rockbox/firmware/pcm_record.c
Miika Pekkarinen a85044bf9e New scheduler, with priorities for swcodec platforms. Frequent task
switching should be more efficient and tasks are stored in linked
lists to eliminate unnecessary task switching to improve performance.
Audio should no longer skip on swcodec targets caused by too CPU
hungry UI thread or background threads.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@10958 a1c6a512-1295-4272-9138-f99709370657
2006-09-16 16:18:11 +00:00

1168 lines
32 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 by Linus Nielsen Feltzing
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "config.h"
#include "debug.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 "uda1380.h"
#endif
#ifdef 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 */
/***************************************************************************/
static bool is_recording; /* We are recording */
static bool is_paused; /* We have paused */
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 */
static int wav_file;
static char recording_filename[MAX_PATH];
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;
#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
/****************************************************************************
use 2 circular buffers of same size:
rec_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
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
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)
****************************************************************************/
#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 */
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 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);
/***************************************************************************/
static struct event_queue pcmrec_queue;
static long pcmrec_stack[2*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
#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
/*******************************************************************/
/* Functions that are not executing in the pcmrec_thread first */
/*******************************************************************/
/* Creates pcmrec_thread */
void pcm_rec_init(void)
{
queue_init(&pcmrec_queue, true);
create_thread(pcmrec_thread, pcmrec_stack, sizeof(pcmrec_stack),
pcmrec_thread_name IF_PRIO(, PRIORITY_RECORDING));
}
int audio_get_encoder_id(void)
{
return audio_enc_id;
}
/* Initializes recording:
* - Set up the UDA1380/TLV320 for recording
* - Prepare for DMA transfers
*/
void audio_init_recording(unsigned int buffer_offset)
{
(void)buffer_offset;
init_done = false;
queue_post(&pcmrec_queue, PCMREC_INIT, 0);
while(!init_done)
sleep_thread(1);
}
void audio_close_recording(void)
{
close_done = false;
queue_post(&pcmrec_queue, PCMREC_CLOSE, 0);
while(!close_done)
sleep_thread(1);
audio_remove_encoder();
}
unsigned long pcm_rec_status(void)
{
unsigned long ret = 0;
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)
ret |= AUDIO_STATUS_PRERECORD;
return ret;
}
int pcm_rec_current_bitrate(void)
{
return avrg_bit_rate;
}
unsigned long audio_recorded_time(void)
{
if (is_recording)
{
if (is_paused)
return pause_start_time - record_start_time;
else
return current_tick - record_start_time;
}
return 0;
}
unsigned long audio_num_recorded_bytes(void)
{
if (is_recording)
return num_rec_bytes;
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 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)
{
int i = 0;
unsigned long measured_rate;
const int upper_bound = sizeof(spdif_sample_rates)/sizeof(long) - 1;
/* 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.
*/
measured_rate = (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;
if (diff2 > diff1) --i;
}
return i;
}
#endif
#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;
}
#endif
#endif
/**
* 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)
{
/* TODO: */
(void)editable;
/* 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;
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;
#ifdef HAVE_SPDIF_IN
case AUDIO_SRC_SPDIF:
/* Int. when 6 samples in FIFO. PDIR2 source = ebu1RcvData */
DATAINCONTROL = 0xc038;
break;
#endif /* HAVE_SPDIF_IN */
}
sample_rate = frequency;
/* Monitoring: route the signals through the coldfire audio interface. */
SET_IIS_PLAY(0x800); /* Reset before reprogram */
#ifdef HAVE_SPDIF_IN
if (source == AUDIO_SRC_SPDIF)
{
/* 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 */
}
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]);
}
/**
* Note that microphone is mono, only left value is used
* See {uda1380,tlv320}_set_recvol() for exact ranges.
