Nyaaori/rockbox
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
rockbox/apps/plugins/test_codec.c
Michael Sevakis d37bf24d90 Enable setting of global output samplerate on certain targets. 
Replaces the NATIVE_FREQUENCY constant with a configurable frequency.

The user may select 48000Hz if the hardware supports it. The default is
still 44100Hz and the minimum is 44100Hz. The setting is located in the
playback settings, under "Frequency".

"Frequency" was duplicated in english.lang for now to avoid having to
fix every .lang file for the moment and throwing everything out of sync
because of the new play_frequency feature in features.txt. The next
cleanup should combine it with the one included for recording and
generalize the ID label.

If the hardware doesn't support 48000Hz, no setting will be available.

On particular hardware where very high rates are practical and desireable,
the upper bound can be extended by patching.

The PCM mixer can be configured to play at the full hardware frequency
range. The DSP core can configure to the hardware minimum up to the
maximum playback setting (some buffers must be reserved according to
the maximum rate).

If only 44100Hz is supported or possible on a given target for playback,
using the DSP and mixer at other samperates is possible if the hardware
offers them.

Change-Id: I6023cf0c0baa8bc6292b6919b4dd3618a6a25622
Reviewed-on: http://gerrit.rockbox.org/479
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
2013-07-06 04:22:04 +02:00

1029 lines
26 KiB
C
Raw Blame History

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2007 Dave Chapman
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "plugin.h"
#include "lib/pluginlib_touchscreen.h"
#include "lib/pluginlib_exit.h"
#include "lib/pluginlib_actions.h"
/* this set the context to use with PLA */
static const struct button_mapping *plugin_contexts[] = { pla_main_ctx };
#define TESTCODEC_EXITBUTTON PLA_EXIT
#define TESTCODEC_EXITBUTTON2 PLA_CANCEL
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
static unsigned int boost =1;
static const struct opt_items boost_settings[2] = {
{ "No", -1 },
{ "Yes", -1 },
};
#endif
/* Log functions copied from test_disk.c */
static int line = 0;
static int max_line = 0;
static int log_fd = -1;
static void log_close(void)
{
if (log_fd >= 0)
rb->close(log_fd);
}
static bool log_init(bool use_logfile)
{
int h;
char logfilename[MAX_PATH];
rb->lcd_getstringsize("A", NULL, &h);
max_line = LCD_HEIGHT / h;
line = 0;
rb->lcd_clear_display();
rb->lcd_update();
if (use_logfile) {
log_close();
rb->create_numbered_filename(logfilename, HOME_DIR, "test_codec_log_", ".txt",
2 IF_CNFN_NUM_(, NULL));
log_fd = rb->open(logfilename, O_RDWR|O_CREAT|O_TRUNC, 0666);
return log_fd >= 0;
}
return true;
}
static void log_text(char *text, bool advance)
{
rb->lcd_puts(0, line, text);
rb->lcd_update();
if (advance)
{
if (++line >= max_line)
line = 0;
if (log_fd >= 0)
rb->fdprintf(log_fd, "%s\n", text);
}
}
struct wavinfo_t
{
int fd;
int samplerate;
int channels;
int sampledepth;
int stereomode;
int totalsamples;
};
static void* audiobuf;
static void* codec_mallocbuf;
static size_t audiosize;
static size_t audiobufsize;
static int offset;
static int fd;
/* Our local implementation of the codec API */
static struct codec_api ci;
struct test_track_info {
struct mp3entry id3; /* TAG metadata */
