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-rw-r--r--apps/SOURCES2
-rw-r--r--apps/codecs.c13
-rw-r--r--apps/codecs.h5
-rw-r--r--apps/codecs/wav.c13
-rw-r--r--apps/debug_menu.c10
-rw-r--r--apps/dsp.c195
-rw-r--r--apps/main.c3
-rw-r--r--apps/pcmbuf.c104
-rw-r--r--apps/pcmbuf.h7
-rw-r--r--apps/playback.c507
-rw-r--r--apps/playback.h1
-rw-r--r--apps/playlist.c7
-rw-r--r--apps/talk.c111
-rw-r--r--apps/talk.h1
-rw-r--r--firmware/mp3_playback.c16
-rw-r--r--firmware/sound.c7
-rw-r--r--uisimulator/common/stubs.c2
17 files changed, 774 insertions, 230 deletions
diff --git a/apps/SOURCES b/apps/SOURCES
index c1f3dfc88e..e8fd2d2ddb 100644
--- a/apps/SOURCES
+++ b/apps/SOURCES
@@ -22,7 +22,7 @@ settings_menu.c
sleeptimer.c
sound_menu.c
status.c
-#ifndef SIMULATOR
+#if !defined(SIMULATOR) || CONFIG_HWCODEC == MASNONE
talk.c
#endif
tree.c
diff --git a/apps/codecs.c b/apps/codecs.c
index b1d30868bc..004d4681f7 100644
--- a/apps/codecs.c
+++ b/apps/codecs.c
@@ -54,7 +54,7 @@
#ifdef SIMULATOR
#if CONFIG_HWCODEC == MASNONE
-static unsigned char codecbuf[CODEC_SIZE];
+unsigned char codecbuf[CODEC_SIZE];
#endif
void *sim_codec_load_ram(char* codecptr, int size,
void* ptr2, int bufwrap, int *pd);
@@ -68,6 +68,8 @@ extern void* plugin_get_audio_buffer(int *buffer_size);
static int codec_test(int api_version, int model, int memsize);
+struct codec_api ci_voice;
+
struct codec_api ci = {
CODEC_API_VERSION,
codec_test,
@@ -247,7 +249,8 @@ struct codec_api ci = {
NULL,
};
-int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap)
+int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap,
+ struct codec_api *api)
{
enum codec_status (*codec_start)(const struct codec_api* api);
int status;
@@ -277,7 +280,7 @@ int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap)
#endif /* SIMULATOR */
invalidate_icache();
- status = codec_start(&ci);
+ status = codec_start(api);
#ifdef SIMULATOR
sim_codec_close(pd);
#endif
@@ -285,7 +288,7 @@ int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap)
return status;
}
-int codec_load_file(const char *plugin)
+int codec_load_file(const char *plugin, struct codec_api *api)
{
char msgbuf[80];
int fd;
@@ -309,7 +312,7 @@ int codec_load_file(const char *plugin)
return CODEC_ERROR;
}
- return codec_load_ram(codecbuf, (size_t)rc, NULL, 0);
+ return codec_load_ram(codecbuf, (size_t)rc, NULL, 0, api);
}
static int codec_test(int api_version, int model, int memsize)
diff --git a/apps/codecs.h b/apps/codecs.h
index b2cf9e5a1e..3b8e1d8394 100644
--- a/apps/codecs.h
+++ b/apps/codecs.h
@@ -331,8 +331,9 @@ struct codec_api {
/* defined by the codec loader (codec.c) */
#if CONFIG_HWCODEC == MASNONE
-int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap);
-int codec_load_file(const char* codec);
+int codec_load_ram(char* codecptr, int size, void* ptr2, int bufwrap,
+ struct codec_api *api);
+int codec_load_file(const char* codec, struct codec_api *api);
#endif
/* defined by the codec */
diff --git a/apps/codecs/wav.c b/apps/codecs/wav.c
index 527d33d286..d2ae7bd6c3 100644
--- a/apps/codecs/wav.c
+++ b/apps/codecs/wav.c
@@ -71,15 +71,12 @@ enum codec_status codec_start(struct codec_api* api)
return CODEC_ERROR;
}
- while (!rb->taginfo_ready)
+ while (!*rb->taginfo_ready)
rb->yield();
-
- if (rb->id3->frequency != NATIVE_FREQUENCY) {
- rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
- rb->configure(CODEC_DSP_ENABLE, (bool *)true);
- } else {
- rb->configure(CODEC_DSP_ENABLE, (bool *)false);
- }
+
+ /* Always enable DSP to support voice ui. */
+ rb->configure(CODEC_DSP_ENABLE, (bool *)true);
+ rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
/* FIX: Correctly parse WAV header - we assume canonical 44-byte header */
diff --git a/apps/debug_menu.c b/apps/debug_menu.c
index 86e02a1e84..b4cd05699c 100644
--- a/apps/debug_menu.c
+++ b/apps/debug_menu.c
@@ -208,8 +208,8 @@ bool dbg_audio_thread(void)
}
#else
extern size_t audiobuffer_free;
-extern int codecbuflen;
-extern int codecbufused;
+extern int filebuflen;
+extern int filebufused;
extern int track_count;
static int ticks, boost_ticks;
@@ -260,12 +260,12 @@ bool dbg_audio_thread(void)
bufsize-audiobuffer_free, HORIZONTAL);
line++;
- snprintf(buf, sizeof(buf), "codec: %d/%d", codecbufused, codecbuflen);
+ snprintf(buf, sizeof(buf), "codec: %d/%d", filebufused, filebuflen);
lcd_puts(0, line++, buf);
/* Playable space left */
- scrollbar(0, line*8, LCD_WIDTH, 6, codecbuflen, 0,
- codecbufused, HORIZONTAL);
+ scrollbar(0, line*8, LCD_WIDTH, 6, filebuflen, 0,
+ filebufused, HORIZONTAL);
line++;
snprintf(buf, sizeof(buf), "track count: %d", track_count);
diff --git a/apps/dsp.c b/apps/dsp.c
index 2ba7fb99f6..8064a881a3 100644
--- a/apps/dsp.c
+++ b/apps/dsp.c
@@ -132,9 +132,12 @@ struct dither_data
long random;
};
-static struct dsp_config dsp IDATA_ATTR;
+static struct dsp_config dsp_conf[2] IDATA_ATTR;
static struct dither_data dither_data[2] IDATA_ATTR;
-static struct resample_data resample_data[2] IDATA_ATTR;
+static struct resample_data resample_data[2][2] IDATA_ATTR;
+
+extern int current_codec;
+struct dsp_config *dsp;
/* The internal format is 32-bit samples, non-interleaved, stereo. This
* format is similar to the raw output from several codecs, so the amount
@@ -155,20 +158,20 @@ static int convert_to_internal(char* src[], int count, long* dst[])
{
count = MIN(SAMPLE_BUF_SIZE / 2, count);
- if ((dsp.sample_depth <= NATIVE_DEPTH)
- || (dsp.stereo_mode == STEREO_INTERLEAVED))
+ if ((dsp->sample_depth <= NATIVE_DEPTH)
+ || (dsp->stereo_mode == STEREO_INTERLEAVED))
{
dst[0] = &sample_buf[0];
- dst[1] = (dsp.stereo_mode == STEREO_MONO)
+ dst[1] = (dsp->stereo_mode == STEREO_MONO)
? dst[0] : &sample_buf[SAMPLE_BUF_SIZE / 2];
}
else
{
dst[0] = (long*) src[0];
- dst[1] = (long*) ((dsp.stereo_mode == STEREO_MONO) ? src[0] : src[1]);
+ dst[1] = (long*) ((dsp->stereo_mode == STEREO_MONO) ? src[0] : src[1]);
}
- if (dsp.sample_depth <= NATIVE_DEPTH)
+ if (dsp->sample_depth <= NATIVE_DEPTH)
{
short* s0 = (short*) src[0];
long* d0 = dst[0];
@@ -176,7 +179,7 @@ static int convert_to_internal(char* src[], int count, long* dst[])
int scale = WORD_SHIFT;
int i;
- if (dsp.stereo_mode == STEREO_INTERLEAVED)
+ if (dsp->stereo_mode == STEREO_INTERLEAVED)
{
for (i = 0; i < count; i++)
{
@@ -184,7 +187,7 @@ static int convert_to_internal(char* src[], int count, long* dst[])
*d1++ = *s0++ << scale;
}
}
- else if (dsp.stereo_mode == STEREO_NONINTERLEAVED)
+ else if (dsp->stereo_mode == STEREO_NONINTERLEAVED)
{
short* s1 = (short*) src[1];
@@ -202,7 +205,7 @@ static int convert_to_internal(char* src[], int count, long* dst[])
}
}
}
- else if (dsp.stereo_mode == STEREO_INTERLEAVED)
+ else if (dsp->stereo_mode == STEREO_INTERLEAVED)
{
long* s0 = (long*) src[0];
long* d0 = dst[0];
@@ -216,18 +219,18 @@ static int convert_to_internal(char* src[], int count, long* dst[])
}
}
- if (dsp.stereo_mode == STEREO_NONINTERLEAVED)
+ if (dsp->stereo_mode == STEREO_NONINTERLEAVED)
{
- src[0] += count * dsp.sample_bytes;
- src[1] += count * dsp.sample_bytes;
+ src[0] += count * dsp->sample_bytes;
