Commit 040b6ca6 authored by Josh Coalson's avatar Josh Coalson
Browse files

add dither support

parent aad7aa72
......@@ -32,6 +32,7 @@
#endif
#include "FLAC/all.h"
#include "plugin_common/all.h"
#include "configure.h"
#include "wrap_id3.h"
#include "charset.h"
......@@ -41,6 +42,8 @@
#endif
#define min(x,y) ((x)<(y)?(x):(y))
#define FLAC__DO_DITHER
typedef struct {
FLAC__bool abort_flag;
FLAC__bool is_playing;
......@@ -77,7 +80,7 @@ InputPlugin flac_ip =
{
NULL,
NULL,
"FLAC Player v" FLAC__VERSION_STRING,
"Reference FLAC Player v" FLAC__VERSION_STRING,
FLAC_XMMS__init,
FLAC_XMMS__aboutbox,
FLAC_XMMS__configure,
......@@ -102,9 +105,9 @@ InputPlugin flac_ip =
};
#define SAMPLES_PER_WRITE 512
static FLAC__byte reservoir_[FLAC__MAX_BLOCK_SIZE * 2 * 2 * 2]; /* *2 for max bytes-per-sample, *2 for max channels, another *2 for overflow */
static FLAC__byte output_[FLAC__MAX_BLOCK_SIZE * 2 * 2]; /* *2 for max bytes-per-sample, *2 for max channels */
static unsigned reservoir_samples_ = 0;
static FLAC__int32 reservoir_[FLAC__MAX_BLOCK_SIZE * 2/*for overflow*/ * FLAC_PLUGIN__MAX_SUPPORTED_CHANNELS];
static FLAC__byte sample_buffer_[SAMPLES_PER_WRITE * FLAC_PLUGIN__MAX_SUPPORTED_CHANNELS * (24/8)]; /* (24/8) for max bytes per sample */
static unsigned wide_samples_in_reservoir_ = 0;
static FLAC__FileDecoder *decoder_ = 0;
static file_info_struct file_info_;
static pthread_t decode_thread_;
......@@ -123,7 +126,7 @@ void FLAC_XMMS__init()
flac_cfg.tag_override = FALSE;
g_free(flac_cfg.tag_format);
flac_cfg.convert_char_set = FALSE;
cfg = xmms_cfg_open_default_file();
xmms_cfg_read_boolean(cfg, "flac", "tag_override", &flac_cfg.tag_override);
......@@ -134,10 +137,10 @@ void FLAC_XMMS__init()
xmms_cfg_read_boolean(cfg, "flac", "convert_char_set", &flac_cfg.convert_char_set);
if(!xmms_cfg_read_string(cfg, "flac", "file_char_set", &flac_cfg.file_char_set))
flac_cfg.file_char_set = get_current_charset();
flac_cfg.file_char_set = FLAC_plugin__charset_get_current();
if(!xmms_cfg_read_string(cfg, "flac", "user_char_set", &flac_cfg.user_char_set))
flac_cfg.user_char_set = get_current_charset();
flac_cfg.user_char_set = FLAC_plugin__charset_get_current();
decoder_ = FLAC__file_decoder_new();
}
......@@ -157,8 +160,13 @@ void FLAC_XMMS__play_file(char *filename)
{
FILE *f;
gchar *ret;
#ifdef FLAC__DO_DITHER
const AFormat output_format = FMT_S16_NE;
#else
const AFormat output_format = file_info_.sample_format;
#endif
reservoir_samples_ = 0;
wide_samples_in_reservoir_ = 0;
audio_error_ = false;
file_info_.abort_flag = false;
file_info_.is_playing = false;
......@@ -177,7 +185,7 @@ void FLAC_XMMS__play_file(char *filename)
file_info_.is_playing = true;
if(flac_ip.output->open_audio(file_info_.sample_format, file_info_.sample_rate, file_info_.channels) == 0) {
if(flac_ip.output->open_audio(output_format, file_info_.sample_rate, file_info_.channels) == 0) {
audio_error_ = true;
safe_decoder_finish_(decoder_);
return;
......@@ -272,7 +280,7 @@ void *play_loop_(void *arg)
while(file_info_.is_playing) {
if(!file_info_.eof) {
while(reservoir_samples_ < SAMPLES_PER_WRITE) {
while(wide_samples_in_reservoir_ < SAMPLES_PER_WRITE) {
if(FLAC__file_decoder_get_state(decoder_) == FLAC__FILE_DECODER_END_OF_FILE) {
file_info_.eof = true;
break;
......@@ -284,24 +292,28 @@ void *play_loop_(void *arg)
break;
}
}
if(reservoir_samples_ > 0) {
if(wide_samples_in_reservoir_ > 0) {
const unsigned channels = file_info_.channels;
const unsigned bytes_per_sample = (file_info_.bits_per_sample+7)/8;
unsigned i, n = min(reservoir_samples_, SAMPLES_PER_WRITE), delta;
unsigned bytes = n * bytes_per_sample * channels;
for(i = 0; i < bytes; i++)
output_[i] = reservoir_[i];
delta = i;
for( ; i < reservoir_samples_*bytes_per_sample*channels; i++)
const unsigned bits_per_sample = file_info_.bits_per_sample;
#ifdef FLAC__DO_DITHER
const unsigned target_bps = 16;
#else
const unsigned target_bps = bits_per_sample;
#endif
const unsigned n = min(wide_samples_in_reservoir_, SAMPLES_PER_WRITE);
const unsigned delta = n * channels;