size_t filesize; /* File total length */
};
static struct test_track_info track;
static bool use_dsp;
static bool checksum;
static uint32_t crc32;
static volatile unsigned int elapsed;
static volatile bool codec_playing;
static volatile enum codec_command_action codec_action;
static volatile long endtick;
static volatile long rebuffertick;
struct wavinfo_t wavinfo;
static unsigned char wav_header[44] =
{
'R','I','F','F', // 0 - ChunkID
0,0,0,0, // 4 - ChunkSize (filesize-8)
'W','A','V','E', // 8 - Format
'f','m','t',' ', // 12 - SubChunkID
16,0,0,0, // 16 - SubChunk1ID // 16 for PCM
1,0, // 20 - AudioFormat (1=16-bit)
0,0, // 22 - NumChannels
0,0,0,0, // 24 - SampleRate in Hz
0,0,0,0, // 28 - Byte Rate (SampleRate*NumChannels*(BitsPerSample/8)
0,0, // 32 - BlockAlign (== NumChannels * BitsPerSample/8)
16,0, // 34 - BitsPerSample
'd','a','t','a', // 36 - Subchunk2ID
0,0,0,0 // 40 - Subchunk2Size
};
static inline void int2le32(unsigned char* buf, int32_t x)
{
buf[0] = (x & 0xff);
buf[1] = (x & 0xff00) >> 8;
buf[2] = (x & 0xff0000) >> 16;
buf[3] = (x & 0xff000000) >>24;
}
static inline void int2le24(unsigned char* buf, int32_t x)
{
buf[0] = (x & 0xff);
buf[1] = (x & 0xff00) >> 8;
buf[2] = (x & 0xff0000) >> 16;
}
static inline void int2le16(unsigned char* buf, int16_t x)
{
buf[0] = (x & 0xff);
buf[1] = (x & 0xff00) >> 8;
}
static unsigned char *wavbuffer;
static unsigned char *dspbuffer;
static int dspbuffer_count;
void init_wav(char* filename)
{
wavinfo.totalsamples = 0;
wavinfo.fd = rb->creat(filename, 0666);
if (wavinfo.fd >= 0)
{
/* Write WAV header - we go back and fill in the details at the end */
rb->write(wavinfo.fd, wav_header, sizeof(wav_header));
}
}
void close_wav(void)
{
int filesize = rb->filesize(wavinfo.fd);
int channels = (wavinfo.stereomode == STEREO_MONO) ? 1 : 2;
int bps = 16; /* TODO */
/* We assume 16-bit, Stereo */
rb->lseek(wavinfo.fd,0,SEEK_SET);
int2le32(wav_header+4, filesize-8); /* ChunkSize */
int2le16(wav_header+22, channels);
int2le32(wav_header+24, wavinfo.samplerate);
int2le32(wav_header+28, wavinfo.samplerate * channels * (bps / 8)); /* ByteRate */
int2le16(wav_header+32, channels * (bps / 8));
int2le32(wav_header+40, filesize - 44); /* Subchunk2Size */
rb->write(wavinfo.fd, wav_header, sizeof(wav_header));
rb->close(wavinfo.fd);
}
/* Returns buffer to malloc array. Only codeclib should need this. */
static void* codec_get_buffer(size_t *size)
{
*size = CODEC_SIZE;
return codec_mallocbuf;
}
static int process_dsp(const void *ch1, const void *ch2, int count)
{
struct dsp_buffer src;
src.remcount = count;
src.pin[0] = ch1;
src.pin[1] = ch2;
src.proc_mask = 0;
struct dsp_buffer dst;
dst.remcount = 0;
dst.p16out = (int16_t *)dspbuffer;
dst.bufcount = dspbuffer_count;
while (1)
{
int old_remcount = dst.remcount;
rb->dsp_process(ci.dsp, &src, &dst);
if (dst.bufcount <= 0 ||
(src.remcount <= 0 && dst.remcount <= old_remcount))
{
/* Dest is full or no input left and DSP purged */
break;
}
}
return dst.remcount;
}
/* Null output */
static void pcmbuf_insert_null(const void *ch1, const void *ch2, int count)
{
if (use_dsp)
process_dsp(ch1, ch2, count);
/* Prevent idle poweroff */
rb->reset_poweroff_timer();
}
/*
* Helper function used when the file is larger then the available memory.
* Rebuffers the file by setting the start of the audio buffer to be
* new_offset and filling from there.
