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[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 06 Aug '09

06 Aug '09

Here's another appl_ptr related patch to the alsa soc mpc5200 ac97 driver. This time the patch is to fix a problem with not correctly handling appl_ptr wrapping back around to the beginning of the buffer. I haven't yet seen a symptom related to this. This came up when I was chatting with Jon Smirl about another bug I was examining. Right now there is another problem that is preventing me from playing really long audio streams. Let me know if this is the wrong approach. - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

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[PATCH] ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
by Daniel Mack 05 Aug '09

05 Aug '09

Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model and let the I2C functions only call the SoC core function for resume and suspend, which do nothing currently but will do later. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 27 +++++++++++++++++++++++---- 1 files changed, 23 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b822..ca1e24a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -806,15 +806,30 @@ static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + return snd_soc_suspend_device(codec->dev); } static int cs4270_i2c_resume(struct i2c_client *client) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; + + return snd_soc_resume_device(codec->dev); +} + +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) +{ + struct snd_soc_codec *codec = cs4270_codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -840,6 +855,8 @@ static int cs4270_i2c_resume(struct i2c_client *client) #else #define cs4270_i2c_suspend NULL #define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* @@ -868,7 +885,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -- 1.6.3.3

1 0

[alsa-devel] [PATCH - amixer] listen to D-Bus for volume-up, volume-down and mute signals.
by Łukasz Stelmach 05 Aug '09