+ src[1] += count * dsp->sample_bytes;
}
- else if (dsp.stereo_mode == STEREO_INTERLEAVED)
+ else if (dsp->stereo_mode == STEREO_INTERLEAVED)
{
- src[0] += count * dsp.sample_bytes * 2;
+ src[0] += count * dsp->sample_bytes * 2;
}
else
{
- src[0] += count * dsp.sample_bytes;
+ src[0] += count * dsp->sample_bytes;
}
return count;
@@ -310,29 +313,33 @@ static inline int resample(long* src[], int count)
{
long new_count;
- if (dsp.frequency != NATIVE_FREQUENCY)
+ if (dsp->frequency != NATIVE_FREQUENCY)
{
long* d0 = &resample_buf[0];
/* Only process the second channel if needed. */
long* d1 = (src[0] == src[1]) ? d0
: &resample_buf[RESAMPLE_BUF_SIZE / 2];
- if (dsp.frequency < NATIVE_FREQUENCY)
+ if (dsp->frequency < NATIVE_FREQUENCY)
{
- new_count = upsample(d0, src[0], count, &resample_data[0]);
+ new_count = upsample(d0, src[0], count,
+ &resample_data[current_codec][0]);
if (d0 != d1)
{
- upsample(d1, src[1], count, &resample_data[1]);
+ upsample(d1, src[1], count,
+ &resample_data[current_codec][1]);
}
}
else
{
- new_count = downsample(d0, src[0], count, &resample_data[0]);
+ new_count = downsample(d0, src[0], count,
+ &resample_data[current_codec][0]);
if (d0 != d1)
{
- downsample(d1, src[1], count, &resample_data[1]);
+ downsample(d1, src[1], count,
+ &resample_data[current_codec][1]);
}
}
@@ -389,8 +396,8 @@ static long dither_sample(long sample, long bias, long mask,
/* Clip and quantize */
- min = dsp.clip_min;
- max = dsp.clip_max;
+ min = dsp->clip_min;
+ max = dsp->clip_max;
sample = clip_sample(sample, min, max);
output = clip_sample(output, min, max) & ~mask;
@@ -407,13 +414,13 @@ static long dither_sample(long sample, long bias, long mask,
*/
static void apply_gain(long* src[], int count)
{
- if (dsp.replaygain)
+ if (dsp->replaygain)
{
long* s0 = src[0];
long* s1 = src[1];
long* d0 = &sample_buf[0];
long* d1 = (s0 == s1) ? d0 : &sample_buf[SAMPLE_BUF_SIZE / 2];
- long gain = dsp.replaygain;
+ long gain = dsp->replaygain;
long s;
long i;
@@ -442,11 +449,11 @@ static void write_samples(short* dst, long* src[], int count)
{
long* s0 = src[0];
long* s1 = src[1];
- int scale = dsp.frac_bits + 1 - NATIVE_DEPTH;
+ int scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
- if (dsp.dither_enabled)
+ if (dsp->dither_enabled)
{
- long bias = (1L << (dsp.frac_bits - NATIVE_DEPTH));
+ long bias = (1L << (dsp->frac_bits - NATIVE_DEPTH));
long mask = (1L << scale) - 1;
while (count-- > 0)
@@ -459,8 +466,8 @@ static void write_samples(short* dst, long* src[], int count)
}
else
{
- long min = dsp.clip_min;
- long max = dsp.clip_max;
+ long min = dsp->clip_min;
+ long max = dsp->clip_max;
while (count-- > 0)
{
@@ -482,10 +489,13 @@ long dsp_process(char* dst, char* src[], long size)
{
long* tmp[2];
long written = 0;
- long factor = (dsp.stereo_mode != STEREO_MONO) ? 2 : 1;
+ long factor;
int samples;
- size /= dsp.sample_bytes * factor;
+ dsp = &dsp_conf[current_codec];
+
+ factor = (dsp->stereo_mode != STEREO_MONO) ? 2 : 1;
+ size /= dsp->sample_bytes * factor;
INIT();
dsp_set_replaygain(false);
@@ -513,21 +523,23 @@ long dsp_process(char* dst, char* src[], long size)
/* dsp_input_size MUST be called afterwards */
long dsp_output_size(long size)
{
- if (dsp.sample_depth > NATIVE_DEPTH)
+ dsp = &dsp_conf[current_codec];
+
+ if (dsp->sample_depth > NATIVE_DEPTH)
{
size /= 2;
}
- if (dsp.frequency != NATIVE_FREQUENCY)
+ if (dsp->frequency != NATIVE_FREQUENCY)
{
size = (long) ((((unsigned long) size * NATIVE_FREQUENCY)
- + (dsp.frequency - 1)) / dsp.frequency);
+ + (dsp->frequency - 1)) / dsp->frequency);
}
/* round to the next multiple of 2 (these are shorts) */
size = (size + 1) & ~1;
- if (dsp.stereo_mode == STEREO_MONO)
+ if (dsp->stereo_mode == STEREO_MONO)
{
size *= 2;
}
@@ -547,25 +559,28 @@ long dsp_output_size(long size)
*/
long dsp_input_size(long size)
{
+ dsp = &dsp_conf[current_codec];
+
/* convert to number of output stereo samples. */
size /= 2;
/* Mono means we need half input samples to fill the output buffer */
- if (dsp.stereo_mode == STEREO_MONO)
+ if (dsp->stereo_mode == STEREO_MONO)
size /= 2;
/* size is now the number of resampled input samples. Convert to
original input samples. */
- if (dsp.frequency != NATIVE_FREQUENCY)
+ if (dsp->frequency != NATIVE_FREQUENCY)
{
/* Use the real resampling delta =
- * (unsigned long) dsp.frequency * 65536 / NATIVE_FREQUENCY, and
+ * (unsigned long) dsp->frequency * 65536 / NATIVE_FREQUENCY, and
* round towards zero to avoid buffer overflows. */
- size = ((unsigned long)size * resample_data[0].delta) >> 16;
+ size = ((unsigned long)size *
+ resample_data[current_codec][0].delta) >> 16;
}
/* Convert back to bytes. */
- if (dsp.sample_depth > NATIVE_DEPTH)
+ if (dsp->sample_depth > NATIVE_DEPTH)
size *= 4;
else
size *= 2;
@@ -575,90 +590,96 @@ long dsp_input_size(long size)
int dsp_stereo_mode(void)
{
- return dsp.stereo_mode;
+ dsp = &dsp_conf[current_codec];
+
+ return dsp->stereo_mode;
}
bool dsp_configure(int setting, void *value)
{
+ dsp = &dsp_conf[current_codec];
+
switch (setting)
{
case DSP_SET_FREQUENCY:
- memset(resample_data, 0, sizeof(resample_data));
+ memset(&resample_data[current_codec][0], 0,
+ sizeof(struct resample_data) * 2);
/* Fall through!!! */
case DSP_SWITCH_FREQUENCY:
- dsp.frequency = ((int) value == 0) ? NATIVE_FREQUENCY : (int) value;
- resample_data[0].delta = resample_data[1].delta =
- (unsigned long) dsp.frequency * 65536 / NATIVE_FREQUENCY;
+ dsp->frequency = ((int) value == 0) ? NATIVE_FREQUENCY : (int) value;
+ resample_data[current_codec][0].delta =
+ resample_data[current_codec][1].delta =
+ (unsigned long) dsp->frequency * 65536 / NATIVE_FREQUENCY;
break;
case DSP_SET_CLIP_MIN:
- dsp.clip_min = (long) value;
+ dsp->clip_min = (long) value;
break;
case DSP_SET_CLIP_MAX:
- dsp.clip_max = (long) value;
+ dsp->clip_max = (long) value;
break;
case DSP_SET_SAMPLE_DEPTH:
- dsp.sample_depth = (long) value;
+ dsp->sample_depth = (long) value;
- if (dsp.sample_depth <= NATIVE_DEPTH)
+ if (dsp->sample_depth <= NATIVE_DEPTH)
{
- dsp.frac_bits = WORD_FRACBITS;
- dsp.sample_bytes = sizeof(short);
- dsp.clip_max = ((1 << WORD_FRACBITS) - 1);
- dsp.clip_min = -((1 << WORD_FRACBITS));
+ dsp->frac_bits = WORD_FRACBITS;
+ dsp->sample_bytes = sizeof(short);
+ dsp->clip_max = ((1 << WORD_FRACBITS) - 1);
+ dsp->clip_min = -((1 << WORD_FRACBITS));
}
else
{
- dsp.frac_bits = (long) value;
- dsp.sample_bytes = sizeof(long);
+ dsp->frac_bits = (long) value;
+ dsp->sample_bytes = sizeof(long);
}
break;
case DSP_SET_STEREO_MODE:
- dsp.stereo_mode = (long) value;
+ dsp->stereo_mode = (long) value;
break;
case DSP_RESET:
- dsp.dither_enabled = false;
- dsp.stereo_mode = STEREO_NONINTERLEAVED;
- dsp.clip_max = ((1 << WORD_FRACBITS) - 1);
- dsp.clip_min = -((1 << WORD_FRACBITS));
- dsp.track_gain = 0;
- dsp.album_gain = 0;
- dsp.track_peak = 0;
- dsp.album_peak = 0;
- dsp.frequency = NATIVE_FREQUENCY;
- dsp.sample_depth = NATIVE_DEPTH;
- dsp.frac_bits = WORD_FRACBITS;
- dsp.new_gain = true;
+ dsp->dither_enabled = false;
+ dsp->stereo_mode = STEREO_NONINTERLEAVED;