*/
static int fill_buffer(int new_offset){
size_t n, bytestoread;
long temp = *rb->current_tick;
rb->lseek(fd,new_offset,SEEK_SET);
if(new_offset + audiobufsize <= track.filesize)
bytestoread = audiobufsize;
else
bytestoread = track.filesize-new_offset;
n = rb->read(fd, audiobuf,bytestoread);
if (n != bytestoread)
{
log_text("Read failed.",true);
DEBUGF("read fail: got %d bytes, expected %d\n", (int)n, (int)audiobufsize);
rb->backlight_on();
if (fd >= 0)
{
rb->close(fd);
}
return -1;
}
offset = new_offset;
/*keep track of how much time we spent buffering*/
rebuffertick += *rb->current_tick-temp;
return 0;
}
/* WAV output or calculate crc32 of output*/
static void pcmbuf_insert_wav_checksum(const void *ch1, const void *ch2, int count)
{
/* Prevent idle poweroff */
rb->reset_poweroff_timer();
if (use_dsp) {
count = process_dsp(ch1, ch2, count);
wavinfo.totalsamples += count;
#ifdef ROCKBOX_BIG_ENDIAN
unsigned char* p = dspbuffer;
int i;
for (i = 0; i < count; i++) {
int2le16(p,*(int16_t *)p);
p += 2;
int2le16(p,*(int16_t *)p);
p += 2;
}
#endif
if (checksum) {
crc32 = rb->crc_32(dspbuffer, count * 2 * sizeof (int16_t), crc32);
} else {
rb->write(wavinfo.fd, dspbuffer, count * 2 * sizeof (int16_t));
}
}
else
{
const int16_t* data1_16;
const int16_t* data2_16;
const int32_t* data1_32;
const int32_t* data2_32;
unsigned char* p = wavbuffer;
const int scale = wavinfo.sampledepth - 15;
const int dc_bias = 1 << (scale - 1);
if (wavinfo.sampledepth <= 16) {
data1_16 = ch1;
data2_16 = ch2;
switch(wavinfo.stereomode)
{
case STEREO_INTERLEAVED:
while (count--) {
int2le16(p,*data1_16++);
p += 2;
int2le16(p,*data1_16++);
p += 2;
}
break;
case STEREO_NONINTERLEAVED:
while (count--) {
int2le16(p,*data1_16++);
p += 2;
int2le16(p,*data2_16++);
p += 2;
}
break;
case STEREO_MONO:
while (count--) {
int2le16(p,*data1_16++);
p += 2;
}
break;
}
} else {
data1_32 = ch1;
data2_32 = ch2;
switch(wavinfo.stereomode)
{
case STEREO_INTERLEAVED:
while (count--) {
int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
p += 2;
int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
p += 2;
}
break;
case STEREO_NONINTERLEAVED:
while (count--) {
int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
p += 2;
int2le16(p, clip_sample_16((*data2_32++ + dc_bias) >> scale));
p += 2;
}
break;
case STEREO_MONO:
while (count--) {
int2le16(p, clip_sample_16((*data1_32++ + dc_bias) >> scale));
p += 2;
}
break;
}
}
wavinfo.totalsamples += count;
if (checksum)
crc32 = rb->crc_32(wavbuffer, p - wavbuffer, crc32);
else
rb->write(wavinfo.fd, wavbuffer, p - wavbuffer);
} /* else */
}
/* Set song position in WPS (value in ms). */
static void set_elapsed(unsigned long value)
{
elapsed = value;
ci.id3->elapsed = value;
}
/* Read next <size> amount bytes from file buffer to <ptr>.
