Setting Cache-Control to "no-cache" tells a client to always make a
request to check if the version in the client cache is still valid
(response code 403 not modified).
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().
Also changes relative seeking behavior:
- seeking behind the the current track only switches to the previous
track, if we are not more than 3 seconds into the current track,
otherwise starts current track from the beginning
- seeking beyond the current track will start the next track from the
beginning
This implementation uses a tmpfile for storage of the artwork, plus it uses
artwork_get() which means that it scales the image as requested by the client.
It also does not create a tmpfile unless we actually receive artwork.
When pipe playback is started, but no data is written to the pipe, the input
loop would bring the cpu to 100%. This fix limits the loop like it was before
player refactor.
This adds a new settings component for user configurable options that
can be changed through the JSON API.
The settings are stored in the admin db table and not in the conf-file.
Since it is unknown how to do real sync on Chromecast, this commit instead adds
a primitive delay to the stream, so that it is at least somewhat closer to
Airplay/local audio.
Also some cleanup of unused stuff.
* [db,jsonapi] case insensitive directory/file listing
* [jsonapi] file listing of playlist uses same VPATH ordering as per directory and files
* [db,jsonapi] sorting via existing S_VPATH
* [db] replace LOWER with COLLATE NOCASE
pb_suspend() + pb_resume() during track changes made the playback status
incorrect, i.e. pb_session.source_list/playing_now would not match what the
input was actually writing. This attempts to solve it by resetting the
session when pb_suspend() is called, so that the input, input_buffer and
source_list come into sync.
If playback was paused during the very last part of the track, the rest of the
track would be read into the input buffer and the input would be closed. With
this commit the input will not be reopened.
Also allow input_flush to be called with null argument.
