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* - Fix moving of sink inputs between sinksLennart Poettering2008-05-091-1/+1
| | | | | | | | | - Don't write more than a single buffer size in the ALSA driver at a time, to give the clients time to fill up the memblockq again - Add API for querying the requested latency of a sink input/source output - Drop get_letancy() from vtable of sinks/sources git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2392 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Yes, yet another evil all-in-one commit of intervowen changes. I suck.Lennart Poettering2008-05-011-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * Drop "state" directory, fold that into "runtime directory" * No longer automatically rewind when a new stream connects * Rework sound file stream, to cause a rewind on initialisation, shorten _pop() code a bit * Fix reference counting of pa_socket_server in the protocol implementations * Rework daemon initialization code to be compatible with non-SUID-root setups where RLIMIT_RTPRIO is non-zero * Print warning if RT/HP is enabled in the config, but due to missing caps, rlimits, policy we cannot enable it. * Fix potential memory leak in pa_open_config_file() * Add pa_find_config_file() which works much like pa_open_config_file() but doesn't actually open the config file in question. Just searches for it. * Add portable pa_is_path_absolute() * Add pa_close_all() and use it on daemon startup to close leaking file descriptors (inspired from what I did for libdaemon) * Add pa_unblock_sigs() and use it on daemon startup to unblock all signals (inspired from libdaemon, too) * Add pa_reset_sigs() and use it on daemon startup to reset all signal handlers (inspired from libdaemon as well) * Implement pa_set_env() * Define RLIMIT_RTTIME and friends if not defined by glibc * Add pa_strempty() * rename state testing macros to include _IS_, to make clearer that they are no states, but testing macros * Implement pa_source_output_set_requested_latency_within_thread() to be able to forward latency info to sources from within the IO thread * Similar for sink inputs * generelize since_underrun counter in sink inputs to "playing_for" and "underrun_for". Use only this for ignore potential rewind requests over underruns * Add new native protocol message PLAYBACK_STREAM_MESSAGE_STARTED for notification about the end of an underrun * Port native protocol to use underrun_for/playing_for which is maintained by the sink input anyway * Pass underrun_for/playing_for in timing info to client * Drop pa_sink_skip() since it breaks underrun detection code * Move PID file and unix sockets to the runtime dir (i.e. ~/.pulse). This fixes a potention DoS attack from other users stealing dirs in /tmp from us so that we cannot take them anymore) * Allow setting of more resource limits from the config file. Set RTTIME by default * Streamline daemon startup code * Rework algorithm to find default configuration files * If run in system mode use "system.pa" instead of "default.pa" as default script file * Change ladspa sink to use pa_clamp_samples() for clamping samples * Teach module-null-sink how to deal with rewinding * Try to support ALSA devices with no implicit channel map. Synthesize one by padding with PA_CHANNEL_POSITION_AUX channels. This is not tested since I lack hardware with these problems. * Make use of time smoother in the client libraries. * Add new pa_stream_is_corked() and pa_stream_set_started_callback() functions to public API * Since our native socket moved, add some code for finding sockets created by old versions of PA. This should ease upgrades git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2329 fefdeb5f-60dc-0310-8127-8f9354f1896f
* define minimal and maximal wakeup/sleep times; check for underrun condition ↵Lennart Poettering2008-04-231-4/+31
| | | | | | only once during buffer fillup git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2312 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Big pile of interdependant changes:Lennart Poettering2008-04-231-112/+161
| | | | | | | | | | | | | | * Fix a deadlock when an asyncq overflows and an RT thread needed to wait until space became available again while the main thread was waiting for a operation to complete and thus didn't free any new items. Now, if the asyncq overflows, queue those items temporarily, and return immediately. Then, when the queue becomes writable again, flush it. * Modify pa_thread_mq_init() to also set up pa_rtpoll events properly for the MQ * Some more pa_bool_t'ization * Unify more common code between alsa-sink and alsa-source * The upper limit for the tsched watermark is max_use minus one frame * make module-alsa-source work * make the alsa modules use pa_alsa_build_pollfd() now * fix detection of dB scale for alsa-source git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2308 fefdeb5f-60dc-0310-8127-8f9354f1896f
