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authorMiika Pekkarinen <miipekk@ihme.org>2005-07-07 07:15:05 +0000
committerMiika Pekkarinen <miipekk@ihme.org>2005-07-07 07:15:05 +0000
commit3eb962d13bd4ca8c29ab33c41428a44e644e59ec (patch)
tree6ad56c24745868d9c56a56dc1278cfde4f5573ff /apps
parent8d3855eb536d4b8f1459c9b2da3beb5a0ac328b4 (diff)
downloadrockbox-3eb962d13bd4ca8c29ab33c41428a44e644e59ec.tar.gz
rockbox-3eb962d13bd4ca8c29ab33c41428a44e644e59ec.zip
PCM buffering fixes. Made a temporary workaround for playback glitch
bug (see the patch). git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7049 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps')
-rw-r--r--apps/codecs.c3
-rw-r--r--apps/codecs/mpa.c20
-rw-r--r--apps/playback.c49
3 files changed, 39 insertions, 33 deletions
diff --git a/apps/codecs.c b/apps/codecs.c
index d3a9d9e9c1..400e7fbfcf 100644
--- a/apps/codecs.c
+++ b/apps/codecs.c
@@ -277,6 +277,9 @@ int codec_load_file(const char *plugin)
277 int fd; 277 int fd;
278 int rc; 278 int rc;
279 279
280 /* zero out codec buffer to ensure a properly zeroed bss area */
281 memset(codecbuf, 0, CODEC_SIZE);
282
280 fd = open(plugin, O_RDONLY); 283 fd = open(plugin, O_RDONLY);
281 if (fd < 0) { 284 if (fd < 0) {
282 snprintf(msgbuf, sizeof(msgbuf)-1, "Couldn't load codec: %s", plugin); 285 snprintf(msgbuf, sizeof(msgbuf)-1, "Couldn't load codec: %s", plugin);
diff --git a/apps/codecs/mpa.c b/apps/codecs/mpa.c
index 5cf4eb8730..5d6f7d29ad 100644
--- a/apps/codecs/mpa.c
+++ b/apps/codecs/mpa.c
@@ -39,12 +39,6 @@ void abort(void) {
39 39
40 40
41#define INPUT_CHUNK_SIZE 8192 41#define INPUT_CHUNK_SIZE 8192
42#define OUTPUT_BUFFER_SIZE 65536 /* Must be an integer multiple of 4. */
43
44unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
45unsigned char *OutputPtr;
46unsigned char *GuardPtr = NULL;
47const unsigned char *OutputBufferEnd = OutputBuffer + OUTPUT_BUFFER_SIZE;
48 42
49mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR; 43mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
50unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR; 44unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR;
@@ -115,7 +109,6 @@ enum codec_status codec_start(struct codec_api* api)
115 109
116 first_frame = false; 110 first_frame = false;
117 file_end = 0; 111 file_end = 0;
118 OutputPtr = OutputBuffer;
119 112
120 while (!*ci->taginfo_ready) 113 while (!*ci->taginfo_ready)
121 ci->yield(); 114 ci->yield();
@@ -195,7 +188,7 @@ enum codec_status codec_start(struct codec_api* api)
195 } 188 }
196 else if(MAD_RECOVERABLE(Stream.error)) 189 else if(MAD_RECOVERABLE(Stream.error))
197 { 190 {
198 if(Stream.error!=MAD_ERROR_LOSTSYNC || Stream.this_frame!=GuardPtr) 191 if(Stream.error!=MAD_ERROR_LOSTSYNC)
199 { 192 {
200 // rb->splash(HZ*1, true, "Recoverable...!"); 193 // rb->splash(HZ*1, true, "Recoverable...!");
201 } 194 }
@@ -209,9 +202,9 @@ enum codec_status codec_start(struct codec_api* api)
209 Status=1; 202 Status=1;
210 break; 203 break;
211 } 204 }
205 break ;
212 } 206 }
213 if (Stream.next_frame) 207
214 ci->advance_buffer_loc((void *)Stream.next_frame);
215 file_end = false; 208 file_end = false;
216 /* ?? Do we need the timer module? */ 209 /* ?? Do we need the timer module? */
217 // mad_timer_add(&Timer,Frame.header.duration); 210 // mad_timer_add(&Timer,Frame.header.duration);
@@ -222,7 +215,7 @@ enum codec_status codec_start(struct codec_api* api)
222 /* We skip start_skip number of samples here, this should only happen for 215 /* We skip start_skip number of samples here, this should only happen for
223 very first frame in the stream. */ 216 very first frame in the stream. */
224 /* TODO: possible for start_skip to exceed one frames worth of samples? */ 217 /* TODO: possible for start_skip to exceed one frames worth of samples? */
225 218
226 if (MAD_NCHANNELS(&Frame.header) == 2) { 219 if (MAD_NCHANNELS(&Frame.header) == 2) {
227 if (current_stereo_mode != STEREO_NONINTERLEAVED) { 220 if (current_stereo_mode != STEREO_NONINTERLEAVED) {