05 Aug '09

The patch introduces D-Bus support in amixer. With --dbus option given amixer works continiously (like with the --stdin) waiting for org.freedesktop.Hal.Device.Condition with the first argument "ButtonPressed" and the second being one of: "volume-up", "volume-down" and "mute". These are emmited by the HAL input helper. A single amixer process can controll one mixer item. There are options to choose a specific item to controll, an input device to listen and a single adjustment step size. Signed-off-by: Łukasz Stelmach <stlman(a)poczta.fm> --- amixer/Makefile.am | 3 +- amixer/amixer.1 | 18 ++++++ amixer/amixer.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.in | 12 ++++ 4 files changed, 203 insertions(+), 1 deletions(-) diff --git a/amixer/Makefile.am b/amixer/Makefile.am index fcd0e81..fa951a6 100644 --- a/amixer/Makefile.am +++ b/amixer/Makefile.am @@ -1,5 +1,6 @@ INCLUDES = -I$(top_srcdir)/include -LDADD = -lm +LDADD = -lm @DBUS_LIBS@ +CFLAGS += @DBUS_CFLAGS@ # LDFLAGS = -static # CFLAGS += -g -Wall diff --git a/amixer/amixer.1 b/amixer/amixer.1 index b1ac323..bc160be 100644 --- a/amixer/amixer.1 +++ b/amixer/amixer.1 @@ -88,6 +88,24 @@ Select the card number to control. The device name created from this parameter has syntax 'hw:N' where N is specified card number. .TP +\fI\-\-dbus[=\fISCONTROL\fP] + +Connect to system D-Bus and wait for \fIorg.freedesktop.Hal.Device.Condition\fP +signals with ButtonPressed as the first argument and one of: \fIvolume-up\fP, +\fIvolume-down\fP or \fImute\fP as the second one. Control the selected +\fISCONTROL\fP item. The default item is 'Master,0'. + +.TP +\fI\-\-dbus-path=<\fIUDI\fP> + +Choose a particular input device that emits control events. + +.TP +\fI\-\-dbus-step=<\fIN\fP> + +Set an amount by wich to increase or decrease volume upon event. + +.TP \fI\-D\fP device Select the device name to control. The default control name is 'default'. diff --git a/amixer/amixer.c b/amixer/amixer.c index 9620721..31de674 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -18,6 +18,8 @@ * */ +#include "aconfig.h" + #include <stdio.h> #include <stdlib.h> #include <string.h> @@ -31,6 +33,10 @@ #include <sys/poll.h> #include "amixer.h" +#ifdef HAVE_DBUS +#include <dbus/dbus.h> +#endif /* HAVE_DBUS */ + #define LEVEL_BASIC (1<<0) #define LEVEL_INACTIVE (1<<1) #define LEVEL_ID (1<<2) @@ -43,6 +49,12 @@ static int ignore_error = 0; static struct snd_mixer_selem_regopt smixer_options; static char card[64] = "default"; +#ifdef HAVE_DBUS +static char dbus_sctl[64] = "Master,0"; +static char *dbus_path = NULL; +static char dbus_step[8] = "1"; +#endif /* HAVE_DBUS */ + static void error(const char *fmt,...) { va_list va; @@ -60,6 +72,11 @@ static int help(void) printf("\nAvailable options:\n"); printf(" -h,--help this help\n"); printf(" -c,--card N select the card\n"); +#ifdef HAVE_DBUS + printf(" --dbus[=sID] Connect to the system D-Bus and wait for volume messages\n"); + printf(" --dbus-path UDI Choose a particular input device (udi)\n"); + printf(" --dbus-step N Set the step size (default: 1)\n"); +#endif /* HAVE_DBUS */ printf(" -D,--device N select the device, default '%s'\n", card); printf(" -d,--debug debug mode\n"); printf(" -n,--nocheck do not perform range checking\n"); @@ -1887,15 +1904,147 @@ static int exec_stdin(void) return 0; } +#ifdef HAVE_DBUS + +#define OPT_DBUS 0x81 +#define OPT_DBUS_PATH 0x82 +#define OPT_DBUS_STEP 0x83 + +void* get_dbus_arg(DBusMessageIter* args, int type, void** out) { + if (type != dbus_message_iter_get_arg_type(args)) + return NULL; + dbus_message_iter_get_basic(args, out); + dbus_message_iter_next(args); + return *out; +} + +int exec_dbus(void) +{ + char *sargv[MAX_ARGS]; + int sargc; + const char* s; + int stepl; + DBusError error; + DBusMessage* msg; + DBusConnection *conn; + DBusMessageIter args; + + stepl = strlen(dbus_step); + dbus_step[stepl+1] = '\0'; + + dbus_error_init (&error); + conn = dbus_bus_get (DBUS_BUS_SYSTEM, &error); + + if(!conn) { + fprintf(stderr, "dbus:%s: %s\n", + error.name, error.message); + return 1; + } + + dbus_bus_add_match(conn, + "type='signal'," + "interface='org.freedesktop.Hal.Device'," + "member='Condition'", + &error); + /* The first message is a NameOwnerChanged signal, ignore it. */ + msg = dbus_connection_pop_message(conn); + dbus_message_unref(msg); + + if(dbus_error_is_set(&error)) { + fprintf(stderr, "dbus:match error:%s\n", + error.message); + return 1; + } + + while (1) { + sargv[0] = dbus_sctl; sargc = 1; + + if (!