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FS#10504: Make mpegplayer audio thread use correct sample count

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@22280 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Jeffrey Goode 2009-08-12 18:28:30 +00:00
parent 0dc2fb5760
commit d5592a6710
1 changed files with 12 additions and 3 deletions
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15
apps/plugins/mpegplayer/audio_thread.c
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@ -643,8 +643,8 @@ static void audio_thread(void)
struct pcm_frame_header *dst_hdr = pcm_output_get_buffer();
const char *src[2] =
{ (char *)synth.pcm.samples[0], (char *)synth.pcm.samples[1] };
int out_count = (synth.pcm.length * CLOCK_RATE
+ (td.samplerate - 1)) / td.samplerate;
int out_count = rb->dsp_output_count(td.dsp, (synth.pcm.length *
CLOCK_RATE + (td.samplerate - 1)) / td.samplerate);
ssize_t size = sizeof(*dst_hdr) + out_count*4;
/* Wait for required amount of free buffer space */
@ -657,8 +657,17 @@ static void audio_thread(void)
goto message_process;
}
int inp_count = rb->dsp_input_count(td.dsp, out_count);
if (inp_count <= 0)
break;
/* Input size has grown, no error, just don't write more than length */
if (inp_count > synth.pcm.length)
inp_count = synth.pcm.length;
out_count = rb->dsp_process(td.dsp, dst_hdr->data, src,
synth.pcm.length);
inp_count);
if (out_count <= 0)
break;