When switching between tracks with different sample rates, the probe queue blocks before the pipeline can emit EOS.
This prevents the track change from proceeding without manual intervention. This appears to be because the queue
element doesn't handle the rate change correctly (either due to buffer length, or cap negotiation).
The queue2 element however does handle this without blocking indefinitely.
Let's the user see the error message what failed instead of Clementine crashing.
Also don't do gst_object_unref unless bin is set.
This fixes GStreamer-CRITICAL gst_object_unref: assertion 'object != NULL' failed
When the network connection changes while playing an HTTP stream, I always get the "Server does not support seeking." error from GStreamer.
It seems like GStreamer tries to seek on reconnect, which fails, an propagates the error to Clementine which in turn ceases playback with
the error message handed through from GStreamer, even though there is now a perfectly fine network connection again.
As a workaround, try to reload the stream when this error occurs.
fixes#5116
In the new version of gstreamer, alsasink supports floating samples, so it seems to be bypassing audioconvert.
Integer samples make downmixing work correctly.
Fixes#1747
The pipeline has the caps for the analyzer applied in the wrong place. This results in the audio output being limited to 16 bit regardless of the input file.
This change also cleans up the mono/sample rate caps as well.
Two problems here:
- the first was that "StartPlaybackLater" wasn't called from the thread which created SpotifyServer, so the timer never started.
- then the playback sometimes failed or started with an offset: just hack to ignore sourcedrained signal in this case.
This change provides the ability to set a fixed pipeline sample rate as an alternate to automatically negotiating it.
This can be useful on systems with sound cards that work at a fixed rate, as well as it can triage issues (on Windows)
where changing tracks hangs due to a problem with gstreamer's caps negotiation.
Gstreamer was failing to link the pipeline if 32bit could not be enabled.
We should just let gst autonegotiate the bit depth of the pipeline, which it does with mono disabled anyways.
Spotify doesn't work with this fix anymore, gstreamer throws `gst_segment_to_stream_time: assertion 'segment->format == format' failed`. Using `audio/*` for caps doesn't work either, the channes property is ignored. An `if (url_.scheme == "spotify")` would work, but maybe there is a more elegant solution.
This reverts commit 8799222d64.
SourceDrainedCallback() has a valid next url and calls TransitionToNext(). This sets a new decode bin for the pipeline. The pipeline then buffers the new track and sends GST_MESSAGE_BUFFERING. BufferingMessageReceived() then pauses the pipeline because it's buffering. But we are buffering the next track and not the current one, so the pipeline has still data and doesn't need to be paused.
This is a regression in the upgrade to gstreamer1.0.
The gst_buffer_unref() is incorrect as it removes the buffer when it is still needed by the chunker.
Forcing the pointer to be null prevents it from segfaulting but causes it to skip all chunks in the buffer, dropping the framerate and causing a worse case of #4321.
Removing these 2 lines restores original functionality.
- Rewrite gstspectrum (1.0) to use FFTW (2x faster) and emit raw magnitude
values (not log scaled).
- Rewrite the moodbar generation code to be somewhat understandable, and
do it in Clementine instead of gstreamer.
This is functional but pretty hacky.
And, as noted in the comments, there is a small delay (depends, but usually several seconds) to have the seek taken into account. But IMHO it's better than nothing.
Fixes#2503
There was a possibility of a dbz when a buffer sent to the
analyzer was shorter than 1ms long, such as what may happen at the end
of a track when stopping. This patch guards against this.
The chunks are now determined by the density of data in the buffer
to the length of audio in the buffer. The chunk length can change
size so that the audio that is analysed is exactly what is being played
at the instant the frame is requested.
The analyzers are sent new buffers of audio data to process each time
they pass through the gst pipeline. Different file formats and bit depths/
sample rates can change the size of these buffers, in some cases making them
large and therefore infrequent. This causes choppiness in the analyzer
as it is not getting new data with every frame. This patch chunks the buffers
coming off the pipeline to correspond with the framerate of the analyzer.
If "fade out on stop" is enabled, the "stop after this track" feature
would stop not stop the on current track, but instead start playing
the next track and fade out on that immediately. This patch disables
fadeout when the engine is stopped by HandleStopAfter().
This at least compiles against gstreamer 1.2.
Things that work:
* Playing audio
* Automatically completing tags
Things that don't work
* Spotify
* Moodbar
Things I haven't tested
* Transcoding