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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/plugins/mpegplayer/audio_thread.c
View file

@ -36,6 +36,7 @@ struct audio_thread_data
unsigned samplerate; /* Current stream sample rate */
int nchannels; /* Number of audio channels */
struct dsp_config *dsp; /* The DSP we're using */
struct dsp_buffer src; /* Current audio data for DSP processing */
};
/* The audio thread is stolen from the core codec thread */
@ -479,12 +480,13 @@ static void audio_thread(void)
/* We need this here to init the EMAC for Coldfire targets */
init_mad();
td.dsp = (struct dsp_config *)rb->dsp_configure(NULL, DSP_MYDSP,
CODEC_IDX_AUDIO);
td.dsp = rb->dsp_get_config(CODEC_IDX_AUDIO);
#ifdef HAVE_PITCHSCREEN
rb->sound_set_pitch(PITCH_SPEED_100);
rb->dsp_set_timestretch(PITCH_SPEED_100);
#endif
rb->dsp_configure(td.dsp, DSP_RESET, 0);
rb->dsp_configure(td.dsp, DSP_FLUSH, 0);
rb->dsp_configure(td.dsp, DSP_SET_SAMPLE_DEPTH, MAD_F_FRACBITS);
goto message_wait;
@ -631,43 +633,53 @@ static void audio_thread(void)
STEREO_MONO : STEREO_NONINTERLEAVED);
}
td.src.remcount = synth.pcm.length;
td.src.pin[0] = synth.pcm.samples[0];
td.src.pin[1] = synth.pcm.samples[1];
td.src.proc_mask = 0;
td.state = TSTATE_RENDER_WAIT;
/* Add a frame of audio to the pcm buffer. Maximum is 1152 samples. */
render_wait:
if (synth.pcm.length > 0)
rb->yield();
while (1)
{
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;
unsigned char *out_buf;
ssize_t size = out_count*4;
struct dsp_buffer dst;
dst.remcount = 0;
dst.bufcount = MAX(td.src.remcount, 1024);
ssize_t size = dst.bufcount * 2 * sizeof(int16_t);
/* Wait for required amount of free buffer space */
while ((out_buf = pcm_output_get_buffer(&size)) == NULL)
while ((dst.p16out = pcm_output_get_buffer(&size)) == NULL)
{
/* Wait one frame */
int timeout = out_count*HZ / td.samplerate;
int timeout = dst.bufcount*HZ / td.samplerate;
str_get_msg_w_tmo(&audio_str, &td.ev, MAX(timeout, 1));
if (td.ev.id != SYS_TIMEOUT)
goto message_process;
}
out_count = rb->dsp_process(td.dsp, out_buf, src, synth.pcm.length);
dst.bufcount = size / (2 * sizeof (int16_t));
rb->dsp_process(td.dsp, &td.src, &dst);
if (out_count <= 0)
if (dst.remcount > 0)
{
/* Make this data available to DMA */
pcm_output_commit_data(dst.remcount * 2 * sizeof(int16_t),
audio_queue.curr->time);
/* As long as we're on this timestamp, the time is just
incremented by the number of samples */
audio_queue.curr->time += dst.remcount;
}
else if (td.src.remcount <= 0)
{
break;
/* Make this data available to DMA */
pcm_output_commit_data(out_count*4, audio_queue.curr->time);
/* As long as we're on this timestamp, the time is just
incremented by the number of samples */
audio_queue.curr->time += out_count;
}
}
rb->yield();
} /* end decoding loop */
}