rockbox/apps/pcmbuf.c, branch wolf3d

Fix a problem with audio not starting on a list of short files

2017-04-06T23:32:35+00:00

Forced audio start was left out when a third codec attempts to
start a second track transition. Only one pending transition is
allowed at a time. There wouldn't be enough PCM in the buffer to
trigger audio playback and audio would just return without giving
the pcm buffer a kick.

Fixes FS#13100 - Player failed on short tracks

Change-Id: I338b0b12022c591930451fd5ed26a2a73008835f

Fix previous errors for targets with no crossfade.

2017-01-05T08:22:47+00:00

Change-Id: I1c10f5588f3fe06c7071f3f07b68e3d552c44f05

Playback: Fix problems with crossfade on short tracks.

2017-01-05T07:37:14+00:00

Addresses issues brought up in this thread:
http://forums.rockbox.org/index.php/topic,51605.0.html

While we're at it, improve the quality with a sample-level fader.

Change-Id: I73dde60d6858a1c9042812e26d490739e3906a1e

Enable setting of global output samplerate on certain targets.

2013-07-06T02:22:04+00:00

Replaces the NATIVE_FREQUENCY constant with a configurable frequency.

The user may select 48000Hz if the hardware supports it. The default is
still 44100Hz and the minimum is 44100Hz. The setting is located in the
playback settings, under "Frequency".

"Frequency" was duplicated in english.lang for now to avoid having to
fix every .lang file for the moment and throwing everything out of sync
because of the new play_frequency feature in features.txt. The next
cleanup should combine it with the one included for recording and
generalize the ID label.

If the hardware doesn't support 48000Hz, no setting will be available.

On particular hardware where very high rates are practical and desireable,
the upper bound can be extended by patching.

The PCM mixer can be configured to play at the full hardware frequency
range. The DSP core can configure to the hardware minimum up to the
maximum playback setting (some buffers must be reserved according to
the maximum rate).

If only 44100Hz is supported or possible on a given target for playback,
using the DSP and mixer at other samperates is possible if the hardware
offers them.

Change-Id: I6023cf0c0baa8bc6292b6919b4dd3618a6a25622
Reviewed-on: http://gerrit.rockbox.org/479
Reviewed-by: Michael Sevakis 
Tested-by: Michael Sevakis

Fix a bug in pcmbuf.c when doing offset with modulus.

2013-06-28T06:17:58+00:00

Causes the track change to go in the wrong buffer (and even be missed
by playback) if the current descriptor for the write index is descriptor
0 because the offset used is "-1" upon track change.

Got nailed by implicit conversion of the % operator dividend to unsigned.

Change-Id: I32538db801ac9d790c8b1b5bd041b09ad4b64d2e

Fix whitespace for next commit

2013-06-28T06:17:17+00:00

Change-Id: I8773ff7275372882321ebf2e66462781735cddc6

Make rbcodec/dsp includes more specific.

2012-04-29T21:31:30+00:00

Change-Id: Idb6af40df26f5b8499a40e8b98602261ef227044

Fundamentally rewrite much of the audio DSP.

2012-04-29T08:00:56+00:00

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 
Tested-by: Michael Sevakis

Fix up some more type stuff in pcmbuf.c.

2012-04-27T20:51:54+00:00

(Never seem to find it all at once :-)

Change-Id: I4b3d145c6d90be13f9afc8a72d8d87a92de88d88

Revise the PCM callback system after adding multichannel audio.

2012-03-03T06:23:38+00:00

Additional status callback is added to pcm_play/rec_data instead of
using a special function to set it. Status includes DMA error
reporting to the status callback. Playback and recording callback
become more alike except playback uses "const void **addr" (because
the data should not be altered) and recording  uses "void **addr".
"const" is put in place throughout where appropriate.

Most changes are fairly trivial. One that should be checked in
particular because it isn't so much is telechips, if anyone cares to
bother. PP5002 is not so trivial either but that tested as working.

Change-Id: I4928d69b3b3be7fb93e259f81635232df9bd1df2
Reviewed-on: http://gerrit.rockbox.org/166
Reviewed-by: Michael Sevakis 
Tested-by: Michael Sevakis