rockbox/apps/recorder/pcm_record.c, branch working

recording: Fix sloppy return value check. This caused an OOM panic.

2013-12-24T09:20:13+00:00

Change-Id: I30bbc5623d81ad7cbbee52a439fb5909af2761b7

playback,talk: Share audiobuffer via core_alloc_maximum().

2013-12-23T11:17:38+00:00

This fixes the radioart crash that was the result of buffering.c working
on a freed buffer at the same time as buflib (radioart uses buffering.c for the
images). With this change the buffer is owned by buflib exclusively so this
cannot happen.

As a result, audio_get_buffer() doesn't exist anymore. Callers should call
core_alloc_maximum() directly. This buffer needs to be protected as usual
against movement if necessary (previously it was not protected at all which
cased the radioart crash), To get most of it they can adjust the willingness of
the talk engine to give its buffer away (at the expense of disabling voice
interface) with the new talk_buffer_set_policy() function.

Change-Id: I52123012208d04967876a304451d634e2bef3a33

SWCODEC recording: Boost CPU frequency before doing flush on stop

2013-07-09T11:36:22+00:00

If CPU is not boosted for some reason already, then the stop flush
can take longer than it really ought to.

Change-Id: I0572cc83067749e9945b3eb825f976db21d914f9

Fix strlcpy warnings in 4888131.

2013-06-29T23:27:21+00:00

Change-Id: Ibd355ee1cf39d1a4b07def94634f0316468bf727

Update software recording engine to latest codec interface.

2013-06-29T22:40:27+00:00

Basically, just give it a good rewrite.

Software codec recording can be implemented in a more straightforward
and simple manner and made more robust through the better codec
control now available.

Encoded audio buffer uses a packed format instead of fixed-size
chunks and uses smaller data headers leading to more efficient usage.
The greatest benefit is with a VBR format like wavpack which needs
to request a maximum size but only actually ends up committing part
of that request.

No guard buffers are used for either PCM or encoded audio. PCM is
read into the codec's provided buffer and mono conversion done at
that time in the core if required. Any highly-specialized sample
conversion is still done within the codec itself, such as 32-bit
(wavpack) or interleaved mono (mp3).

There is no longer a separate filename array. All metadata goes
onto the main encoded audio buffer, eliminating any predermined
file limit on the buffer as well as not wasting the space for
unused path queue slots.

The core and codec interface is less awkward and a bit more sensible.
Some less useful interface features were removed. Threads are kept
on narrow code paths ie. the audio thread never calls encoding
functions and the codec thread never calls file functions as before.

Codecs no longer call file functions directly. Writes are buffered
in the core and data written to storage in larger chunks to speed up
flushing of data. In fact, codecs are no longer aware of the stream
being a file at all and have no access to the fd.

SPDIF frequency detection no longer requires a restart of recording
or plugging the source before entering the screen. It will poll
for changes and update when stopped or prerecording (which does
discard now-invalid prerecorded data).

I've seen to it that writing a proper header on full disk works
when the format makes it reasonably practical to do so. Other cases
may have incorrect data sizes but sample info will be in tact. File
left that way may play anyway.

mp3_enc.codec acquires the ability to write 'Info' headers with LAME
tags to make it gapless (bonus).

Change-Id: I670685166d5eb32ef58ef317f50b8af766ceb653
Reviewed-on: http://gerrit.rockbox.org/493
Reviewed-by: Michael Sevakis 
Tested-by: Michael Sevakis

Fix some whitespace in files changed in following commit.

2013-06-29T22:40:09+00:00

Change-Id: Ie3f43e43076e0dcae9a10f1b0b9e4698b398acee
Reviewed-on: http://gerrit.rockbox.org/492
Reviewed-by: Michael Sevakis 
Tested-by: Michael Sevakis

Some corrections after 5857c44.

2013-05-31T08:13:39+00:00

Playback needs to receive a couple of settings-related messages even
when not playing.

Put the message reply back where it was when loading an encoder for
recording.

Change-Id: I8cc80f46e42a0afd119991d698510e1ebef38ead

Refactor audio thread to run both recording and playback.

2013-05-31T07:20:35+00:00

Eliminates the pcmrec thread and keeps playback and recording engine
operation mutually-exclusive.

audio_thread.c contains the audio thread which branches to the
correct engine depending upon the request. It also handles the main
audio initialization.

Moves pcm_init into main.c just before dsp_init because I don't want
that one in audio_init in the new file.

(Also makes revision df6e1bc pointless ;)

Change-Id: Ifc1db24404e6d8dd9ac42d9f4dfbc207aa9a26e1

pcm_record: Track initialization state

2013-05-30T22:42:59+00:00

It should not access audio hardware and change settings unless it has
been initialized first and given control of it.

Change-Id: I5004602d7caa604ded751f6838b792d1ff24b3fb

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