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SWCODEC: Stop clicks between tracks when resampler is active by only switching the DSP frequency and not resetting the resampler at track boundaries. Will make sure DSP is correctly flushed at dicontinuities but don't hear any problems currently.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@11600 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Sevakis 2006-11-26 12:02:47 +00:00
parent 4049d44b03
commit bbef13eddf
16 changed files with 23 additions and 15 deletions
Download patch file Download diff file Expand all files Collapse all files

2
apps/codecs/a52.c
View file

@ -154,7 +154,7 @@ next_track:
while (!ci->taginfo_ready)
ci->yield();
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(ci->id3->frequency));
/* Intialise the A52 decoder and check for success */
state = a52_init(0);

2
apps/codecs/aac.c
View file

@ -90,7 +90,7 @@ next_track:
sound_samples_done = ci->id3->offset;
ci->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(rb->id3->frequency));
codec_set_replaygain(rb->id3);
stream_create(&input_stream,ci);

2
apps/codecs/adx.c
View file

@ -161,7 +161,7 @@ next_track:
bufoff = chanstart;
/* setup pcm buffer format */
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(ci->id3->frequency));
if (channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_INTERLEAVED);
} else if (channels == 1) {

4
apps/codecs/aiff.c
View file

@ -81,6 +81,7 @@ enum codec_status codec_start(struct codec_api *api)
ci->memset(iedata, 0, iend - iedata);
#endif
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
@ -182,8 +183,7 @@ next_track:
goto done;
}
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(ci->id3->frequency));
if (num_channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);

2
apps/codecs/alac.c
View file

@ -78,7 +78,7 @@ enum codec_status codec_start(struct codec_api* api)
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
ci->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(rb->id3->frequency));
codec_set_replaygain(rb->id3);
stream_create(&input_stream,ci);

2
apps/codecs/flac.c
View file

@ -470,7 +470,7 @@ enum codec_status codec_start(struct codec_api* api)
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(ci->id3->frequency));
codec_set_replaygain(ci->id3);
if (samplesdone) {

2
apps/codecs/mpa.c
View file

@ -110,7 +110,7 @@ next_track:
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
ci->configure(DSP_SET_FREQUENCY, (int *)ci->id3->frequency);
ci->configure(DSP_SWITCH_FREQUENCY, (int *)ci->id3->frequency);
current_frequency = ci->id3->frequency;
codec_set_replaygain(ci->id3);

2
apps/codecs/mpc.c
View file

@ -121,7 +121,7 @@ next_track:
goto done;
}
frequency = info.sample_freq / 1000;
ci->configure(DSP_SET_FREQUENCY, (long *)(long)info.sample_freq);
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(long)info.sample_freq);
/* set playback engine up for correct number of channels */
/* NOTE: current musepack format only allows for stereo files

2
apps/codecs/shorten.c
View file

@ -98,7 +98,7 @@ next_track:
}
ci->id3->frequency = sc.sample_rate;
ci->configure(DSP_SET_FREQUENCY, (long *)(long)(sc.sample_rate));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(long)(sc.sample_rate));
if (sc.sample_rate) {
ci->id3->length = (sc.totalsamples / sc.sample_rate) * 1000;

2
apps/codecs/sid.c
View file

@ -1267,7 +1267,7 @@ next_track:
/* Make use of 44.1khz */
ci->configure(DSP_SET_FREQUENCY, (long *)44100);
ci->configure(DSP_SWITCH_FREQUENCY, (long *)44100);
/* Sample depth is 28 bit host endian */
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
/* Mono output */

2
apps/codecs/vorbis.c
View file

@ -85,7 +85,7 @@ bool vorbis_set_codec_parameters(OggVorbis_File *vf)
return false;
}
rb->configure(DSP_SET_FREQUENCY, (int *)rb->id3->frequency);
rb->configure(DSP_SWITCH_FREQUENCY, (int *)rb->id3->frequency);
codec_set_replaygain(rb->id3);
if (vi->channels == 2) {

2
apps/codecs/wav.c
View file

@ -397,7 +397,7 @@ next_track:
goto done;
}
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(ci->id3->frequency));
if (channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_INTERLEAVED);
} else if (channels == 1) {

2
apps/codecs/wavpack.c
View file

@ -74,7 +74,7 @@ enum codec_status codec_start(struct codec_api* api)
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
ci->configure(DSP_SWITCH_FREQUENCY, (long *)(ci->id3->frequency));
codec_set_replaygain(ci->id3);
/* Create a decoder instance */

6
apps/dsp.c
View file

@ -1055,6 +1055,12 @@ bool dsp_configure(int setting, void *value)
dsp->new_gain = true;
break;
case DSP_FLUSH:
memset(&resample_data[current_codec], 0,
sizeof (struct resample_data));
dither_init();
break;
case DSP_SET_TRACK_GAIN:
dsp->track_gain = (long) value;
dsp->new_gain = true;

1
apps/dsp.h
View file

@ -39,6 +39,7 @@ enum {
DSP_SET_SAMPLE_DEPTH,
DSP_SET_STEREO_MODE,
DSP_RESET,
DSP_FLUSH,
DSP_SET_TRACK_GAIN,
DSP_SET_ALBUM_GAIN,
DSP_SET_TRACK_PEAK,

1
apps/playback.c
View file

@ -1679,6 +1679,7 @@ static void codec_seek_complete_callback(void)
{
/* If this is not a seamless seek, clear the buffer */
pcmbuf_play_stop();
dsp_configure(DSP_FLUSH, NULL);
/* If playback was not 'deliberately' paused, unpause now */
if (!paused)