* * don't increase tsched_watermark on underrun without limits Lennart Poettering2008-04-221-48/+10
| | | | | | | | | | * fix the watermark when we change the latency * fix latency measurement * move rewinding code into its own function * make use of new function pa_alsa_recover_from_poll() were applicable git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2299 fefdeb5f-60dc-0310-8127-8f9354f1896f
* - Change meaning of special values of latency request: 0 -> "minimal ↵Lennart Poettering2008-04-221-1/+4
| | | | | | | | | | | latency, please"; (pa_usec_t)-1 -> "don't care" - Remove "source" word from monitor source description - Increase default tsched watermark to 20ms again - For the first iteration after snd_pcm_start() halve the sleep time as workaround for USB devices with quick starts git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2291 fefdeb5f-60dc-0310-8127-8f9354f1896f
* initialize properties for ALSA sinks/sources more elaborately, re #277Lennart Poettering2008-04-201-12/+1
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2287 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Big pile of dependant changes:Lennart Poettering2008-04-201-21/+30
| | | | | | | | | | | | | | | | | | * Change pa_memblockq to carry silence memchunk instead of memblock and adapt all users * Add new call pa_sink_input_get_silence() to get the suitable silence block for a sink input * Implement monitoring sources properly by adding a delay queue to even out rewinds * Remove pa_{sink|source}_ping() becaused unnecessary these days and not used * Fix naming of various rewind related functions. Downstream is now _request_rewind(), upstream is _process_rewind() * Fix volume adjustments for a single stream in pa_sink_render() * Properly handle prebuf-style buffer underruns in pa_sink_input * Don't allow rewinding to more than the last underrun * Rework default buffering metrics selection for native protocol * New functions pa_memblockq_prebuf_active(), pa_memblockq_silence() * add option "mixer_reset=" to module-alsa-sink * Other cleanups git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2283 fefdeb5f-60dc-0310-8127-8f9354f1896f
* redirect alsa errors to normal PA log system; export buffer settings in ↵Lennart Poettering2008-04-111-2/+9
| | | | | | device props git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2246 fefdeb5f-60dc-0310-8127-8f9354f1896f
* init min/max latency properly; fix avail_min updatingLennart Poettering2008-04-091-2/+4
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2227 fefdeb5f-60dc-0310-8127-8f9354f1896f
* call snd_pcm_hwsync() expclicitly before we access any of the status fields, ↵Lennart Poettering2008-04-071-32/+78
| | | | | | since this seems to be necessary. try to find the right mixer device via the card index git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2221 fefdeb5f-60dc-0310-8127-8f9354f1896f
* propery calculate min_avail in frames instead of bytes. don't use device_id= ↵Lennart Poettering2008-04-041-1/+1
| | | | | | parameter in alsa modules if parameter wasn't specified git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2218 fefdeb5f-60dc-0310-8127-8f9354f1896f
* - deprecate autoload stuffLennart Poettering2008-04-031-48/+158
| | | | | | | | | | | | - allow setting of the requested latency of a sink input/source output before _put() is called - allow sinks/sources to have a "minimal" latency which applies to all requested latencies by sink inputs/source outputs - add new client library flags PA_STREAM_ADJUST_LATENCY, PA_STREAM_START_MUTED - allow client library to fill in 0 to buffer_attr fields - update module-alsa-source following module-alsa-sink - other cleanups and fixes git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2215 fefdeb5f-60dc-0310-8127-8f9354f1896f
* commit glitch-free workLennart Poettering2008-03-151-88/+233
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/branches/glitch-free@2124 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Rework ALSA mixer channel detection code. This time we actually care about ↵Lennart Poettering2007-11-131-22/+32