(msg = dbus_connection_pop_message(conn))) { + dbus_connection_read_write(conn, -1); + continue; + } + + s = dbus_message_get_interface(msg); + if(strncmp("org.freedesktop.Hal.Device", s, 26)) { + if(debugflag) + fprintf(stderr, "dbus:unsupported interface:%s\n", s); + goto badmessage; + } + + if(dbus_path && + (s=dbus_message_get_path(msg)) && + strncmp(dbus_path, s, strlen(s))) { + if(debugflag) + fprintf(stderr, "dbus:not the selected udi:%s\n", s); + goto badmessage; + } + + if(!dbus_message_iter_init(msg, &args)) { + if(debugflag) + fprintf(stderr, "dbus:message has no arguments\n"); + goto badmessage; + } + + if (strncmp(get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s), "ButtonPressed", 13)) { + if(debugflag) + fprintf(stderr, "dbus:not a ButtonPressed event\n"); + goto badmessage; + } + + if (!get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s)) { + if(debugflag) + fprintf(stderr, "dbus:invalid argument type\n"); + goto badmessage; + } + + /* static int sset(unsigned int argc, char *argv[], int roflag, int keep_handle) */ + if (!strncmp(s, "volume-up", 9)) { + if(debugflag) + fprintf(stderr, "dbus:volume-up\n"); + dbus_step[stepl]='+'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "volume-down", 11)) { + if(debugflag) + fprintf(stderr, "dbus:volume-down\n"); + dbus_step[stepl]='-'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "mute", 4)) { + if(debugflag) + fprintf(stderr, "dbus:mute (toggle)\n"); + sargv[1] = "toggle"; + } else { + if(debugflag) + fprintf(stderr, "dbus:invalid argument value\n"); + goto badmessage; + } + sset(++sargc, sargv, 0, 1); +badmessage: + dbus_message_unref(msg); + } + + if(conn) + dbus_connection_unref(conn); + return 0; +} +#endif /* HAVE_DBUS */ int main(int argc, char *argv[]) { int morehelp, level = 0; int read_stdin = 0; +#ifdef HAVE_DBUS + int read_dbus = 0; + int t; +#endif /* HAVE_DBUS */ static const struct option long_option[] = { {"help", 0, NULL, 'h'}, {"card", 1, NULL, 'c'}, +#ifdef HAVE_DBUS + {"dbus", optional_argument, NULL, OPT_DBUS}, + {"dbus-path", 1, NULL, OPT_DBUS_PATH}, + {"dbus-step", 1, NULL, OPT_DBUS_STEP}, +#endif /* HAVE_DBUS */ {"device", 1, NULL, 'D'}, {"quiet", 0, NULL, 'q'}, {"inactive", 0, NULL, 'i'}, @@ -1964,6 +2113,24 @@ int main(int argc, char *argv[]) case 's': read_stdin = 1; break; +#ifdef HAVE_DBUS + case OPT_DBUS: + if(optarg) { + strncpy(dbus_sctl, optarg, sizeof(dbus_sctl)-1); + dbus_sctl[sizeof(dbus_sctl)-1] = '\0'; + } + read_dbus = 1; + break; + case OPT_DBUS_PATH: + dbus_path=optarg; + break; + case OPT_DBUS_STEP: + t=atoi(optarg); + if(t > 0 && t <= 999999) { + snprintf(dbus_step, sizeof(dbus_step), "%d", atoi(optarg)); + } + break; +#endif /* HAVE_DBUS */ default: fprintf(stderr, "Invalid switch or option needs an argument.\n"); morehelp++; @@ -1978,6 +2145,10 @@ int main(int argc, char *argv[]) if (read_stdin) return exec_stdin(); +#ifdef HAVE_DBUS + if(read_dbus) + return exec_dbus(); +#endif /* HAVE_DBUS */ if (argc - optind <= 0) { return selems(LEVEL_BASIC | level) ? 1 : 0; } diff --git a/configure.in b/configure.in index 1349ff3..717d6c5 100644 --- a/configure.in +++ b/configure.in @@ -7,6 +7,9 @@ AM_INIT_AUTOMAKE(alsa-utils, 1.0.20) AM_GNU_GETTEXT([external]) AM_GNU_GETTEXT_VERSION([0.15]) +dnl required versions of other packages +m4_define([dbus_required_version], [1.2.0]) + dnl Checks for programs. dnl try to gues cross-compiler if not set @@ -74,6 +77,15 @@ AC_ARG_ENABLE(alsamixer, no) alsamixer=false ;; *) AC_MSG_ERROR(bad value ${enableval} for --enable-alsamixer) ;; esac],[alsamixer=true]) +AC_ARG_WITH(dbus, [ --with-dbus Enable D-Bus support in amixer]) + if test "x$with_dbus" != xno; then + PKG_CHECK_MODULES(DBUS, dbus-1 >= dbus_required_version, + have_dbus=yes, + have_dbus=no) + fi + if test "x$have_dbus" = xyes; then + AC_DEFINE(HAVE_DBUS, 1, [Define to 1 if amixer is to support D-Bus]) + fi fi AM_CONDITIONAL(ALSAMIXER, test x$alsamixer = xtrue) -- 1.6.3.3 ---------------------------------------------------------------------- Zostan Dziewczyna lub Chlopakiem lata! 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[alsa-devel] Question to sophisticated alsa developer
by Andreas Bießmann 05 Aug '09

