x4;
-#endif
-
- /* Fill the FIFO - we assume there are enough bytes in the
- pcm buffer to fill the 32-byte FIFO. */
- while (p_size > 0) {
- if (FIFO_FREE_COUNT < 2) {
- /* Enable interrupt */
-#if CONFIG_CPU == PP5020
- IISCONFIG |= 0x2;
-#elif CONFIG_CPU == PP5002
- IISFIFO_CFG |= (1<<9);
-#endif
- return;
- }
-
- IISFIFO_WR = (*(p++))<<16;
- IISFIFO_WR = (*(p++))<<16;
- p_size-=4;
- }
-#elif (CONFIG_CPU == PNX0101 || CONFIG_CPU == S3C2440) /* End wmcodecs */
- /* nothing yet */
-#endif
- } else {
-#if (CONFIG_CPU != PNX0101 && CONFIG_CPU != S3C2440)
- size_t next_size;
- unsigned char *next_start;
- void (*get_more)(unsigned char**, size_t*) = callback_for_more;
- logf("unpause, no data waiting");
- if (get_more)
- get_more(&next_start, &next_size);
- if (next_start && next_size)
- dma_start(next_start, next_size);
- else
- {
- dma_stop();
- logf("unpause attempted, no data");
- }
-#endif
- }
- } else {
- logf("pause");
-
-#ifdef CPU_COLDFIRE
- /* Disable DMA peripheral request. */
- DCR0 &= ~DMA_EEXT;
- SET_IIS_CONFIG(IIS_RESET | IIS_DEFPARM(pcm_freq));
-#ifdef HAVE_SPDIF_OUT
- EBU1CONFIG = IIS_RESET | EBU_DEFPARM;
-#endif
-#elif defined(HAVE_WM8975) || defined(HAVE_WM8758) \
- || defined(HAVE_WM8731) || defined(HAVE_WM8721)
-#if CONFIG_CPU == PP5020
- /* Disable the interrupt */
- IISCONFIG &= ~0x2;
- /* Disable playback FIFO */
- IISCONFIG &= ~0x20000000;
-#elif CONFIG_CPU == PP5002
- /* Disable the interrupt */
- IISFIFO_CFG &= ~(1<<9);
- /* Disable playback FIFO */
- IISCONFIG &= ~0x4;
-#endif
-
- disable_fiq();
-#elif (CONFIG_CPU == PNX0101 || CONFIG_CPU == S3C2440) /* End wmcodecs */
- /* nothing yet */
-#endif
- }
- } /* pcm_playing && needs_change */
-}
-
-bool pcm_is_playing(void) {
- return pcm_playing;
-}
-
-bool pcm_is_paused(void) {
- return pcm_paused;
-}
-
-
-#if defined(CPU_COLDFIRE)
-/* Peaks ahead in the DMA buffer based upon the calling period to
- attempt to compensate for the delay. Keeps a moving average of
- length four. */
-void pcm_calculate_peaks(int *left, int *right)
-{
- unsigned long samples;
- unsigned long *addr, *end;
- long peak_p, peak_n;
-
- static unsigned long last_peak_tick = 0;
- static unsigned long frame_period = 0;
-
- /* Throttled peak ahead based on calling period */
- unsigned long period = current_tick - last_peak_tick;
-
- /* Keep reasonable limits on period */
- if (period < 1)
- period = 1;
- else if (period > HZ/5)
- period = HZ/5;
-
- frame_period = (3*frame_period + period) >> 2;
-
- last_peak_tick = current_tick;
-
- if (!pcm_playing || pcm_paused)
- {
- peak_left = peak_right = 0;
- goto peak_done;
- }
-
- samples = (BCR0 & 0xffffff) >> 2;
- addr = (long *)(SAR0 & ~3);
- samples = MIN(frame_period*44100/HZ, samples);
- end = addr + samples;
- peak_p = peak_n = 0;
-
- if (left && right)
- {
- if (samples > 0)
- {
- long peak_rp = 0, peak_rn = 0;
-
- do
- {
- long value = *addr;
- long ch;
-
- ch = value >> 16;
- if (ch > peak_p) peak_p = ch;
- else if (ch < peak_n) peak_n = ch;
-
- ch = (short)value;
- if (ch > peak_rp) peak_rp = ch;
- else if (ch < peak_rn) peak_rn = ch;
-
- addr += 4;
- }
- while (addr < end);
-
- peak_left = MAX(peak_p, -peak_n);
- peak_right = MAX(peak_rp, -peak_rn);
- }
- }
- else if (left || right)
- {
- if (samples > 0)
- {
- if (left)
- {
- /* Put left channel in low word */
- addr = (long *)((short *)addr - 1);
- end = (long *)((short *)end - 1);
- }
-
- do
- {
- long value = *(short *)addr;
-
- if (value > peak_p) peak_p = value;
- else if (value < peak_n) peak_n = value;
-
- addr += 4;
- }
- while (addr < end);
-
- if (left)
- peak_left = MAX(peak_p, -peak_n);
- else
- peak_right = MAX(peak_p, -peak_n);
- }
- }
-
-peak_done:
- if (left)
- *left = peak_left;
-
- if (right)
- *right = peak_right;
-}
-#else
/*
* This function goes directly into the DMA buffer to calculate the left and
* right peak values. To avoid missing peaks it tries to look forward two full
@@ -1037,4 +693,94 @@ void pcm_calculate_peaks(int *left, int *right)
}
#endif
}
+
#endif /* CPU_COLDFIRE */
+
+/****************************************************************************
+ * Functions that do not require targeted implementation but only a targeted
+ * interface
+ */
+
+/* Common code to pcm_play_data and pcm_play_pause
+ Returns true if DMA playback was started, else false. */
+bool pcm_play_data_start(pcm_more_callback_type get_more,
+ unsigned char *start, size_t size)
+{
+ if (!(start && size))
+ {
+ size = 0;
+ if (get_more)
+ get_more(&start, &size);
+ }
+
+ if (start && size)
+ {
+ pcm_play_dma_start(start, size);
+ return true;
+ }
+
+ return false;
+}
+
+void pcm_play_data(pcm_more_callback_type get_more,
+ unsigned char *start, size_t size)
+{
+ pcm_callback_for_more = get_more;
+
+ if (pcm_play_data_start(get_more, start, size) && pcm_paused)
+ {
+ pcm_paused = false;
+ pcm_play_pause(false);
+ }
+}
+
+void pcm_play_pause(bool play)
+{
+ bool needs_change = pcm_paused == play;
+
+ /* This needs to be done ahead of the rest to prevent infinite
+ recursion from pcm_play_data */
+ pcm_paused = !play;
+
+ if (pcm_playing && needs_change)
+ {
+ if (play)
+ {
+ if (pcm_get_bytes_waiting())
+ {
+ logf("unpause");
+ pcm_play_pause_unpause();
+ }
+ else
+ {
+ logf("unpause, no data waiting");
+ if (!pcm_play_data_start(pcm_callback_for_more, NULL, 0))
+ {
+ pcm_play_dma_stop();
+ logf("unpause attempted, no data");
+ }
+ }
+ }
+ else
+ {
+ logf("pause");
+ pcm_play_pause_pause();
+ }
+ } /* pcm_playing && needs_change */
+}
+
+void pcm_play_stop(void)
+{
+ if (pcm_playing)
+ pcm_play_dma_stop();
+}
+
+bool pcm_is_playing(void)
+{
+ return pcm_playing;
+}
+
+bool pcm_is_paused(void)
+{
+ return pcm_paused;
+}
diff --git a/firmware/pcm_record.c b/firmware/pcm_record.c
index 2785d4b1b1..25f1f1ef64 100644
--- a/firmware/pcm_record.c
+++ b/firmware/pcm_record.c
@@ -16,199 +16,238 @@
* KIND, either express or implied.
*
****************************************************************************/
-
-#include "config.h"
-#include "debug.h"
+#include "system.h"
+#include "kernel.h"
+#include "logf.h"
#include "panic.h"
#include "thread.h"
-
-#include <kernel.h>
-#include <stdio.h>
-#include <string.h>
-#include <stdarg.h>
#include <string.h>
-
-#include "cpu.h"
-#include "i2c.h"
-#include "power.h"
-#ifdef HAVE_UDA1380
+#include "ata.h"
+#include "usb.h"
+#if defined(HAVE_UDA1380)
#include "uda1380.h"
-#endif
-#ifdef HAVE_TLV320
+#include "general.h"
+#elif defined(HAVE_TLV320)
#include "tlv320.h"
#endif
-#include "system.h"
-#include "usb.h"
-
#include "buffer.h"
#include "audio.h"
-#include "button.h"
-#include "file.h"
-#include "sprintf.h"
-#include "logf.h"
-#include "button.h"
-#include "lcd.h"
-#include "lcd-remote.h"
-#include "pcm_playback.h"
#include "sound.h"
#include "id3.h"
-#include "pcm_record.h"
-
-extern int boost_counter; /* used for boost check */
/***************************************************************************/
+/**
+ * APIs implemented in the target tree portion:
+ * Public -
+ * pcm_init_recording
+ * pcm_close_recording
+ * pcm_rec_mux
+ * Semi-private -
+ * pcm_rec_dma_start
+ * pcm_rec_dma_stop
+ */
+
+/** These items may be implemented target specifically or need to
+ be shared semi-privately **/
+
+/* the registered callback function for when more data is available */
+pcm_more_callback_type pcm_callback_more_ready = NULL;
+/* DMA transfer in is currently active */
+bool pcm_recording = false;
+
+/* APIs implemented in the target-specific portion */
+void pcm_rec_dma_start(const void *addr, size_t size);
+void pcm_rec_dma_stop(void);
+
+/** General recording state **/
static bool is_recording; /* We are recording */
static bool is_paused; /* We have paused */
+static bool is_stopping; /* We are currently stopping */
static bool is_error; /* An error has occured */
-static unsigned long num_rec_bytes; /* Num bytes recorded */
-static unsigned long num_file_bytes; /* Num bytes written to current file */
-static int error_count; /* Number of DMA errors */
-static unsigned long num_pcm_samples; /* Num pcm samples written to current file */
-
-static long record_start_time; /* current_tick when recording was started */
-static long pause_start_time; /* current_tick when pause was started */
-static unsigned int sample_rate; /* Sample rate at time of recording start */
-static int rec_source; /* Current recording source */
+/** Stats on encoded data for current file **/
+static size_t num_rec_bytes; /* Num bytes recorded */
+static unsigned long num_rec_samples; /* Number of PCM samples recorded */
-static int wav_file;
-static char recording_filename[MAX_PATH];
+/** Stats on encoded data for all files from start to stop **/
+static unsigned long long accum_rec_bytes; /* total size written to chunks */
+static unsigned long long accum_pcm_samples; /* total pcm count processed */
-static volatile bool init_done, close_done, record_done;
-static volatile bool stop_done, pause_done, resume_done, new_file_done;
-
-static int peak_left, peak_right;
+/* Keeps data about current file and is sent as event data for codec */
+static struct enc_file_event_data rec_fdata IDATA_ATTR =
+{
+ .chunk = NULL,
+ .new_enc_size = 0,
+ .new_num_pcm = 0,
+ .rec_file = -1,
+ .num_pcm_samples = 0
+};
-#ifdef IAUDIO_X5
-#define SET_IIS_PLAY(x) IIS1CONFIG = (x);
-#define SET_IIS_REC(x) IIS1CONFIG = (x);
-#else
-#define SET_IIS_PLAY(x) IIS2CONFIG = (x);
-#define SET_IIS_REC(x) IIS1CONFIG = (x);
-#endif
+/** These apply to current settings **/
+static int rec_source; /* current rec_source setting */
+static int rec_frequency; /* current frequency setting */
+static unsigned long sample_rate; /* Sample rate in HZ */
+static int num_channels; /* Current number of channels */
+static struct encoder_config enc_config; /* Current encoder configuration */
/****************************************************************************
- use 2 circular buffers of same size:
- rec_buffer=DMA output buffer: chunks (8192 Bytes) of raw pcm audio data
+ use 2 circular buffers:
+ pcm_buffer=DMA output buffer: chunks (8192 Bytes) of raw pcm audio data
enc_buffer=encoded audio buffer: storage for encoder output data
Flow:
- 1. when entering recording_screen DMA feeds the ringbuffer rec_buffer
+ 1. when entering recording_screen DMA feeds the ringbuffer pcm_buffer
2. if enough pcm data are available the encoder codec does encoding of pcm
chunks (4-8192 Bytes) into ringbuffer enc_buffer in codec_thread
3. pcmrec_callback detects enc_buffer 'near full' and writes data to disk
- Functions calls:
- 1.main: codec_load_encoder(); start the encoder
- 2.encoder: enc_get_inputs(); get encoder buffsize, mono/stereo, quality
- 3.encoder: enc_set_parameters(); set the encoder parameters (max.chunksize)
- 4.encoder: enc_get_wav_data(); get n bytes of unprocessed pcm data
- 5.encoder: enc_wavbuf_near_empty();if true: reduce cpu_boost
+ Functions calls (basic encoder steps):
+ 1.main: audio_load_encoder(); start the encoder
+ 2.encoder: enc_get_inputs(); get encoder recording settings
+ 3.encoder: enc_set_parameters(); set the encoder parameters
+ 4.encoder: enc_get_pcm_data(); get n bytes of unprocessed pcm data
+ 5.encoder: enc_pcm_buf_near_empty(); if 1: reduce cpu_boost
6.encoder: enc_alloc_chunk(); get a ptr to next enc chunk
7.encoder: <process enc chunk> compress and store data to enc chunk
8.encoder: enc_free_chunk(); inform main about chunk process finished
9.encoder: repeat 4. to 8.