+ dsp->clip_max = ((1 << WORD_FRACBITS) - 1);
+ dsp->clip_min = -((1 << WORD_FRACBITS));
+ dsp->track_gain = 0;
+ dsp->album_gain = 0;
+ dsp->track_peak = 0;
+ dsp->album_peak = 0;
+ dsp->frequency = NATIVE_FREQUENCY;
+ dsp->sample_depth = NATIVE_DEPTH;
+ dsp->frac_bits = WORD_FRACBITS;
+ dsp->new_gain = true;
break;
case DSP_DITHER:
memset(dither_data, 0, sizeof(dither_data));
- dsp.dither_enabled = (bool) value;
+ dsp->dither_enabled = (bool) value;
break;
case DSP_SET_TRACK_GAIN:
- dsp.track_gain = (long) value;
- dsp.new_gain = true;
+ dsp->track_gain = (long) value;
+ dsp->new_gain = true;
break;
case DSP_SET_ALBUM_GAIN:
- dsp.album_gain = (long) value;
- dsp.new_gain = true;
+ dsp->album_gain = (long) value;
+ dsp->new_gain = true;
break;
case DSP_SET_TRACK_PEAK:
- dsp.track_peak = (long) value;
- dsp.new_gain = true;
+ dsp->track_peak = (long) value;
+ dsp->new_gain = true;
break;
case DSP_SET_ALBUM_PEAK:
- dsp.album_peak = (long) value;
- dsp.new_gain = true;
+ dsp->album_peak = (long) value;
+ dsp->new_gain = true;
break;
default:
@@ -670,11 +691,13 @@ bool dsp_configure(int setting, void *value)
void dsp_set_replaygain(bool always)
{
- if (always || dsp.new_gain)
+ dsp = &dsp_conf[current_codec];
+
+ if (always || dsp->new_gain)
{
long gain = 0;
- dsp.new_gain = false;
+ dsp->new_gain = false;
if (global_settings.replaygain || global_settings.replaygain_noclip)
{
@@ -682,8 +705,8 @@ void dsp_set_replaygain(bool always)
if (global_settings.replaygain)
{
- gain = (global_settings.replaygain_track || !dsp.album_gain)
- ? dsp.track_gain : dsp.album_gain;
+ gain = (global_settings.replaygain_track || !dsp->album_gain)
+ ? dsp->track_gain : dsp->album_gain;
if (global_settings.replaygain_preamp)
{
@@ -694,8 +717,8 @@ void dsp_set_replaygain(bool always)
}
}
- peak = (global_settings.replaygain_track || !dsp.album_peak)
- ? dsp.track_peak : dsp.album_peak;
+ peak = (global_settings.replaygain_track || !dsp->album_peak)
+ ? dsp->track_peak : dsp->album_peak;
if (gain == 0)
{
@@ -718,6 +741,6 @@ void dsp_set_replaygain(bool always)
}
/* Store in S8.23 format to simplify calculations. */
- dsp.replaygain = gain >> 1;
+ dsp->replaygain = gain >> 1;
}
}
diff --git a/apps/main.c b/apps/main.c
index 90be703c6b..55897de5f5 100644
--- a/apps/main.c
+++ b/apps/main.c
@@ -129,6 +129,9 @@ void init(void)
global_settings.mdb_enable,
global_settings.superbass);
button_clear_queue(); /* Empty the keyboard buffer */
+#if CONFIG_HWCODEC == MASNONE
+ talk_init();
+#endif
}
#else
diff --git a/apps/pcmbuf.c b/apps/pcmbuf.c
index 691f8d5a19..5f78901a56 100644
--- a/apps/pcmbuf.c
+++ b/apps/pcmbuf.c
@@ -34,6 +34,7 @@
#include "buffer.h"
#include "settings.h"
#include "audio.h"
+#include "dsp.h"
#define CHUNK_SIZE PCMBUF_GUARD
/* Must be a power of 2 */
@@ -86,9 +87,11 @@ struct pcmbufdesc
volatile int pcmbuf_read_index;
volatile int pcmbuf_write_index;
int pcmbuf_unplayed_bytes;
+int pcmbuf_mix_used_bytes;
int pcmbuf_watermark;
void (*pcmbuf_watermark_event)(int bytes_left);
static int last_chunksize;
+static long mixpos = 0;
static void pcmbuf_boost(bool state)
{
@@ -173,6 +176,7 @@ bool pcmbuf_add_chunk(void *addr, int size, void (*callback)(void))
pcmbuffers[pcmbuf_write_index].callback = callback;
pcmbuf_write_index = (pcmbuf_write_index+1) & NUM_PCM_BUFFERS_MASK;
pcmbuf_unplayed_bytes += size;
+ pcmbuf_mix_used_bytes = MAX(0, pcmbuf_mix_used_bytes - size);
return true;
}
else
@@ -254,6 +258,7 @@ void pcmbuf_play_stop(void)
pcm_play_stop();
last_chunksize = 0;
pcmbuf_unplayed_bytes = 0;
+ pcmbuf_mix_used_bytes = 0;
pcmbuf_read_index = 0;
pcmbuf_write_index = 0;
audiobuffer_pos = 0;
@@ -297,6 +302,13 @@ void pcmbuf_flush_audio(void)
crossfade_init = true;
}
+/* Force playback. */
+void pcmbuf_play_start(void)
+{
+ if (!pcm_is_playing() && pcmbuf_unplayed_bytes)
+ pcm_play_data(pcmbuf_callback);
+}
+
void pcmbuf_flush_fillpos(void)
{
int copy_n;
@@ -562,6 +574,98 @@ bool pcmbuf_insert_buffer(char *buf, long length)
return true;
}
+/* Generates a constant square wave sound with a given frequency
+ in Hertz for a duration in milliseconds. */
+void pcmbuf_beep(int frequency, int duration, int amplitude)
+{
+ int state = 0, count = 0;
+ int interval = NATIVE_FREQUENCY / frequency;
+ int pos;
+ short *buf = (short *)audiobuffer;
+ int bufsize = pcmbuf_size / 2;
+
+ /* FIXME: Should start playback. */
+ //if (pcmbuf_unplayed_bytes * 1000 < 4 * NATIVE_FREQUENCY * duration)
+ // return ;
+
+ pos = (audiobuffer_pos - pcmbuf_unplayed_bytes) / 2;
+ if (pos < 0)
+ pos += bufsize;
+
+ duration = NATIVE_FREQUENCY / 1000 * duration;
+ while (duration-- > 0)
+ {
+ if (state) {
+ buf[pos] = MIN(MAX(buf[pos] + amplitude, -32768), 32767);
+ if (++pos >= bufsize)
+ pos = 0;
+ buf[pos] = MIN(MAX(buf[pos] + amplitude, -32768), 32767);
+ } else {
+ buf[pos] = MIN(MAX(buf[pos] - amplitude, -32768), 32767);
+ if (++pos >= bufsize)
+ pos = 0;
+ buf[pos] = MIN(MAX(buf[pos] - amplitude, -32768), 32767);
+ }
+
+ if (++count >= interval)
+ {
+ count = 0;
+ if (state)
+ state = 0;
+ else
+ state = 1;
+ }
+ pos++;
+ if (pos >= bufsize)
+ pos = 0;
+ }
+}
+
+/* Returns pcm buffer usage in percents (0 to 100). */
+int pcmbuf_usage(void)
+{
+ return pcmbuf_unplayed_bytes * 100 / pcmbuf_size;
+}
+
+int pcmbuf_mix_usage(void)
+{
+ return pcmbuf_mix_used_bytes * 100 / pcmbuf_unplayed_bytes;
+}
+
+void pcmbuf_reset_mixpos(void)
+{
+ int bufsize = pcmbuf_size / 2;
+
+ pcmbuf_mix_used_bytes = 0;
+ mixpos = (audiobuffer_pos - pcmbuf_unplayed_bytes) / 2;
+ if (mixpos < 0)
+ mixpos += bufsize;
+ if (mixpos >= bufsize)
+ mixpos -= bufsize;
+}
+
+void pcmbuf_mix(char *buf, long length)
+{
+ short *ibuf = (short *)buf;
+ short *obuf = (short *)audiobuffer;
+ int bufsize = pcmbuf_size / 2;
+
+ if (pcmbuf_mix_used_bytes == 0)
+ pcmbuf_reset_mixpos();
+
+ pcmbuf_mix_used_bytes += length;
+ length /= 2;
+
+ while (length-- > 0) {
+ obuf[mixpos] = MIN(MAX(obuf[mixpos] + *ibuf*4, -32768), 32767);
+
+ ibuf++;
+ mixpos++;
+ if (mixpos >= bufsize)
+ mixpos = 0;
+ }
+}
+
void pcmbuf_crossfade_enable(bool on_off)
{
crossfade_enabled = on_off;
diff --git a/apps/pcmbuf.h b/apps/pcmbuf.h
index 6381dbc27e..f2533defe6 100644
--- a/apps/pcmbuf.h
+++ b/apps/pcmbuf.h
@@ -36,6 +36,7 @@ void pcmbuf_set_watermark(int numbytes, void (*callback)(int bytes_left));
void pcmbuf_set_boost_mode(bool state);
bool pcmbuf_is_lowdata(void);
void pcmbuf_flush_audio(void);
+void pcmbuf_play_start(void);
bool pcmbuf_crossfade_init(int type);
void pcmbuf_add_event(void (*event_handler)(void));
unsigned int pcmbuf_get_latency(void);
@@ -45,4 +46,10 @@ void* pcmbuf_request_buffer(long length, long *realsize);
bool pcmbuf_is_crossfade_enabled(void);
void pcmbuf_crossfade_enable(bool on_off);
+int pcmbuf_usage(void);
+int pcmbuf_mix_usage(void);
+void pcmbuf_beep(int frequency, int duration, int amplitude);
+void pcmbuf_reset_mixpos(void);
+void pcmbuf_mix(char *buf, long length);
+
#endif
diff --git a/apps/playback.c b/apps/playback.c
index e12b01ee55..8829757949 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -60,18 +60,20 @@
#include "misc.h"
#include "sound.h"