05 Aug '09

Hi all, I currently test alsa playback on AVR32 board with atmel's abdac driver. My environment is 2.6.30.4 customized kernel (customization for my own board), rootfs and toolchain made of customized buildroot from atmel. I use alsa-lib 1.0.20 with patch [1] applied and alsa-utils with patch [2] applied. The alsa code does compile and work fine (aside from my problem described some lines later). My test case is a sine wave file with 440Hz tone, samplerate is 17089 (cause of samplerate abdac driver offers and to avoid resampling in alsa-lib) playing with aplay. This works fine, I see a 440Hz sine wave on my oscilloscope and there is no effect as described later on. My problem is when I'm enabling dmix layer to run concurrent aplay processes I always get a click at the end of my last played sine wave file. My Oscilloscope shows me that there is sometimes a short part of a single or somtimes multiple sine period(s) moved down (or up) to mid value (I can supply a screenshot of oscilloscope, if this description is not clear enough). Mostly this effect is about 500 us long. The sine wave is missing the information at this position, it is no delay of the analog value at this position. If I overlay a correct sine wave the following part after the problematic position matches exactly the overlayed sine wave. I investigated that this problem always occur somewhere at the end of dmix. When I play a long "null" sample and start repeated play of my short sine wave sample the effect is not there while the null sample is playing. After ending of null sample the described effect appears instantly on each end of my repeated test sample. Here is my asound.conf to enable dmix layer for alsa: ---8<--- pcm.!default { type plug slave.pcm "dmixer" } ctl.mixer0 { type hw card 0 } pcm.!plughw { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0" format S16_BE rate 17089 } bindings { 0 0 1 1 } slowptr false } ctl.dmixer { type hw card 0 } --->8--- Maybe there is already a bloomer in. I have tried to change slowptr stuff in pcm.dmixer without any effect to my problem. I also tried to change buffer sizes in dmixer's slave but these changes had no effect. I also tried some changes in kernel driver for ABDAC. I played a bit with periods_min and period_bytes_min without effect to my problem. The buffer_size in "Direct Stream Mixing PCM" is fixed to the value in "hw:0". But if I double the periods_min in Kernel (default: 6 periods_min, 64 periods_max) for "hw:0" the maximum value for buffer_size in "Direct Stream Mixing PCM" is fixed to 8192. The "Route conversion" layer has always the buffer_size value of the next layer. This is 8192 for doubled periods_min in kernel and dmix enabled and 12288 for doubled periods_min and dmix disabled. Any advice to this "feature"? Here is the output of aplay -v for dmix enabled: ---8<--- # aplay -v sin_440_17089_1sec.wav Playing WAVE 'sin_440_17089_1sec.wav' : Signed 16 bit Little Endian, Rate 17089 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_BE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 8192 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 8192 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'Atmel ABDAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1610612736 silence_threshold: 0 silence_size : 1610612736 boundary : 1610612736 appl_ptr : 0 hw_ptr : 0 --->8--- And here the output of aplay -v without dmix: ---8<--- # aplay -v sin_440_17089_1sec.wav Playing WAVE 'sin_440_17089_1sec.wav' : Signed 16 bit Little Endian, Rate 17089 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_BE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 12288 stop_threshold : 12288 silence_threshold: 0 silence_size : 0 boundary : 1610612736 Slave: Hardware PCM card 0 'Atmel ABDAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_BE subformat : STD channels : 2 rate : 17089 exact rate : 17089 (17089/1) msbits : 16 buffer_size : 12288 period_size : 1024 period_time : 59921 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 12288 stop_threshold : 12288 silence_threshold: 0 silence_size : 0 boundary : 1610612736 appl_ptr : 0 hw_ptr : 0 --->8--- Well, currently I have no more ideas where to dig for my problem. I do need a mixing layer to allow two or three concurrent aplay instances at the same time but the output in this case is (atm) worse. Has anyone some advice where to search for a solution to my problem. Or has anybody some advice to use another feature of alsa to allow concurrent aplay processes? regards Andreas Bießmann [1] http://git.buildroot.net/buildroot/tree/package/multimedia/alsa-lib/alsa-li… [2] http://git.buildroot.net/buildroot/tree/package/multimedia/alsa-utils/alsa-…