- A.main: enc_set_header_callback(); create the current format header (file)
+ A.pcmrec: enc_events_callback(); called for certain events
****************************************************************************/
-#define NUM_CHUNKS 256 /* Power of 2 */
-#define CHUNK_SIZE 8192 /* Power of 2 */
-#define MAX_FEED_SIZE 20000 /* max pcm size passed to encoder */
-#define CHUNK_MASK (NUM_CHUNKS * CHUNK_SIZE - 1)
-#define WRITE_THRESHOLD (44100 * 5 / enc_samp_per_chunk) /* 5sec */
-#define GET_CHUNK(x) (long*)(&rec_buffer[x])
-#define GET_ENC_CHUNK(x) (long*)(&enc_buffer[enc_chunk_size*(x)])
-
-static int audio_enc_id; /* current encoder id */
-static unsigned char *rec_buffer; /* Circular recording buffer */
-static unsigned char *enc_buffer; /* Circular encoding buffer */
-static unsigned char *enc_head_buffer; /* encoder header buffer */
-static int enc_head_size; /* used size in header buffer */
-static int write_pos; /* Current chunk pos for DMA writing */
-static int read_pos; /* Current chunk pos for encoding */
-static long pre_record_ticks;/* pre-record time expressed in ticks */
-static int enc_wr_index; /* Current encoding chunk write index */
-static int enc_rd_index; /* Current encoding chunk read index */
-static int enc_chunk_size; /* maximum encoder chunk size */
+
+/** buffer parameters where incoming PCM data is placed **/
+#define PCM_NUM_CHUNKS 256 /* Power of 2 */
+#define PCM_CHUNK_SIZE 8192 /* Power of 2 */
+#define PCM_CHUNK_MASK (PCM_NUM_CHUNKS*PCM_CHUNK_SIZE - 1)
+
+#define GET_PCM_CHUNK(offset) ((long *)(pcm_buffer + (offset)))
+#define GET_ENC_CHUNK(index) ENC_CHUNK_HDR(enc_buffer + enc_chunk_size*(index))
+
+#define INC_ENC_INDEX(index) \
+ { if (++index >= enc_num_chunks) index = 0; }
+#define DEC_ENC_INDEX(index) \
+ { if (--index < 0) index = enc_num_chunks - 1; }
+
+static size_t rec_buffer_size; /* size of available buffer */
+static unsigned char *pcm_buffer; /* circular recording buffer */
+static unsigned char *enc_buffer; /* circular encoding buffer */
+static volatile int dma_wr_pos; /* current DMA write pos */
+static int pcm_rd_pos; /* current PCM read pos */
+static volatile bool dma_lock; /* lock DMA write position */
+static unsigned long pre_record_ticks;/* pre-record time in ticks */
+static int enc_wr_index; /* encoder chunk write index */
+static int enc_rd_index; /* encoder chunk read index */
static int enc_num_chunks; /* number of chunks in ringbuffer */
-static int enc_buffer_size; /* encode buffer size */
-static int enc_channels; /* 1=mono 2=stereo */
-static int enc_quality; /* mp3: 64,96,128,160,192,320 kBit */
-static int enc_samp_per_chunk;/* pcm samples per encoder chunk */
+static size_t enc_chunk_size; /* maximum encoder chunk size */
+static size_t enc_data_size; /* maximum data size for encoder */
+static unsigned long enc_sample_rate; /* sample rate used by encoder */
static bool wav_queue_empty; /* all wav chunks processed? */
-static unsigned long avrg_bit_rate; /* average bit rates from chunks */
-static unsigned long curr_bit_rate; /* cumulated bit rates from chunks */
-static unsigned long curr_chunk_cnt; /* number of processed chunks */
-void (*enc_set_header_callback)(void *head_buffer, int head_size,
- int num_pcm_samples, bool is_file_header);
+/** file flushing **/
+static int write_threshold; /* max chunk limit for data flush */
+static int panic_threshold; /* boost thread prio when here */
+static int spinup_time = -1;/* last ata_spinup_time */
+
+/** encoder events **/
+static void (*enc_events_callback)(enum enc_events event, void *data);
+
+/** Path queue for files to write **/
+#define FNQ_MIN_NUM_PATHS 16 /* minimum number of paths to hold */
+static unsigned char *fn_queue; /* pointer to first filename */
+static ssize_t fnq_size; /* capacity of queue in bytes */
+static int fnq_rd_pos; /* current read position */
+static int fnq_wr_pos; /* current write position */
/***************************************************************************/
static struct event_queue pcmrec_queue;
-static long pcmrec_stack[2*DEFAULT_STACK_SIZE/sizeof(long)];
+static long pcmrec_stack[3*DEFAULT_STACK_SIZE/sizeof(long)];
static const char pcmrec_thread_name[] = "pcmrec";
static void pcmrec_thread(void);
-static void pcmrec_dma_start(void);
-static void pcmrec_dma_stop(void);
-static void close_wave(void);
-/* Event IDs */
-#define PCMREC_INIT 1 /* Enable recording */
-#define PCMREC_CLOSE 2
+/* Event values which are also single-bit flags */
+#define PCMREC_INIT 0x00000001 /* enable recording */
+#define PCMREC_CLOSE 0x00000002
-#define PCMREC_START 3 /* Start a new recording */
-#define PCMREC_STOP 4 /* Stop the current recording */
-#define PCMREC_PAUSE 10
-#define PCMREC_RESUME 11
-#define PCMREC_NEW_FILE 12
-#define PCMREC_SET_GAIN 13
+#define PCMREC_START 0x00000004 /* start recording (when stopped) */
+#define PCMREC_STOP 0x00000008 /* stop the current recording */
+#define PCMREC_PAUSE 0x00000010 /* pause the current recording */
+#define PCMREC_RESUME 0x00000020 /* resume the current recording */
+#define PCMREC_NEW_FILE 0x00000040 /* start new file (when recording) */
+#define PCMREC_SET_GAIN 0x00000080
+#define PCMREC_FLUSH_NUM 0x00000100 /* flush a number of files out */
+#define PCMREC_FINISH_STOP 0x00000200 /* finish the stopping recording */
-/*******************************************************************/
-/* Functions that are not executing in the pcmrec_thread first */
-/*******************************************************************/
+/* mask for signaling events */
+static volatile long pcm_thread_event_mask;
-/* Creates pcmrec_thread */
-void pcm_rec_init(void)
+static void pcm_thread_sync_post(long event, void *data)
{
- queue_init(&pcmrec_queue, true);
- create_thread(pcmrec_thread, pcmrec_stack, sizeof(pcmrec_stack),
- pcmrec_thread_name IF_PRIO(, PRIORITY_RECORDING));
-}
+ pcm_thread_event_mask &= ~event;
+ queue_post(&pcmrec_queue, event, data);
+ while(!(event & pcm_thread_event_mask))
+ yield();
+} /* pcm_thread_sync_post */
+static inline void pcm_thread_signal_event(long event)
+{
+ pcm_thread_event_mask |= event;
+} /* pcm_thread_signal_event */
-int audio_get_encoder_id(void)
+static inline void pcm_thread_unsignal_event(long event)
{
- return audio_enc_id;
-}
+ pcm_thread_event_mask &= ~event;
+} /* pcm_thread_unsignal_event */
-/* Initializes recording:
- * - Set up the UDA1380/TLV320 for recording
- * - Prepare for DMA transfers
- */
+static inline bool pcm_thread_event_state(long signaled, long unsignaled)
+{
+ return ((signaled | unsignaled) & pcm_thread_event_mask) == signaled;
+} /* pcm_thread_event_state */
-void audio_init_recording(unsigned int buffer_offset)
+static void pcm_thread_wait_for_stop(void)
{
- (void)buffer_offset;
+ if (is_stopping)
+ {
+ logf("waiting for stop to finish");
+ while (is_stopping)
+ yield();
+ }
+} /* pcm_thread_wait_for_stop */
- init_done = false;
- queue_post(&pcmrec_queue, PCMREC_INIT, 0);
+/*******************************************************************/
+/* Functions that are not executing in the pcmrec_thread first */
+/*******************************************************************/
- while(!init_done)
- sleep_thread(1);
-}
-
-void audio_close_recording(void)
+/* Callback for when more data is ready */
+static void pcm_rec_have_more(unsigned char **data, size_t *size)
{
- close_done = false;
- queue_post(&pcmrec_queue, PCMREC_CLOSE, 0);
+ if (*size != 0)
+ {
+ /* some error condition */
+ if (*size == DMA_REC_ERROR_DMA)
+ {
+ /* Flush recorded data to disk and stop recording */
+ queue_post(&pcmrec_queue, PCMREC_STOP, NULL);
+ return;
+ }
+ /* else try again next transmission */
+ }
+ else if (!dma_lock)
+ {
+ /* advance write position */
+ dma_wr_pos = (dma_wr_pos + PCM_CHUNK_SIZE) & PCM_CHUNK_MASK;
+ }
- while(!close_done)
- sleep_thread(1);
+ *data = (unsigned char *)GET_PCM_CHUNK(dma_wr_pos);
+ *size = PCM_CHUNK_SIZE;
+} /* pcm_rec_have_more */
- audio_remove_encoder();
-}
+/** pcm_rec_* group **/
+void pcm_rec_error_clear(void)
+{
+ is_error = false;
+} /* pcm_rec_error_clear */
unsigned long pcm_rec_status(void)
{
@@ -216,165 +255,223 @@ unsigned long pcm_rec_status(void)
if (is_recording)
ret |= AUDIO_STATUS_RECORD;
+
if (is_paused)
ret |= AUDIO_STATUS_PAUSE;
+
if (is_error)
ret |= AUDIO_STATUS_ERROR;
- if (!is_recording && pre_record_ticks && init_done && !close_done)
+
+ if (!is_recording && pre_record_ticks &&
+ pcm_thread_event_state(PCMREC_INIT, PCMREC_CLOSE))
ret |= AUDIO_STATUS_PRERECORD;
return ret;
-}
+} /* pcm_rec_status */
int pcm_rec_current_bitrate(void)
{
- return avrg_bit_rate;
-}
+ if (accum_pcm_samples == 0)
+ return 0;
-unsigned long audio_recorded_time(void)
+ return (int)(8*accum_rec_bytes*enc_sample_rate / (1000*accum_pcm_samples));
+} /* pcm_rec_current_bitrate */
+
+int pcm_rec_encoder_afmt(void)
{
- if (is_recording)
+ return enc_config.afmt;
+} /* pcm_rec_encoder_afmt */
+
+int pcm_rec_rec_format(void)
{
- if (is_paused)
- return pause_start_time - record_start_time;
- else
- return current_tick - record_start_time;
- }
+ return afmt_rec_format[enc_config.afmt];
+} /* pcm_rec_rec_format */
- return 0;
-}
+unsigned long pcm_rec_sample_rate(void)
+{
+ /* Which is better ?? */
+#if 0
+ return enc_sample_rate;
+#endif
+ return sample_rate;
+} /* audio_get_sample_rate */
-unsigned long audio_num_recorded_bytes(void)
+/**
+ * Creates pcmrec_thread
+ */
+void pcm_rec_init(void)
{
- if (is_recording)
- return num_rec_bytes;
+ queue_init(&pcmrec_queue, true);
+ create_thread(pcmrec_thread, pcmrec_stack, sizeof(pcmrec_stack),
+ pcmrec_thread_name, PRIORITY_RECORDING);
+} /* pcm_rec_init */
+/** audio_* group **/
+
+void audio_init_recording(unsigned int buffer_offset)
+{
+ (void)buffer_offset;
+ pcm_thread_wait_for_stop();
+ pcm_thread_sync_post(PCMREC_INIT, NULL);
+} /* audio_init_recording */
+
+void audio_close_recording(void)
+{
+ pcm_thread_wait_for_stop();
+ pcm_thread_sync_post(PCMREC_CLOSE, NULL);
+ /* reset pcm to defaults (playback only) */
+ pcm_set_frequency(-1);
+ pcm_set_monitor(-1);
+ pcm_set_rec_source(-1);
+#ifdef HAVE_TLV320
+ /* tlv320 screeches if left at 88.2 with no inputs */
+ pcm_apply_settings(true);
+#endif
+ audio_remove_encoder();
+} /* audio_close_recording */
+
+unsigned long audio_recorded_time(void)
+{
+ if (!is_recording || enc_sample_rate == 0)
return 0;
-}
-#ifdef HAVE_SPDIF_IN
-/* Only the last six of these are standard rates, but all sample rates are
- * possible, so we support some other common ones as well.