#include "metadata.h"
+#include "talk.h"
-static volatile bool codec_loaded;
+static volatile bool audio_codec_loaded;
+static volatile bool voice_codec_loaded;
static volatile bool playing;
static volatile bool paused;
-#define CODEC_VORBIS "/.rockbox/codecs/vorbis.codec";
-#define CODEC_MPA_L3 "/.rockbox/codecs/mpa.codec";
-#define CODEC_FLAC "/.rockbox/codecs/flac.codec";
-#define CODEC_WAV "/.rockbox/codecs/wav.codec";
-#define CODEC_A52 "/.rockbox/codecs/a52.codec";
-#define CODEC_MPC "/.rockbox/codecs/mpc.codec";
-#define CODEC_WAVPACK "/.rockbox/codecs/wavpack.codec";
+#define CODEC_VORBIS "/.rockbox/codecs/vorbis.codec"
+#define CODEC_MPA_L3 "/.rockbox/codecs/mpa.codec"
+#define CODEC_FLAC "/.rockbox/codecs/flac.codec"
+#define CODEC_WAV "/.rockbox/codecs/wav.codec"
+#define CODEC_A52 "/.rockbox/codecs/a52.codec"
+#define CODEC_MPC "/.rockbox/codecs/mpc.codec"
+#define CODEC_WAVPACK "/.rockbox/codecs/wavpack.codec"
#define AUDIO_FILL_CYCLE (1024*256)
#define AUDIO_DEFAULT_WATERMARK (1024*512)
@@ -96,6 +98,10 @@ static volatile bool paused;
#define MALLOC_BUFSIZE (512*1024)
#define GUARD_BUFSIZE (8*1024)
+/* As defined in plugin.lds */
+#define CODEC_IRAM_ORIGIN 0x10010000
+#define CODEC_IRAM_SIZE 0x8000
+
extern bool audio_is_initialized;
/* Buffer control thread. */
@@ -108,19 +114,39 @@ static struct event_queue codec_queue;
static long codec_stack[(DEFAULT_STACK_SIZE + 0x2500)/sizeof(long)] IDATA_ATTR;
static const char codec_thread_name[] = "codec";
+/* Voice codec thread. */
+static struct event_queue voice_codec_queue;
+/* Not enough IRAM for this. */
+static long voice_codec_stack[(DEFAULT_STACK_SIZE + 0x2500)/sizeof(long)];
+static const char voice_codec_thread_name[] = "voice codec";
+
static struct mutex mutex_bufferfill;
+static struct mutex mutex_codecthread;
+
+static struct mp3entry id3_voice;
+
+#define CODEC_IDX_AUDIO 0
+#define CODEC_IDX_VOICE 1
+
+static char *voicebuf;
+static int voice_remaining;
+static bool voice_is_playing;
+static void (*voice_getmore)(unsigned char** start, int* size);
/* Is file buffer currently being refilled? */
static volatile bool filling;
+volatile int current_codec;
+extern unsigned char codecbuf[];
+
/* Ring buffer where tracks and codecs are loaded. */
-static char *codecbuf;
+static char *filebuf;
/* Total size of the ring buffer. */
-int codecbuflen;
+int filebuflen;
/* Bytes available in the buffer. */
-int codecbufused;
+int filebufused;
/* Ring buffer read and write indexes. */
static volatile int buf_ridx;
@@ -153,6 +179,7 @@ static struct track_info *cur_ti;
/* Codec API including function callbacks. */
extern struct codec_api ci;
+extern struct codec_api ci_voice;
/* When we change a song and buffer is not in filling state, this
variable keeps information about whether to go a next/previous track. */
@@ -174,6 +201,59 @@ static bool v1first = false;
static void mp3_set_elapsed(struct mp3entry* id3);
int mp3_get_file_pos(void);
+static void do_swap(int idx_old, int idx_new)
+{
+#ifndef SIMULATOR
+ unsigned char *iram_p = (unsigned char *)(CODEC_IRAM_ORIGIN);
+ unsigned char *iram_buf[2];
+#endif
+ unsigned char *dram_buf[2];
+
+
+#ifndef SIMULATOR
+ iram_buf[0] = &filebuf[filebuflen];
+ iram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE];
+ memcpy(iram_buf[idx_old], iram_p, CODEC_IRAM_SIZE);
+ memcpy(iram_p, iram_buf[idx_new], CODEC_IRAM_SIZE);
+#endif
+
+ dram_buf[0] = &filebuf[filebuflen+CODEC_IRAM_SIZE*2];
+ dram_buf[1] = &filebuf[filebuflen+CODEC_IRAM_SIZE*2+CODEC_SIZE];
+ memcpy(dram_buf[idx_old], codecbuf, CODEC_SIZE);
+ memcpy(codecbuf, dram_buf[idx_new], CODEC_SIZE);
+}
+
+static void swap_codec(void)
+{
+ int last_codec;
+
+ logf("swapping codec:%d", current_codec);
+
+ /* We should swap codecs' IRAM contents and code space. */
+ do_swap(current_codec, !current_codec);
+
+ last_codec = current_codec;
+ current_codec = !current_codec;
+
+ /* Release the semaphore and force a task switch. */
+ mutex_unlock(&mutex_codecthread);
+ sleep(1);
+
+ /* Waiting until we are ready to run again. */
+ mutex_lock(&mutex_codecthread);
+
+ /* Check if codec swap did not happen. */
+ if (current_codec != last_codec)
+ {
+ logf("no codec switch happened!");
+ do_swap(current_codec, !current_codec);
+ current_codec = !current_codec;
+ }
+
+ invalidate_icache();
+ logf("codec resuming:%d", current_codec);
+}
+
bool codec_pcmbuf_insert_split_callback(void *ch1, void *ch2,
long length)
{
@@ -209,12 +289,41 @@ bool codec_pcmbuf_insert_split_callback(void *ch1, void *ch2,
pcmbuf_flush_buffer(0);
DEBUGF("Warning: dsp_input_size(%ld=dsp_output_size(%ld))=%ld <= 0\n",
output_size, length, input_size);
- /* should we really continue, or should we break? */
+ /* should we really continue, or should we break?
+ * We should probably continue because calling pcmbuf_flush_buffer(0)
+ * will wrap the buffer if it was fully filled and so next call to
+ * pcmbuf_request_buffer should give the requested output_size. */
continue;
}
output_size = dsp_process(dest, src, input_size);
- pcmbuf_flush_buffer(output_size);
+
+ /* Hotswap between audio and voice codecs as necessary. */
+ switch (current_codec)
+ {
+ case CODEC_IDX_AUDIO:
+ pcmbuf_flush_buffer(output_size);
+ if (voice_is_playing && pcmbuf_usage() > 30
+ && pcmbuf_mix_usage() < 20)
+ {
+ cpu_boost(true);
+ swap_codec();
+ cpu_boost(false);
+ }
+ break ;
+
+ case CODEC_IDX_VOICE:
+ if (audio_codec_loaded) {
+ pcmbuf_mix(dest, output_size);
+ if ((pcmbuf_usage() < 10)
+ || pcmbuf_mix_usage() > 70)
+ swap_codec();
+ } else {
+ pcmbuf_flush_buffer(output_size);
+ }
+ break ;
+ }
+
length -= input_size;
}
@@ -241,6 +350,9 @@ bool codec_pcmbuf_insert_callback(char *buf, long length)
void* get_codec_memory_callback(long *size)
{
*size = MALLOC_BUFSIZE;
+ if (voice_codec_loaded)
+ return &audiobuf[talk_get_bufsize()];
+
return &audiobuf[0];
}
@@ -248,7 +360,7 @@ void codec_set_elapsed_callback(unsigned int value)
{
unsigned int latency;
- if (ci.stop_codec)
+ if (ci.stop_codec || current_codec == CODEC_IDX_VOICE)
return ;
latency = pcmbuf_get_latency();
@@ -265,7 +377,7 @@ void codec_set_offset_callback(unsigned int value)
{
unsigned int latency;
- if (ci.stop_codec)
+ if (ci.stop_codec || current_codec == CODEC_IDX_VOICE)
return ;
latency = pcmbuf_get_latency() * cur_ti->id3.bitrate / 8;
@@ -283,7 +395,7 @@ long codec_filebuf_callback(void *ptr, long size)
int copy_n;
int part_n;
- if (ci.stop_codec || !playing)
+ if (ci.stop_codec || !playing || current_codec == CODEC_IDX_VOICE)
return 0;
copy_n = MIN((off_t)size, (off_t)cur_ti->available + cur_ti->filerem);
@@ -297,26 +409,78 @@ long codec_filebuf_callback(void *ptr, long size)
if (copy_n == 0)
return 0;
- part_n = MIN(copy_n, codecbuflen - buf_ridx);
- memcpy(buf, &codecbuf[buf_ridx], part_n);
+ part_n = MIN(copy_n, filebuflen - buf_ridx);
+ memcpy(buf, &filebuf[buf_ridx], part_n);
if (part_n < copy_n) {
- memcpy(&buf[part_n], &codecbuf[0], copy_n - part_n);
+ memcpy(&buf[part_n], &filebuf[0], copy_n - part_n);
}
buf_ridx += copy_n;
- if (buf_ridx >= codecbuflen)
- buf_ridx -= codecbuflen;
