aboutsummaryrefslogtreecommitdiffstats
path: root/src/audioOutputs (follow)
Commit message (Collapse)AuthorAgeFilesLines
* shout: removed typedefs on structs and plugin methodsMax Kellermann2008-09-124-73/+65
| | | | | | | Don't typedef the structs at all. It is easier to forward-declare this way. Don't typedef methods. They are used exactly once, a few lines below.
* shout: added mp3 encoderEric Wollesen2008-09-123-0/+196
| | | | | | | | | | | | | [mk: moved this patch after "Refactor and cleanup of shout Ogg and MP3 audio outputs". The original commit message follows, although it is outdated:] Creation of shout_mp3 audio output plugin. Basically I just copied the existing shout plugin and replaced ogg with lame. Uses lame for mp3 encoding. Next step is to pull common functionality out of each shout plugin and share it between them. Configuration options for "shout_mp3" are the same as for "shout".
* shout: introduce pluggable encoder APIEric Wollesen2008-09-123-55/+177
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | I've perhaps gone a bit overboard, but here's the current rundown: Both Ogg and MP3 use the "shout" audio output plugin. The shout audio output plugin itself has two new plugins, one for the Ogg encoder, and another for the MP3 (LAME) encoder. Configuration for an Ogg stream doesn't change. For an MP3 stream, configuration is the same as Ogg, with two exceptions. First, you must specify the optional "encoding" parameter, which should be set to "mp3". See mpd.conf(5) for more details. Second, the "quality" parameter is reversed for LAME, such that 1 is high quality for LAME, whereas 10 is high quality for Ogg. I've decomposed the code so that all libshout related operations are done in audioOutput_shout.c, all Ogg specific functions are in audioOutput_shout_ogg.c, and of course then all LAME specific functions are handled in audioOutput_shout_mp3.c. To develop encoder plugins for the shout audio output plugin, I basically just mimicked the plugin system used for audio outputs. This might be overkill, but hopefully if anyone ever wants to support some other sort of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will hopefully be all set. The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. As for metadata, I'm pretty sure they'll both work. I wrote up a test scaffold that would create a fake tag, and tell the plugin to send it out to the stream every few seconds. It seemed to work fine. Of course, if something does break, I'll be glad to fix it. Lastly, I've renamed lots of things into snake_case, in keeping with normalperson's wishes in that regard. [mk: moved the MP3 patch after this one. Splitted this patch into several parts; the others were already applied before this one. Fixed a bunch GCC warnings and wrong whitespace modifications. Made it compile with mpd-mk by adapting to its prototypes]
* shout: send shout metadataEric Wollesen2008-09-123-5/+19
| | | | | | | | | Support sending metadata to a shout server using shout_metadata_new() and shout_metadata_add(). The Ogg Vorbis encoder does not support this currently. [mk: this patch was separated from Eric's patch "Refactor and cleanup of shout Ogg and MP3 audio outputs", I added a description]
* shout: added struct _ogg_vorbis_dataMax Kellermann2008-09-122-56/+73
| | | | | | Preparing the merge of Eric Wollesen's patch "Refactor and cleanup of shout Ogg and MP3 audio outputs": we declare one of the struct types here, to make the merge smoother.
* shout: added shout_bufferEric Wollesen2008-09-123-46/+102
| | | | | | | | | | | | The Ogg encoder is slightly less optimal under this configuration. It used to send shout data directly out of its ogg_page structures. Now, in the interest of encapsulation, it copies the data from its ogg_page structures into a buffer provided by the shout audio output plugin (see audioOutput_shout_ogg.c, line 77.) I suspect the performance impact is negligible. [mk: this patch and its description was separated from Eric's patch "Refactor and cleanup of shout Ogg and MP3 audio outputs"]
* shout: moved code to audioOutput_shout_ogg.cMax Kellermann2008-09-123-153/+210
| | | | | | | | | | | Begin dividing audioOutput_shout.c: move everything OGG Vorbis related to audioOutput_shout_ogg.c. The header audioOutput_shout.h has to keep its dependency on vorbis/vorbisenc.h, because it needs the vorbis encoder types. For this patch, we have to export several internal functions with generic names to the ABI; these will be removed later when the encoder plugin patches are merged.
* shout: moved declarations to audioOutput_shout.hMax Kellermann2008-09-122-40/+69
| | | | | Prepare the split of the shout plugin into multiple sources: move all important declarations to audioOutput_shout.h.
* shout: removed commented codeMax Kellermann2008-09-121-4/+0
| | | | | | Remove unused code which is in comments. Remove that comment about "stolen code", since the plugin has changed much, and it isn't obvious which parts are derived.
* shout: no CamelCaseMax Kellermann2008-09-121-194/+194
|
* shout: use reqAudioFormat instead of outAudioFormatMax Kellermann2008-09-111-1/+1
| | | | | In the plugin's init() function, outAudioFormat is simply a copy of reqAudioFormat. Use reqAudioFormat instead of outAudioFormat here.