- */
-static unsigned long spdif_sample_rates[] = {
- 8000, 11025, 12000, 16000, 22050, 24000,
- 32000, 44100, 48000, 64000, 88200, 96000
-};
+ /* return actual recorded time a la encoded data even if encoder rate
+ doesn't match the pcm rate */
+ return (long)(HZ*(unsigned long long)num_rec_samples / enc_sample_rate);
+} /* audio_recorded_time */
-/* Return SPDIF sample rate. Since we base our reading on the actual SPDIF
- * sample rate (which might be a bit inaccurate), we round off to the closest
- * sample rate that is supported by SPDIF.
- */
-unsigned long audio_get_spdif_sample_rate(void)
+unsigned long audio_num_recorded_bytes(void)
{
- int i = 0;
- unsigned long measured_rate;
- const int upper_bound = sizeof(spdif_sample_rates)/sizeof(long) - 1;
+ if (!is_recording)
+ return 0;
+
+ return num_rec_bytes;
+} /* audio_num_recorded_bytes */
+#ifdef HAVE_SPDIF_IN
+/* Return current SPDIF sample rate */
+static unsigned long measure_spdif_sample_rate(void)
+{
/* The following formula is specified in MCF5249 user's manual section
- * 17.6.1. The 3*(1 << 13) part will need changing if the setup of the
- * PHASECONFIG register is ever changed. The 128 divide is because of the
- * fact that the SPDIF clock is the sample rate times 128.
+ * 17.6.1. The 128 divide is because of the fact that the SPDIF clock is
+ * the sample rate times 128. Keep "3*(1 << 13)" part in sync with
+ * PHASECONFIG setup in pcm_init_recording in pcm-coldfire.c.
*/
- measured_rate = (unsigned long)((unsigned long long)FREQMEAS*CPU_FREQ/
+ return (unsigned long)((unsigned long long)FREQMEAS*CPU_FREQ /
((1 << 15)*3*(1 << 13))/128);
- /* Find which SPDIF sample rate we're closest to. */
- while (spdif_sample_rates[i] < measured_rate && i < upper_bound) ++i;
- if (i > 0 && i < upper_bound)
- {
- long diff1 = measured_rate - spdif_sample_rates[i - 1];
- long diff2 = spdif_sample_rates[i] - measured_rate;
+} /* measure_spdif_sample_rate */
- if (diff2 > diff1) --i;
- }
- return i;
-}
-#endif
+/**
+ * Return SPDIF sample rate index in audio_master_sampr_list. Since we base
+ * our reading on the actual SPDIF sample rate (which might be a bit
+ * inaccurate), we round off to the closest sample rate that is supported by
+ * SPDIF.
+ */
+int audio_get_spdif_sample_rate(void)
+{
+ unsigned long measured_rate = measure_spdif_sample_rate();
+ /* Find which SPDIF sample rate we're closest to. */
+ return round_value_to_list32(measured_rate, audio_master_sampr_list,
+ SAMPR_NUM_FREQ, false);
+} /* audio_get_spdif_sample_rate */
-#if 0
-/* not needed atm */
#ifdef HAVE_SPDIF_POWER
static bool spdif_power_setting;
void audio_set_spdif_power_setting(bool on)
{
spdif_power_setting = on;
-}
+} /* audio_set_spdif_power_setting */
+
+bool audio_get_spdif_power_setting(void)
+{
+ return spdif_power_setting;
+} /* audio_get_spdif_power_setting */
#endif
+
+void audio_spdif_set_monitor(int monitor_spdif)
+{
+ EBU1CONFIG = 0x800; /* Reset before reprogram */
+
+ if (monitor_spdif > 0)
+ {
+#ifdef HAVE_SPDIF_POWER
+ EBU1CONFIG = spdif_power_setting ? (1 << 2) : 0;
+ /* Input source is EBUin1, Feed-through monitoring if desired */
+#else
+ EBU1CONFIG = (1 << 2);
+ /* Input source is EBUin1, Feed-through monitoring */
#endif
+ }
+ else if (monitor_spdif == 0)
+ {
+ /* SCLK2, TXSRC = IIS1recv, validity, normal operation */
+ EBU1CONFIG = (7 << 12) | (4 << 8) | (1 << 5) | (5 << 2);
+ }
+} /* audio_spdif_set_monitor */
+
+#endif /* HAVE_SPDIF_IN */
/**
* Sets recording parameters
- *
- * This functions starts feeding the CPU with audio data over the I2S bus
*/
-void audio_set_recording_options(int frequency, int quality,
- int source, int channel_mode,
- bool editable, int prerecord_time)
+void audio_set_recording_options(struct audio_recording_options *options)
{
- /* TODO: */
- (void)editable;
+ pcm_thread_wait_for_stop();
- /* NOTE: Coldfire UDA based recording does not yet support anything other
- * than 44.1kHz sampling rate, so we limit it to that case here now. SPDIF
- * based recording will overwrite this value with the proper sample rate in
- * audio_record(), and will not be affected by this.
- */
- frequency = 44100;
- enc_quality = quality;
- rec_source = source;
- enc_channels = channel_mode == CHN_MODE_MONO ? 1 : 2;
- pre_record_ticks = prerecord_time * HZ;
+ /* stop DMA transfer */
+ dma_lock = true;
+ pcm_stop_recording();
- switch (source)
- {
- case AUDIO_SRC_MIC:
- case AUDIO_SRC_LINEIN:
-#ifdef HAVE_FMRADIO_IN
- case AUDIO_SRC_FMRADIO:
-#endif
- /* Generate int. when 6 samples in FIFO, PDIR2 src = IIS1recv */
- DATAINCONTROL = 0xc020;
- break;
+ rec_frequency = options->rec_frequency;
+ rec_source = options->rec_source;
+ num_channels = options->rec_channels == 1 ? 1 : 2;
+ pre_record_ticks = options->rec_prerecord_time * HZ;
+ enc_config = options->enc_config;
+ enc_config.afmt = rec_format_afmt[enc_config.rec_format];
#ifdef HAVE_SPDIF_IN
- case AUDIO_SRC_SPDIF:
- /* Int. when 6 samples in FIFO. PDIR2 source = ebu1RcvData */
- DATAINCONTROL = 0xc038;
- break;
-#endif /* HAVE_SPDIF_IN */
+ if (rec_source == AUDIO_SRC_SPDIF)
+ {
+ /* must measure SPDIF sample rate before configuring codecs */
+ unsigned long sr = measure_spdif_sample_rate();
+ /* round to master list for SPDIF rate */
+ int index = round_value_to_list32(sr, audio_master_sampr_list,
+ SAMPR_NUM_FREQ, false);
+ sample_rate = audio_master_sampr_list[index];
+ /* round to HW playback rates for monitoring */
+ index = round_value_to_list32(sr, hw_freq_sampr,
+ HW_NUM_FREQ, false);
+ pcm_set_frequency(hw_freq_sampr[index]);
+ /* encoders with a limited number of rates do their own rounding */
+ }
+ else
+#endif
+ {
+ /* set sample rate from frequency selection */
+ sample_rate = rec_freq_sampr[rec_frequency];
+ pcm_set_frequency(sample_rate);
}
- sample_rate = frequency;
-
- /* Monitoring: route the signals through the coldfire audio interface. */
+ pcm_set_monitor(rec_source);
+ pcm_set_rec_source(rec_source);
- SET_IIS_PLAY(0x800); /* Reset before reprogram */
+ /* apply pcm settings to hardware */
+ pcm_apply_settings(true);
-#ifdef HAVE_SPDIF_IN
- if (source == AUDIO_SRC_SPDIF)
+ if (audio_load_encoder(enc_config.afmt))
{
- /* SCLK2 = Audioclk/4 (can't use EBUin clock), TXSRC = EBU1rcv, 64 bclk/wclk */
- IIS2CONFIG = (6 << 12) | (7 << 8) | (4 << 2);
- /* S/PDIF feed-through already configured */
+ /* start DMA transfer */
+ pcm_record_data(pcm_rec_have_more, NULL, 0);
+ /* do unlock after starting to prevent preincrement of dma_wr_pos */
+ dma_lock = pre_record_ticks == 0;
}
else
{
- /* SCLK2 follow IIS1 (UDA clock), TXSRC = IIS1rcv, 64 bclk/wclk */
- IIS2CONFIG = (8 << 12) | (4 << 8) | (4 << 2);
-
- EBU1CONFIG = 0x800; /* Reset before reprogram */
- /* SCLK2, TXSRC = IIS1recv, validity, normal operation */
- EBU1CONFIG = (7 << 12) | (4 << 8) | (1 << 5) | (5 << 2);
- }
-#else
- /* SCLK2 follow IIS1 (UDA clock), TXSRC = IIS1rcv, 64 bclk/wclk */
- SET_IIS_PLAY( (8 << 12) | (4 << 8) | (4 << 2) );
-#endif
-
- audio_load_encoder(rec_quality_info_afmt[quality]);
+ logf("set rec opt: enc load failed");
+ is_error = true;
}
-
+} /* audio_set_recording_options */
/**
* Note that microphone is mono, only left value is used
@@ -391,8 +488,7 @@ void audio_set_recording_gain(int left, int right, int type)
#elif defined (HAVE_TLV320)
tlv320_set_recvol(left, right, type);
#endif
-}
-
+} /* audio_set_recording_gain */
/**
* Start recording
@@ -401,585 +497,882 @@ void audio_set_recording_gain(int left, int right, int type)
*/
void audio_record(const char *filename)
{
+ logf("audio_record: %s", filename);
+
+ pcm_thread_wait_for_stop();
+ pcm_thread_sync_post(PCMREC_START, (void *)filename);
+
+ logf("audio_record_done");
+} /* audio_record */
+
+void audio_new_file(const char *filename)
+{
+ logf("audio_new_file: %s", filename);
+
+ pcm_thread_wait_for_stop();
+ pcm_thread_sync_post(PCMREC_NEW_FILE, (void *)filename);
+
+ logf("audio_new_file done");
+} /* audio_new_file */
+
+void audio_stop_recording(void)
+{
+ logf("audio_stop_recording");
+
+ pcm_thread_wait_for_stop();
+
if (is_recording)
- {
- logf("record while recording");
- return;
- }
+ dma_lock = true; /* fix DMA write ptr at current position */
+
+ pcm_thread_sync_post(PCMREC_STOP, NULL);
+
+ logf("audio_stop_recording done");
+} /* audio_stop_recording */
+
