aboutsummaryrefslogtreecommitdiffstats
Commit message (Collapse)AuthorAgeFilesLines
...
* shout: static encoder plugin listMax Kellermann2008-09-121-35/+15
| | | | | | Shout encoder plugins are known at compile time. There is no reason to use a complex data structure as "List" to manage them at runtime - just put the pointers into a static array.
* 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 Wollesen2008-09-126-0/+340
| | | | | | | | | | | | | [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 The original commit message follows, although it is outdated:] Creation of shout_mp3 audio output plugin.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 Wollesen2008-09-126-73/+209
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Kellermann2008-09-124-153/+211
| | | | | | | | | | | 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 Kellermann2008-09-123-40/+70
| | | | | 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
|
* output: copy reqAudioFormat to outAudioFormat only if not yet openMax Kellermann2008-09-111-1/+7
| | | | | | If the output device is already open, it may have modified outAudioFormat; in this case, outAudioFormat is still valid, and does not need an overwrite.
* output: don't initialize inAudioFormat, outAudioFormatMax Kellermann2008-09-111-4/+0
| | | | | | As long as the device isn't open, both attributes are not used. Since they will both be initialized in audio_output_open(), we do not need the initialization in audio_output_init().
* 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().
* output: removed audio_output.sameInAndOutFormatsMax Kellermann2008-09-113-8/+2
| | | | | Eliminate sameInAndOutFormats and check with audio_format_equals() each time it this information is needed. Another 4 bytes saved.
* output: removed audio_output.convertAudioFormatMax Kellermann2008-09-113-7/+3
| | | | | Instead of checking convertAudioFormat, we can simply check if reqAudioFormat is defined. This saves 4 bytes in the struct.
* audio: removed commented codeMax Kellermann2008-09-101-12/+0
| | | | We have git..
* audio: added assertionsMax Kellermann2008-09-101-0/+5
|
* audio: make audio_configFormat a static variableMax Kellermann2008-09-101-8/+5
| | | | | | Save one allocation, since the whole audio_format struct is nearly the same size as the pointer to it. Check audio_format_defined(af) instead of af!=NULL.
* audio: don't free uninitialized audio_bufferMax Kellermann2008-09-101-5/+6
| | | | | | | free(NULL) isn't explicitly forbidden, but isn't exactly good style. Check the rare case that the audio buffer isn't initialized yet in closeAudioDevice(). In this case, we also don't have to call flushAudioBuffer().
* audio: added function audio_buffer_resize()Max Kellermann2008-09-101-4/+22
| | | | | | To make openAudioDevice() smaller and more readable, move code to a static function. Also don't use realloc(), since the old value of the buffer isn't needed anymore, saving a memcpy().
* audio: moved static variables into struct audio_bufferMax Kellermann2008-09-101-26/+30
| | | | | | | There are too many static variables in audio.c - organize all properties of the audio buffer in a struct. The current audio format is also a property of the buffer, since it describes the buffer's data format.
* audio: don't allow isCurrentAudioFormat(NULL)Max Kellermann2008-09-101-3/+5
| | | | | | Passing NULL to this function doesn't make sense, and complicates its implementation. The one caller which might pass NULL should rather check this.
* audio: removed isAudioDeviceOpen()Max Kellermann2008-09-102-7/+0
| | | | The function isAudioDeviceOpen() is never used.
* audio_format: added audio_format_clear() and audio_format_defined()Max Kellermann2008-09-102-1/+13
| | | | | | | | | audio_format_clear() sets an audio_format struct to an cleared (undefined) state, which is both faster and smaller than memset(0). audio_format_defined() checks if the audio_format struct actually has a defined value (i.e. non-zero). Both can be used to avoid pointers to audio_format, replacing the "NULL" value with an "undefined" audio_format.
* client: simplified client_read()Max Kellermann2008-09-101-3/+5
| | | | Remove one comparison by changing branch order.
* client: client_input_received() returns 0Max Kellermann2008-09-101-4/+2
| | | | | | | | | Since the caller chain doesn't care about the return value (except for COMMAND_RETURN_KILL, COMMAND_RETURN_CLOSE), just return 0 if there is nothing special. This saves one local variable initialization, and one access to it. Also remove one unreachable "return 1" from client_read().
* client: check for COMMAND_RETURN_CLOSEMax Kellermann2008-09-101-15/+14
| | | | | | | Don't close the client within client_process_line(), return COMMAND_RETURN_CLOSE instead. This is the signal for the caller chain to actually close it. This makes dealing with the client pointer a lot safer, since the caller always knows whether it is still valid.
* client: renamed local variable "selret" to "ret"Max Kellermann2008-09-101-4/+4
| | | | It's easier to reuse the variable if it has a more generic name.
* client: moved CLOSE/KILL check after client_process_line()Max Kellermann2008-09-101-4/+3
| | | | Don't update client data if it is going to be closed anyway.
* audio: moved cmpAudioFormat() to audio_format.hMax Kellermann2008-09-095-22/+14
| | | | | Rename it to audio_format_equals() and return "true" if they are equal.
* audio: replaced copyAudioFormat() with simple assignmentMax Kellermann2008-09-094-21/+9
| | | | | | | | | The "!src" check in copyAudioFormat() used to hide bugs - one should never pass NULL to it. There is one caller which might pass NULL, add a check in this caller. Instead of doing mempcy(), we can simply assign the structures, which looks more natural.
* output: renamed the functions in output_control.cMax Kellermann2008-09-094-34/+34
| | | | Getting rid of CamcelCase, again.
* output: moved code from audioOutput.c to output_control.cMax Kellermann2008-09-097-216/+281
| | | | | Similar to decoder_control.c, output_control.c will provide functions for controlling the output thread (which will be implemented later).
* output: renamed method namesMax Kellermann2008-09-092-34/+23
| | | | No CamelCase. Also don't declare typedefs for the methods.
* output: removed keepAudioOutputAlive() declarationMax Kellermann2008-09-091-1/+0
| | | | This function is declared, but is neither used nor implemented.
* timer: constant pointersMax Kellermann2008-09-093-3/+3
| | | | | The audio_format argument to timer_new() should be constant, because it is not modified. The same is true for ShoutData.audioFormat.
* storedPlaylist: correctly expand path when writingEric Wong2008-09-091-5/+6
| | | | | Otherwise we'd be writing to whatever directory that mpd is running in.
* 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]
* strset: fix duplicate valuesMax Kellermann2008-09-081-1/+1
| | | | | Due to a minor typo, the string set had duplicate values, because strset_add() didn't check the base slot properly.
* use strset.h instead of tagTracker.hMax Kellermann2008-09-086-163/+58
| | | | | | | With a large music database, the linear string collection in tagTracker.c becomes very slow. We implemented that in a quick'n'dirty fashion when we removed tree.c, and now we rewrite it using the fast hashed string set.
* added string set libraryMax Kellermann2008-09-083-0/+195
| | | | | | | | "struct strset" is a hashed string set: you can add strings to this library, and it stores them as a set of unique strings. You can get the size of the set, and you can enumerate through all values. This will be used to replace the linear tagTracker library.
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-03 16:33:08 +0000