+ if (buf_ridx >= filebuflen)
+ buf_ridx -= filebuflen;
ci.curpos += copy_n;
cur_ti->available -= copy_n;
- codecbufused -= copy_n;
+ filebufused -= copy_n;
return copy_n;
}
+void* voice_request_data(long *realsize, long reqsize)
+{
+ while (queue_empty(&voice_codec_queue) && (voice_remaining == 0
+ || voicebuf == NULL) && !ci_voice.stop_codec)
+ {
+ yield();
+ if (audio_codec_loaded && (pcmbuf_usage() < 30
+ || !voice_is_playing || voicebuf == NULL))
+ {
+ swap_codec();
+ }
+ if (!voice_is_playing)
+ sleep(HZ/16);
+
+ if (voice_remaining)
+ {
+ voice_is_playing = true;
+ break ;
+ }
+
+ if (voice_getmore != NULL)
+ {
+ voice_getmore((unsigned char **)&voicebuf, (int *)&voice_remaining);
+
+ if (!voice_remaining)
+ {
+ voice_is_playing = false;
+ /* Force pcm playback. */
+ pcmbuf_play_start();
+ }
+ }
+ }
+
+ if (reqsize < 0)
+ reqsize = 0;
+
+ voice_is_playing = true;
+ *realsize = voice_remaining;
+ if (*realsize > reqsize)
+ *realsize = reqsize;
+
+ if (*realsize == 0)
+ return NULL;
+
+ return voicebuf;
+}
+
void* codec_request_buffer_callback(long *realsize, long reqsize)
{
long part_n;
-
+
+ /* Voice codec. */
+ if (current_codec == CODEC_IDX_VOICE) {
+ return voice_request_data(realsize, reqsize);
+ }
+
if (ci.stop_codec || !playing) {
*realsize = 0;
return NULL;
@@ -335,22 +499,22 @@ void* codec_request_buffer_callback(long *realsize, long reqsize)
}
}
- part_n = MIN((int)*realsize, codecbuflen - buf_ridx);
+ part_n = MIN((int)*realsize, filebuflen - buf_ridx);
if (part_n < *realsize) {
part_n += GUARD_BUFSIZE;
if (part_n < *realsize)
*realsize = part_n;
- memcpy(&codecbuf[codecbuflen], &codecbuf[0], *realsize -
- (codecbuflen - buf_ridx));
+ memcpy(&filebuf[filebuflen], &filebuf[0], *realsize -
+ (filebuflen - buf_ridx));
}
- return (char *)&codecbuf[buf_ridx];
+ return (char *)&filebuf[buf_ridx];
}
static bool rebuffer_and_seek(int newpos)
{
int fd;
-
+
logf("Re-buffering song");
mutex_lock(&mutex_bufferfill);
@@ -367,7 +531,7 @@ static bool rebuffer_and_seek(int newpos)
/* Clear codec buffer. */
audio_invalidate_tracks();
- codecbufused = 0;
+ filebufused = 0;
buf_ridx = buf_widx = 0;
cur_ti->filerem = cur_ti->filesize - newpos;
cur_ti->filepos = newpos;
@@ -390,6 +554,15 @@ static bool rebuffer_and_seek(int newpos)
void codec_advance_buffer_callback(long amount)
{
+ if (current_codec == CODEC_IDX_VOICE) {
+ //logf("voice ad.buf:%d", amount);
+ amount = MAX(0, MIN(amount, voice_remaining));
+ voicebuf += amount;
+ voice_remaining -= amount;
+
+ return ;
+ }
+
if (amount > cur_ti->available + cur_ti->filerem)
amount = cur_ti->available + cur_ti->filerem;
@@ -400,10 +573,10 @@ void codec_advance_buffer_callback(long amount)
}
buf_ridx += amount;
- if (buf_ridx >= codecbuflen)
- buf_ridx -= codecbuflen;
+ if (buf_ridx >= filebuflen)
+ buf_ridx -= filebuflen;
cur_ti->available -= amount;
- codecbufused -= amount;
+ filebufused -= amount;
ci.curpos += amount;
codec_set_offset_callback(ci.curpos);
}
@@ -411,15 +584,18 @@ void codec_advance_buffer_callback(long amount)
void codec_advance_buffer_loc_callback(void *ptr)
{
long amount;
-
- amount = (int)ptr - (int)&codecbuf[buf_ridx];
+
+ if (current_codec == CODEC_IDX_VOICE)
+ amount = (int)ptr - (int)voicebuf;
+ else
+ amount = (int)ptr - (int)&filebuf[buf_ridx];
codec_advance_buffer_callback(amount);
}
off_t codec_mp3_get_filepos_callback(int newtime)
{
off_t newpos;
-
+
cur_ti->id3.elapsed = newtime;
newpos = mp3_get_file_pos();
@@ -429,7 +605,10 @@ off_t codec_mp3_get_filepos_callback(int newtime)
bool codec_seek_buffer_callback(off_t newpos)
{
int difference;
-
+
+ if (current_codec == CODEC_IDX_VOICE)
+ return false;
+
if (newpos < 0)
newpos = 0;
@@ -457,11 +636,11 @@ bool codec_seek_buffer_callback(off_t newpos)
/* Seeking inside buffer space. */
logf("seek: -%d", difference);
- codecbufused += difference;
+ filebufused += difference;
cur_ti->available += difference;
buf_ridx -= difference;
if (buf_ridx < 0)
- buf_ridx = codecbuflen + buf_ridx;
+ buf_ridx = filebuflen + buf_ridx;
ci.curpos -= difference;
if (!pcmbuf_is_crossfade_active())
pcmbuf_play_stop();
@@ -473,8 +652,11 @@ static void set_filebuf_watermark(int seconds)
{
long bytes;
+ if (current_codec == CODEC_IDX_VOICE)
+ return ;
+
bytes = MAX((int)cur_ti->id3.bitrate * seconds * (1000/8), conf_watermark);
- bytes = MIN(bytes, codecbuflen / 2);
+ bytes = MIN(bytes, filebuflen / 2);
conf_watermark = bytes;
}
@@ -540,7 +722,7 @@ void yield_codecs(void)
sleep(5);
while ((pcmbuf_is_crossfade_active() || pcmbuf_is_lowdata())
&& !ci.stop_codec && playing && queue_empty(&audio_queue)
- && codecbufused > (128*1024))
+ && filebufused > (128*1024))
yield();
}
@@ -552,18 +734,18 @@ void strip_id3v1_tag(void)
int tagptr;
bool found = true;
- if (codecbufused >= 128)
+ if (filebufused >= 128)
{
tagptr = buf_widx - 128;
if (tagptr < 0)
- tagptr += codecbuflen;
+ tagptr += filebuflen;
for(i = 0;i < 3;i++)
{
- if(tagptr >= codecbuflen)
- tagptr -= codecbuflen;
+ if(tagptr >= filebuflen)
+ tagptr -= filebuflen;
- if(codecbuf[tagptr] != tag[i])
+ if(filebuf[tagptr] != tag[i])
{
found = false;
break;
@@ -578,7 +760,7 @@ void strip_id3v1_tag(void)
logf("Skipping ID3v1 tag\n");
buf_widx -= 128;
tracks[track_widx].available -= 128;
- codecbufused -= 128;
+ filebufused -= 128;
}
}
}
@@ -603,9 +785,9 @@ void audio_fill_file_buffer(void)
if (fill_bytesleft == 0)
break ;
- rc = MIN(conf_filechunk, codecbuflen - buf_widx);
+ rc = MIN(conf_filechunk, filebuflen - buf_widx);
rc = MIN(rc, fill_bytesleft);
- rc = read(current_fd, &codecbuf[buf_widx], rc);
+ rc = read(current_fd, &filebuf[buf_widx], rc);
if (rc <= 0) {
tracks[track_widx].filerem = 0;
strip_id3v1_tag();
@@ -613,13 +795,13 @@ void audio_fill_file_buffer(void)
}
buf_widx += rc;
- if (buf_widx >= codecbuflen)
- buf_widx -= codecbuflen;
+ if (buf_widx >= filebuflen)
+ buf_widx -= filebuflen;
i += rc;
tracks[track_widx].available += rc;
tracks[track_widx].filerem -= rc;
tracks[track_widx].filepos += rc;
- codecbufused += rc;
+ filebufused += rc;
fill_bytesleft -= rc;
}
@@ -725,15 +907,15 @@ bool loadcodec(const char *trackname, bool start_play)
while (i < size) {
yield_codecs();
- copy_n = MIN(conf_filechunk, codecbuflen - buf_widx);
- rc = read(fd, &codecbuf[buf_widx], copy_n);
+ copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
+ rc = read(fd, &filebuf[buf_widx], copy_n);
if (rc < 0)
return false;
buf_widx += rc;
- codecbufused += rc;
+ filebufused += rc;
fill_bytesleft -= rc;
- if (buf_widx >= codecbuflen)
- buf_widx -= codecbuflen;
+ if (buf_widx >= filebuflen)
+ buf_widx -= filebuflen;
i += rc;
}
close(fd);
@@ -840,20 +1022,23 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
tracks[track_widx].playlist_offset = peek_offset;
last_peek_offset = peek_offset;
- if (buf_widx >= codecbuflen)
- buf_widx -= codecbuflen;
+ if (buf_widx >= filebuflen)
+ buf_widx -= filebuflen;
/* Set default values */
if (start_play) {
+ int last_codec = current_codec;
+ current_codec = CODEC_IDX_AUDIO;
conf_bufferlimit = 0;
conf_watermark = AUDIO_DEFAULT_WATERMARK;
conf_filechunk = AUDIO_DEFAULT_FILECHUNK;