*
* @param type 0=line-in (radio), 1=mic
*
*/
void audio_set_recording_gain(int left, int right, int type)
{
//logf("rcmrec: t=%d l=%d r=%d", type, left, right);
#if defined(HAVE_UDA1380)
uda1380_set_recvol(left, right, type);
#elif defined (HAVE_TLV320)
tlv320_set_recvol(left, right, type);
#endif
}
/**
* Start recording
*
* Use audio_set_recording_options first to select recording options
*/
void audio_record(const char *filename)
{
if (is_recording)
{
logf("record while recording");
return;
}
strncpy(recording_filename, filename, MAX_PATH - 1);
recording_filename[MAX_PATH - 1] = 0;
#ifdef HAVE_SPDIF_IN
if (rec_source == AUDIO_SRC_SPDIF)
sample_rate = audio_get_spdif_sample_rate();
#endif
record_done = false;
queue_post(&pcmrec_queue, PCMREC_START, 0);
while(!record_done)
sleep_thread(1);
}
void audio_new_file(const char *filename)
{
logf("pcm_new_file");
new_file_done = false;
strncpy(recording_filename, filename, MAX_PATH - 1);
recording_filename[MAX_PATH - 1] = 0;
queue_post(&pcmrec_queue, PCMREC_NEW_FILE, 0);
while(!new_file_done)
sleep_thread(1);
logf("pcm_new_file done");
}
/**
*
*/
void audio_stop_recording(void)
{
if (!is_recording)
return;
logf("pcm_stop");
is_paused = true; /* fix pcm write ptr at current position */
stop_done = false;
queue_post(&pcmrec_queue, PCMREC_STOP, 0);
while(!stop_done)
sleep_thread(1);
logf("pcm_stop done");
}
void audio_pause_recording(void)
{
if (!is_recording)
{
logf("pause when not recording");
return;
}
if (is_paused)
{
logf("pause when paused");
return;
}
pause_done = false;
queue_post(&pcmrec_queue, PCMREC_PAUSE, 0);
while(!pause_done)
sleep_thread(1);
}
void audio_resume_recording(void)
{
if (!is_paused)
{
logf("resume when not paused");
return;
}
resume_done = false;
queue_post(&pcmrec_queue, PCMREC_RESUME, 0);
while(!resume_done)
sleep_thread(1);
}
/* 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 (left)
*left = peak_left;
if (right)
*right = peak_right;
peak_left = 0;
peak_right = 0;
}
/***************************************************************************/
/* Functions that executes in the context of pcmrec_thread */
/***************************************************************************/
/**
* 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.
*
*/
static void pcmrec_callback(bool flush)
{
int i, num_ready, size_yield;
long *enc_chunk, chunk_size;
if (!is_recording && !flush)
return;
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);
/* near full state reached: less than 5sec remaining space */
if (enc_num_chunks - num_ready < WRITE_THRESHOLD || flush)
{
bool must_boost = (boost_counter ? false : true);
logf("writing: %d (%d)", num_ready, flush);
if(must_boost)
cpu_boost(true);
size_yield = 0;
for (i=0; i<num_ready; i++)
{
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);
if (enc_set_header_callback != NULL)
enc_set_header_callback(enc_chunk, enc_chunk_size,
num_pcm_samples, false);
if (write(wav_file, enc_chunk, chunk_size) != chunk_size)
{
close_wave();
if(must_boost)
cpu_boost(false);
logf("pcmrec: write err");
is_error = true;
break;
}
num_file_bytes += chunk_size;
num_pcm_samples += enc_samp_per_chunk;
size_yield += chunk_size;
if (size_yield >= 32768)
{ /* yield when 32kB written */
size_yield = 0;
yield();
}
enc_rd_index = (enc_rd_index + 1) % enc_num_chunks;
}
if(must_boost)
cpu_boost(false);
/* sync file */
fsync(wav_file);
logf("done");
}
}
/* Abort dma transfer */
static void pcmrec_dma_stop(void)
{
DCR1 = 0;
error_count++;
DSR1 = 1; /* Clear interrupt */
IPR |= (1<<15); /* Clear pending interrupt request */
logf("dma1 stopped");
}
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 */
