HAVE_IO_PRIORITY was defined for native targets with dircache.
It is already effectively disabled for the most part since dircache no
longer lowers its thread's I/O priority. It existed primarily for the
aforementioned configuration.
Change-Id: Ia04935305397ba14df34647c8ea29c2acaea92aa
Invalid event data would be accessed if a play message isn't queued
which will cause crash problems.
It came about in the addition of time-based resume.
Change-Id: I1d5219064e2bf552b4183e9db4e7b380ffbe7a67
add_event_ex is added that takes an extra user_data pointer. This pointer is
passed to the callback (add_event and add_event_ex have slightly different
callbacks types). All callbacks also get the event id passed. Events added
with add_event_ex must be removed with remove_event_ex because the user_data
pointer must match in addition to the callback pointer.
On the other add_event is simplified to omit the oneshort parameter which
was almost always false (still there with add_event_ex).
As a side effect the ata_idle_notify callbacks are changed as well, they
do not take a data parameter anymore which was always NULL anyway.
This commit also adds some documentation to events.h
Change-Id: I13e29a0f88ef908f175b376d83550f9e0231f772
This complements offset-based resume and playback start funcionality.
The implementation is global on both HWCODEC and SWCODEC.
Basically, if either the specified elapsed or offset are non-zero,
it indicates a mid-track resume.
To resume by time only, set elapsed to nonzero and offset to zero.
To resume by offset only, set offset to nonzero and elapsed to zero.
Which one the codec uses and which has priority is up to the codec;
however, using an elapsed time covers more cases:
* Codecs not able to use an offset such as VGM or other atomic
formats
* Starting playback at a nonzero elapsed time from a source that
contains no offset, such as a cuesheet
The change re-versions pretty much everything from tagcache to nvram.
Change-Id: Ic7aebb24e99a03ae99585c5e236eba960d163f38
Reviewed-on: http://gerrit.rockbox.org/516
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested: Michael Sevakis <jethead71@rockbox.org>
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
* Remove explicit tracking of elapsed time of previous track.
* Remove function to obtain auto skip flag.
* Most playback events now carry the extra information instead and
pass 'struct track_event *' for data.
* Tweak scrobbler to use PLAYBACK_EVENT_TRACK_FINISH, which makes
it cleaner and removes the struct mp3entry.
Change-Id: I500d2abb4056a32646496efc3617406e36811ec5
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 <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
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>
Further decouples voice_thread.c from other playback areas. Also allows
other audio sources, such as FM radio, to be attenuated when voice is
playing by implementing a callback.
Defined as another playback event rather than a new event class:
PLAYBACK_EVENT_VOICE_PLAYING
Change-Id: I2e3e218be6cd6bebbf39e7883a8c0e4ed42b62bb
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
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
...by default where they would be interpreted as valid but not actually
be which would cause calls to buffering while it was not initialized.
Add BUFFER_EVENT_BUFFER_RESET to inform users of buffering that the
buffer is being reinitialized. Basically, this wraps all the
functionality being provided by three events (...START_PLAYBACK,
RECORDING_EVENT_START, RECORDING_EVENT_STOP) into one for radioart.c,
the only user of those events (perhaps remove them?) and closes some
loopholes.
Change-Id: I99ec46b9b5fb4e36605db5944c60ed986163db3a
Buffers are not allocated and thread is not created until the first
call where voice is required.
Adds a different callback (sync_callback) to buflib so that other
sorts of synchonization are possible, such as briefly locking-out the
PCM callback for a buffer move. It's sort of a messy addition but it
is needed so voice decoding won't have to be stopped when its buffer
is moved.
Change-Id: I4d4d8c35eed5dd15fb7ee7df9323af3d036e92b3
Moved to playback.c, since it doesn't use metadata from the music file.
Change-Id: I5c3ad7750d94b36754f64eb302f96ec163785cb9
Reviewed-on: http://gerrit.rockbox.org/142
Reviewed-by: Nils Wallménius <nils@rockbox.org>
This function has been changed to rbcodec_format_is_atomic, which
doesn't require an enum from the kernel.
Change-Id: I1d537605087fe130a9b545509d7b8a340806dbf2
Reviewed-on: http://gerrit.rockbox.org/141
Reviewed-by: Nils Wallménius <nils@rockbox.org>
Tested-by: Nils Wallménius <nils@rockbox.org>
When enabled, if the user has set "Start File Browser Here" (config.cfg:
start directory) to anything other than root and "Auto-Change Directory"
is set to "Yes" or "Random", the directory returned when an auto change
is required will be constrained to the value of "start directory" or below.
Change-Id: Iaab773868c4cab5a54f6ae67bdb22e84642a9e4b
Reviewed-on: http://gerrit.rockbox.org/182
Reviewed-by: Nick Peskett <rockbox@peskett.co.uk>
Tested-by: Nick Peskett <rockbox@peskett.co.uk>
Now all threads need to ack the connection like on real target, dircache is unloaded and playback stops accordingly.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31009 a1c6a512-1295-4272-9138-f99709370657
* shrinking now considers freespace just before the alloc-to-be-shrinked,
that means less (or sometimes none at all) is taken from the audio buffer.