dsp_configure(DSP_RESET, 0);
ci.configure(CODEC_DSP_ENABLE, false);
+ current_codec = last_codec;
}
/* Load the codec. */
- tracks[track_widx].codecbuf = &codecbuf[buf_widx];
+ tracks[track_widx].codecbuf = &filebuf[buf_widx];
if (!loadcodec(trackname, start_play)) {
close(fd);
/* Stop buffer filling if codec load failed. */
@@ -870,7 +1055,7 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
}
return false;
}
- // tracks[track_widx].filebuf = &codecbuf[buf_widx];
+ // tracks[track_widx].filebuf = &filebuf[buf_widx];
tracks[track_widx].start_pos = 0;
/* Get track metadata if we don't already have it. */
@@ -933,10 +1118,10 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
if (fill_bytesleft == 0)
break ;
- copy_n = MIN(conf_filechunk, codecbuflen - buf_widx);
+ copy_n = MIN(conf_filechunk, filebuflen - buf_widx);
copy_n = MIN(size - i, copy_n);
copy_n = MIN((int)fill_bytesleft, copy_n);
- rc = read(fd, &codecbuf[buf_widx], copy_n);
+ rc = read(fd, &filebuf[buf_widx], copy_n);
if (rc < copy_n) {
logf("File error!");
tracks[track_widx].filesize = 0;
@@ -945,12 +1130,12 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
return false;
}
buf_widx += rc;
- if (buf_widx >= codecbuflen)
- buf_widx -= codecbuflen;
+ if (buf_widx >= filebuflen)
+ buf_widx -= filebuflen;
i += rc;
tracks[track_widx].available += rc;
tracks[track_widx].filerem -= rc;
- codecbufused += rc;
+ filebufused += rc;
fill_bytesleft -= rc;
}
@@ -997,10 +1182,10 @@ void audio_play_start(int offset)
track_ridx = 0;
buf_ridx = 0;
buf_widx = 0;
- codecbufused = 0;
+ filebufused = 0;
pcmbuf_set_boost_mode(true);
- fill_bytesleft = codecbuflen;
+ fill_bytesleft = filebuflen;
filling = true;
last_peek_offset = -1;
if (audio_load_track(offset, true, 0)) {
@@ -1089,7 +1274,7 @@ void initialize_buffer_fill(void)
int cur_idx, i;
- fill_bytesleft = codecbuflen - codecbufused;
+ fill_bytesleft = filebuflen - filebufused;
cur_ti->start_pos = ci.curpos;
pcmbuf_set_boost_mode(true);
@@ -1124,7 +1309,7 @@ void initialize_buffer_fill(void)
void audio_check_buffer(void)
{
/* Start buffer filling as necessary. */
- if ((codecbufused > conf_watermark || !queue_empty(&audio_queue)
+ if ((filebufused > conf_watermark || !queue_empty(&audio_queue)
|| !playing || ci.stop_codec || ci.reload_codec) && !filling)
return ;
@@ -1132,8 +1317,8 @@ void audio_check_buffer(void)
/* Limit buffering size at first run. */
if (conf_bufferlimit && fill_bytesleft > conf_bufferlimit
- - codecbufused) {
- fill_bytesleft = MAX(0, conf_bufferlimit - codecbufused);
+ - filebufused) {
+ fill_bytesleft = MAX(0, conf_bufferlimit - filebufused);
}
/* Try to load remainings of the file. */
@@ -1169,27 +1354,27 @@ void audio_update_trackinfo(void)
{
if (new_track >= 0) {
buf_ridx += cur_ti->available;
- codecbufused -= cur_ti->available;
+ filebufused -= cur_ti->available;
cur_ti = &tracks[track_ridx];
buf_ridx += cur_ti->codecsize;
- codecbufused -= cur_ti->codecsize;
- if (buf_ridx >= codecbuflen)
- buf_ridx -= codecbuflen;
+ filebufused -= cur_ti->codecsize;
+ if (buf_ridx >= filebuflen)
+ buf_ridx -= filebuflen;
if (!filling)
pcmbuf_set_boost_mode(false);
} else {
buf_ridx -= ci.curpos + cur_ti->codecsize;
- codecbufused += ci.curpos + cur_ti->codecsize;
+ filebufused += ci.curpos + cur_ti->codecsize;
cur_ti->available = cur_ti->filesize;
cur_ti = &tracks[track_ridx];
buf_ridx -= cur_ti->filesize;
- codecbufused += cur_ti->filesize;
+ filebufused += cur_ti->filesize;
cur_ti->available = cur_ti->filesize;
if (buf_ridx < 0)
- buf_ridx = codecbuflen + buf_ridx;
+ buf_ridx = filebuflen + buf_ridx;
}
ci.filesize = cur_ti->filesize;
@@ -1220,7 +1405,7 @@ static void audio_stop_playback(void)
current_fd = -1;
}
pcmbuf_play_stop();
- while (codec_loaded)
+ while (audio_codec_loaded)
yield();
pcm_play_pause(true);
track_count = 0;
@@ -1267,6 +1452,11 @@ static int get_codec_base_type(int type)
bool codec_request_next_track_callback(void)
{
+ if (current_codec == CODEC_IDX_VOICE) {
+ voice_remaining = 0;
+ return !ci_voice.stop_codec;
+ }
+
if (ci.stop_codec || !playing)
return false;
@@ -1309,8 +1499,8 @@ bool codec_request_next_track_callback(void)
if (--track_ridx < 0)
track_ridx = MAX_TRACK-1;
if (tracks[track_ridx].filesize == 0 ||
- codecbufused+ci.curpos+tracks[track_ridx].filesize
- /*+ (off_t)tracks[track_ridx].codecsize*/ > codecbuflen) {
+ filebufused+ci.curpos+tracks[track_ridx].filesize
+ /*+ (off_t)tracks[track_ridx].codecsize*/ > filebuflen) {
logf("Loading from disk...");
new_track = 0;
last_index = -1;
@@ -1379,10 +1569,10 @@ void audio_invalidate_tracks(void)
track_widx = track_ridx;
/* Mark all other entries null (also buffered wrong metadata). */
audio_clear_track_entries(false);
- codecbufused = cur_ti->available;
+ filebufused = cur_ti->available;
buf_widx = buf_ridx + cur_ti->available;
- if (buf_widx >= codecbuflen)
- buf_widx -= codecbuflen;
+ if (buf_widx >= filebuflen)
+ buf_widx -= filebuflen;
read_next_metadata();
}
@@ -1436,7 +1626,7 @@ void audio_thread(void)
ci.reload_codec = false;
ci.seek_time = 0;
pcmbuf_crossfade_init(CROSSFADE_MODE_CROSSFADE);
- while (codec_loaded)
+ while (audio_codec_loaded)
yield();
audio_play_start((int)ev.data);
playlist_update_resume_info(audio_current_track());
@@ -1462,11 +1652,13 @@ void audio_thread(void)
case AUDIO_NEXT:
logf("audio_next");
+ pcmbuf_beep(5000, 100, 5000);
initiate_track_change(1);
break ;
case AUDIO_PREV:
logf("audio_prev");
+ pcmbuf_beep(5000, 100, 5000);
initiate_track_change(-1);
break;
@@ -1514,8 +1706,11 @@ void codec_thread(void)
switch (ev.id) {
case CODEC_LOAD_DISK:
ci.stop_codec = false;
- codec_loaded = true;
- status = codec_load_file((char *)ev.data);
+ audio_codec_loaded = true;
+ mutex_lock(&mutex_codecthread);
+ current_codec = CODEC_IDX_AUDIO;
+ status = codec_load_file((char *)ev.data, &ci);
+ mutex_unlock(&mutex_codecthread);
break ;
case CODEC_LOAD:
@@ -1531,10 +1726,13 @@ void codec_thread(void)
}
ci.stop_codec = false;
- wrap = (int)&codecbuf[codecbuflen] - (int)cur_ti->codecbuf;
- codec_loaded = true;
- status = codec_load_ram(cur_ti->codecbuf, codecsize,
- &codecbuf[0], wrap);
+ wrap = (int)&filebuf[filebuflen] - (int)cur_ti->codecbuf;
+ audio_codec_loaded = true;
+ mutex_lock(&mutex_codecthread);
+ current_codec = CODEC_IDX_AUDIO;
+ status = codec_load_ram(cur_ti->codecbuf, codecsize,
+ &filebuf[0], wrap, &ci);
+ mutex_unlock(&mutex_codecthread);
break ;
#ifndef SIMULATOR
@@ -1545,7 +1743,7 @@ void codec_thread(void)
#endif
}
- codec_loaded = false;
+ audio_codec_loaded = false;
switch (ev.id) {
case CODEC_LOAD_DISK:
@@ -1569,6 +1767,83 @@ void codec_thread(void)
}
}
+static void reset_buffer(void)
+{
+ filebuf = &audiobuf[MALLOC_BUFSIZE];
+ filebuflen = audiobufend - audiobuf - pcmbuf_get_bufsize()
+ - PCMBUF_GUARD - MALLOC_BUFSIZE - GUARD_BUFSIZE;
+
+ if (talk_get_bufsize() && voice_codec_loaded)
+ {
+ filebuf = &filebuf[talk_get_bufsize()];
+ filebuflen -= 2*CODEC_IRAM_SIZE + 2*CODEC_SIZE + talk_get_bufsize();
+ }
+}
+
+void voice_codec_thread(void)
+{
+ struct event ev;
+ int status;
+