* shout: copy the audio_format, instead of taking a pointerMax Kellermann2008-09-111-13/+12
| | | | | | | Storing pointers to immutable audio_format structs isn't worth it, because the struct itself isn't much larger than the pointer. Since the shout plugin requires the user to configure a fixed audio format, we can simply copy it in myShout_initDriver().
* timer: constant pointersMax Kellermann2008-09-091-1/+1
| | | | | The audio_format argument to timer_new() should be constant, because it is not modified. The same is true for ShoutData.audioFormat.
* alsa: use blocking instead of non-blocking writeEric Wong2008-09-091-1/+6
| | | | | | | | | | | | The way we used non-blocking mode was HORRIBLE. It was non-blocking to ALSA, but we end up blocking in a busy loop that does absolutely NOTHING but retry. We don't check for playback cancellation (like we do in decoders) or anything. This is seriously broken and I can imagine it affects people on fast CPUs more because we do asynchronous output buffering and our ALSA device will always have data ready.
* alsa: snd_pcm_sw_params_set_xfer_align is deprecatedEric Wong2008-09-081-5/+0
| | | | | Lets not use deprecated functions. It's apparently possible to not care about the sw_params stuff at all!
* alsa: only run snd_config_update_free_global once atexitEric Wong2008-09-081-3/+7
| | | | | | | | | This is safer than the patch in http://www.musicpd.org/mantis/view.php?id=1542 with multiple audio outputs enabled. Sadly, I only noticed that patch/problem when I googled for "snd_config_update_free_global"
* alsa: move bitformat reading code out of the wayEric Wong2008-09-081-16/+12
|
* alsa: avoid unnecessary heap usage if we don't set a device nameEric Wong2008-09-081-11/+12
|
* alsa: get rid of the needless canPause flagEric Wong2008-09-081-3/+0
| | | | | We never use it for anything anyways as we release the device entirely on pause.
* alsa: capitalize "ALSA" consistently in messagesEric Wong2008-09-081-8/+8
| | | | That's the name of this project.
* alsa: optimistically try resuming from suspendEric Wong2008-09-081-6/+4
| | | | | | | | | | | | | | Apparently snd_pcm_hw_params_can_resume() can return false even though my hardware does in fact support resuming. So stop carrying that value in the canResume flag and just try to resume when we're in the suspended state; falling back to snd_pcm_prepare only if resuming fails. libao does something similar on resume, too. While we're at it, use the E() macro which will enable us to have better error reporting. [mk: remove the E() macro stuff]
* output: const plugin structuresMax Kellermann2008-09-0810-10/+10
| | | | | Since the plugin struct is never modified, we should store it in constant locations.
* output: renamed typedef AudioOutput to struct audio_outputMax Kellermann2008-09-0710-79/+87
| | | | | Also rename AudioOutputPlugin to struct audio_output_plugin, and use forward declarations to reduce include dependencies.
* output: added output_api.hMax Kellermann2008-09-0710-26/+18
| | | | | Just like decoder_api.h, output_api.h provides the audio output API which is used by the plugins.
* audio_format: converted typedef AudioFormat to struct audio_formatMax Kellermann2008-09-075-6/+6
| | | | | Get rid of CamelCase, and don't use a typedef, so we can forward-declare it, and unclutter the include dependencies.
* fix -Wcast-qual -Wwrite-strings warningsMax Kellermann2008-09-072-9/+10
| | | | | | | | | The previous patch enabled these warnings. In Eric's branch, they were worked around with a generic deconst_ptr() function. There are several places where we can add "const" to pointers, and in others, libraries want non-const strings. In the latter, convert string literals to "static char[]" variables - this takes the same space, and seems safer than deconsting a string literal.
* tag: fix the shout and oggflac pluginsMax Kellermann2008-08-291-8/+8
| | | | | | During the tag library refactoring, the shout plugin was disabled, and I forgot about adapting it to the new API. Apply the same fixes to the oggflac decoder plugin.
* tag: renamed MpdTag and MpdTagItem to struct tag, struct mpd_tag_itemMax Kellermann2008-08-291-2/+2
| | | | | Getting rid of CamelCase; not having typedefs also allows us to forward-declare the structures.
* made "sample_size" static constMax Kellermann2008-08-261-1/+1
| | | | | sample_size is a variable which is computed at compile time. Declare it "static const", so the compiler can optimize it away.
* moved jack configuration to the JackData structMax Kellermann2008-08-261-49/+64
| | | | | | Storing local configuration in global (static) variables is obviously a bad idea. Move all those variables into the JackData struct, including the locks.
