| Commit message (Collapse) | Author | Age | Files | Lines |
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GLib 2.16 was released more than 4 years ago. Let's remove some cruft
from the glib_compat.h header, and avoid new cruft to it.
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This plugin is horrible code, I mean it. Last year, I tried hard to
fix it, but I figured would take less time to do a full rewrite.
Given that I don't even have any device that supports RAOP, I can't do
that properly. After 16 months, nobody volunteered for fixing it.
Hereby, I delete it, because having no RAOP plugin is better than
having this mess. Sorry.
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.. and not in_frame_size, because this relates to the frame size being
sent to ALSA. pcm_export_source_size() will then turn it back into
the in_frame_size scale.
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Move code from the ALSA output plugin.
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The hard-coded "3 bytes" was wrong because it ignored the number of
channels.
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Conflicts:
src/output/osx_plugin.c
src/text_input_stream.c
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Add the encoder_plugin method end(). This is important for the
recorder plugin.
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The existing buffer implementation has a major flaw: it is unable to
re-fill the buffer until it has been consumed completely, leading to
many occasions where the render callback needs to generate silence,
just because the play() implementation was unable to append more
data. The fifo_buffer library handles that well.
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DSD-over-USB should not be a MPD core format, because it is not a
"natural" format; it is just a temnporary over-the-wire format. This
format has been implemented in pcm_export, and does not need to be
supported by pcm_convert.
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For 32 bit DSD-over-USB support.
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Prepare for removing SAMPLE_FORMAT_DSD_OVER_USB.
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Give the caller more control, prepare for DSD-over-USB improvements.
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Make it in_frame_size and out_frame_size, to account for packing.
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For simplicity, the MPD core should not have to deal with packing. It
is rarely used, and those plugins that need it should use the
pcm_export library instead.
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Converts padded 24 bit samples to packed 24 bit samples. Will replace
the packed S24 sample format, which is not used internally.
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Improving oss_reopen() by using the very same value that was used
initially.
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Let the caller configure the audio_format object.
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Merge three functions into one and call get_bitformat() only once.
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Eliminate support for reverse endian samples from the MPD core. This
moves a lot of complexity to the plugins that really need it (only
ALSA and CDIO currently).
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Don't use audio_format.reverse_endian.
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Don't use audio_format.reverse_endian.
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This format is unused since the DSDIFF decoder plugin now reverses the
bit order.
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Make sure we get a UTF-8 encoded string.
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raop_session_free must be called from raop_output_finish,
not from raop_output_remove.
In raop_output_remove, do close the ntp_server & control port.
Signed-off-by: Kurt Van Dijck <kurt.van.dijck@skynet.be>
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Basic support for Direct Stream Digital. No conversion yet, and no
decoder/output plugin support.
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Conflicts:
NEWS
configure.ac
src/decoder/ffmpeg_decoder_plugin.c
test/read_tags.c
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pcm_buffer_get() cannot ever return NULL.
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Keep the device open as long as the output is enabled, but initialize
it only when playback starts.
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When we don't have enough data, generate some silence, hoping the
input buffer will fill soon. Reducing the render buffer size is not
legal.
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Blocking inside the render callback is forbidden, and this sleep call
didn't make any sense.
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This plugin's use of the "Timer" library was wrong; it added the same
amount of virtual data in every iteration in _play(), but did not
actually play something. This created an artificial, but useless,
delay. This patch implements the method _cancel(), and implements
hard-coded sleep values. This is only slightly better, but does not
attempt to look sane.
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