Will return number of bytes read or 0 if end of file. */
static size_t read_filebuf(void *ptr, size_t size)
{
if (ci.curpos > (off_t)track.filesize)
{
return 0;
} else {
size_t realsize = MIN(track.filesize-ci.curpos,size);
/* check if we have enough bytes ready*/
if(realsize >(audiobufsize - (ci.curpos-offset)))
{
/*rebuffer so that we start at ci.curpos*/
fill_buffer(ci.curpos);
}
rb->memcpy(ptr, audiobuf + (ci.curpos-offset), realsize);
ci.curpos += realsize;
return realsize;
}
}
/* Request pointer to file buffer which can be used to read
<realsize> amount of data. <reqsize> tells the buffer system
how much data it should try to allocate. If <realsize> is 0,
end of file is reached. */
static void* request_buffer(size_t *realsize, size_t reqsize)
{
*realsize = MIN(track.filesize-ci.curpos,reqsize);
/*check if we have enough bytes ready - requested > bufsize-currentbufpos*/
if(*realsize>(audiobufsize - (ci.curpos-offset)))
{
/*rebuffer so that we start at ci.curpos*/
fill_buffer(ci.curpos);
}
return (audiobuf + (ci.curpos-offset));
}
/* Advance file buffer position by <amount> amount of bytes. */
static void advance_buffer(size_t amount)
{
ci.curpos += amount;
ci.id3->offset = ci.curpos;
}
/* Seek file buffer to position <newpos> beginning of file. */
static bool seek_buffer(size_t newpos)
{
ci.curpos = newpos;
return true;
}
/* Codec should call this function when it has done the seeking. */
static void seek_complete(void)
{
/* Do nothing */
}
/* Codec calls this to know what it should do next. */
static enum codec_command_action get_command(intptr_t *param)
{
rb->yield();
return codec_action;
(void)param;
}
/* Some codecs call this to determine whether they should loop. */
static bool loop_track(void)
{
return false;
}
static void set_offset(size_t value)
{
ci.id3->offset = value;
}
/* Configure different codec buffer parameters. */
static void configure(int setting, intptr_t value)
{
if (use_dsp)
rb->dsp_configure(ci.dsp, setting, value);
switch(setting)
{
case DSP_SET_FREQUENCY:
DEBUGF("samplerate=%d\n",(int)value);
if (use_dsp) {
wavinfo.samplerate = rb->dsp_configure(
ci.dsp, DSP_GET_OUT_FREQUENCY, 0);
} else {
wavinfo.samplerate = (int)value;
}
break;
case DSP_SET_SAMPLE_DEPTH:
DEBUGF("sampledepth = %d\n",(int)value);
wavinfo.sampledepth = use_dsp ? 16 : (int)value;
break;
case DSP_SET_STEREO_MODE:
DEBUGF("Stereo mode = %d\n",(int)value);
wavinfo.stereomode = use_dsp ? STEREO_INTERLEAVED : (int)value;
break;
}
}
static void init_ci(void)
{
/* --- Our "fake" implementations of the codec API functions. --- */
ci.dsp = rb->dsp_get_config(CODEC_IDX_AUDIO);
ci.codec_get_buffer = codec_get_buffer;
if (wavinfo.fd >= 0 || checksum) {
ci.pcmbuf_insert = pcmbuf_insert_wav_checksum;
} else {
ci.pcmbuf_insert = pcmbuf_insert_null;
}
ci.set_elapsed = set_elapsed;
ci.read_filebuf = read_filebuf;
ci.request_buffer = request_buffer;
ci.advance_buffer = advance_buffer;
ci.seek_buffer = seek_buffer;
ci.seek_complete = seek_complete;
ci.set_offset = set_offset;
ci.configure = configure;
ci.get_command = get_command;
ci.loop_track = loop_track;
/* --- "Core" functions --- */
/* kernel/ system */
ci.sleep = rb->sleep;
ci.yield = rb->yield;