228 ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED); 221 ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
@@ -241,6 +234,11 @@ enum codec_status codec_start(struct codec_api* api)
241 } 234 }
242 start_skip = 0; /* not very elegant, and might want to keep this value */ 235 start_skip = 0; /* not very elegant, and might want to keep this value */
243 236
237 if (Stream.next_frame)
238 ci->advance_buffer_loc((void *)Stream.next_frame);
239 else
240 ci->advance_buffer(size);
241
244 samplesdone += Synth.pcm.length; 242 samplesdone += Synth.pcm.length;
245 samplecount -= Synth.pcm.length; 243 samplecount -= Synth.pcm.length;
246 ci->set_elapsed(samplesdone / (frequency_divider / 10)); 244 ci->set_elapsed(samplesdone / (frequency_divider / 10));
diff --git a/apps/playback.c b/apps/playback.c
index 13c66a43b5..f9caff1c4f 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -659,12 +659,12 @@ void audio_fill_file_buffer(void)
659 buf_widx -= codecbuflen; 659 buf_widx -= codecbuflen;
660 i += rc; 660 i += rc;
661 tracks[track_widx].available += rc; 661 tracks[track_widx].available += rc;
662 tracks[track_widx].filerem -= rc;
663 tracks[track_widx].filepos += rc;
662 codecbufused += rc; 664 codecbufused += rc;
663 fill_bytesleft -= rc; 665 fill_bytesleft -= rc;
664 } 666 }
665 667
666 tracks[track_widx].filerem -= i;
667 tracks[track_widx].filepos += i;
668 /*logf("Filled:%d/%d", tracks[track_widx].available, 668 /*logf("Filled:%d/%d", tracks[track_widx].available,
669 tracks[track_widx].filerem);*/ 669 tracks[track_widx].filerem);*/
670} 670}
@@ -890,26 +890,29 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
890 890
891 /* Starting playback from an offset is only support in MPA at the moment */ 891 /* Starting playback from an offset is only support in MPA at the moment */
892 if (offset > 0) { 892 if (offset > 0) {
893 if ((tracks[track_widx].id3.codectype==AFMT_MPA_L2) || 893 switch (tracks[track_widx].id3.codectype) {
894 (tracks[track_widx].id3.codectype==AFMT_MPA_L3)) { 894 case AFMT_MPA_L2:
895 lseek(fd, offset, SEEK_SET); 895 case AFMT_MPA_L3:
896 tracks[track_widx].id3.offset = offset; 896 lseek(fd, offset, SEEK_SET);
897 mp3_set_elapsed(&tracks[track_widx].id3); 897 tracks[track_widx].id3.offset = offset;
898 tracks[track_widx].filepos = offset; 898 mp3_set_elapsed(&tracks[track_widx].id3);
899 tracks[track_widx].filerem = tracks[track_widx].filesize - offset; 899 tracks[track_widx].filepos = offset;
900 ci.curpos = offset; 900 tracks[track_widx].filerem = tracks[track_widx].filesize - offset;
901 tracks[track_widx].start_pos = offset; 901 ci.curpos = offset;
902 } 902 tracks[track_widx].start_pos = offset;
903 else if (tracks[track_widx].id3.codectype==AFMT_WAVPACK) { 903 break;
904 lseek(fd, offset, SEEK_SET); 904
905 tracks[track_widx].id3.offset = offset; 905 case AFMT_WAVPACK:
906 tracks[track_widx].id3.elapsed = tracks[track_widx].id3.length / 2; 906 lseek(fd, offset, SEEK_SET);
907 tracks[track_widx].filepos = offset; 907 tracks[track_widx].id3.offset = offset;
908 tracks[track_widx].filerem = tracks[track_widx].filesize - offset; 908 tracks[track_widx].id3.elapsed = tracks[track_widx].id3.length / 2;
909 ci.curpos = offset; 909 tracks[track_widx].filepos = offset;
910 tracks[track_widx].start_pos = offset; 910 tracks[track_widx].filerem = tracks[track_widx].filesize - offset;
911 } 911 ci.curpos = offset;
912 } 912 tracks[track_widx].start_pos = offset;
913 break;
914 }
915 }
913 916
914 if (start_play) { 917 if (start_play) {
915 track_count++; 918 track_count++;
@@ -1795,6 +1798,8 @@ void audio_init(void)
1795 track_buffer_callback = NULL; 1798 track_buffer_callback = NULL;
1796 track_unbuffer_callback = NULL; 1799 track_unbuffer_callback = NULL;
1797 track_changed_callback = NULL; 1800 track_changed_callback = NULL;
1801 /* Just to prevent cur_ti never be anything random. */
1802 cur_ti = &tracks[0];
1798 1803
1799 logf("abuf:%0x", PCMBUF_SIZE); 1804 logf("abuf:%0x", PCMBUF_SIZE);
1800 logf("fbuf:%0x", codecbuflen); 1805 logf("fbuf:%0x", codecbuflen);