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[alsa-devel] pyalsa: synchronizing queue with MIDI clock events
by Christopher Arndt 05 Aug '09

05 Aug '09

Hi everybody, this is my first post on this list, so please excuse me, should it not be the right place to post this. I started exploring programming the ALSA sequencer via the pyalsa Python interface, i.e the alsaseq module. I wrote a small utility that reads MIDI events from one port, filters/processes them and writes it to another. This allows for routing by channel, note range, controller number etc. and to filter out, replace or add MIDI events. Now to synchronize the queue tempo to incoming MIDI clock events, to be able implement synchronized delays or arpeggiators, I use the method in the code shown below. This works ok, but I notice that the measured BPM oscillates +-1 BPM around the value displayed by the clock source (my synth's sequencer) if I measure/average only a few (~10) ticks or do not round the result to an integer value. Is this the right approach? What is a sensible number of ticks to take into account for measurement? Should I use another reference timer than Python's time.time() function? Is pyalsa generally suitable for this kind of application even on older/weaker hardware (e.g. a NSLU2)? Any comments or suggestions for improvement would be very much appreciated! Chris class MidiProcessor(object): def run(self): # This is simplified to only show the general logic while True: events = self.sequencer.receive_events( timeout=RECEIVE_TIMEOUT, maxevents=1) for event in events: if event.type == SEQ_EVENT_CLOCK: self.sync_queue(event) else: # do other MIDI processing / routing def sync_queue(self, event): """Sync queue tempo to incoming MIDI clock events.""" # list to collect the timestamps of the last few ticks lt = self._last_ticks lt.append(time.time()) ltlen = len(lt) if ltlen > 1: # calculate & set bpm: calculate difference between # the times the last few ticks were received and average # all results avg_delta = sum( [y-x for x,y in zip(lt, lt[1:])]) / (ltlen-1) # tick length is a 24th of a quarter note bpm = round(60 / avg_delta / 24) if bpm != self.bpm: self.bpm = bpm self.sequencer.queue_tempo(self.queue, tempo=int(6e7 / self.bpm), ppq=self.ppq) # only remember last 24 received ticks # (length of a quarter note) if ltlen > 24: lt.pop(0)

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[alsa-devel] [PATCH 3/3] ASoC: add machine driver for i.mx27_visstrim_m10 board
by javier Martin 05 Aug '09