/* strings and memory */
ci.strcpy = rb->strcpy;
ci.strlen = rb->strlen;
ci.strcmp = rb->strcmp;
ci.strcat = rb->strcat;
ci.memset = rb->memset;
ci.memcpy = rb->memcpy;
ci.memmove = rb->memmove;
ci.memcmp = rb->memcmp;
ci.memchr = rb->memchr;
#if defined(DEBUG) || defined(SIMULATOR)
ci.debugf = rb->debugf;
#endif
#ifdef ROCKBOX_HAS_LOGF
ci.logf = rb->logf;
#endif
ci.qsort = rb->qsort;
#ifdef RB_PROFILE
ci.profile_thread = rb->profile_thread;
ci.profstop = rb->profstop;
ci.profile_func_enter = rb->profile_func_enter;
ci.profile_func_exit = rb->profile_func_exit;
#endif
ci.commit_dcache = rb->commit_dcache;
ci.commit_discard_dcache = rb->commit_discard_dcache;
ci.commit_discard_idcache = rb->commit_discard_idcache;
#if NUM_CORES > 1
ci.create_thread = rb->create_thread;
ci.thread_thaw = rb->thread_thaw;
ci.thread_wait = rb->thread_wait;
ci.semaphore_init = rb->semaphore_init;
ci.semaphore_wait = rb->semaphore_wait;
ci.semaphore_release = rb->semaphore_release;
#endif
#if defined(CPU_ARM) && (CONFIG_PLATFORM & PLATFORM_NATIVE)
ci.__div0 = rb->__div0;
#endif
}
static void codec_thread(void)
{
const char* codecname;
int res;
codecname = rb->get_codec_filename(track.id3.codectype);
/* Load the codec */
res = rb->codec_load_file(codecname, &ci);
if (res >= 0)
{
/* Decode the file */
res = rb->codec_run_proc();
}
/* Clean up */
rb->codec_close();
/* Signal to the main thread that we are done */
endtick = *rb->current_tick - rebuffertick;
codec_playing = false;
}
static enum plugin_status test_track(const char* filename)
{
size_t n;
enum plugin_status res = PLUGIN_ERROR;
long starttick;
long ticks;
unsigned long speed;
unsigned long duration;
const char* ch;
char str[MAX_PATH];
offset=0;
/* Display filename (excluding any path)*/
ch = rb->strrchr(filename, '/');
if (ch==NULL)
ch = filename;
else
ch++;
rb->snprintf(str,sizeof(str),"%s",ch);
log_text(str,true);
log_text("Loading...",false);
fd = rb->open(filename,O_RDONLY);
if (fd < 0)
{
log_text("Cannot open file",true);
goto exit;
}
track.filesize = rb->filesize(fd);
/* Clear the id3 struct */
rb->memset(&track.id3, 0, sizeof(struct mp3entry));
if (!rb->get_metadata(&(track.id3), fd, filename))
{
log_text("Cannot read metadata",true);
goto exit;
}
if (track.filesize > audiosize)
{
audiobufsize=audiosize;
} else
{
audiobufsize=track.filesize;
}
n = rb->read(fd, audiobuf, audiobufsize);
if (n != audiobufsize)
{
log_text("Read failed.",true);
goto exit;
}
/* Initialise the function pointers in the codec API */
init_ci();
/* Prepare the codec struct for playing the whole file */
ci.filesize = track.filesize;
ci.id3 = &track.id3;
ci.curpos = 0;
if (use_dsp) {
rb->dsp_configure(ci.dsp, DSP_RESET, 0);
rb->dsp_configure(ci.dsp, DSP_FLUSH, 0);
}
if (checksum)
crc32 = 0xffffffff;
rebuffertick=0;
starttick = *rb->current_tick;
codec_playing = true;
codec_action = CODEC_ACTION_NULL;
rb->codec_thread_do_callback(codec_thread, NULL);
/* Wait for codec thread to die */
while (codec_playing)
{
int button = pluginlib_getaction(HZ, plugin_contexts,
ARRAYLEN(plugin_contexts));
if ((button == TESTCODEC_EXITBUTTON) || (button == TESTCODEC_EXITBUTTON2))
{
codec_action = CODEC_ACTION_HALT;
break;
}
