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Fundamentally rewrite much of the audio DSP.

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Creates a standard buffer passing, local data passing and messaging
system for processing stages. Stages can be moved to their own source
files to reduce clutter and ease assimilation of new ones. dsp.c
becomes dsp_core.c which supports an engine and framework for effects.

Formats and change notifications are passed along with the buffer so
that they arrive at the correct time at each stage in the chain
regardless of the internal delays of a particular one.

Removes restrictions on the number of samples that can be processed at
a time and it pays attention to destination buffer size restrictions
without having to limit input count, which also allows pcmbuf to
remain fuller and safely set its own buffer limits as it sees fit.
There is no longer a need to query input/output counts given a certain
number of input samples; just give it the sizes of the source and
destination buffers.

Works in harmony with stages that are not deterministic in terms of
sample input/output ratio (like both resamplers but most notably
the timestretch). As a result it fixes quirks with timestretch hanging
up with certain settings and it now operates properly throughout its
full settings range.
Change-Id: Ib206ec78f6f6c79259c5af9009fe021d68be9734
Reviewed-on: http://gerrit.rockbox.org/200
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
This commit is contained in:
Michael Sevakis 2012-03-27 19:52:15 -04:00
parent c9c1349773
commit c9bcbe202d
56 changed files with 4823 additions and 2998 deletions
Download patch file Download diff file Expand all files Collapse all files

56
apps/codec_thread.c
View file

@ -213,49 +213,41 @@ void codec_thread_do_callback(void (*fn)(void), unsigned int *id)
static void codec_pcmbuf_insert_callback(
const void *ch1, const void *ch2, int count)
{
const char *src[2] = { ch1, ch2 };
struct dsp_buffer src;
while (count > 0)
src.remcount = count;
src.pin[0] = ch1;
src.pin[1] = ch2;
src.proc_mask = 0;
while (1)
{
int out_count = dsp_output_count(ci.dsp, count);
int inp_count;
char *dest;
struct dsp_buffer dst;
dst.remcount = 0;
dst.bufcount = MAX(src.remcount, 1024); /* Arbitrary min request */
while (1)
while ((dst.p16out = pcmbuf_request_buffer(&dst.bufcount)) == NULL)
{
if ((dest = pcmbuf_request_buffer(&out_count)) != NULL)
break;
cancel_cpu_boost();
/* It will be awhile before space is available but we want
/* It may be awhile before space is available but we want
"instant" response to any message */
queue_wait_w_tmo(&codec_queue, NULL, HZ/20);
if (!queue_empty(&codec_queue) &&
codec_check_queue__have_msg() < 0)
{
dsp_configure(ci.dsp, DSP_FLUSH, 0); /* Discontinuity */
return;
}
}
/* Get the real input_size for output_size bytes, guarding
* against resampling buffer overflows. */
inp_count = dsp_input_count(ci.dsp, out_count);
dsp_process(ci.dsp, &src, &dst);
if (inp_count <= 0)
return;
/* Input size has grown, no error, just don't write more than length */
if (inp_count > count)
inp_count = count;
out_count = dsp_process(ci.dsp, dest, src, inp_count);
if (out_count <= 0)
return;
pcmbuf_write_complete(out_count, ci.id3->elapsed, ci.id3->offset);
count -= inp_count;
if (dst.remcount > 0)
pcmbuf_write_complete(dst.remcount, ci.id3->elapsed, ci.id3->offset);
else if (src.remcount <= 0)
break; /* No input remains and DSP purged */
}
}
@ -352,10 +344,7 @@ static void codec_seek_complete_callback(void)
static void codec_configure_callback(int setting, intptr_t value)
{
if (!dsp_configure(ci.dsp, setting, value))
{
logf("Illegal key: %d", setting);
}
dsp_configure(ci.dsp, setting, value);
}
static enum codec_command_action
@ -611,8 +600,7 @@ static void NORETURN_ATTR codec_thread(void)
void codec_thread_init(void)
{
/* Init API */
ci.dsp = (struct dsp_config *)dsp_configure(NULL, DSP_MYDSP,
CODEC_IDX_AUDIO);
ci.dsp = dsp_get_config(CODEC_IDX_AUDIO);
ci.codec_get_buffer = codec_get_buffer_callback;
ci.pcmbuf_insert = codec_pcmbuf_insert_callback;
ci.set_elapsed = audio_codec_update_elapsed;