05 Aug '09

This adds support for i.mx27_visstrim_sm10 board machine driver which uses an i.mx27 processor plus a wm8974 codec. It has been tested on a visstrim_sm10 board. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/Kconfig | 8 + sound/soc/imx/Makefile | 4 + sound/soc/imx/mx27vis_wm8974.c | 317 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 329 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 886dadd..2c6f568 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -9,5 +9,13 @@ config SND_MX1_MX2_SOC config SND_MXC_SOC_SSI tristate +config SND_SOC_MX27VIS_WM8974 + tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" + depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 + select SND_MXC_SOC_SSI + select SND_SOC_WM8974 + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with WM8974. diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 6552cb2..c2ffd2c 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -4,3 +4,7 @@ snd-soc-mxc-ssi-objs := mxc-ssi.o obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o + +# i.MX Machine Support +snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o +obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c new file mode 100644 index 0000000..e4dcb53 --- /dev/null +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -0,0 +1,317 @@ +/* + * mx27vis_wm8974.c -- SoC audio for mx27vis + * + * Copyright 2009 Vista Silicon S.L. + * Author: Javier Martin + * javier.martin(a)vista-silicon.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + + +#include "../codecs/wm8974.h" +#include "mx1_mx2-pcm.h" +#include "mxc-ssi.h" +#include <mach/gpio.h> +#include <mach/iomux.h> + +#define IGNORED_ARG 0 + + +static struct snd_soc_card mx27vis; + +/** + * This function connects SSI1 (HPCR1) as slave to + * SSI1 external signals (PPCR1) + * As slave, HPCR1 must set TFSDIR and TCLKDIR as inputs from + * port 4 + */ +void audmux_connect_1_4(void) +{ + pr_debug("AUDMUX: normal operation mode\n"); + /* Reset HPCR1 and PPCR1 */ + + DAM_HPCR1 = 0x00000000; + DAM_PPCR1 = 0x00000000; + + /* set to synchronous */ + DAM_HPCR1 |= AUDMUX_HPCR_SYN; + DAM_PPCR1 |= AUDMUX_PPCR_SYN; + + + /* set Rx sources 1 <--> 4 */ + DAM_HPCR1 |= AUDMUX_HPCR_RXDSEL(3); /* port 4 */ + DAM_PPCR1 |= AUDMUX_PPCR_RXDSEL(0); /* port 1 */ + + /* set Tx frame and Clock direction and source 4 --> 1 output */ + DAM_HPCR1 |= AUDMUX_HPCR_TFSDIR | AUDMUX_HPCR_TCLKDIR; + DAM_HPCR1 |= AUDMUX_HPCR_TFCSEL(3); /* TxDS and TxCclk from port 4 */ + + return; +} + +static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0, fmt = 0, mclk = 0; + int ret = 0; + + /* + * The WM8974 is better at generating accurate audio clocks than the + * MX27 SSI controller, so we will use it as master when we can. + */ + switch (params_rate(params)) { + case 8000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + mclk = WM8974_MCLKDIV_12; + pll_out = 24576000; + break; + case 16000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + pll_out = 12288000; + break; + case 48000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + pll_out = 12288000; + break; + case 96000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 12288000; + break; + case 11025: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_16; + pll_out = 11289600; + break; + case 22050: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_8; + pll_out = 11289600; + break; + case 44100: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + mclk = WM8974_MCLKDIV_2; + pll_out = 11289600; + break; + case 88200: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = codec_dai->ops->set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from codec DAI configuration\n"); + return ret; + } + + /* set cpu DAI configuration */ + ret = cpu_dai->ops->set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from cpu DAI