rb->snprintf(str,sizeof(str),"%d of %d",elapsed,(int)track.id3.length);
log_text(str,false);
}
ticks = endtick - starttick;
/* Be sure it is done */
rb->codec_thread_do_callback(NULL, NULL);
rb->backlight_on();
log_text(str,true);
if (codec_action == CODEC_ACTION_HALT)
{
/* User aborted test */
}
else if (checksum)
{
rb->snprintf(str, sizeof(str), "CRC32 - %08x", (unsigned)crc32);
log_text(str,true);
}
else if (wavinfo.fd < 0)
{
/* Display benchmark information */
rb->snprintf(str,sizeof(str),"Decode time - %d.%02ds",(int)ticks/100,(int)ticks%100);
log_text(str,true);
duration = track.id3.length / 10;
rb->snprintf(str,sizeof(str),"File duration - %d.%02ds",(int)duration/100,(int)duration%100);
log_text(str,true);
if (ticks > 0)
speed = duration * 10000 / ticks;
else
speed = 0;
rb->snprintf(str,sizeof(str),"%d.%02d%% realtime",(int)speed/100,(int)speed%100);
log_text(str,true);
#if (CONFIG_PLATFORM & PLATFORM_NATIVE)
/* show effective clockrate in MHz needed for realtime decoding */
if (speed > 0)
{
int freq;
freq = *rb->cpu_frequency;
speed = freq / speed;
rb->snprintf(str,sizeof(str),"%d.%02dMHz needed for realtime",
(int)speed/100,(int)speed%100);
log_text(str,true);
}
#endif
}
res = PLUGIN_OK;
exit:
rb->backlight_on();
if (fd >= 0)
{
rb->close(fd);
}
return res;
}
#ifdef HAVE_TOUCHSCREEN
void cleanup(void)
{
rb->screens[0]->set_viewport(NULL);
}
#endif
void plugin_quit(void)
{
int btn;
#ifdef HAVE_TOUCHSCREEN
static struct touchbutton button[] = {{
.action = ACTION_STD_OK,
.title = "OK",
/* viewport runtime initialized, rest false/NULL */
}};
struct viewport *vp = &button[0].vp;
struct screen *lcd = rb->screens[SCREEN_MAIN];
rb->viewport_set_defaults(vp, SCREEN_MAIN);
const int border = 10;
const int height = 50;
lcd->set_viewport(vp);
/* button matches the bottom center in the grid */
vp->x = lcd->lcdwidth/3;
vp->width = lcd->lcdwidth/3;
vp->height = height;
vp->y = lcd->lcdheight - height - border;
touchbutton_draw(button, ARRAYLEN(button));
lcd->update_viewport();
if (rb->touchscreen_get_mode() == TOUCHSCREEN_POINT)
{
while (codec_action != CODEC_ACTION_HALT &&
touchbutton_get(button, ARRAYLEN(button)) != ACTION_STD_OK);
}
else
#endif
do {
btn = pluginlib_getaction(TIMEOUT_BLOCK, plugin_contexts,
ARRAYLEN(plugin_contexts));
exit_on_usb(btn);
} while ((codec_action != CODEC_ACTION_HALT)
&& (btn != TESTCODEC_EXITBUTTON)
&& (btn != TESTCODEC_EXITBUTTON2));
}
/* plugin entry point */
enum plugin_status plugin_start(const void* parameter)
{
int result, selection = 0;
enum plugin_status res = PLUGIN_OK;
int scandir;
struct dirent *entry;
DIR* dir;
char* ch;
char dirpath[MAX_PATH];
char filename[MAX_PATH];
size_t buffer_size;
if (parameter == NULL)
{
rb->splash(HZ*2, "No File");
return PLUGIN_ERROR;
}
wavbuffer = rb->plugin_get_buffer(&buffer_size);
dspbuffer = wavbuffer + buffer_size / 2;
dspbuffer_count = (buffer_size - (dspbuffer - wavbuffer)) /
(2 * sizeof (int16_t));
codec_mallocbuf = rb->plugin_get_audio_buffer(&audiosize);
/* Align codec_mallocbuf to pointer size, tlsf wants that */
codec_mallocbuf = (void*)(((intptr_t)codec_mallocbuf +
sizeof(intptr_t)-1) & ~(sizeof(intptr_t)-1));
audiobuf = SKIPBYTES(codec_mallocbuf, CODEC_SIZE);