+ current_codec = CODEC_IDX_AUDIO;
+ voice_codec_loaded = false;
+ while (1) {
+ status = 0;
+ queue_wait(&voice_codec_queue, &ev);
+ switch (ev.id) {
+ case CODEC_LOAD_DISK:
+ logf("Loading voice codec");
+ audio_stop_playback();
+ mutex_lock(&mutex_codecthread);
+ current_codec = CODEC_IDX_VOICE;
+ dsp_configure(DSP_RESET, 0);
+ ci.configure(CODEC_DSP_ENABLE, (bool *)true);
+ voice_remaining = 0;
+ voice_getmore = NULL;
+ voice_codec_loaded = true;
+ reset_buffer();
+ ci_voice.stop_codec = false;
+
+ status = codec_load_file((char *)ev.data, &ci_voice);
+
+ logf("Voice codec finished");
+ audio_stop_playback();
+ mutex_unlock(&mutex_codecthread);
+ current_codec = CODEC_IDX_AUDIO;
+ voice_codec_loaded = false;
+ reset_buffer();
+ break ;
+
+#ifndef SIMULATOR
+ case SYS_USB_CONNECTED:
+ usb_acknowledge(SYS_USB_CONNECTED_ACK);
+ usb_wait_for_disconnect(&voice_codec_queue);
+ break ;
+#endif
+ }
+ }
+}
+
+void voice_init(void)
+{
+ while (voice_codec_loaded)
+ {
+ logf("Terminating voice codec");
+ ci_voice.stop_codec = true;
+ if (current_codec != CODEC_IDX_VOICE)
+ swap_codec();
+ sleep(1);
+ }
+
+ if (!talk_get_bufsize())
+ return ;
+
+ logf("Starting voice codec");
+ queue_post(&voice_codec_queue, CODEC_LOAD_DISK, (void *)CODEC_MPA_L3);
+ while (!voice_codec_loaded)
+ sleep(1);
+}
+
struct mp3entry* audio_current_track(void)
{
// logf("audio_current_track");
@@ -1620,7 +1895,7 @@ void audio_stop(void)
{
logf("audio_stop");
queue_post(&audio_queue, AUDIO_STOP, 0);
- while (playing || codec_loaded)
+ while (playing || audio_codec_loaded)
yield();
}
@@ -1805,6 +2080,16 @@ int mp3_get_file_pos(void)
return pos;
}
+void mp3_play_data(const unsigned char* start, int size,
+ void (*get_more)(unsigned char** start, int* size))
+{
+ voice_getmore = get_more;
+ voicebuf = (unsigned char *)start;
+ voice_remaining = size;
+ voice_is_playing = true;
+ pcmbuf_reset_mixpos();
+}
+
void audio_set_buffer_margin(int setting)
{
int lookup[] = {5, 15, 30, 60, 120, 180, 300, 600};
@@ -1827,7 +2112,7 @@ void audio_set_crossfade(int type)
offset = cur_ti->id3.offset;
if (type == CROSSFADE_MODE_OFF)
- seconds = 0;
+ seconds = 1;
/* Buffer has to be at least 2s long. */
seconds += 2;
@@ -1843,12 +2128,13 @@ void audio_set_crossfade(int type)
if (was_playing)
splash(0, true, str(LANG_RESTARTING_PLAYBACK));
pcmbuf_init(size);
- pcmbuf_crossfade_enable(seconds > 2);
- codecbuflen = audiobufend - audiobuf - pcmbuf_get_bufsize()
- - PCMBUF_GUARD - MALLOC_BUFSIZE - GUARD_BUFSIZE;
+ pcmbuf_crossfade_enable(type != CROSSFADE_MODE_OFF);
+ reset_buffer();
logf("abuf:%dB", pcmbuf_get_bufsize());
- logf("fbuf:%dB", codecbuflen);
+ logf("fbuf:%dB", filebuflen);
+ voice_init();
+
/* Restart playback. */
if (was_playing) {
audio_play(offset);
@@ -1856,7 +2142,7 @@ void audio_set_crossfade(int type)
/* Wait for the playback to start again (and display the splash
screen during that period. */
playing = true;
- while (playing && !codec_loaded)
+ while (playing && !audio_codec_loaded)
yield();
}
}
@@ -1884,13 +2170,17 @@ void test_unbuffer_event(struct mp3entry *id3, bool last_track)
void audio_init(void)
{
+ static bool voicetagtrue = true;
+
logf("audio api init");
pcm_init();
- codecbufused = 0;
+ filebufused = 0;
filling = false;
- codecbuf = &audiobuf[MALLOC_BUFSIZE];
+ current_codec = CODEC_IDX_AUDIO;
+ filebuf = &audiobuf[MALLOC_BUFSIZE];
playing = false;
- codec_loaded = false;
+ audio_codec_loaded = false;
+ voice_is_playing = false;
paused = false;
track_changed = false;
current_fd = -1;
@@ -1918,12 +2208,25 @@ void audio_init(void)
ci.set_offset = codec_set_offset_callback;
ci.configure = codec_configure_callback;
+ memcpy(&ci_voice, &ci, sizeof(struct codec_api));
+ memset(&id3_voice, 0, sizeof(struct mp3entry));
+ ci_voice.taginfo_ready = &voicetagtrue;
+ ci_voice.id3 = &id3_voice;
+ ci_voice.pcmbuf_insert = codec_pcmbuf_insert_callback;
+ id3_voice.frequency = 11200;
+ id3_voice.length = 1000000L;
+
mutex_init(&mutex_bufferfill);
+ mutex_init(&mutex_codecthread);
+
queue_init(&audio_queue);
queue_init(&codec_queue);
+ queue_init(&voice_codec_queue);
create_thread(codec_thread, codec_stack, sizeof(codec_stack),
codec_thread_name);
+ create_thread(voice_codec_thread, voice_codec_stack,
+ sizeof(voice_codec_stack), voice_codec_thread_name);
create_thread(audio_thread, audio_stack, sizeof(audio_stack),
audio_thread_name);
}
diff --git a/apps/playback.h b/apps/playback.h
index 7ed9a4b700..a5b64ba0e3 100644
--- a/apps/playback.h
+++ b/apps/playback.h
@@ -72,6 +72,7 @@ void audio_set_track_buffer_event(void (*handler)(struct mp3entry *id3,
void audio_set_track_unbuffer_event(void (*handler)(struct mp3entry *id3,
bool last_track));
void audio_invalidate_tracks(void);
+void voice_init(void);
#endif
diff --git a/apps/playlist.c b/apps/playlist.c
index 68fd8be369..bd443e4f38 100644
--- a/apps/playlist.c
+++ b/apps/playlist.c
@@ -1413,9 +1413,14 @@ int playlist_resume(void)
};
/* use mp3 buffer for maximum load speed */
+#if CONFIG_HWCODEC != MASNONE
talk_buffer_steal(); /* we use the mp3 buffer, need to tell */
buflen = (audiobufend - audiobuf);
buffer = audiobuf;
+#else
+ buflen = (audiobufend - audiobuf - talk_get_bufsize());
+ buffer = &audiobuf[talk_get_bufsize()];
+#endif
empty_playlist(playlist, true);
@@ -1827,7 +1832,9 @@ int playlist_start(int start_index, int offset)
struct playlist_info* playlist = &current_playlist;
playlist->index = start_index;
+#if CONFIG_HWCODEC != MASNONE
talk_buffer_steal(); /* will use the mp3 buffer */
+#endif
audio_play(offset);
return 0;
diff --git a/apps/talk.c b/apps/talk.c
index a896ca3a1a..b417046a61 100644
--- a/apps/talk.c
+++ b/apps/talk.c
@@ -32,7 +32,11 @@
#include "lang.h"
#include "talk.h"
#include "id3.h"
+#include "logf.h"
#include "bitswap.h"
+#if CONFIG_HWCODEC == MASNONE
+#include "playback.h"
+#endif
/***************** Constants *****************/
@@ -88,6 +92,7 @@ static int filehandle; /* global, so the MMC variant can keep the file open */
static unsigned char* p_silence; /* VOICE_PAUSE clip, used for termination */
static long silence_len; /* length of the VOICE_PAUSE clip */
static unsigned char* p_lastclip; /* address of latest clip, for silence add */
+static unsigned long voicefile_size = 0; /* size of the loaded voice file */
/***************** Private prototypes *****************/
@@ -114,10 +119,28 @@ static int open_voicefile(void)
}
snprintf(buf, sizeof(buf), ROCKBOX_DIR LANG_DIR "/%s.voice", p_lang);
-
+
return open(buf, O_RDONLY);
}
+int talk_get_bufsize(void)
+{
+ return voicefile_size;
+}
+
+#ifdef SIMULATOR
+static unsigned short BSWAP16(unsigned short value)
+{
+ return (value >> 8) | (value << 8);
+}
+
+static unsigned long BSWAP32(unsigned long value)
+{
+ unsigned long hi = BSWAP16(value >> 16);
+ unsigned long lo = BSWAP16(value & 0xffff);
+ return (lo << 16) | hi;
+}
+#endif
/* load the voice file into the mp3 buffer */
static void load_voicefile(void)
@@ -125,6 +148,10 @@ static void load_voicefile(void)
int load_size;
int got_size;
int file_size;
+#if CONFIG_HWCODEC == MASNONE
+ int length, i;