configuration\n"); + return ret; + } + + /* Put DC field of STCCR to 1 (not zero) */ + ret = cpu_dai->ops->set_tdm_slot(cpu_dai, 0, 2); + + /* set the SSI system clock as input */ + ret = cpu_dai->ops->set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Error when setting system SSI clk\n"); + return ret; + } + + /* set codec BCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_BCLKDIV, bclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting BCLK division\n"); + return ret; + } + + + /* codec PLL input is 25 MHz */ + ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, + 25000000, pll_out); + if (ret < 0) { + printk(KERN_ERR "Error when setting PLL input\n"); + return ret; + } + + /*set codec MCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_MCLKDIV, mclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting MCLK division\n"); + return ret; + } + + return 0; +} + +static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); +} + +/* + * mx27vis WM8974 HiFi DAI opserations. + */ +static struct snd_soc_ops mx27vis_hifi_ops = { + .hw_params = mx27vis_hifi_hw_params, + .hw_free = mx27vis_hifi_hw_free, +}; + + +static int mx27vis_suspend(struct platform_device *pdev, pm_message_t state) +{ + return 0; +} + +static int mx27vis_resume(struct platform_device *pdev) +{ + return 0; +} + +static int mx27vis_probe(struct platform_device *pdev) +{ + int ret = 0; + + ret = get_ssi_clk(0, &pdev->dev); + + if (ret < 0) { + printk(KERN_ERR "%s: cant get ssi clock\n", __func__); + return ret; + } + + + return 0; +} + +static int mx27vis_remove(struct platform_device *pdev) +{ + put_ssi_clk(0); + return 0; +} + +static struct snd_soc_dai_link mx27vis_dai[] = { +{ /* Hifi Playback*/ + .name = "WM8974", + .stream_name = "WM8974 HiFi", + .cpu_dai = &imx_ssi_pcm_dai[0], + .codec_dai = &wm8974_dai, + .ops = &mx27vis_hifi_ops, +}, +}; + +static struct snd_soc_card mx27vis = { + .name = "mx27vis", + .platform = &mx1_mx2_soc_platform, + .probe = mx27vis_probe, + .remove = mx27vis_remove, + .suspend_pre = mx27vis_suspend, + .resume_post = mx27vis_resume, + .dai_link = mx27vis_dai, + .num_links = ARRAY_SIZE(mx27vis_dai), +}; + +static struct snd_soc_device mx27vis_snd_devdata = { + .card = &mx27vis, + .codec_dev = &soc_codec_dev_wm8974, +}; + +static struct platform_device *mx27vis_snd_device; + +/* Temporal definition of board specific behaviour */ +void gpio_ssi_active(int ssi_num) +{ + int ret = 0; + + unsigned int ssi1_pins[] = { + PC20_PF_SSI1_FS, + PC21_PF_SSI1_RXD, + PC22_PF_SSI1_TXD, + PC23_PF_SSI1_CLK, + }; + unsigned int ssi2_pins[] = { + PC24_PF_SSI2_FS, + PC25_PF_SSI2_RXD, + PC26_PF_SSI2_TXD, + PC27_PF_SSI2_CLK, + }; + if (ssi_num == 0) + ret = mxc_gpio_setup_multiple_pins(ssi1_pins, + ARRAY_SIZE(ssi1_pins), "USB OTG"); + else + ret = mxc_gpio_setup_multiple_pins(ssi2_pins, + ARRAY_SIZE(ssi2_pins), "USB OTG"); + if (ret) + printk(KERN_ERR "Error requesting ssi %x pins\n", ssi_num); +} + + +static int __init mx27vis_init(void) +{ + int ret; + + mx27vis_snd_device = platform_device_alloc("soc-audio", -1); + if (!mx27vis_snd_device) + return -ENOMEM; + + platform_set_drvdata(mx27vis_snd_device, &mx27vis_snd_devdata); + mx27vis_snd_devdata.dev = &mx27vis_snd_device->dev; + ret = platform_device_add(mx27vis_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(mx27vis_snd_device); + } + + /* WM8974 uses SSI1 (HPCR1) via AUDMUX port 4 for audio (PPCR1) */ + gpio_ssi_active(0); + audmux_connect_1_4(); + + return ret; +} + +static void __exit mx27vis_exit(void) +{ + /* We should call some "ssi_gpio_inactive()" properly */ +} + +module_init(mx27vis_init); +module_exit(mx27vis_exit); + + +MODULE_AUTHOR("Javier Martin, javier.martin(a)vista-silicon.com"); +MODULE_DESCRIPTION("ALSA SoC WM8974 mx27vis"); +MODULE_LICENSE("GPL"); --- -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