audiosize -= CODEC_SIZE;
rb->lcd_clear_display();
rb->lcd_update();
#ifdef HAVE_TOUCHSCREEN
rb->touchscreen_set_mode(rb->global_settings->touch_mode);
#endif
enum
{
SPEED_TEST = 0,
SPEED_TEST_DIR,
WRITE_WAV,
SPEED_TEST_WITH_DSP,
SPEED_TEST_DIR_WITH_DSP,
WRITE_WAV_WITH_DSP,
CHECKSUM,
CHECKSUM_DIR,
QUIT,
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
BOOST,
#endif
};
MENUITEM_STRINGLIST(
menu, "test_codec", NULL,
"Speed test",
"Speed test folder",
"Write WAV",
"Speed test with DSP",
"Speed test folder with DSP",
"Write WAV with DSP",
"Checksum",
"Checksum folder",
"Quit",
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
"Boosting",
#endif
);
show_menu:
rb->lcd_clear_display();
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
menu:
#endif
result = rb->do_menu(&menu, &selection, NULL, false);
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
if (result == BOOST)
{
rb->set_option("Boosting", &boost, INT,
boost_settings, 2, NULL);
goto menu;
}
#endif
if (result == QUIT)
{
res = PLUGIN_OK;
goto exit;
}
scandir = 0;
/* Map test runs with checksum calcualtion to standard runs
* SPEED_TEST and SPEED_TEST_DIR and set the 'checksum' flag. */
if ((checksum = (result == CHECKSUM ||
result == CHECKSUM_DIR)))
result -= 6;
/* Map test runs with DSP to standard runs SPEED_TEST,
* SPEED_TEST_DIR and WRITE_WAV and set the 'use_dsp' flag. */
if ((use_dsp = (result >= SPEED_TEST_WITH_DSP &&
result <= WRITE_WAV_WITH_DSP)))
result -= 3;
if (result == SPEED_TEST) {
wavinfo.fd = -1;
log_init(false);
} else if (result == SPEED_TEST_DIR) {
wavinfo.fd = -1;
scandir = 1;
/* Only create a log file when we are testing a folder */
if (!log_init(true)) {
rb->splash(HZ*2, "Cannot create logfile");
res = PLUGIN_ERROR;
goto exit;
}
} else if (result == WRITE_WAV) {
log_init(false);
init_wav("/test.wav");
if (wavinfo.fd < 0) {
rb->splash(HZ*2, "Cannot create /test.wav");
res = PLUGIN_ERROR;
goto exit;
}
} else if (result == MENU_ATTACHED_USB) {
res = PLUGIN_USB_CONNECTED;
goto exit;
} else if (result < 0) {
res = PLUGIN_OK;
goto exit;
}
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
if (boost)
rb->cpu_boost(true);
#endif
if (scandir) {
/* Test all files in the same directory as the file selected by the
user */
rb->strlcpy(dirpath,parameter,sizeof(dirpath));
ch = rb->strrchr(dirpath,'/');
ch[1]=0;
DEBUGF("Scanning directory \"%s\"\n",dirpath);
dir = rb->opendir(dirpath);
if (dir) {
entry = rb->readdir(dir);
while (entry) {
struct dirinfo info = rb->dir_get_info(dir, entry);
if (!(info.attribute & ATTR_DIRECTORY)) {
rb->snprintf(filename,sizeof(filename),"%s%s",dirpath,entry->d_name);
test_track(filename);
if (codec_action == CODEC_ACTION_HALT)
break;
log_text("", true);
}
/* Read next entry */
entry = rb->readdir(dir);
}
rb->closedir(dir);
}
} else {
/* Just test the file */
res = test_track(parameter);
/* Close WAV file (if there was one) */
if (wavinfo.fd >= 0) {
close_wav();
log_text("Wrote /test.wav",true);
}
}
#ifdef HAVE_ADJUSTABLE_CPU_FREQ
if (boost)
rb->cpu_boost(false);
#endif
plugin_quit();
rb->button_clear_queue();
goto show_menu;
exit:
log_close();
return res;
}
Reference in a new issue View git blame Copy permalink