+void audio_pause_recording(void)
+{
+ logf("audio_pause_recording");
- strncpy(recording_filename, filename, MAX_PATH - 1);
- recording_filename[MAX_PATH - 1] = 0;
+ pcm_thread_wait_for_stop();
-#ifdef HAVE_SPDIF_IN
- if (rec_source == AUDIO_SRC_SPDIF)
- sample_rate = audio_get_spdif_sample_rate();
-#endif
+ if (is_recording)
+ dma_lock = true; /* fix DMA write ptr at current position */
- record_done = false;
- queue_post(&pcmrec_queue, PCMREC_START, 0);
+ pcm_thread_sync_post(PCMREC_PAUSE, NULL);
+ logf("audio_pause_recording done");
+} /* audio_pause_recording */
- while(!record_done)
- sleep_thread(1);
-}
+void audio_resume_recording(void)
+{
+ logf("audio_resume_recording");
+ pcm_thread_wait_for_stop();
+ pcm_thread_sync_post(PCMREC_RESUME, NULL);
-void audio_new_file(const char *filename)
+ logf("audio_resume_recording done");
+} /* audio_resume_recording */
+
+/***************************************************************************/
+/* */
+/* Functions that execute in the context of pcmrec_thread */
+/* */
+/***************************************************************************/
+
+/** Filename Queue **/
+
+/* returns true if the queue is empty */
+static inline bool pcmrec_fnq_is_empty(void)
{
- logf("pcm_new_file");
+ return fnq_rd_pos == fnq_wr_pos;
+} /* pcmrec_fnq_is_empty */
+
+/* empties the filename queue */
+static inline void pcmrec_fnq_set_empty(void)
+{
+ fnq_rd_pos = fnq_wr_pos;
+} /* pcmrec_fnq_set_empty */
- new_file_done = false;
+/* returns true if the queue is full */
+static bool pcmrec_fnq_is_full(void)
+{
+ ssize_t size = fnq_wr_pos - fnq_rd_pos;
+ if (size < 0)
+ size += fnq_size;
- strncpy(recording_filename, filename, MAX_PATH - 1);
- recording_filename[MAX_PATH - 1] = 0;
+ return size >= fnq_size - MAX_PATH;
+} /* pcmrec_fnq_is_full */
- queue_post(&pcmrec_queue, PCMREC_NEW_FILE, 0);
+/* queue another filename - will overwrite oldest one if full */
+static bool pcmrec_fnq_add_filename(const char *filename)
+{
+ strncpy(fn_queue + fnq_wr_pos, filename, MAX_PATH);
- while(!new_file_done)
- sleep_thread(1);
+ if ((fnq_wr_pos += MAX_PATH) >= fnq_size)
+ fnq_wr_pos = 0;
- logf("pcm_new_file done");
-}
+ if (fnq_rd_pos != fnq_wr_pos)
+ return true;
-/**
- *
- */
-void audio_stop_recording(void)
+ /* queue full */
+ if ((fnq_rd_pos += MAX_PATH) >= fnq_size)
+ fnq_rd_pos = 0;
+
+ return true;
+} /* pcmrec_fnq_add_filename */
+
+/* replace the last filename added */
+static bool pcmrec_fnq_replace_tail(const char *filename)
{
- if (!is_recording)
- return;
+ int pos;
+
+ if (pcmrec_fnq_is_empty())
+ return false;
+
+ pos = fnq_wr_pos - MAX_PATH;
+ if (pos < 0)
+ pos = fnq_size - MAX_PATH;
+
+ strncpy(fn_queue + pos, filename, MAX_PATH);
+
+ return true;
+} /* pcmrec_fnq_replace_tail */
- logf("pcm_stop");
+/* pulls the next filename from the queue */
+static bool pcmrec_fnq_get_filename(char *filename)
+{
+ if (pcmrec_fnq_is_empty())
+ return false;
+
+ if (filename)
+ strncpy(filename, fn_queue + fnq_rd_pos, MAX_PATH);
- is_paused = true; /* fix pcm write ptr at current position */
- stop_done = false;
- queue_post(&pcmrec_queue, PCMREC_STOP, 0);
+ if ((fnq_rd_pos += MAX_PATH) >= fnq_size)
+ fnq_rd_pos = 0;
- while(!stop_done)
- sleep_thread(1);
+ return true;
+} /* pcmrec_fnq_get_filename */
- logf("pcm_stop done");
-}
+/* close the file number pointed to by fd_p */
+static void pcmrec_close_file(int *fd_p)
+{
+ if (*fd_p < 0)
+ return; /* preserve error */
-void audio_pause_recording(void)
+ close(*fd_p);
+ *fd_p = -1;
+} /* pcmrec_close_file */
+
+/** Data Flushing **/
+
+/**
+ * called after callback to update sizes if codec changed the amount of data
+ * a chunk represents
+ */
+static inline void pcmrec_update_sizes_inl(size_t prev_enc_size,
+ unsigned long prev_num_pcm)
{
- if (!is_recording)
+ if (rec_fdata.new_enc_size != prev_enc_size)
{
- logf("pause when not recording");
- return;
+ ssize_t size_diff = rec_fdata.new_enc_size - prev_enc_size;
+ num_rec_bytes += size_diff;
+ accum_rec_bytes += size_diff;
}
- if (is_paused)
+
+ if (rec_fdata.new_num_pcm != prev_num_pcm)
{
- logf("pause when paused");
- return;
+ unsigned long pcm_diff = rec_fdata.new_num_pcm - prev_num_pcm;
+ num_rec_samples += pcm_diff;
+ accum_pcm_samples += pcm_diff;
}
-
- pause_done = false;
- queue_post(&pcmrec_queue, PCMREC_PAUSE, 0);
+} /* pcmrec_update_sizes_inl */
- while(!pause_done)
- sleep_thread(1);
-}
+/* don't need to inline every instance */
+static void pcmrec_update_sizes(size_t prev_enc_size,
+ unsigned long prev_num_pcm)
+{
+ pcmrec_update_sizes_inl(prev_enc_size, prev_num_pcm);
+} /* pcmrec_update_sizes */
-void audio_resume_recording(void)
+static void pcmrec_start_file(void)
{
- if (!is_paused)
+ size_t enc_size = rec_fdata.new_enc_size;
+ unsigned long num_pcm = rec_fdata.new_num_pcm;
+ int curr_rec_file = rec_fdata.rec_file;
+ char filename[MAX_PATH];
+
+ /* must always pull the filename that matches with this queue */
+ if (!pcmrec_fnq_get_filename(filename))
{
- logf("resume when not paused");
- return;
+ logf("start file: fnq empty");
+ *filename = '\0';
+ is_error = true;
+ }
+ else if (is_error)
+ {
+ logf("start file: is_error already");
+ }
+ else if (curr_rec_file >= 0)
+ {
+ /* Any previous file should have been closed */
+ logf("start file: file already open");
+ is_error = true;
}
- resume_done = false;
- queue_post(&pcmrec_queue, PCMREC_RESUME, 0);
+ if (is_error)
+ rec_fdata.chunk->flags |= CHUNKF_ERROR;
- while(!resume_done)
- sleep_thread(1);
-}
+ /* encoder can set error flag here and should increase
+ enc_new_size and pcm_new_size to reflect additional
+ data written if any */
+ rec_fdata.filename = filename;
+ enc_events_callback(ENC_START_FILE, &rec_fdata);
+
+ if (!is_error && (rec_fdata.chunk->flags & CHUNKF_ERROR))
+ {
+ logf("start file: enc error");
+ is_error = true;
+ }
-/* return peaks as int, so convert from short first
- note that peak values are always positive */
-void pcm_rec_get_peaks(int *left, int *right)
+ if (is_error)
+ {
+ pcmrec_close_file(&curr_rec_file);
+ /* Write no more to this file */
+ rec_fdata.chunk->flags |= CHUNKF_END_FILE;
+ }
+ else
+ {
+ pcmrec_update_sizes(enc_size, num_pcm);
+ }
+
+ rec_fdata.chunk->flags &= ~CHUNKF_START_FILE;
+} /* pcmrec_start_file */
+
+static inline void pcmrec_write_chunk(void)
{
- if (left)
- *left = peak_left;
- if (right)
- *right = peak_right;
- peak_left = 0;
- peak_right = 0;
-}
+ size_t enc_size = rec_fdata.new_enc_size;
+ unsigned long num_pcm = rec_fdata.new_num_pcm;
-/***************************************************************************/
-/* Functions that executes in the context of pcmrec_thread */
-/***************************************************************************/
+ if (is_error)
+ rec_fdata.chunk->flags |= CHUNKF_ERROR;
+
+ enc_events_callback(ENC_WRITE_CHUNK, &rec_fdata);
+
+ if ((long)rec_fdata.chunk->flags >= 0)
+ {
+ pcmrec_update_sizes_inl(enc_size, num_pcm);
+ }
+ else if (!is_error)
+ {
+ logf("wr chk enc error %d %d",
+ rec_fdata.chunk->enc_size, rec_fdata.chunk->num_pcm);
+ is_error = true;
+ }
+} /* pcmrec_write_chunk */
+
+static void pcmrec_end_file(void)
+{
+ /* all data in output buffer for current file will have been
+ written and encoder can now do any nescessary steps to
+ finalize the written file */
+ size_t enc_size = rec_fdata.new_enc_size;
+ unsigned long num_pcm = rec_fdata.new_num_pcm;
+
+ enc_events_callback(ENC_END_FILE, &rec_fdata);
+
+ if (!is_error)
+ {
+ if (rec_fdata.chunk->flags & CHUNKF_ERROR)
+ {
+ logf("end file: enc error");
+ is_error = true;
+ }
+ else
+ {
+ pcmrec_update_sizes(enc_size, num_pcm);
+ }
+ }
+
+ /* Force file close if error */
+ if (is_error)
+ pcmrec_close_file(&rec_fdata.rec_file);
+
+ rec_fdata.chunk->flags &= ~CHUNKF_END_FILE;
+} /* pcmrec_end_file */
/**
* Process the chunks
*
* This function is called when queue_get_w_tmo times out.
*
- * Other functions can also call this function with flush = true when
- * they want to save everything in the buffers to disk.
+ * Set flush_num to the number of files to flush to disk.
+ * flush_num = -1 to flush all available chunks to disk.
+ * flush_num = 0 normal write thresholding
+ * flush_num = 1 or greater - all available chunks of current file plus
+ * flush_num file starts if first chunk has been processed.