/* Start the DMA transfer.. */
#ifdef HAVE_SPDIF_IN
INTERRUPTCLEAR = 0x03c00000;
#endif
/* 16Byte transfers prevents from sporadic errors during cpu_boost() */
DCR1 = DMA_INT | DMA_EEXT | DMA_CS | DMA_DINC | DMA_DSIZE(3) | DMA_START;
logf("dma1 started");
}
/* DMA1 Interrupt is called when the DMA has finished transfering a chunk */
void DMA1(void) __attribute__ ((interrupt_handler, section(".icode")));
void DMA1(void)
{
int res = DSR1;
DSR1 = 1; /* Clear interrupt */
if (res & 0x70)
{
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;
/* 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 */
{
INTERRUPTCLEAR = 0x03c00000;
error_count++;
logf("spdif err");
DAR1 = (unsigned long)GET_CHUNK(write_pos); /* Destination address */
BCR1 = CHUNK_SIZE;
}
#endif
else
{
long peak_l, peak_r;
long *ptr, j;
ptr = GET_CHUNK(write_pos);
if (!is_paused) /* advance write position */
write_pos = (write_pos + CHUNK_SIZE) & CHUNK_MASK;
DAR1 = (unsigned long)GET_CHUNK(write_pos); /* Destination address */
BCR1 = CHUNK_SIZE;
peak_l = peak_r = 0;
/* only peak every 4th sample */
for (j=0; j<CHUNK_SIZE/4; j+=4)
{
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
}
peak_left = (int)(peak_l >> 16);
peak_right = (int)(peak_r >> 16);
}
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);
if (wav_file < 0)
{
wav_file = -1;
logf("rec: create failed: %d", wav_file);
is_error = true;
return -1;
}
/* add main file header (enc_head_size=0 for encoders without) */
if (enc_head_size != write(wav_file, enc_head_buffer, enc_head_size))
{
close(wav_file);
wav_file = -1;
logf("rec: write failed");
is_error = true;
return -1;
}
return 0;
}
/* 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;
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));
enc_set_header_callback(head, size_read, num_pcm_samples, true);
lseek(wav_file, 0, SEEK_SET);
write(wav_file, head, size_read);
}
close(wav_file);
wav_file = -1;
}
}
static void pcmrec_start(void)
{
long max_pre_chunks, pre_ticks, max_pre_ticks;
logf("pcmrec_start");
if (is_recording)
{
logf("already recording");
record_done = true;
return;
}
if (wav_file != -1)
close_wave();
if (start_wave() != 0)
{
/* failed to create the file */
record_done = true;
return;
}
/* 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 +
enc_num_chunks) % enc_num_chunks;
record_start_time = current_tick - pre_ticks;
num_rec_bytes = enc_num_chunks * CHUNK_SIZE;
num_file_bytes = 0;
num_pcm_samples = 0;
pause_start_time = 0;
is_paused = false;
is_recording = true;
record_done = true;
}
static void pcmrec_stop(void)
{
logf("pcmrec_stop");
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();
}
is_paused = false;
stop_done = true;
logf("pcmrec_stop done");
}
static void pcmrec_new_file(void)
{
logf("pcmrec_new_file");
if (!is_recording)
{
logf("not recording");
new_file_done = true;
return;
}
/* 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;
if (is_paused)
pause_start_time = record_start_time;
/* Flush what we got in buffers to file */
pcmrec_callback(true);
close_wave();
num_rec_bytes = 0;
num_file_bytes = 0;
num_pcm_samples = 0;
/* start the new file */
if (start_wave() != 0)
{
logf("new_file failed");
pcmrec_stop();
}
new_file_done = true;
logf("pcmrec_new_file done");
}
static void pcmrec_pause(void)
{
logf("pcmrec_pause");
if (!is_recording)
{
logf("pause: not recording");
pause_done = true;
return;
}
pause_start_time = current_tick;
is_paused = true;
pause_done = true;
logf("pcmrec_pause done");
}
static void pcmrec_resume(void)
{
logf("pcmrec_resume");
if (!is_paused)
{
logf("resume: not paused");
resume_done = true;
return;