| | | | | | the channel names the ALSA mixer exports for us git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2055 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Completely rework ALSA device selection code: choose the device to open ↵Lennart Poettering2007-11-131-39/+30
| | | | | | depending on the requested number of channels and channel map. In most cases it will now suffice to set default-channels=6 to enable 5.1 sound for all devices that support it git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2050 fefdeb5f-60dc-0310-8127-8f9354f1896f
* tag modules that may only be loaded once at most especially, and enforce ↵Lennart Poettering2007-11-091-4/+5
| | | | | | that in the module loader git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2043 fefdeb5f-60dc-0310-8127-8f9354f1896f
* make rtprio and nice level actually configurableLennart Poettering2007-11-011-2/+2
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2014 fefdeb5f-60dc-0310-8127-8f9354f1896f
* merge 'lennart' branch back into trunk.Lennart Poettering2007-10-281-262/+651
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1971 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Handle suspended alsa devices. Based on patch by ranma. (closes #26)Pierre Ossman2007-03-021-0/+39
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* Add copyright notices to all relevant files. (based on svn log)Pierre Ossman2007-02-131-0/+3
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1426 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Huge trailing whitespace cleanup. Let's keep the tree pure from here on,Pierre Ossman2007-01-041-29/+29
| | | | | | | mmmkay? git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1418 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Revert r1404 and keep it on a development branch until it is fully tested.Pierre Ossman2006-11-061-7/+3
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1409 fefdeb5f-60dc-0310-8127-8f9354f1896f
* rework memory block management to be thread-safe and mostly lock-free.Lennart Poettering2006-09-261-3/+7
| | | | | | | | | | | | | | | | | | | | | pa_memblock is now an opaque structure. Access to its fields is now done through various accessor functions in a thread-safe manner. pa_memblock_acquire() and pa_memblock_release() are now used to access the attached audio data. Why? To allow safe manipulation of the memory pointer maintained by the memory block. Internally _acquire() and _release() maintain a reference counter. Please do not confuse this reference counter whith the one maintained by pa_memblock_ref()/_unref()! As a side effect this patch removes all direct usages of AO_t and replaces it with pa_atomic_xxx based code. This stuff needs some serious testing love. Especially if threads are actively used. git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1404 fefdeb5f-60dc-0310-8127-8f9354f1896f
* remove all occurences of Lennart Poettering2006-08-181-19/+19
| | | | | | | | | | | | pa_logXXX(__FILE__": and replace them by pa_logXXX(" git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1272 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Rework memory management to allow shared memory data transfer. The central ideaLennart Poettering2006-08-181-1/+1
| | | | | | | | | | | | | | | is to allocate all audio memory blocks from a per-process memory pool which is available as read-only SHM segment to other local processes. Then, instead of writing the actual audio data to the socket just write references to this shared memory pool. To work optimally all memory blocks should now be of type PA_MEMBLOCK_POOL or PA_MEMBLOCK_POOL_EXTERNAL. The function pa_memblock_new() now generates memory blocks of this type by default. git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1266 fefdeb5f-60dc-0310-8127-8f9354f1896f
* make use of pa_sink_used_by()/pa_source_used_by() wherever applicableLennart Poettering2006-08-121-2/+1
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1227 fefdeb5f-60dc-0310-8127-8f9354f1896f
* generate default sink/source names from the device files they belong toLennart Poettering2006-08-121-3/+14
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1223 fefdeb5f-60dc-0310-8127-8f9354f1896f
* don't set the sink/source descriptions manually, use the new functions ↵Lennart Poettering2006-08-111-1/+3
| | | | | | pa_{sink,source}_set_description() instead git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1205 fefdeb5f-60dc-0310-8127-8f9354f1896f
* shorten sink/source device descriptions a littleLennart Poettering2006-08-111-1/+1