*
*/
-static void pcmrec_callback(bool flush)
+static void pcmrec_flush(unsigned flush_num)
{
- int i, num_ready, size_yield;
- long *enc_chunk, chunk_size;
-
- if (!is_recording && !flush)
- return;
+ static unsigned long last_flush_tick = 0;
+ unsigned long start_tick;
+ int num_ready, num;
+ int prio;
+ int i;
num_ready = enc_wr_index - enc_rd_index;
if (num_ready < 0)
num_ready += enc_num_chunks;
- /* calculate an estimate of recorded bytes */
- num_rec_bytes = num_file_bytes + num_ready * /* enc_chunk_size */
- ((avrg_bit_rate * 1000 / 8 * enc_samp_per_chunk + 22050) / 44100);
+ num = num_ready;
- /* near full state reached: less than 5sec remaining space */
- if (enc_num_chunks - num_ready < WRITE_THRESHOLD || flush)
+ if (flush_num == 0)
{
- logf("writing: %d (%d)", num_ready, flush);
-
- cpu_boost_id(true, CPUBOOSTID_PCMRECORD);
+ if (!is_recording)
+ return;
- size_yield = 0;
- for (i=0; i<num_ready; i++)
+ if (ata_spinup_time != spinup_time)
{
- enc_chunk = GET_ENC_CHUNK(enc_rd_index);
- chunk_size = *enc_chunk++;
-
- /* safety net: if size entry got corrupted => limit */
- if (chunk_size > (long)(enc_chunk_size - sizeof(long)))
- chunk_size = enc_chunk_size - sizeof(long);
+ /* spinup time has changed, calculate new write threshold */
+ logf("new t spinup : %d", ata_spinup_time);
+ unsigned long st = spinup_time = ata_spinup_time;
+
+ /* write at 5s + st remaining in enc_buffer */
+ if (st < 2*HZ)
+ st = 2*HZ; /* my drive is usually < 250 ticks :) */
+ else if (st > 10*HZ)
+ st = 10*HZ;
+
+ write_threshold = enc_num_chunks -
+ (int)(((5ull*HZ + st)*4ull*sample_rate + (enc_chunk_size-1)) /
+ (enc_chunk_size*HZ));
+
+ if (write_threshold < 0)
+ write_threshold = 0;
+ else if (write_threshold > panic_threshold)
+ write_threshold = panic_threshold;
+
+ logf("new wr thresh: %d", write_threshold);
+ }
- if (enc_set_header_callback != NULL)
- enc_set_header_callback(enc_chunk, enc_chunk_size,
- num_pcm_samples, false);
+ if (num_ready < write_threshold)
+ return;
- if (write(wav_file, enc_chunk, chunk_size) != chunk_size)
- {
- close_wave();
- logf("pcmrec: write err");
- is_error = true;
- break;
- }
+ /* if we're getting called too much and this isn't forced,
+ boost stat */
+ if (current_tick - last_flush_tick < HZ/2)
+ num = panic_threshold;
+ }
- num_file_bytes += chunk_size;
- num_pcm_samples += enc_samp_per_chunk;
- size_yield += chunk_size;
+ start_tick = current_tick;
+ prio = -1;
- if (size_yield >= 32768)
- { /* yield when 32kB written */
- size_yield = 0;
- yield();
- }
+ logf("writing: %d (%d)", num_ready, flush_num);
+
+ cpu_boost_id(true, CPUBOOSTID_PCMRECORD);
- enc_rd_index = (enc_rd_index + 1) % enc_num_chunks;
+ for (i=0; i<num_ready; i++)
+ {
+ if (prio == -1 && (num >= panic_threshold ||
+ current_tick - start_tick > 10*HZ))
+ {
+ /* losing ground - boost priority until finished */
+ logf("pcmrec: boost priority");
+ prio = thread_set_priority(NULL, thread_get_priority(NULL)-1);
}
- /* sync file */
- fsync(wav_file);
+ rec_fdata.chunk = GET_ENC_CHUNK(enc_rd_index);
+ rec_fdata.new_enc_size = rec_fdata.chunk->enc_size;
+ rec_fdata.new_num_pcm = rec_fdata.chunk->num_pcm;
- cpu_boost_id(false, CPUBOOSTID_PCMRECORD);
+ if (rec_fdata.chunk->flags & CHUNKF_START_FILE)
+ {
+ pcmrec_start_file();
+ if (--flush_num == 0)
+ i = num_ready; /* stop on next loop - must write this
+ chunk if it has data */
+ }
- logf("done");
- }
-}
+ pcmrec_write_chunk();
-/* Abort dma transfer */
-static void pcmrec_dma_stop(void)
-{
- DCR1 = 0;
+ if (rec_fdata.chunk->flags & CHUNKF_END_FILE)
+ pcmrec_end_file();
- error_count++;
+ INC_ENC_INDEX(enc_rd_index);
- DSR1 = 1; /* Clear interrupt */
- IPR |= (1<<15); /* Clear pending interrupt request */
+ if (is_error)
+ break;
- logf("dma1 stopped");
-}
+ if (prio == -1)
+ {
+ num = enc_wr_index - enc_rd_index;
+ if (num < 0)
+ num += enc_num_chunks;
+ }
-static void pcmrec_dma_start(void)
-{
- DAR1 = (unsigned long)GET_CHUNK(write_pos); /* Destination address */
- SAR1 = (unsigned long)&PDIR2; /* Source address */
- BCR1 = CHUNK_SIZE; /* Bytes to transfer */
+ /* no yielding, the file apis called in the codecs do that */
+ } /* end for */
- /* Start the DMA transfer.. */
-#ifdef HAVE_SPDIF_IN
- INTERRUPTCLEAR = 0x03c00000;
-#endif
+ /* sync file */
+ if (rec_fdata.rec_file >= 0)
+ fsync(rec_fdata.rec_file);
+
+ cpu_boost_id(false, CPUBOOSTID_PCMRECORD);
- /* 16Byte transfers prevents from sporadic errors during cpu_boost() */
- DCR1 = DMA_INT | DMA_EEXT | DMA_CS | DMA_DINC | DMA_DSIZE(3) | DMA_START;
+ if (prio != -1)
+ {
+ /* return to original priority */
+ logf("pcmrec: unboost priority");
+ thread_set_priority(NULL, prio);
+ }
- logf("dma1 started");
-}
+ last_flush_tick = current_tick; /* save tick when we left */
+ logf("done");
+} /* pcmrec_flush */
-/* DMA1 Interrupt is called when the DMA has finished transfering a chunk */
-void DMA1(void) __attribute__ ((interrupt_handler, section(".icode")));
-void DMA1(void)
+/**
+ * Marks a new stream in the buffer and gives the encoder a chance for special
+ * handling of transition from one to the next. The encoder may change the
+ * chunk that ends the old stream by requesting more chunks and similiarly for
+ * the new but must always advance the position though the interface. It can
+ * later reject any data it cares to when writing the file but should mark the
+ * chunk so it can recognize this. ENC_WRITE_CHUNK event must be able to accept
+ * a NULL data pointer without error as well.
+ */
+static void pcmrec_new_stream(const char *filename, /* next file name */
+ unsigned long flags, /* CHUNKF_* flags */
+ int pre_index) /* index for prerecorded data */
{
- int res = DSR1;
+ logf("pcmrec_new_stream");
- DSR1 = 1; /* Clear interrupt */
+ struct enc_buffer_event_data data;
+ bool (*fnq_add_fn)(const char *) = NULL;
+ struct enc_chunk_hdr *start = NULL;
- if (res & 0x70)
+ int get_chunk_index(struct enc_chunk_hdr *chunk)
{
- DCR1 = 0; /* Stop DMA transfer */
- error_count++;
-
- logf("dma1 err: 0x%x", res);
-
- DAR1 = (unsigned long)GET_CHUNK(write_pos); /* Destination address */
- BCR1 = CHUNK_SIZE;
- DCR1 = DMA_INT | DMA_EEXT | DMA_CS | DMA_DINC | DMA_START;
+ return ((char *)chunk - (char *)enc_buffer) / enc_chunk_size;
+ }
- /* Flush recorded data to disk and stop recording */
- queue_post(&pcmrec_queue, PCMREC_STOP, NULL);
- }
-#ifdef HAVE_SPDIF_IN
- else if ((rec_source == AUDIO_SRC_SPDIF) &&
- (INTERRUPTSTAT & 0x01c00000)) /* valnogood, symbolerr, parityerr */
+ struct enc_chunk_hdr * get_prev_chunk(int index)
{
- INTERRUPTCLEAR = 0x03c00000;
- error_count++;
+ DEC_ENC_INDEX(index);
+ return GET_ENC_CHUNK(index);
+ }
- logf("spdif err");
+ data.pre_chunk = NULL;
+ data.chunk = GET_ENC_CHUNK(enc_wr_index);
- DAR1 = (unsigned long)GET_CHUNK(write_pos); /* Destination address */
- BCR1 = CHUNK_SIZE;
- }
-#endif
- else
+ /* end chunk */
+ if (flags & CHUNKF_END_FILE)
{
- long peak_l, peak_r;
- long *ptr, j;
-
- ptr = GET_CHUNK(write_pos);
+ data.chunk->flags &= CHUNKF_START_FILE | CHUNKF_END_FILE;
- if (!is_paused) /* advance write position */
- write_pos = (write_pos + CHUNK_SIZE) & CHUNK_MASK;
+ if (data.chunk->flags & CHUNKF_START_FILE)
+ {
+ /* cannot start and end on same unprocessed chunk */
+ logf("file end on start");
+ flags &= ~CHUNKF_END_FILE;
+ }
+ else if (enc_rd_index == enc_wr_index)
+ {
+ /* all data flushed but file not ended - chunk will be left
+ empty */
+ logf("end on dead end");
+ data.chunk->flags = 0;
+ data.chunk->enc_size = 0;
+ data.chunk->num_pcm = 0;
+ data.chunk->enc_data = NULL;
+ INC_ENC_INDEX(enc_wr_index);
+ data.chunk = GET_ENC_CHUNK(enc_wr_index);
+ }
+ else
+ {
+ struct enc_chunk_hdr *last = get_prev_chunk(enc_wr_index);
- DAR1 = (unsigned long)GET_CHUNK(write_pos); /* Destination address */
- BCR1 = CHUNK_SIZE;
+ if (last->flags & CHUNKF_END_FILE)
+ {
+ /* end already processed and marked - can't end twice */
+ logf("file end again");
+ flags &= ~CHUNKF_END_FILE;
+ }
+ }
+ }
- peak_l = peak_r = 0;
+ /* start chunk */
+ if (flags & CHUNKF_START_FILE)
+ {
+ bool pre = flags & CHUNKF_PRERECORD;
- /* only peak every 4th sample */
- for (j=0; j<CHUNK_SIZE/4; j+=4)
+ if (pre)
{
- long value = ptr[j];
-#ifdef ROCKBOX_BIG_ENDIAN
- if (value > peak_l) peak_l = value;
- else if (-value > peak_l) peak_l = -value;
-
- value <<= 16;
- if (value > peak_r) peak_r = value;
- else if (-value > peak_r) peak_r = -value;
-#else
- if (value > peak_r) peak_r = value;
- else if (-value > peak_r) peak_r = -value;
-
- value <<= 16;
- if (value > peak_l) peak_l = value;
- else if (-value > peak_l) peak_l = -value;
-#endif
+ logf("stream prerecord start");
+ start = data.pre_chunk = GET_ENC_CHUNK(pre_index);
+ start->flags &= CHUNKF_START_FILE | CHUNKF_PRERECORD;
+ }
+ else
+ {
+ logf("stream normal start");
+ start = data.chunk;
+ start->flags &= CHUNKF_START_FILE;
}
- peak_left = (int)(peak_l >> 16);
- peak_right = (int)(peak_r >> 16);
+ /* if encoder hasn't yet processed the last start - abort the start
+ of the previous file queued or else it will be empty and invalid */
+ if (start->flags & CHUNKF_START_FILE)
+ {
+ logf("replacing fnq tail: %s", filename);
+ fnq_add_fn = pcmrec_fnq_replace_tail;
+ }
+ else
+ {
+ logf("adding filename: %s", filename);
+ fnq_add_fn = pcmrec_fnq_add_filename;
+ }
}
- IPR |= (1<<15); /* Clear pending interrupt request */
-}
-
-/* Create WAVE file and write header */
-/* Sets returns 0 if success, -1 on failure */
-static int start_wave(void)
-{
- wav_file = open(recording_filename, O_RDWR|O_CREAT|O_TRUNC);
+ data.flags = flags;
+ enc_events_callback(ENC_REC_NEW_STREAM, &data);
- if (wav_file < 0)
+ if (flags & CHUNKF_END_FILE)
{
- wav_file = -1;
- logf("rec: create failed: %d", wav_file);
- is_error = true;
- return -1;
+ int i = get_chunk_index(data.chunk);
+ get_prev_chunk(i)->flags |= CHUNKF_END_FILE;
}
-
- /* add main file header (enc_head_size=0 for encoders without) */
- if (enc_head_size != write(wav_file, enc_head_buffer, enc_head_size))
+
+ if (start)
{
- close(wav_file);
- wav_file = -1;
- logf("rec: write failed");
- is_error = true;
- return -1;
- }
+ if (!(flags & CHUNKF_PRERECORD))
+ {
+ /* get stats on data added to start - sort of a prerecord operation */
+ int i = get_chunk_index(data.chunk);
+ struct enc_chunk_hdr *chunk = data.chunk;
- return 0;
-}
+ logf("start data: %d %d", i, enc_wr_index);
-/* Update header and set correct length values */
-static void close_wave(void)
-{
- unsigned char head[100]; /* assume maximum 100 bytes for file header */
- int size_read;
+ num_rec_bytes = 0;
+ num_rec_samples = 0;
- if (wav_file != -1)
- {
- /* update header before closing the file (wav+wv encoder will do) */
- if (enc_set_header_callback != NULL)
- {
- lseek(wav_file, 0, SEEK_SET);
- /* try to read the head size (but we'll accept less) */
- size_read = read(wav_file, head, sizeof(head));
+ while (i != enc_wr_index)
+ {
+ num_rec_bytes += chunk->enc_size;
+ num_rec_samples += chunk->num_pcm;
+ INC_ENC_INDEX(i);
+ chunk = GET_ENC_CHUNK(i);
+ }
+
+ start->flags &= ~CHUNKF_START_FILE;
+ start = data.chunk;
+ }
+
+ start->flags |= CHUNKF_START_FILE;
- enc_set_header_callback(head, size_read, num_pcm_samples, true);
- lseek(wav_file, 0, SEEK_SET);
- write(wav_file, head, size_read);
+ /* flush one file out if full and adding */
+ if (fnq_add_fn == pcmrec_fnq_add_filename && pcmrec_fnq_is_full())
+ {
+ logf("fnq full: flushing 1");
+ pcmrec_flush(1);
}
- close(wav_file);
- wav_file = -1;
+
+ fnq_add_fn(filename);
}
-}
+} /* pcmrec_new_stream */
-static void pcmrec_start(void)
+/** event handlers for pcmrec thread */
+
+/* PCMREC_INIT */
+static void pcmrec_init(void)
{
- long max_pre_chunks, pre_ticks, max_pre_ticks;
+ rec_fdata.rec_file = -1;
+
+ /* pcm FIFO */
+ dma_lock = true;
+ pcm_rd_pos = 0;
+ dma_wr_pos = 0;
+
+ /* encoder FIFO */
+ enc_wr_index = 0;
+ enc_rd_index = 0;
+
+ /* filename queue */
+ fnq_rd_pos = 0;
+ fnq_wr_pos = 0;
+
+ /* stats */
+ num_rec_bytes = 0;
+ num_rec_samples = 0;
+ accum_rec_bytes = 0;
+ accum_pcm_samples = 0;
+
+ pcm_thread_unsignal_event(PCMREC_CLOSE);
+ is_recording = false;
+ is_paused = false;
+ is_stopping = false;
+ is_error = false;
+
+ pcm_buffer = audio_get_recording_buffer(&rec_buffer_size);
+ /* Line align pcm_buffer 2^4=16 bytes */
+ pcm_buffer = (unsigned char *)ALIGN_UP_P2((unsigned)pcm_buffer, 4);
+ enc_buffer = pcm_buffer + ALIGN_UP_P2(PCM_NUM_CHUNKS*PCM_CHUNK_SIZE +
+ PCM_MAX_FEED_SIZE, 2);
+
+ pcm_init_recording();
+ pcm_thread_signal_event(PCMREC_INIT);
+} /* pcmrec_init */
+
+/* PCMREC_CLOSE */
+static void pcmrec_close(void)
+{
+ dma_lock = true;
+ pcm_close_recording();
+ pcm_thread_unsignal_event(PCMREC_INIT);
+ pcm_thread_signal_event(PCMREC_CLOSE);
+} /* pcmrec_close */
+
+/* PCMREC_START */
+static void pcmrec_start(const char *filename)
+{
+ unsigned long pre_sample_ticks;
+ int rd_start;
- logf("pcmrec_start");
+ logf("pcmrec_start: %s", filename);
if (is_recording)
{
logf("already recording");
- record_done = true;
- return;
+ goto already_recording;
}
- if (wav_file != -1)
- close_wave();
+ /* reset stats */
+ num_rec_bytes = 0;
+ num_rec_samples = 0;
+ accum_rec_bytes = 0;
+ accum_pcm_samples = 0;
+ spinup_time = -1;
+
+ rd_start = enc_wr_index;
+ pre_sample_ticks = 0;
- if (start_wave() != 0)
+ if (pre_record_ticks)
{
- /* failed to create the file */
- record_done = true;
- return;
- }
+ int i;
- /* calculate maximum available chunks & resulting ticks */
- max_pre_chunks = (enc_wr_index - enc_rd_index +
- enc_num_chunks) % enc_num_chunks;
- if (max_pre_chunks > enc_num_chunks - WRITE_THRESHOLD)
- max_pre_chunks = enc_num_chunks - WRITE_THRESHOLD;
- max_pre_ticks = max_pre_chunks * HZ * enc_samp_per_chunk / 44100;
-
- /* limit prerecord if not enough data available */
- pre_ticks = pre_record_ticks > max_pre_ticks ?
