Age | Commit message (Collapse) | Author | Files | Lines |
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Shouldn’t make a difference right now. Just in case…
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libav 9.12 and ffmpeg 2.2 have been tested.
Here’s why: My mp4 “parser” *cough* never was a mp4 parser in the sense
that it actually understood the file format. Instead it grepped the
input stream for “magic” strings (section identifiers). That alone
should be sufficient to throw away the code and rewrite it. Additionally
libfaad2 has not been updated for ages. I guess it was abandoned in
favor of libav/ffmpeg.
With libav/ffmpeg, which we support both as long as the API’s don’t
diverge too much, pianobar gains fast and reliable AAC and MP3 decoding
without bothering too much about the details. Most users will have it
installed already. On my own machine libav consumes about 2/3 CPU time
compared to the previous solution when playing AAC. Unfortunately memory
usage doubled and my attempts to disable unused protocols/formats/codec
failed due to libav’s API limitations.
While cleaning up a small detail regarding the eventcmd API has changed
too: Song duration and position are measured in seconds instead of
milliseconds now. Since libav/ffmpeg keeps track of accurate timing the
precision pianobar keeps track of can be reduced, while still being
sufficient for most users.
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1) Make sure that multiple bad playlists in a row don’t result in a
temporary ban
2) Ignore songs skipped because the playlist timed out after pausing for
too long
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Player thread now sets its status correctly. Closes #360.
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Workaround for #355, fixes commit
2c516503d2cb81dd156afc24677ac4bf3caefceb.
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See #352.
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Add commands that always play and always pause, in addition to the
current toggle pause command.
Closes #342.
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Decoding errors are usually not fatal, so we can recover by skipping the
broken frame. This also fixes invalid memory reads caused by
sampleSizeCurr >= sampleSizeN. See issue #304.
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Pandora limited the audio formats we can request explicitly. Don’t rely
on this and use the formats available with the given partner
credentials. Closes #271.
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Closes #250.
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This reverts commit 7df9371491e96a99c1e463f7787aede352ac5a37.
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This reverts commit f6dffd1822404522b8354ac453a911a0d98bfc61.
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1) Realtime signals don’t exist in OS X
2) libao apparently locks a mutex in ao_play, which is locked again in
ao_close and causes a deadlock
Closes #256, although I’m considering the option of partially reverting
7df9371491e96a99c1e463f7787aede352ac5a37, as this introduces more nasty
behavior than it solves.
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No more mutex locking/checking for quit condition. Should (slightly)
increase responsiveness of the player thread. Closes #250.
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32kbit/s. Closes #233.
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Move player buffer to heap, a few const’s here and there, increase
header buffer size, fix comments.
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as requested by rfc 2616
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Closes #88
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Again. And hopefully the last time.
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Fixes issue #66 (audio playback on big-endian machines).
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A few byteswaps replaced by conditional byteswaps. Thanks to Andrew Peng
for providing access to a powerpc machine.
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Fixes bug #16.
ao_sample_format structure has been updated, matrix must be NULL.
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Under certain circumstances (fast station switching e.g.) a new playlist
was not fetched, because the player thread returned 0x1 (== libao error)
and the main thread NULLed curStation.
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Thanks to jpfuentes2@github
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mp3 and mp3-hifi player functions are the same
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Get bitrate from header of first decoded frame now.
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I missed some #includes. Documentation had to be updated, too.
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Enhances commit 279f3ba93b729684e54300352c59523bd05e54bc
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Don't parse Content-Length header, curl does this too; CURLE_RECV_ERROR
is recoverable too
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printf wrapper introduced, hopefully cleaned up ui
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...and small cleanups.
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Introduced in (last) commit a2aee035072a5d346c187a890539f72c6d5167a0.
Don't divide by zero :)
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Time is now counted in milliseconds, not aac frames. Calculations seem
to be inaccurate sometimes (about +-2 seconds).
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