* jack: removed unused macrosMax Kellermann2008-08-261-10/+0
|
* jack: don't set audioOutput->data=NULLMax Kellermann2008-08-261-5/+5
| | | | | | | There is only one caller of freeJackData() left: jack_finishDriver(). This function is called by the mpd core, and is called exactly once for every successful jack_initDriver(). We do not need to clear audioOutput->data, since this variable is invalidated anyway.
* jack: initialize JackData in jack_initDriver()Max Kellermann2008-08-261-6/+2
| | | | | | | | Over the lifetime of the jack AudioOutput object, we want a single valid JackData object, so we can persistently store data there (configuration etc.). Allocate JackData in jack_initDriver(). After that, we can safely remove all audioOutput->data==NULL checks (and replace them with assertions).
* jack: added freeJackClient()Max Kellermann2008-08-261-13/+25
| | | | | | | No need to destroy the JackData object when an error occurs, since jack_finishDriver() already frees it. Only deinitialize the jack library, introduce freeJackClient() for that, and move code from freeJackData().
* jack: initialize jd->client after !jd checkMax Kellermann2008-08-261-5/+5
| | | | | | Prepare the next patch: make the "!jd" check independent of the jd->client initialization. This way we can change the "jd" initialization semantics later.
* jack: eliminate superfluous freeJackData() callsMax Kellermann2008-08-261-6/+0
| | | | | | | connect_jack() invokes freeJackData() in every error handler, although its caller also invokes this function after a failure. We can save a lot of lines in connect_jack() by removing these redundant freeJackData() invocations.
* converted MpdTagItem.type to an enumMax Kellermann2008-08-261-0/+2
| | | | | Don't use CPP macros when you can use C enum... this also allows better type checking.
* enable -Wpointer-arith, -Wstrict-prototypesMax Kellermann2008-08-264-6/+8
| | | | | | Also enable -Wunused-parameter - this forces us to add the gcc "unused" attribute to a lot of parameters (mostly library callback functions), but it's worth it during code refactorizations.
* jack: initialize audioOutput->dataMax Kellermann2008-06-131-1/+3
| | | | | | | | | | | | | Initialize audioOutput->data with NULL in jack_initDriver(). Previously, this was never initialized, although the other functions relied on it being NULL prior to jack_openDevice(). This patch addresses bug 0001641[1]. In contrast to the patch provided by the bug reporter, it moves the initialization before the "!param" check. [1] - http://musicpd.org/mantis/view.php?id=1641 git-svn-id: https://svn.musicpd.org/mpd/trunk@7375 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fixed another shadow warningMax Kellermann2008-04-121-11/+11
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@7307 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* use size_t and constant pointer in ao pluginsMax Kellermann2008-04-1210-31/+59
| | | | | | | The audio output plugins should get a constant pointer, because they must not modify the buffer. Since the size is a non-negative buffer size in bytes, we should change its type to size_t. git-svn-id: https://svn.musicpd.org/mpd/trunk@7293 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* yet more unsigned integersMax Kellermann2008-04-121-1/+1
| | | git-svn-id: https://svn.musicpd.org/mpd/trunk@7287 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* whitespace cleanupMax Kellermann2008-04-121-7/+7
| | | | | Clean up some space indentations, replace with tabs. git-svn-id: https://svn.musicpd.org/mpd/trunk@7239 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* eliminated duplicate initializationMax Kellermann2008-03-262-4/+4
| | | | | | | Local variables which are never read before the first assignment don't need initialization. Saves a few bytes of text. Also don't reset variables which are never read until function return. git-svn-id: https://svn.musicpd.org/mpd/trunk@7199 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* parse/initialize with the correct data typeMax Kellermann2008-03-261-1/+1
| | | | | | | When we expect an integer as result, why would we use the double precision floating point parser? strtol() is a better match, although we should probably check for overflows... git-svn-id: https://svn.musicpd.org/mpd/trunk@7198 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* explicitly downcastMax Kellermann2008-03-262-3/+3
| | | | | | | | Tools like "sparse" check for missing downcasts, since implicit cast may be dangerous. Although that does not change the compiler result, it may make the code more readable (IMHO), because you always see when there may be data cut off. git-svn-id: https://svn.musicpd.org/mpd/trunk@7196 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fix -Wconst warningsMax Kellermann2008-02-051-6/+11
| | | | | | [ew: cleaned up the dirty union hack a bit] Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@7180 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fixed "comparison between signed and unsigned"Max Kellermann2008-01-261-1/+1
| | | | | Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@7146 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fixed -Wshadow warningsMax Kellermann2008-01-261-44/+44
| | | | | Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@7143 09075e82-0dd4-0310-85a5-a0d7c8717e4f
* fixed invalid C prototypesMax Kellermann2008-01-261-1/+1
| | | | | Signed-off-by: Eric Wong <normalperson@yhbt.net> git-svn-id: https://svn.musicpd.org/mpd/trunk@7142 09075e82-0dd4-0310-85a5-a0d7c8717e4f
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-26 17:33:16 +0000