- max_pre_ticks : pre_record_ticks;
- max_pre_chunks = 44100 * pre_ticks / HZ / enc_samp_per_chunk;
- enc_rd_index = (enc_wr_index - max_pre_chunks +
+ /* calculate number of available chunks */
+ unsigned long avail_pre_chunks = (enc_wr_index - enc_rd_index +
enc_num_chunks) % enc_num_chunks;
+ /* overflow at 974 seconds of prerecording at 44.1kHz */
+ unsigned long pre_record_sample_ticks = enc_sample_rate*pre_record_ticks;
+
+ /* Get exact measure of recorded data as number of samples aren't
+ nescessarily going to be the max for each chunk */
+ for (i = rd_start; avail_pre_chunks-- > 0;)
+ {
+ struct enc_chunk_hdr *chunk;
+ unsigned long chunk_sample_ticks;
+
+ DEC_ENC_INDEX(i);
+
+ chunk = GET_ENC_CHUNK(i);
+
+ /* must have data to be counted */
+ if (chunk->enc_data == NULL)
+ continue;
- record_start_time = current_tick - pre_ticks;
+ chunk_sample_ticks = chunk->num_pcm*HZ;
- num_rec_bytes = enc_num_chunks * CHUNK_SIZE;
- num_file_bytes = 0;
- num_pcm_samples = 0;
- pause_start_time = 0;
+ rd_start = i;
+ pre_sample_ticks += chunk_sample_ticks;
+ num_rec_bytes += chunk->enc_size;
+ num_rec_samples += chunk->num_pcm;
+
+ /* stop here if enough already */
+ if (pre_sample_ticks >= pre_record_sample_ticks)
+ break;
+ }
+
+ accum_rec_bytes = num_rec_bytes;
+ accum_pcm_samples = num_rec_samples;
+ }
+
+ enc_rd_index = rd_start;
+
+ /* filename queue should be empty */
+ if (!pcmrec_fnq_is_empty())
+ {
+ logf("fnq: not empty!");
+ pcmrec_fnq_set_empty();
+ }
+ dma_lock = false;
is_paused = false;
is_recording = true;
- record_done = true;
-}
+ pcmrec_new_stream(filename,
+ CHUNKF_START_FILE |
+ (pre_sample_ticks > 0 ? CHUNKF_PRERECORD : 0),
+ enc_rd_index);
+
+already_recording:
+ pcm_thread_signal_event(PCMREC_START);
+ logf("pcmrec_start done");
+} /* pcmrec_start */
+
+/* PCMREC_STOP */
static void pcmrec_stop(void)
{
logf("pcmrec_stop");
- if (is_recording)
+ if (!is_recording)
{
- /* wait for encoding finish */
- is_paused = true;
- while(!wav_queue_empty)
- sleep_thread(1);
-
- is_recording = false;
-
- /* Flush buffers to file */
- pcmrec_callback(true);
- close_wave();
+ logf("not recording");
+ goto not_recording_or_stopping;
+ }
+ else if (is_stopping)
+ {
+ logf("already stopping");
+ goto not_recording_or_stopping;
}
- is_paused = false;
- stop_done = true;
+ is_stopping = true;
+ dma_lock = true; /* lock dma write position */
+ queue_post(&pcmrec_queue, PCMREC_FINISH_STOP, NULL);
+not_recording_or_stopping:
+ pcm_thread_signal_event(PCMREC_STOP);
logf("pcmrec_stop done");
-}
+} /* pcmrec_stop */
-static void pcmrec_new_file(void)
+/* PCMREC_FINISH_STOP */
+static void pcmrec_finish_stop(void)
{
- logf("pcmrec_new_file");
+ logf("pcmrec_finish_stop");
- if (!is_recording)
+ if (!is_stopping)
{
- logf("not recording");
- new_file_done = true;
- return;
+ logf("not stopping");
+ goto not_stopping;
}
- /* Since pcmrec_callback() blocks until the data has been written,
- here is a good approximation when recording to the new file starts
- */
- record_start_time = current_tick;
+ /* flush all available data first to avoid overflow while waiting
+ for encoding to finish */
+ pcmrec_flush(-1);
- if (is_paused)
- pause_start_time = record_start_time;
+ /* wait for encoder to finish remaining data */
+ if (!is_error)
+ {
+ while (!wav_queue_empty)
+ yield();
+ }
- /* Flush what we got in buffers to file */
- pcmrec_callback(true);
+ /* end stream at last data */
+ pcmrec_new_stream(NULL, CHUNKF_END_FILE, 0);
- close_wave();
-
- num_rec_bytes = 0;
- num_file_bytes = 0;
- num_pcm_samples = 0;
+ /* flush anything else encoder added */
+ pcmrec_flush(-1);
+
+ /* remove any pending file start not yet processed - should be at
+ most one at enc_wr_index */
+ pcmrec_fnq_get_filename(NULL);
+ /* encoder should abort any chunk it was in midst of processing */
+ GET_ENC_CHUNK(enc_wr_index)->flags = CHUNKF_ABORT;
- /* start the new file */
- if (start_wave() != 0)
+ /* filename queue should be empty */
+ if (!pcmrec_fnq_is_empty())
{
- logf("new_file failed");
- pcmrec_stop();
+ logf("fnq: not empty!");
+ pcmrec_fnq_set_empty();
}
- new_file_done = true;
- logf("pcmrec_new_file done");
-}
+ /* be absolutely sure the file is closed */
+ if (is_error)
+ pcmrec_close_file(&rec_fdata.rec_file);
+ rec_fdata.rec_file = -1;
+
+ is_recording = false;
+ is_paused = false;
+ is_stopping = false;
+ dma_lock = pre_record_ticks == 0;
+
+not_stopping:
+ logf("pcmrec_finish_stop done");
+} /* pcmrec_finish_stop */
+/* PCMREC_PAUSE */
static void pcmrec_pause(void)
{
logf("pcmrec_pause");
if (!is_recording)
{
- logf("pause: not recording");
- pause_done = true;
- return;
+ logf("not recording");
+ goto not_recording_or_paused;
+ }
+ else if (is_paused)
+ {
+ logf("already paused");
+ goto not_recording_or_paused;
}
- pause_start_time = current_tick;
+ dma_lock = true; /* fix DMA write pointer at current position */
is_paused = true;
- pause_done = true;
+not_recording_or_paused:
+ pcm_thread_signal_event(PCMREC_PAUSE);
logf("pcmrec_pause done");
-}
-
+} /* pcmrec_pause */
+/* PCMREC_RESUME */
static void pcmrec_resume(void)
{
logf("pcmrec_resume");
- if (!is_paused)
+ if (!is_recording)
{
- logf("resume: not paused");
- resume_done = true;
- return;
+ logf("not recording");
+ goto not_recording_or_not_paused;
+ }
+ else if (!is_paused)
+ {
+ logf("not paused");
+ goto not_recording_or_not_paused;
}
is_paused = false;
is_recording = true;
+ dma_lock = false;
- /* Compensate for the time we have been paused */
- if (pause_start_time)
- {
- record_start_time += current_tick - pause_start_time;
- pause_start_time = 0;
- }
-
- resume_done = true;
+not_recording_or_not_paused:
+ pcm_thread_signal_event(PCMREC_RESUME);
logf("pcmrec_resume done");
-}
+} /* pcmrec_resume */
-/**
- * audio_init_recording calls this function using PCMREC_INIT
- *
- */
-static void pcmrec_init(void)
+/* PCMREC_NEW_FILE */
+static void pcmrec_new_file(const char *filename)
{
- wav_file = -1;
- read_pos = 0;
- write_pos = 0;
- enc_wr_index = 0;
- enc_rd_index = 0;
+ logf("pcmrec_new_file: %s", filename);
- avrg_bit_rate = 0;
- curr_bit_rate = 0;
- curr_chunk_cnt = 0;
-
- peak_left = 0;
- peak_right = 0;
+ if (!is_recording)
+ {
+ logf("not recording");
+ goto not_recording;
+ }
num_rec_bytes = 0;
- num_file_bytes = 0;
- num_pcm_samples = 0;
- record_start_time = 0;
- pause_start_time = 0;
-
- close_done = false;
- is_recording = false;
- is_paused = false;
- is_error = false;
-
- rec_buffer = (unsigned char*)(((long)audiobuf + 15) & ~15);
- enc_buffer = rec_buffer + NUM_CHUNKS * CHUNK_SIZE + MAX_FEED_SIZE;
- /* 8000Bytes at audiobufend */
- enc_buffer_size = audiobufend - enc_buffer - 8000;
-
- SET_IIS_PLAY(0x800); /* Stop any playback */
- AUDIOGLOB |= 0x180; /* IIS1 fifo auto sync = on, PDIR2 auto sync = on */
- DATAINCONTROL = 0xc000; /* Generate Interrupt when 6 samples in fifo */
-
- DIVR1 = 55; /* DMA1 is mapped into vector 55 in system.c */
- DMACONFIG = 1; /* DMA0Req = PDOR3, DMA1Req = PDIR2 */
- DMAROUTE = (DMAROUTE & 0xffff00ff) | DMA1_REQ_AUDIO_2;
- ICR7 = 0x1c; /* Enable interrupt at level 7, priority 0 */
- IMR &= ~(1<<15); /* bit 15 is DMA1 */
-
-#ifdef HAVE_SPDIF_IN
- PHASECONFIG = 0x34; /* Gain = 3*2^13, source = EBUIN */
-#endif
- pcmrec_dma_start();
-
- init_done = 1;
-}
+ num_rec_samples = 0;
-static void pcmrec_close(void)
-{
- DMAROUTE = (DMAROUTE & 0xffff00ff);
- ICR7 = 0x00; /* Disable interrupt */
- IMR |= (1<<15); /* bit 15 is DMA1 */
+ pcmrec_new_stream(filename,
+ CHUNKF_START_FILE | CHUNKF_END_FILE,
+ 0);
- pcmrec_dma_stop();
-
- /* Reset PDIR2 data flow */
- DATAINCONTROL = 0x200;
- close_done = true;
- init_done = false;
-}
+not_recording:
+ pcm_thread_signal_event(PCMREC_NEW_FILE);
+ logf("pcmrec_new_file done");
+} /* pcmrec_new_file */
+static void pcmrec_thread(void) __attribute__((noreturn));
static void pcmrec_thread(void)
{
struct event ev;
logf("thread pcmrec start");
- error_count = 0;
-
while(1)
{
- queue_wait_w_tmo(&pcmrec_queue, &ev, HZ / 4);
+ if (is_recording)
+ {
+ /* Poll periodically to flush data */
+ queue_wait_w_tmo(&pcmrec_queue, &ev, HZ/5);
+
+ if (ev.id == SYS_TIMEOUT)
+ {
+ pcmrec_flush(0); /* flush if getting full */
+ continue;
+ }
+ }
+ else
+ {
+ /* Not doing anything - sit and wait for commands */
+ queue_wait(&pcmrec_queue, &ev);
+ }
switch (ev.id)
{
@@ -992,13 +1385,17 @@ static void pcmrec_thread(void)
break;
case PCMREC_START:
- pcmrec_start();
+ pcmrec_start((const char *)ev.data);
break;
case PCMREC_STOP:
pcmrec_stop();
break;
+ case PCMREC_FINISH_STOP:
+ pcmrec_finish_stop();