* core_available() now searches for the best free space, instead of simply the end,
i.e. it will not return 0 if the audio buffer is allocated and there's free space
before it. It also runs a compaction to ensure maximum contiguous memory.
audio_buffer_available() is also enhanced. It now considers the 256K reserve buffer,
and returns free buflib space instead if the audio buffer is short.
This all fixes the root problem of FS#12344 (Sansa Clip+: PANIC occurred when
dircache is enabled), that alloced from the audio buffer, even if it was very
short and buflib had many more available as free space before it.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31006 a1c6a512-1295-4272-9138-f99709370657
Very frequent start-stop cycles (as caused by frequent core_alloc() calls)
of audio makes the codecs lose the resume position, and this causes playback
from the beginning.
To work around, use queue_post() instead of queue_send() to delay the resume
so that it only resumes once per core_alloc() set.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30900 a1c6a512-1295-4272-9138-f99709370657
audio playback. If it goes below 256K new buflib allocations fail.
This prevents buffer underruns as the new buffer size wasn't actually
checked at all.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30893 a1c6a512-1295-4272-9138-f99709370657
The buflib metadata gets corrupted at the new loation between core_shrink()
and actually applying, the new buffer boundaries (most probably due to yield()).
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30574 a1c6a512-1295-4272-9138-f99709370657
Reuse playback's Q_AUDIO_REMAKE_AUDIO_BUFFER capabilities to set the new
playback buffer, instead of stopping/restarting manual. This strongly
reduces the visibility of the short audio stop.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30480 a1c6a512-1295-4272-9138-f99709370657
This enables the ability to allocate (and free) memory dynamically
without fragmentation, through compaction. This means allocations can move
and fragmentation be reduced. Most changes are preparing Rockbox for this,
which many times means adding a move callback which can temporarily disable
movement when the corresponding code is in a critical section.
For now, the audio buffer allocation has a central role, because it's the one
having allocated most. This buffer is able to shrink itself, for which it
needs to stop playback for a very short moment. For this,
audio_buffer_available() returns the size of the audio buffer which can
possibly be used by other allocations because the audio buffer can shrink.
lastfm scrobbling and timestretch can now be toggled at runtime without
requiring a reboot.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30381 a1c6a512-1295-4272-9138-f99709370657
The buflib memory allocator is handle based and can free and
compact, move or resize memory on demand. This allows to effeciently
allocate memory dynamically without an MMU, by avoiding fragmentation
through memory compaction.
This patch adds the buflib library to the core, along with
convinience wrappers to omit the context parameter. Compaction is
not yet enabled, but will be in a later patch. Therefore, this acts as a
replacement for buffer_alloc/buffer_get_buffer() with the benifit of a debug
menu.
See buflib.h for some API documentation.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30380 a1c6a512-1295-4272-9138-f99709370657
buffer chunks.
* Samples and position indication is closely associated with audio data
instead of compensating by a latency constant. Alleviates problems with
using the elapsed as a track indicator where it could be off by several
steps.
* Timing is accurate throughout track even if resampling for pitch shift,
whereas before it updated during transition latency at the normal 1:1 rate.
* Simpler PCM buffer with a constant chunk size, no linked lists.
In converting crossfade, a minor change was made to not change the WPS until
the fade-in of the incoming track, whereas before it would change upon the
start of the fade-out of the outgoing track possibly having the WPS change
with far too much lead time.
Codec changes are to set elapsed times *before* writing next PCM frame because
time and position data last set are saved in the next committed PCM chunk.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657
Namely, introduce buffer_get_buffer() and buffer_release_buffer().
buffer_get_buffer() aquires all available and grabs a lock, attempting to
call buffer_alloc() or buffer_get_buffer() while this lock is locked will cause
a panicf() (doesn't actually happen, but is for debugging purpose).
buffer_release_buffer() unlocks that lock and can additionally increment the
audiobuf buffer to make an allocation. Pass 0 to only unlock if buffer was
used temporarily only.
buffer_available() is a replacement function to query audiobuflen, i.e. what's
left in the buffer.
Buffer init is moved up in the init chain and handles ipodvideo64mb internally.
Further changes happened to mp3data.c and talk.c as to not call the above API
functions, but get the buffer from callers. The caller is the audio system
which has the buffer lock while mp3data.c and talk mess with the buffer.
mpeg.c now implements some buffer related functions of playback.h, especially
audio_get_buffer(), allowing to reduce #ifdef hell a tiny bit.
audiobuf and audiobufend are local to buffer.c now.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30308 a1c6a512-1295-4272-9138-f99709370657
This is done to make reboot more transparent. If a playlist has ended, there should be no difference between the player doing nothing for ten minutes and the player shutting down after the idle timeout and being restarted.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30226 a1c6a512-1295-4272-9138-f99709370657