}
is_paused = false;
is_recording = true;
/* 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;
logf("pcmrec_resume done");
}
/**
* audio_init_recording calls this function using PCMREC_INIT
*
*/
static void pcmrec_init(void)
{
wav_file = -1;
read_pos = 0;
write_pos = 0;
enc_wr_index = 0;
enc_rd_index = 0;
avrg_bit_rate = 0;
curr_bit_rate = 0;
curr_chunk_cnt = 0;
peak_left = 0;
peak_right = 0;
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;
}
static void pcmrec_close(void)
{
DMAROUTE = (DMAROUTE & 0xffff00ff);
ICR7 = 0x00; /* Disable interrupt */
IMR |= (1<<15); /* bit 15 is DMA1 */
pcmrec_dma_stop();
/* Reset PDIR2 data flow */
DATAINCONTROL = 0x200;
close_done = true;
init_done = false;
}
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);
switch (ev.id)
{
case PCMREC_INIT:
pcmrec_init();
break;
case PCMREC_CLOSE:
pcmrec_close();
break;
case PCMREC_START:
pcmrec_start();
break;
case PCMREC_STOP:
pcmrec_stop();
break;
case PCMREC_PAUSE:
pcmrec_pause();
break;
case PCMREC_RESUME:
pcmrec_resume();
break;
case PCMREC_NEW_FILE:
pcmrec_new_file();
break;
case SYS_TIMEOUT:
pcmrec_callback(false);
break;
case SYS_USB_CONNECTED:
if (!is_recording)
{
pcmrec_close();
usb_acknowledge(SYS_USB_CONNECTED_ACK);
usb_wait_for_disconnect(&pcmrec_queue);
}
break;
}
}
logf("thread pcmrec done");
}
/* Select VINL & VINR source: 0=Line-in, 1=FM Radio */
void pcm_rec_mux(int source)
{
#ifdef IRIVER_H300_SERIES
if(source == 0)
and_l(~0x40000000, &GPIO_OUT); /* Line In */
else
or_l(0x40000000, &GPIO_OUT); /* FM radio */
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 */
or_l(0x00800000, &GPIO_ENABLE);
or_l(0x00800000, &GPIO_FUNCTION);
#elif defined(IAUDIO_X5)
if(source == 0)
or_l((1<<29), &GPIO_OUT); /* Line In */
else
and_l(~(1<<29), &GPIO_OUT); /* FM radio */
or_l((1<<29), &GPIO_ENABLE);
or_l((1<<29), &GPIO_FUNCTION);
/* iAudio x5 */
#endif
}
/****************************************************************************/
/* */
/* following functions will be called by the encoder codec */
/* */
/****************************************************************************/
/* pass the encoder buffer pointer/size, mono/stereo, quality to the encoder */
void enc_get_inputs(int *buffer_size, int *channels, int *quality)
{
*buffer_size = enc_buffer_size;
*channels = enc_channels;
*quality = enc_quality;
}
/* 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 */
}
/* allocate encoder chunk */
unsigned int *enc_alloc_chunk(void)
{
return (unsigned int*)(enc_buffer + enc_wr_index * enc_chunk_size);
}
/* free previously allocated encoder chunk */
void enc_free_chunk(void)
{
unsigned long *enc_chunk;
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;
/* advance enc_wr_index to the next chunk */
enc_wr_index = (enc_wr_index + 1) % enc_num_chunks;
/* buffer full: advance enc_rd_index (for prerecording purpose) */
if (enc_rd_index == enc_wr_index)
{
enc_rd_index = (enc_rd_index + 1) % enc_num_chunks;
}
}
/* checks near empty state on wav input buffer */
int enc_wavbuf_near_empty(void)
{
/* less than 1sec raw data? => unboost encoder */
if (((write_pos - read_pos) & CHUNK_MASK) < 44100*4)
return 1;
else
return 0;
}
/* passes a pointer to next chunk of unprocessed wav data */
char *enc_get_wav_data(int size)
{
char *ptr;
int avail;
/* limit the requested pcm data size */
if(size > MAX_FEED_SIZE)
size = MAX_FEED_SIZE;
avail = (write_pos - read_pos) & CHUNK_MASK;
if (avail >= size)
{
ptr = rec_buffer + read_pos;
read_pos = (read_pos + size) & 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);
wav_queue_empty = false;
return ptr;
}
wav_queue_empty = true;
return NULL;
}