+ break;
+
case PCMREC_PAUSE:
pcmrec_pause();
break;
@@ -1008,11 +1405,11 @@ static void pcmrec_thread(void)
break;
case PCMREC_NEW_FILE:
- pcmrec_new_file();
+ pcmrec_new_file((const char *)ev.data);
break;
- case SYS_TIMEOUT:
- pcmrec_callback(false);
+ case PCMREC_FLUSH_NUM:
+ pcmrec_flush((unsigned)ev.data);
break;
case SYS_USB_CONNECTED:
@@ -1023,140 +1420,267 @@ static void pcmrec_thread(void)
usb_wait_for_disconnect(&pcmrec_queue);
}
break;
- }
- }
+ } /* end switch */
+ } /* end while */
+} /* pcmrec_thread */
- logf("thread pcmrec done");
-}
+/****************************************************************************/
+/* */
+/* following functions will be called by the encoder codec */
+/* */
+/****************************************************************************/
-/* Select VINL & VINR source: 0=Line-in, 1=FM Radio */
-void pcm_rec_mux(int source)
+/* pass the encoder settings to the encoder */
+void enc_get_inputs(struct enc_inputs *inputs)
{
-#ifdef IRIVER_H300_SERIES
- if(source == 0)
- and_l(~0x40000000, &GPIO_OUT); /* Line In */
- else
- or_l(0x40000000, &GPIO_OUT); /* FM radio */
+ inputs->sample_rate = sample_rate;
+ inputs->num_channels = num_channels;
+ inputs->config = &enc_config;
+} /* enc_get_inputs */
- or_l(0x40000000, &GPIO_ENABLE);
- or_l(0x40000000, &GPIO_FUNCTION);
-#elif defined(IRIVER_H100_SERIES)
- if(source == 0)
- and_l(~0x00800000, &GPIO_OUT); /* Line In */
- else
- or_l(0x00800000, &GPIO_OUT); /* FM radio */
+/* set the encoder dimensions (called by encoder codec at initialization and
+ termination) */
+void enc_set_parameters(struct enc_parameters *params)
+{
+ size_t bufsize, resbytes;
- or_l(0x00800000, &GPIO_ENABLE);
- or_l(0x00800000, &GPIO_FUNCTION);
+ logf("enc_set_parameters");
-#elif defined(IAUDIO_X5)
- if(source == 0)
- or_l((1<<29), &GPIO_OUT); /* Line In */
- else
- and_l(~(1<<29), &GPIO_OUT); /* FM radio */
+ if (!params)
+ {
+ logf("reset");
+ /* Encoder is terminating */
+ memset(&enc_config, 0, sizeof (enc_config));
+ enc_sample_rate = 0;
+ return;
+ }
+
+ enc_sample_rate = params->enc_sample_rate;
+ logf("enc sampr:%d", enc_sample_rate);
+
+ pcm_rd_pos = dma_wr_pos;
+
+ enc_config.afmt = params->afmt;
+ /* addition of the header is always implied - chunk size 4-byte aligned */
+ enc_chunk_size =
+ ALIGN_UP_P2(ENC_CHUNK_HDR_SIZE + params->chunk_size, 2);
+ enc_data_size = enc_chunk_size - ENC_CHUNK_HDR_SIZE;
+ enc_events_callback = params->events_callback;
+
+ logf("chunk size:%d", enc_chunk_size);
+
+ /*** Configure the buffers ***/
+
+ /* Layout of recording buffer:
+ * [ax] = possible alignment x multiple
+ * [sx] = possible size alignment of x multiple
+ * |[a16]|[s4]:PCM Buffer+PCM Guard|[s4 each]:Encoder Chunks|->
+ * |[[s4]:Reserved Bytes]|Filename Queue->|[space]|
+ */
+ resbytes = ALIGN_UP_P2(params->reserve_bytes, 2);
+ logf("resbytes:%d", resbytes);
+
+ bufsize = rec_buffer_size - (enc_buffer - pcm_buffer) -
+ resbytes - FNQ_MIN_NUM_PATHS*MAX_PATH;
+
+ enc_num_chunks = bufsize / enc_chunk_size;
+ logf("num chunks:%d", enc_num_chunks);
- or_l((1<<29), &GPIO_ENABLE);
- or_l((1<<29), &GPIO_FUNCTION);
+ /* get real amount used by encoder chunks */
+ bufsize = enc_num_chunks*enc_chunk_size;
+ logf("enc size:%d", bufsize);
+
+ /* panic boost thread priority at 1 second remaining */
+ panic_threshold = enc_num_chunks -
+ (4*sample_rate + (enc_chunk_size-1)) / enc_chunk_size;
+ if (panic_threshold < 0)
+ panic_threshold = 0;
+
+ logf("panic thr:%d", panic_threshold);
+
+ /** set OUT parameters **/
+ params->enc_buffer = enc_buffer;
+ params->buf_chunk_size = enc_chunk_size;
+ params->num_chunks = enc_num_chunks;
+
+ /* calculate reserve buffer start and return pointer to encoder */
+ params->reserve_buffer = NULL;
+ if (resbytes > 0)
+ {
+ params->reserve_buffer = enc_buffer + bufsize;
+ bufsize += resbytes;
+ }
- /* iAudio x5 */
+ /* place filename queue at end of buffer using up whatever remains */
+ fnq_rd_pos = 0; /* reset */
+ fnq_wr_pos = 0; /* reset */
+ fn_queue = enc_buffer + bufsize;
+ fnq_size = pcm_buffer + rec_buffer_size - fn_queue;
+ fnq_size = ALIGN_DOWN(fnq_size, MAX_PATH);
+ logf("fnq files: %d", fnq_size / MAX_PATH);
+
+#if 0
+ logf("ab :%08X", (unsigned long)audiobuf);
+ logf("pcm:%08X", (unsigned long)pcm_buffer);
+ logf("enc:%08X", (unsigned long)enc_buffer);
+ logf("res:%08X", (unsigned long)params->reserve_buffer);
+ logf("fnq:%08X", (unsigned long)fn_queue);
+ logf("end:%08X", (unsigned long)fn_queue + fnq_size);
+ logf("abe:%08X", (unsigned long)audiobufend);
#endif
-}
+ /* init all chunk headers and reset indexes */
+ enc_rd_index = 0;
+ for (enc_wr_index = enc_num_chunks; enc_wr_index > 0; )
+ GET_ENC_CHUNK(--enc_wr_index)->flags = 0;
-/****************************************************************************/
-/* */
-/* following functions will be called by the encoder codec */
-/* */
-/****************************************************************************/
+ logf("enc_set_parameters done");
+} /* enc_set_parameters */
-/* pass the encoder buffer pointer/size, mono/stereo, quality to the encoder */
-void enc_get_inputs(int *buffer_size, int *channels, int *quality)
+/* return encoder chunk at current write position */
+struct enc_chunk_hdr * enc_get_chunk(void)
{
- *buffer_size = enc_buffer_size;
- *channels = enc_channels;
- *quality = enc_quality;
-}
+ struct enc_chunk_hdr *chunk = GET_ENC_CHUNK(enc_wr_index);
+ chunk->flags &= CHUNKF_START_FILE;
-/* set the encoder dimensions (called by encoder codec at initialization) */
-void enc_set_parameters(int chunk_size, int num_chunks, int samp_per_chunk,
- char *head_ptr, int head_size, int enc_id)
-{
- /* set read_pos just in front of current write_pos */
- read_pos = (write_pos - CHUNK_SIZE) & CHUNK_MASK;
-
- enc_rd_index = 0; /* reset */
- enc_wr_index = 0; /* reset */
- enc_chunk_size = chunk_size; /* max chunk size */
- enc_num_chunks = num_chunks; /* total number of chunks */
- enc_samp_per_chunk = samp_per_chunk; /* pcm samples / encoderchunk */
- enc_head_buffer = head_ptr; /* optional file header data (wav) */
- enc_head_size = head_size; /* optional file header data (wav) */
- audio_enc_id = enc_id; /* AFMT_* id */
-}
+ if (!is_recording)
+ chunk->flags |= CHUNKF_PRERECORD;
-/* allocate encoder chunk */
-unsigned int *enc_alloc_chunk(void)
-{
- return (unsigned int*)(enc_buffer + enc_wr_index * enc_chunk_size);
-}
+ return chunk;
+} /* enc_get_chunk */
-/* free previously allocated encoder chunk */
-void enc_free_chunk(void)
+/* releases the current chunk into the available chunks */
+void enc_finish_chunk(void)
{
- unsigned long *enc_chunk;
+ struct enc_chunk_hdr *chunk = GET_ENC_CHUNK(enc_wr_index);
- enc_chunk = GET_ENC_CHUNK(enc_wr_index);
- curr_chunk_cnt++;
-/* curr_bit_rate += *enc_chunk * 44100 * 8 / (enc_samp_per_chunk * 1000); */
- curr_bit_rate += *enc_chunk * 441 * 8 / (enc_samp_per_chunk * 10 );
- avrg_bit_rate = (curr_bit_rate + curr_chunk_cnt / 2) / curr_chunk_cnt;
+ /* encoder may have set error flag or written too much data */
+ if ((long)chunk->flags < 0 || chunk->enc_size > enc_data_size)
+ {
+ is_error = true;
- /* advance enc_wr_index to the next chunk */
- enc_wr_index = (enc_wr_index + 1) % enc_num_chunks;
+#ifdef ROCKBOX_HAS_LOGF
+ if (chunk->enc_size > enc_data_size)
+ {
+ /* illegal to scribble over next chunk */
+ logf("finish chk ovf: %d>%d", chunk->enc_size, enc_data_size);
+ }
+ else
+ {
+ /* encoder set error flag */
+ logf("finish chk enc error");
+ }
+#endif
+ }
+
+ /* advance enc_wr_index to the next encoder chunk */
+ INC_ENC_INDEX(enc_wr_index);
- /* buffer full: advance enc_rd_index (for prerecording purpose) */
- if (enc_rd_index == enc_wr_index)
+ if (enc_rd_index != enc_wr_index)
{
- enc_rd_index = (enc_rd_index + 1) % enc_num_chunks;
+ num_rec_bytes += chunk->enc_size;
+ accum_rec_bytes += chunk->enc_size;
+ num_rec_samples += chunk->num_pcm;
+ accum_pcm_samples += chunk->num_pcm;
}
-}
+ else if (is_recording) /* buffer full */
+ {
+ /* keep current position */
+ logf("enc_buffer ovf");
+ DEC_ENC_INDEX(enc_wr_index);
+ }
+ else
+ {
+ /* advance enc_rd_index for prerecording */
+ INC_ENC_INDEX(enc_rd_index);
+ }
+} /* enc_finish_chunk */
-/* checks near empty state on wav input buffer */
-int enc_wavbuf_near_empty(void)
+/* checks near empty state on pcm input buffer */
+int enc_pcm_buf_near_empty(void)
{
/* less than 1sec raw data? => unboost encoder */
- if (((write_pos - read_pos) & CHUNK_MASK) < 44100*4)
- return 1;
- else
- return 0;
-}