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1201 fefdeb5f-60dc-0310-8127-8f9354f1896f
* ALSA: handle write()/read() errors properly by unloading the driver module. ↵Lennart Poettering2006-08-111-24/+52
| | | | | | This should fix problems when removing USB audio device while pulseaudio is running. git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1194 fefdeb5f-60dc-0310-8127-8f9354f1896f
* try to reduce volume updates in the ALSA sinks/sources: only touch the ↵Lennart Poettering2006-08-071-9/+13
| | | | | | shadowed hw volme if necessary git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1189 fefdeb5f-60dc-0310-8127-8f9354f1896f
* set is_hardware flag for a few hw pluginsLennart Poettering2006-07-161-0/+1
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* fix segfault when module-alsa-source fails to loadLennart Poettering2006-06-201-1/+1
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1049 fefdeb5f-60dc-0310-8127-8f9354f1896f
* big s/polyp/pulse/gLennart Poettering2006-06-191-15/+15
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1033 fefdeb5f-60dc-0310-8127-8f9354f1896f
* * alsa-sink: if "PCM" is not found as mixer track name, fallback to "Master"Lennart Poettering2006-05-301-1/+1
| | | | | | | * alsa-source: if "Capture" is not found as mixer track name, fallback to "Mic" git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@993 fefdeb5f-60dc-0310-8127-8f9354f1896f
* * if an ALSA device doesn't support the channel count requested, use what ↵Lennart Poettering2006-05-301-4/+11
| | | | | | | | | | ALSA suggests instead * if an ALSA device doesn't support the sampling freq requested, use what ALSA suggests and resample if this deviates more than 10% from what we requested * fix segfault freeing an unitialized mixer_fdl field git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@992 fefdeb5f-60dc-0310-8127-8f9354f1896f
* * split pa_cstrerror() into its own file polypcore/core-error.[ch]Lennart Poettering2006-05-251-1/+1
| | | | | | | | * fix building of padsp * remove a warning when compiling padsp.c git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@972 fefdeb5f-60dc-0310-8127-8f9354f1896f
* Wrap strerror() in a function that makes it thread safe and converts thePierre Ossman2006-05-221-1/+2
| | | | | | | output to UTF-8. git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@945 fefdeb5f-60dc-0310-8127-8f9354f1896f
* split polypcore/util.[ch] into polypcore/core-util.[ch] and polyp/util.[ch]Lennart Poettering2006-05-171-1/+1
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* Move xmalloc to the public side (libpolyp).Pierre Ossman2006-05-171-1/+2
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* * modify pa_channel_map_init_auto() to take an extra argument specifying the ↵Lennart Poettering2006-05-161-12/+1
| | | | | | | | | standard to use (ALSA, AIFF, ...) * add some more validity checks to pa_source_new(),pa_sink_new(),pa_sink_input_new(),pa_source_output_new() git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@888 fefdeb5f-60dc-0310-8127-8f9354f1896f
* * set default fragment metrics depending on the sample specs of the device ↵Lennart Poettering2006-05-131-3/+5
| | | | | | | | | in OSS and ALSA * fix fragment size calculation in module-alsa-sink git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@855 fefdeb5f-60dc-0310-8127-8f9354f1896f
* fix fragment size calculation for module-alsa-sourceLennart Poettering2006-05-131-5/+9
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* use default alsa channel map for alsa devicesLennart Poettering2006-05-131-1/+13
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* When a control is removed, all bits are set so we need to test for that first.Pierre Ossman2006-04-281-0/+3
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* support new channel_map argument in sink/source modulesLennart Poettering2006-04-261-3/+13
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* Fall back to software volume if hardware mixer cannot control all channels.Pierre Ossman2006-04-251-36/+26
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@796 fefdeb5f-60dc-0310-8127-8f9354f1896f
* clip volume at PA_VOLUME_NORM for alsa devicesLennart Poettering2006-04-081-2/+12
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* Fix some warnings caused by size_t having varying size.Pierre Ossman2006-03-311-1/+1
| | | | git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@641 fefdeb5f-60dc-0310-8127-8f9354f1896f
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