+ remember to flush in source_start(), since the input won't do it if
input_now_reading has already been closed (e.g. if starting a new track
while playback is at the end of another track)
Player now stops 10 secs after stop command and 10 mins after pause. At
that time the outputs have probably cut the connection themselves, but
that might be ok (needs testing).
* Also call full_cb() from input_wait if buffer is full
* Make read_deficit count missing bytes instead of clock ticks
* Make read_deficit a part of the playback session
The time stamp was getting set too late, because if pos was zero the first
reads then it would be overwritten, but it shouldn't because the loop will
catch up even if the initial reads have zero samples.
* Drop output_sessions, was just a pointer to the actual session anyway
* Drop the old write, flush and stop prototypes
* Some minor changes/renaming
Purpose of this is also to fix a race condition in player.c where it
could try to start two sessions on the same speaker. This could happen
because outputs_device_start() in line 2093 is conditional on device->session
which however is false while a device is starting up.
outputs_playback_start() had the problem that was not consistently invoked: If
for instance local audio playback was running and a Airplay device was then
activated, the raop's playback_start would never be invoked (and vice versa,
of course).
Instead, the player now writes the presentation timestamp every time to the
output, so it doesn't need to keep track of it from the start.
* 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.
Output module can now take input data in multiple quality levels, and
can resample to those output modules that would require a certain quality
level, like raop.c would
Extends the http_client_ctx to hold the response code for a request.
Also adds the content type header, if it was a https request (using
libcurl instead of libevent)
This adds a new timestamp value "db_modified" into the admin db table.
In addition to the existing "db_update" admin value, this value is also
updated if rating, play-/skip-count or seek changes for a
media_info_file (files db table).
This should improve the caching behavior in clients of the JSON API
(especially the player web interface) in refreshing its data if some of
this values changes.
New endpoint is PUT api/library/tracks/[id] and supported query
parameters are:
- rating: with values between 0 and 100
- play_count: with values "reset" (resets play_count and skip_count) or
"increment" (increments play_count)
"Play Date" tag was seconds since 1904 (an Apple Mac HFS+ timestamp), not a
Unix timestamp as we assumed. Seems Apple themselves realised that wasn't a
great idea (+ not a proper plist date type), and therefore provide "Play Date
UTC" as an alternative.
If a file gets updated/rescanned we generally don't want to reset the above
values. This commit adds DB_FLAG_NO_ZERO, which marks a field so that
db_file_update() will only update it if the new value is non-zero (i.e. the
caller probably has a "better" value).
This change means that we will use album_artist even if compilation_artist
is configured, thus compilation_artist will only be used to override artist.
The constraints on songalbumid and songartistid where changed with v20, so we
need to make sure they take effect when upgrading.
This commit tries to do the table recreation like sqlite recommends and without
manually crafted copy queries that are probably prone to errors.
Since we are recreating anyway, this commit also reorders the columns slightly.
It also includes auto-drop/recreation of triggers (should really have been its
own commit) during upgrade, like is already done with indices.
The idea here is to make sure the fixing up of tags is done in a consistent
manner. For strings, this means stuff like trimming and empty strings -> null
are applied the same unless there are special exception rules set. It also
means that defaults are applied the same across structs, e.g. "Unknown artist"
for both mfi->artist and queue_item->artist.
The change is also necessary because we want to remove trimming from the sql
query and instead implement it ourselves.
is not the current playing item.
This happens if the input already switched to the next item in the queue
starting to stream it to the outputs (2 second buffer) while the outputs
are still playing the last seconds of the old item.
Add check against the special file_id DB_MEDIA_FILE_NON_PERSISTENT_ID to
identify if a queue item is not in the library. And always prefer the
artwork url in the queue item before the artwork for the library file.
Some remotes don't respond as expected to the test. Retune will give connection
refused, because the test is made too quickly, before the service is running.
Even if we delay the test it won't work because Retune crashes.
Since the false mdns advertisements are only seen on Airplay, we only do the
test there.
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>
The current logic in httpd_dacp.c cannot handle non persistent items
correctly. The items are always shown with the dummy_mfi with "unkown
artist" etc.
Ommits useless update query for playcount for items that are not in the
library. Also avoids trying to scrobble these items (fixes error log
message "lastfm: Scrobble failed, track id 9999999 is unknown")
The connection test would not catch "No route to host", as this is returned
through the value-result buffer.
This fix might partially solve issue #498.
AirPlay 2 devices like Sonos One and AirPort Express with firmware 7.8
require auth-setup before ANNOUNCE, otherwise they will return 403.
Also fixed a problem where metadata did not get sent when toggling
a speaker on/off if we were playing an endless stream.
Avoid calling cache_daap_add() when hreq->user_agent is NULL
(user agent is not provided by the client), because it will
trigger a segfault when strdup() is called with that NULL pointer.
Fixes#571.
Avahi gives us several ipv6 addresses for an ATV4 that aren't actually
connectable. Here we simply try to make a connection to the address,
and if it is not possible within a timeout we ignore the announcement.
We also now don't start a mdns record browser if the address from the
resolver passes the connection test and isn't link-local.
Changing a speaker volume in a DACP clients results in two dacp requests
(at least in Retune):
setproperty?include-speaker-id=0&dmcp.volume=41
setproperty?speaker-id=198018693182577&dmcp.volume=43
The first request sets the absolute volume for the speaker (player.c -
volume_setabs_speaker). The second request is only triggered if the
volume from the first one is not the new master volume. This second
requests adjusts the relativ volume based on the loudest device (new
master volume) (player.c - volume_setrel_speaker). After the second
request the master volume and the relative volumes are correct (range
from 0 - 100 percent and master volume is 100 percent).
MPD and JSON API clients only set the absolute volume which results in
inconsistent master_volume and relative volumes. The added option to
update the master volume in volume_setabs_speaker/volume_setrel_speaker
makes sure the master volume and the relative volumes are correctly set.
Implements Active-Remote, which is sent to the speaker, so it can use this
to tell us who it is when it makes dacp request with a device-volume update.
These commands are not used by Remote, but some Airplay speakers are
able to make them (e.g. my Sony STR-DN1040).
Also prepare being able to set the device volume property.