+ size_t avail = (dma_wr_pos - pcm_rd_pos) & PCM_CHUNK_MASK;
+ return avail < (sample_rate << 2) ? 1 : 0;
+} /* enc_pcm_buf_near_empty */
/* passes a pointer to next chunk of unprocessed wav data */
-char *enc_get_wav_data(int size)
+/* TODO: this really should give the actual size returned */
+unsigned char * enc_get_pcm_data(size_t size)
{
- char *ptr;
- int avail;
+ size_t avail = (dma_wr_pos - pcm_rd_pos) & PCM_CHUNK_MASK;
/* limit the requested pcm data size */
- if(size > MAX_FEED_SIZE)
- size = MAX_FEED_SIZE;
-
- avail = (write_pos - read_pos) & CHUNK_MASK;
+ if (size > PCM_MAX_FEED_SIZE)
+ size = PCM_MAX_FEED_SIZE;
if (avail >= size)
{
- ptr = rec_buffer + read_pos;
- read_pos = (read_pos + size) & CHUNK_MASK;
+ unsigned char *ptr = pcm_buffer + pcm_rd_pos;
+ pcm_rd_pos = (pcm_rd_pos + size) & PCM_CHUNK_MASK;
/* ptr must point to continous data at wraparound position */
- if (read_pos < size)
- memcpy(rec_buffer + NUM_CHUNKS * CHUNK_SIZE,
- rec_buffer, read_pos);
+ if ((size_t)pcm_rd_pos < size)
+ memcpy(pcm_buffer + PCM_NUM_CHUNKS*PCM_CHUNK_SIZE,
+ pcm_buffer, pcm_rd_pos);
wav_queue_empty = false;
return ptr;
}
+ /* not enough data available - encoder should idle */
wav_queue_empty = true;
return NULL;
-}
+} /* enc_get_pcm_data */
+
+/* puts some pcm data back in the queue */
+size_t enc_unget_pcm_data(size_t size)
+{
+ /* can't let DMA advance write position when doing this */
+ int level = set_irq_level(HIGHEST_IRQ_LEVEL);
+
+ if (pcm_rd_pos != dma_wr_pos)
+ {
+ /* disallow backing up into current DMA write chunk */
+ size_t old_avail = (pcm_rd_pos - dma_wr_pos - PCM_CHUNK_SIZE)
+ & PCM_CHUNK_MASK;
+
+ /* limit size to amount of old data remaining */
+ if (size > old_avail)
+ size = old_avail;
+
+ pcm_rd_pos = (pcm_rd_pos - size) & PCM_CHUNK_MASK;
+ }
+
+ set_irq_level(level);
+
+ return size;
+} /* enc_unget_pcm_data */
+
+/** Low level pcm recording apis **/
+
+/****************************************************************************
+ * Functions that do not require targeted implementation but only a targeted
+ * interface
+ */
+void pcm_record_data(pcm_more_callback_type more_ready,
+ unsigned char *start, size_t size)
+{
+ pcm_callback_more_ready = more_ready;
+
+ if (!(start && size))
+ {
+ size = 0;
+ if (more_ready)
+ more_ready(&start, &size);
+ }
+
+ if (start && size)
+ pcm_rec_dma_start(start, size);
+} /* pcm_record_data */
+
+void pcm_stop_recording(void)
+{
+ if (pcm_recording)
+ pcm_rec_dma_stop();
+} /* pcm_stop_recording */
diff --git a/firmware/system.c b/firmware/system.c
index 242d84d16c..96d5f96602 100644
--- a/firmware/system.c
+++ b/firmware/system.c
@@ -390,8 +390,7 @@ int system_memory_guard(int newmode)
(void)newmode;
return 0;
}
-#elif defined(CPU_COLDFIRE)
-/* system code is in target tree for all coldfire targets */
+
#elif CONFIG_CPU == SH7034
#include "led.h"
#include "system.h"
diff --git a/firmware/target/coldfire/iaudio/x5/system-x5.c b/firmware/target/coldfire/iaudio/x5/system-x5.c
index 6be6d25ce0..30a4f6e71b 100644
--- a/firmware/target/coldfire/iaudio/x5/system-x5.c
+++ b/firmware/target/coldfire/iaudio/x5/system-x5.c
@@ -42,7 +42,7 @@ void set_cpu_frequency(long frequency)
PLLCR &= ~1; /* Bypass mode */
timers_adjust_prescale(CPUFREQ_DEFAULT_MULT, false);
RECALC_DELAYS(CPUFREQ_MAX);
- PLLCR = 0x13442045;
+ PLLCR = 0x03042045 | (PLLCR & 0x70C00000);
CSCR0 = 0x00001180; /* Flash: 4 wait states */
CSCR1 = 0x00000980; /* LCD: 2 wait states */
while(!(PLLCR & 0x80000000)) {}; /* Wait until the PLL has locked.
@@ -60,7 +60,7 @@ void set_cpu_frequency(long frequency)
PLLCR &= ~1; /* Bypass mode */
timers_adjust_prescale(CPUFREQ_DEFAULT_MULT, false);
RECALC_DELAYS(CPUFREQ_NORMAL);
- PLLCR = 0x16430045;
+ PLLCR = 0x06030045 | (PLLCR & 0x70C00000);
CSCR0 = 0x00000580; /* Flash: 1 wait state */
CSCR1 = 0x00000180; /* LCD: 0 wait states */
while(!(PLLCR & 0x80000000)) {}; /* Wait until the PLL has locked.
@@ -77,7 +77,8 @@ void set_cpu_frequency(long frequency)
PLLCR &= ~1; /* Bypass mode */
timers_adjust_prescale(CPUFREQ_DEFAULT_MULT, true);
RECALC_DELAYS(CPUFREQ_DEFAULT);
- PLLCR = 0x10400200; /* Power down PLL, but keep CLSEL and CRSEL */
+ /* Power down PLL, but keep CLSEL and CRSEL */
+ PLLCR = 0x00000200 | (PLLCR & 0x70C00000);
CSCR0 = 0x00000180; /* Flash: 0 wait states */
CSCR1 = 0x00000180; /* LCD: 0 wait states */
DCR = (0x8000 | DEFAULT_REFRESH_TIMER); /* Refresh timer */
diff --git a/firmware/target/coldfire/iriver/system-iriver.c b/firmware/target/coldfire/iriver/system-iriver.c
index 3517788641..43ba4eeed4 100644
--- a/firmware/target/coldfire/iriver/system-iriver.c
+++ b/firmware/target/coldfire/iriver/system-iriver.c
@@ -81,7 +81,7 @@ void set_cpu_frequency(long frequency)
PLLCR &= ~1; /* Bypass mode */
timers_adjust_prescale(CPUFREQ_DEFAULT_MULT, false);
RECALC_DELAYS(CPUFREQ_MAX);
- PLLCR = 0x11c56005;
+ PLLCR = 0x01056005 | (PLLCR & 0x70c00000);
CSCR0 = 0x00001180; /* Flash: 4 wait states */
CSCR1 = 0x00001580; /* LCD: 5 wait states */
#if CONFIG_USBOTG == USBOTG_ISP1362
@@ -108,7 +108,7 @@ void set_cpu_frequency(long frequency)
PLLCR &= ~1; /* Bypass mode */
timers_adjust_prescale(CPUFREQ_DEFAULT_MULT, false);
RECALC_DELAYS(CPUFREQ_NORMAL);
- PLLCR = 0x13c5e005;
+ PLLCR = 0x0305e005 | (PLLCR & 0x70c00000);
CSCR0 = 0x00000580; /* Flash: 1 wait state */
CSCR1 = 0x00000180; /* LCD: 0 wait states */
#if CONFIG_USBOTG == USBOTG_ISP1362
@@ -134,7 +134,8 @@ void set_cpu_frequency(long frequency)
PLLCR &= ~1; /* Bypass mode */
timers_adjust_prescale(CPUFREQ_DEFAULT_MULT, true);
RECALC_DELAYS(CPUFREQ_DEFAULT);
- PLLCR = 0x10c00200; /* Power down PLL, but keep CLSEL and CRSEL */
+ /* Power down PLL, but keep CLSEL and CRSEL */
+ PLLCR = 0x00000200 | (PLLCR & 0x70c00000);
CSCR0 = 0x00000180; /* Flash: 0 wait states */
CSCR1 = 0x00000180; /* LCD: 0 wait states */
#if CONFIG_USBOTG == USBOTG_ISP1362
diff --git a/firmware/target/coldfire/system-coldfire.c b/firmware/target/coldfire/system-coldfire.c
index 66e4feb154..2fc81496db 100644
--- a/firmware/target/coldfire/system-coldfire.c
+++ b/firmware/target/coldfire/system-coldfire.c
@@ -310,3 +310,10 @@ int system_memory_guard(int newmode)
return oldmode;
}
+
+/* allow setting of audio clock related bits */
+void coldfire_set_pllcr_audio_bits(long bits)
+{
+ PLLCR = (PLLCR & ~0x70c00000) | (bits & 0x70c00000);
+}
+
diff --git a/firmware/target/coldfire/system-target.h b/firmware/target/coldfire/system-target.h
index 03852115ad..24e3fb8705 100644
--- a/firmware/target/coldfire/system-target.h
+++ b/firmware/target/coldfire/system-target.h
@@ -110,6 +110,28 @@ static inline unsigned long swap32(unsigned long value)
return value;
}
+static inline unsigned long swap_odd_even32(unsigned long value)
+{
+ /*
+ result[31..24],[15.. 8] = value[23..16],[ 7.. 0]
+ result[23..16],[ 7.. 0] = value[31..24],[15.. 8]
+ */
+ unsigned long mask = 0x00FF00FF;
+
+ asm ( /* val = ABCD */
+ "and.l %[val],%[mask] \n" /* mask = .B.D */
+ "eor.l %[mask],%[val] \n" /* val = A.C. */
+ "lsl.l #8,%[mask] \n" /* mask = B.D. */
+ "lsr.l #8,%[val] \n" /* val = .A.C */
+ "or.l %[mask],%[val] \n" /* val = BADC */
+ : /* outputs */
+ [val] "+d"(value),
+ [mask]"+d"(mask)
+ );
+
+ return value;
+}
+
static inline void invalidate_icache(void)
{
asm volatile ("move.l #0x01000000,%d0\n"
@@ -118,6 +140,13 @@ static inline void invalidate_icache(void)
"movec.l %d0,%cacr");
}
+#ifdef IAUDIO_X5
+#define DEFAULT_PLLCR_AUDIO_BITS 0x10400000
+#else
+#define DEFAULT_PLLCR_AUDIO_BITS 0x10c00000
+#endif
+void coldfire_set_pllcr_audio_bits(long bits);
+
/* 11.2896 MHz */
#define CPUFREQ_DEFAULT_MULT 1
#define CPUFREQ_DEFAULT (CPUFREQ_DEFAULT_MULT * CPU_FREQ)
diff --git a/firmware/thread.c b/firmware/thread.c
index 6a94a52333..4094877742 100644
--- a/firmware/thread.c
+++ b/firmware/thread.c
@@ -711,6 +711,14 @@ int thread_set_priority(struct thread_entry *thread, int priority)
return old_priority;
}
+
+int thread_get_priority(struct thread_entry *thread)
+{
+ if (thread == NULL)
+ thread = cores[CURRENT_CORE].running;
+
+ return thread->priority;
+}
#endif
void init_threads(void)
diff --git a/uisimulator/sdl/lcd-charcell.c b/uisimulator/sdl/lcd-charcell.c
index 8b93653a19..eb94ebf67f 100644
--- a/uisimulator/sdl/lcd-charcell.c
+++ b/uisimulator/sdl/lcd-charcell.c
@@ -181,7 +181,8 @@ void screen_dump(void)
int x, y;
static unsigned char line[BMP_LINESIZE];
- create_numbered_filename(filename, "", "dump_", ".bmp", 4);
+ create_numbered_filename(filename, "", "dump_", ".bmp", 4,
+ IF_CNFN_NUM_(, NULL));
DEBUGF("screen_dump\n");
fd = sim_creat(filename, O_WRONLY);