int bytes = (int)FLAC__plugin_common__pack_pcm_signed_little_endian(sample_buffer_, reservoir_, n, channels, bits_per_sample, target_bps);
unsigned i;
for(i = delta; i < wide_samples_in_reservoir_ * channels; i++)
reservoir_[i-delta] = reservoir_[i];
reservoir_samples_ -= n;
wide_samples_in_reservoir_ -= n;
flac_ip.add_vis_pcm(flac_ip.output->written_time(), file_info_.sample_format, channels, bytes, output_);
flac_ip.add_vis_pcm(flac_ip.output->written_time(), file_info_.sample_format, channels, bytes, sample_buffer_);
while(flac_ip.output->buffer_free() < (int)bytes && file_info_.is_playing && file_info_.seek_to_in_sec == -1)
xmms_usleep(10000);
if(file_info_.is_playing && file_info_.seek_to_in_sec == -1)
flac_ip.output->write_audio(output_, bytes);
flac_ip.output->write_audio(sample_buffer_, bytes);
}
else {
file_info_.eof = true;
......@@ -317,7 +329,7 @@ void *play_loop_(void *arg)
flac_ip.output->flush(file_info_.seek_to_in_sec * 1000);
file_info_.seek_to_in_sec = -1;
file_info_.eof = false;
reservoir_samples_ = 0;
wide_samples_in_reservoir_ = 0;
}
}
}
......@@ -371,32 +383,19 @@ void safe_decoder_delete_(FLAC__FileDecoder *decoder)
FLAC__StreamDecoderWriteStatus write_callback_(const FLAC__FileDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data)
{
file_info_struct *file_info = (file_info_struct *)client_data;
const unsigned bps = file_info->bits_per_sample, channels = file_info->channels, wide_samples = frame->header.blocksize;
unsigned wide_sample, sample, channel;
FLAC__int8 *scbuffer = (FLAC__int8*)reservoir_;
FLAC__int16 *ssbuffer = (FLAC__int16*)reservoir_;
const unsigned channels = file_info->channels, wide_samples = frame->header.blocksize;
unsigned wide_sample, offset_sample, channel;
(void)decoder;
if(file_info->abort_flag)
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
if(bps == 8) {
for(sample = reservoir_samples_*channels, wide_sample = 0; wide_sample < wide_samples; wide_sample++)
for(channel = 0; channel < channels; channel++, sample++)
scbuffer[sample] = (FLAC__int8)buffer[channel][wide_sample];
}
else if(bps == 16) {
for(sample = reservoir_samples_*channels, wide_sample = 0; wide_sample < wide_samples; wide_sample++)
for(channel = 0; channel < channels; channel++, sample++)
ssbuffer[sample] = (FLAC__int16)buffer[channel][wide_sample];
}
else {
file_info->abort_flag = true;
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
for(offset_sample = wide_samples_in_reservoir_ * channels, wide_sample = 0; wide_sample < wide_samples; wide_sample++)
for(channel = 0; channel < channels; channel++, offset_sample++)
reservoir_[offset_sample] = buffer[channel][wide_sample];
reservoir_samples_ += wide_samples;
wide_samples_in_reservoir_ += wide_samples;
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
......@@ -412,16 +411,28 @@ void metadata_callback_(const FLAC__FileDecoder *decoder, const FLAC__StreamMeta
file_info->channels = metadata->data.stream_info.channels;
file_info->sample_rate = metadata->data.stream_info.sample_rate;
#ifdef FLAC__DO_DITHER
if(file_info->bits_per_sample == 16 || file_info->bits_per_sample == 24) {
file_info->sample_format = FMT_S16_LE;
}
else {
/*@@@ need some error here like wa2: MessageBox(mod_.hMainWindow, "ERROR: plugin can only handle 16/24-bit samples\n", "ERROR: plugin can only handle 16/24-bit samples", 0); */
file_info->abort_flag = true;
return;
}
#else
if(file_info->bits_per_sample == 8) {
file_info->sample_format = FMT_S8;
}
else if(file_info->bits_per_sample == 16) {
file_info->sample_format = FMT_S16_NE;
file_info->sample_format = FMT_S16_LE;
}
else {
/*@@@ need some error here like wa2: MessageBox(mod_.hMainWindow, "ERROR: plugin can only handle 8/16-bit samples\n", "ERROR: plugin can only handle 8/16-bit samples", 0); */
file_info->abort_flag = true;
return;
}
#endif
file_info->length_in_msec = file_info->total_samples * 10 / (file_info->sample_rate / 100);
}
}
......@@ -433,4 +444,3 @@ void error_callback_(const FLAC__FileDecoder *decoder, FLAC__StreamDecoderErrorS
if(status != FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC)
file_info->abort_flag = true;
}
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