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6 commits

Author SHA1 Message Date
Thomas Martitz
428f069a1f talk: Remove superflous call to talk_init().
talk_init() is called by settings_apply() already which is called on boot.
Doing it again means loading the voicefile a second time which isn't necessary.

Change-Id: I4decd16401e63bf87338d3163c99d06d69fe3a3c
2014-02-02 19:40:38 +01:00
Michael Sevakis
9b990bdab1 SWCODEC Audio: Add some INIT_ATTR's to get a few bytes back.
Change-Id: Ie7b04ecf3b3535e0ed45a6e0e8d81af89e38378e
2013-06-29 22:29:23 -04:00
Michael Sevakis
4888131972 Update software recording engine to latest codec interface.
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 <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
2013-06-30 00:40:27 +02:00
Michael Sevakis
92f1ecfc7b Just init audio thread event with Q_NULL instead of SYS_TIMEOUT.
What was I thinking? Anyway, it's a little nicer and it's reserved so
shouldn't end up as a case value anywhere.

Change-Id: If745c24951518608e3f4fc51f7419ef07d7cf2ac
2013-05-31 05:17:11 -04:00
Michael Sevakis
344b9d0986 Some corrections after 5857c44.
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
2013-05-31 04:13:39 -04:00
Michael Sevakis
5857c44017 Refactor audio thread to run both recording and playback.
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
2013-05-31 03:20:35 -04:00