| Commit message (Collapse) | Author | Age | Files | Lines |
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Hi,
upon trying to play an MMS stream added to the play list, I got this:
mpd: /tmp/mpd/./src/input_stream.c:85: input_stream_open: Assertion `is->plugin->open == ((void *)0) || is->plugin == plugin' failed.
With the following patch applied, it works perfectly.
Thanks for having implemented MMS support :-).
Best regards,
Peter
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The configure.ac script does not care about the host architecture, it
only cares about the OS. Use ${host_os} instead of ${host} to
simplify the matching expressions.
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Without AC_CANONICAL_HOST, ${host_os} etc. are not defined.
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Added an inline assembly function for the 64 bit multiplication.
Benchmark results on a Pentium II 266 MHz, 512 MB of 24 bit PCM data:
dd if=/dev/zero bs=64k count=8k |
time ./test/software_volume 48000:24:2 >/dev/null
Before this patch 22.94s, after this patch 7.24s.
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This program is a simple utility for debugging and benchmarking the
software volume code.
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The switch syntax is "--enable-test". There are no test programs yet.
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When mp4ff_read_sample() returns a value bigger than zero, it
guarantees that the buffer is set. Remove the check.
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Don't waste any precious memory when the seek_table cannot be used.
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Don't include limits.h, use GLib constants instead.
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Use faacDecInit2() instead of AudioSpecificConfig() to detect the AAC
track in the MP4 file. This has a great advantage: it initializes the
libfaad decoder, which the caller would normally do anyway - but now
we can go without the AudioSpecificConfig() call. When decoder==NULL
(called from mp4_tag_dup()), fall back to a mp4ff_get_track_type()==1
check, like other audio players do.
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Moved the libfaad decoder initialization to mp4_faad_new(), and also
fill the audio_format struct there. This eliminates a little bit of
complexity in mp4_decode().
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Don't wait for the first frame to be decoded. We already have the
sample rate and the channel count from faacDecInit2().
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The function mp4_load_tag() is used only once, and mp4_tag_dup() is a
one-liner. Merge them.
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When a file is not seekable, MPD dropped the audio buffers before even
attempting to seek. This caused noticable sound corruption. Fix:
first attempt to seek, and only if that succeeds, call
audio_output_all_cancel().
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Make some variables more local, and eliminate superfluous ones.
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Instead of returning the sample rate and channel count as separate
values, fill an audio_format struct.
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Don't wait for the first frame to be decoded. We already have the
sample rate and the channel count from faacDecInit().
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The MPD core will never send a SEEK command to a decoder which has
declared to be not seekable.
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Define G_LOG_DOMAIN.
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Replace this plugin's own buffer library with the new decoder_buffer
library.
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Instead of checking if the buffer is empty after adts_find_frame(),
check adts_find_frame()'s return value. This is more robust.
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Moved libfaad API quirks to the wrapper functions faad_decoder_init()
and faad_decoder_decode().
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Instead of writing the song duration into a float pointer, return it
from the function.
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There are no callers which pass NULL here.
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All callers of adts_find_frame() use faad_buffer_fill() before that.
Move that faad_buffer_fill() call into adts_find_frame() instead.
adts_find_frame() will get its own logic for on-demand filling.
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The decoder buffer library may be used by decoder libraries such as
"faad".
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adts_check_frame() must not be called with a buffer length smaller
than 8. We can eliminate that duplicate check, and convert it into an
assertion.
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It's not valid to use the buffer's data without ensuring that the
buffer contains enough data.
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"aac" -> "faad"
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The function is unused, since we added the pcm_buffer library.
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When I implemented the pcm_buffer library, I forgot to set the new
buffer size. This caused a new allocation in each pcm_buffer_get(),
fortunately no memory was leaked.
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Fixed the log domains of the renamed decoders. Added G_LOG_DOMAIN
macros in decoders which don't have one already.
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This plugin is based on "libmpcdec".
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This plugin is based on "libmp4ff".
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Renamed functions and variables.
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The element fileOffset is only written, but never read. It can be
removed safely.
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This plugin uses libvorbis.
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A decoder plugin should be named after the library which is used.
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A decoder plugin should be named after the library which is used.
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If an output plugin requires config.h, it should include it directly.
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Don't include conf.h in mixer_api.h. Use a forward struct declaration
instead.
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Don't include output_api.h in output_internal.h. This change requires
adding missing includes in several sources.
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The method control() is too complicated, and overengineered. Replace
it with two trivial functions: get_volume() and set_volume().
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The output plugin shouldn't know any specifics of the mixer API. Make
it return the mixer object, and let the caller deal with it.
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Similar to the decoder plugin API: added wrapper functions to increase
code readability.
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Initialization and deinitialization first, then tag functions, then
play/cancel/pause.
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If we move the plugin struct to a separate header, we don't have to
include the big fat output_api.h everywhere.
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