Include ALSA's device name in the ALSA modules 'info' logging to help
identify sound devices as seen by the system for assisting config setup
Many configs use ALSA's hw ids to refer to device but ALSA can also use
device names:
laudio: Available ALSA playback mixer(s) on hw:0 CARD=Intel (HDA Intel): 'Master' 'Headphone' 'Speaker' 'PCM' 'Mic' 'Beep'
laudio: Available ALSA playback mixer(s) on hw:1 CARD=E30 (E30): 'E30 '
From the example above can use these ALSA names interchangably:
'hw:0' and 'hw:Intel'
'hw:1' and 'hw:E30'
Moves speaker selection, volume handling and startup to outputs.c, plus adds
the ability to "resurrect" a speaker that disconnects.
The purpose of moving the code is to concentrate device handling in one place.
Also changes how we deal with speaker selection. The player will now generally
not alter a user selection, even if the device fails. The purpose of this is to
maintain selection both if the device briefly fails, and if the user switches
off the device (we stop playback) and later turns it on + starts new playback.
The player will write 24 bit samples using 3 bytes, not 4, so the appropriate
sample format is SND_PCM_FORMAT_S24_3LE, not SND_PCM_FORMAT_S24_LE.
For extra protection we also use snd_pcm_bytes_to_frames() instead of BTOS(),
because that way we can be more certain that the buffer is not too short for
snd_pcm_writei().
* Fix "clicks" during playback, especially on low buffer size devices
Bug had two causes: Trying to write to the prebuf ringbuffer when it was full
and writing new audio to the device without first having drained the prebuf,
thus writing out of order.
* Use snd_pcm_drain() so alsa doesn't report underrun on playback session end
Removes SNDRV_PCM_IOCTL_SYNC_PTR errors
* Fix missing error check of the return value from snd_pcm_avail (now use snd_pcm_avail_delay)
In the output implementations playback_stop() was somewhat redundant,
since device_stop() does the same.
The timer should make sure that we always close outputs (previously
they were in some cases kept open).
The commit also includes some renaming.
* Untie Airtunes stuff further from player and non-Airplay outputs
* Change raop.c to use rtp_common.c (step 1)
* Change heartbeat of player to 100 ticks/sec, since we have untied from
Airtunes 352 samples per packet (which equals 126 ticks/sec at 44100)
Still a lot to be done in the player, since the rtptime's in it don't
are probably broken.
The unconfigurable resync period of 10 seconds was not frequent
enough to keep my own ALSA device in sync with the AirPlay stream.
Now the period is configurable. The default is still at 10
seconds, to prevent any change in behavior unless opted in by
the user.
Currently the adjustment causes a tiny "click" distortion in the
ALSA output, so it is better to make the check as infrequent as
possible, while still being frequent enough to stay in sync
over lengthy sessions of playback.
Added source_sample_rate, target_sample_rate to alsa_session.
This is a first step toward rendering ALSA at a different
sampling rate than the AirPlay stream, so that (a) we will
be able to dynamically adjust the ALSA sampling rate for an
improved sync algorithm, and (b) later, a more generalized
resampling algorithm can accommodate very different hardware
sampling rates like 22050 Hz or 48000 Hz.
Reworked alsa_session_free() so that it can be used to
tear down a partially initialized alsa_session if an
error occurs in the middle of alsa_session_make().
This simplifies the error handling logic in alsa_session_make().
This refactoring will be helpful later when resampling is added,
because more data structures will be dynamically allocated
during initialization.
Signed-off-by: Don Cross <cosinekitty@gmail.com>
Thanks to Denis Denisov and cppcheck for notifying about the below. The leaks
are edge cases, but the warning of dereference of avail in alsa.c points at
a bug that could probably cause actual crashes.
[src/evrtsp/rtsp.c:1352]: (warning) Assignment of function parameter has no effect outside the function. Did you forget dereferencing it?
[src/httpd_daap.c:228]: (error) Memory leak: s
[src/library.c:280]: (warning) %d in format string (no. 2) requires 'int' but the argument type is 'unsigned int'.
[src/library.c:284]: (warning) %d in format string (no. 2) requires 'int' but the argument type is 'unsigned int'.
[src/library/filescanner_playlist.c:251]: (error) Resource leak: fp
[src/library/filescanner_playlist.c:273]: (error) Resource leak: fp
[src/outputs/alsa.c:143]: (warning) Assignment of function parameter has no effect outside the function. Did you forget dereferencing it?
[src/outputs/alsa.c:657]: (warning) Possible null pointer dereference: avail
[src/outputs/dummy.c:75]: (warning) Assignment of function parameter has no effect outside the function. Did you forget dereferencing it?
[src/outputs/fifo.c:245]: (warning) Assignment of function parameter has no effect outside the function. Did you forget dereferencing it?
[src/outputs/raop.c:1806]: (warning) Assignment of function parameter has no effect outside the function. Did you forget dereferencing it?
[src/outputs/raop.c:1371]: (warning) %u in format string (no. 1) requires 'unsigned int' but the argument type is 'signed int'.
[src/outputs/raop.c:1471]: (warning) %u in format string (no. 1) requires 'unsigned int' but the argument type is 'signed int'.
[src/outputs/raop_verification.c:705] -> [src/outputs/raop_verification.c:667]: (warning) Either the condition 'if(len_M)' is redundant or there is possible null pointer dereference: len_M.
Support a separate mixer_device configuration file option for
advanced ALSA configurations. Previously, ALSA local output
happened to work becasue "default" is valid as both a PCM and a
mixer. Now you can separately specify the device name for PCM
output and mixer operations.
In my setup, I am using the following setup:
card = "default:CARD=NVidia"
mixer = "Front"
mixer_device = "hw:CARD=NVidia"