+ unsigned char *buf, temp;
+#endif
filehandle = open_voicefile();
if (filehandle < 0) /* failed to open */
@@ -141,8 +168,20 @@ static void load_voicefile(void)
#endif
got_size = read(filehandle, audiobuf, load_size);
- if (got_size == load_size /* success */
- && ((struct voicefile*)audiobuf)->table /* format check */
+ if (got_size != load_size /* failure */)
+ goto load_err;
+
+#ifdef SIMULATOR
+ logf("Byte swapping voice file");
+ p_voicefile = (struct voicefile*)audiobuf;
+ p_voicefile->version = BSWAP32(p_voicefile->version);
+ p_voicefile->table = BSWAP32(p_voicefile->table);
+ p_voicefile->id1_max = BSWAP32(p_voicefile->id1_max);
+ p_voicefile->id2_max = BSWAP32(p_voicefile->id2_max);
+ p_voicefile = NULL;
+#endif
+
+ if (((struct voicefile*)audiobuf)->table /* format check */
== offsetof(struct voicefile, index))
{
p_voicefile = (struct voicefile*)audiobuf;
@@ -155,7 +194,42 @@ static void load_voicefile(void)
else
goto load_err;
-#ifdef HAVE_MMC
+#ifdef SIMULATOR
+ for (i = 0; i < p_voicefile->id1_max + p_voicefile->id2_max; i++)
+ {
+ struct clip_entry *ce;
+ ce = &p_voicefile->index[i];
+ ce->offset = BSWAP32(ce->offset);
+ ce->size = BSWAP32(ce->size);
+ }
+#endif
+
+ /* Do a bitswap as necessary. */
+#if CONFIG_HWCODEC == MASNONE
+ logf("Bitswapping voice file.");
+ cpu_boost(true);
+ buf = (unsigned char *)(&p_voicefile->index) +
+ (p_voicefile->id1_max + p_voicefile->id2_max) * sizeof(struct clip_entry);
+ length = file_size - offsetof(struct voicefile, index) -
+ (p_voicefile->id1_max - p_voicefile->id2_max) * sizeof(struct clip_entry);
+
+ for (i = 0; i < length; i++)
+ {
+ temp = buf[i];
+ buf[i] = ((temp >> 7) & 0x01)
+ | ((temp >> 5) & 0x02)
+ | ((temp >> 3) & 0x04)
+ | ((temp >> 1) & 0x08)
+ | ((temp << 1) & 0x10)
+ | ((temp << 3) & 0x20)
+ | ((temp << 5) & 0x40)
+ | ((temp << 7) & 0x80);
+ }
+ cpu_boost(false);
+
+#endif
+
+#ifdef HAVE_MMC
/* load the index table, now that we know its size from the header */
load_size = (p_voicefile->id1_max + p_voicefile->id2_max)
* sizeof(struct clip_entry);
@@ -193,7 +267,11 @@ static void mp3_callback(unsigned char** start, int* size)
if (queue[queue_read].len > 0) /* current clip not finished? */
{ /* feed the next 64K-1 chunk */
+#if CONFIG_HWCODEC != MASNONE
sent = MIN(queue[queue_read].len, 0xFFFF);
+#else
+ sent = queue[queue_read].len;
+#endif
*start = queue[queue_read].buf;
*size = sent;
return;
@@ -207,7 +285,11 @@ re_check:
if (QUEUE_LEVEL) /* queue is not empty? */
{ /* start next clip */
+#if CONFIG_HWCODEC != MASNONE
sent = MIN(queue[queue_read].len, 0xFFFF);
+#else
+ sent = queue[queue_read].len;
+#endif
*start = p_lastclip = queue[queue_read].buf;
*size = sent;
curr_hd[0] = p_lastclip[1];
@@ -286,7 +368,7 @@ static int shutup(void)
/* nothing to do, was frame boundary or not our clip */
mp3_play_stop();
queue_write = queue_read = 0; /* reset the queue */
-
+
return 0;
}
@@ -317,7 +399,11 @@ static int queue_clip(unsigned char* buf, long size, bool enqueue)
if (queue_level == 0)
{ /* queue was empty, we have to do the initial start */
p_lastclip = buf;
+#if CONFIG_HWCODEC != MASNONE
sent = MIN(size, 0xFFFF); /* DMA can do no more */
+#else
+ sent = size;
+#endif
mp3_play_data(buf, sent, mp3_callback);
curr_hd[0] = buf[1];
curr_hd[1] = buf[2];
@@ -400,12 +486,19 @@ void talk_init(void)
#else
filehandle = open_voicefile();
has_voicefile = (filehandle >= 0); /* test if we can open it */
+ voicefile_size = 0;
+
if (has_voicefile)
{
+ voicefile_size = filesize(filehandle);
+#if CONFIG_HWCODEC == MASNONE
+ voice_init();
+#endif
close(filehandle); /* close again, this was just to detect presence */
filehandle = -1;
}
#endif
+
}
@@ -432,8 +525,10 @@ int talk_id(long id, bool enqueue)
unsigned char* clipbuf;
int unit;
+#if CONFIG_HWCODEC != MASNONE
if (audio_status()) /* busy, buffer in use */
return -1;
+#endif
if (p_voicefile == NULL && has_voicefile)
load_voicefile(); /* reload needed */
@@ -514,8 +609,10 @@ int talk_number(long n, bool enqueue)
int level = 0; /* mille count */
long mil = 1000000000; /* highest possible "-illion" */
+#if CONFIG_HWCODEC != MASNONE
if (audio_status()) /* busy, buffer in use */
return -1;
+#endif
if (!enqueue)
shutup(); /* cut off all the pending stuff */
@@ -593,8 +690,10 @@ int talk_value(long n, int unit, bool enqueue)
VOICE_HERTZ,
};
+#if CONFIG_HWCODEC != MASNONE
if (audio_status()) /* busy, buffer in use */
return -1;
+#endif
if (unit < 0 || unit >= UNIT_LAST)
unit_id = -1;
@@ -625,8 +724,10 @@ int talk_spell(const char* spell, bool enqueue)
{
char c; /* currently processed char */
+#if CONFIG_HWCODEC != MASNONE
if (audio_status()) /* busy, buffer in use */
return -1;
+#endif
if (!enqueue)
shutup(); /* cut off all the pending stuff */
diff --git a/apps/talk.h b/apps/talk.h
index 213e1803d4..18314e52c5 100644
--- a/apps/talk.h
+++ b/apps/talk.h
@@ -59,6 +59,7 @@ extern const char* const dir_thumbnail_name; /* "_dirname.talk" */
extern const char* const file_thumbnail_ext; /* ".talk" for file voicing */
void talk_init(void);
+int talk_get_bufsize(void); /* get the loaded voice file size */
int talk_buffer_steal(void); /* claim the mp3 buffer e.g. for play/record */
int talk_id(long id, bool enqueue); /* play a voice ID from voicefont */
int talk_file(const char* filename, bool enqueue); /* play a thumbnail from file */
diff --git a/firmware/mp3_playback.c b/firmware/mp3_playback.c
index dfe08e5d3a..3a2fdb4f1a 100644
--- a/firmware/mp3_playback.c
+++ b/firmware/mp3_playback.c
@@ -576,7 +576,6 @@ void mp3_reset_playtime(void)
playstart_tick = current_tick;
}
-
bool mp3_is_playing(void)
{
return playing;
@@ -624,18 +623,11 @@ void mp3_init(int volume, int bass, int treble, int balance, int loudness,
audio_is_initialized = true;
#endif
}
+
void mp3_shutdown(void)
{
/* a dummy */
}
-void mp3_play_data(const unsigned char* start, int size,
- void (*get_more)(unsigned char** start, int* size))
-{
- /* a dummy */
- (void)start;
- (void)size;
- (void)get_more;
-}
void mp3_play_stop(void)
{
@@ -653,4 +645,10 @@ unsigned char* mp3_get_pos(void)
/* a dummy */
return (unsigned char *)0x1234;
}
+
+bool mp3_is_playing(void)
+{
+ return playing;
+}
+
#endif /* CONFIG_HWCODEC == MASNONE */
diff --git a/firmware/sound.c b/firmware/sound.c
index 8fb015c27a..cd772f5e9f 100644
--- a/firmware/sound.c
+++ b/firmware/sound.c
@@ -704,10 +704,3 @@ int sound_get_pitch(void)
}
#endif
-#if CONFIG_HWCODEC == MASNONE
-bool mp3_is_playing(void)
-{
- /* a dummy */
- return false;
-}
-#endif
diff --git a/uisimulator/common/stubs.c b/uisimulator/common/stubs.c
index af6f9653fe..2357f9baea 100644
--- a/uisimulator/common/stubs.c
+++ b/uisimulator/common/stubs.c
@@ -273,6 +273,7 @@ void button_set_flip(bool yesno)
(void)yesno;
}
+#if CONFIG_HWCODEC != MASNONE
void talk_init(void)
{
}
@@ -320,6 +321,7 @@ int talk_spell(char* spell, bool enqueue)
const char* const dir_thumbnail_name = "_dirname.talk";
const char* const file_thumbnail_ext = ".talk";
+#endif
/* FIXME: this shoudn't be a stub, rather the real thing.
I'm afraid on Win32/X11 it'll be hard to kill a thread from outside. */