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[alsa-devel] [PATCH 1/6] pulse: replace manual mainloop by pa_mainloop_iterate()
by Lennart Poettering 05 Aug '09

05 Aug '09

The pa_mainloop_prepare()/_poll()/_dispatch() can be simplified by simply calling pa_mainloop_iterate() which does all this in one call. --- pulse/conf_pulse.c | 10 +--------- 1 files changed, 1 insertions(+), 9 deletions(-) diff --git a/pulse/conf_pulse.c b/pulse/conf_pulse.c index 3f9e071..35dd383 100644 --- a/pulse/conf_pulse.c +++ b/pulse/conf_pulse.c @@ -61,15 +61,7 @@ conf_pulse_hook_load_if_running(snd_config_t * root, snd_config_t * config, goto out; do { - err = pa_mainloop_prepare(loop, -1); - if (err < 0) - goto out; - - err = pa_mainloop_poll(loop); - if (err < 0) - goto out; - - err = pa_mainloop_dispatch(loop); + err = pa_mainloop_iterate(loop, 1, NULL); if (err < 0) goto out; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] [PATCH 6/6] pulse: immediately trigger EIO when connection is dropped
by Lennart Poettering 05 Aug '09

05 Aug '09

When the connection is dropped notify the application immediately instead of waiting until the applications calls into us the next time. This makes "aplay" handle connections shutdown similar to hardware unplugs: an immediate EIO is thrown. --- pulse/pulse.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/pulse/pulse.c b/pulse/pulse.c index 9c05171..a94caaf 100644 --- a/pulse/pulse.c +++ b/pulse/pulse.c @@ -77,10 +77,17 @@ int pulse_wait_operation(snd_pulse_t * p, pa_operation * o) static void context_state_cb(pa_context * c, void *userdata) { + pa_context_state_t state; snd_pulse_t *p = userdata; assert(c); - switch (pa_context_get_state(c)) { + state = pa_context_get_state(c); + + /* When we get disconnected, tell the process */x + if (!PA_CONTEXT_IS_GOOD(state)) + pulse_poll_activate(p); + + switch (state) { case PA_CONTEXT_READY: case PA_CONTEXT_TERMINATED: case PA_CONTEXT_FAILED: -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] [PATCH 5/6] pulse: rework object destruction paths a bit
by Lennart Poettering 05 Aug '09

05 Aug '09

Make sure we deal better with partially initialized structs. Don't check for pointer state before calling free() since free() does that anyway. --- pulse/ctl_pulse.c | 7 ++----- pulse/pcm_pulse.c | 19 ++++++++++--------- 2 files changed, 12 insertions(+), 14 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index 879f260..85863ed 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -606,11 +606,8 @@ static void pulse_close(snd_ctl_ext_t * ext) if (ctl->p) pulse_free(ctl->p); - if (ctl->source) - free(ctl->source); - if (ctl->sink) - free(ctl->sink); - + free(ctl->source); + free(ctl->sink); free(ctl); } diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index 98983f8..02a837e 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -819,21 +819,22 @@ static int pulse_close(snd_pcm_ioplug_t * io) assert(pcm); - pa_threaded_mainloop_lock(pcm->p->mainloop); + if (pcm->p && pcm->p->mainloop) { - if (pcm->stream) { - pa_stream_disconnect(pcm->stream); - pa_stream_unref(pcm->stream); - } + pa_threaded_mainloop_lock(pcm->p->mainloop); - pa_threaded_mainloop_unlock(pcm->p->mainloop); + if (pcm->stream) { + pa_stream_disconnect(pcm->stream); + pa_stream_unref(pcm->stream); + } + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + } if (pcm->p) pulse_free(pcm->p); - if (pcm->device) - free(pcm->device); - + free(pcm->device); free(pcm); return 0; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 4/6] pulse: unify stream/context state checks
by Lennart Poettering 05 Aug '09

05 Aug '09

Unify (and simplify) the paths that check for the validity of a stream/context: always call into check_stream()/pulse_check_connection() when applicable instead of rolling our own checks each time. Also check for validity of mainloop before locking it. --- pulse/ctl_pulse.c | 72 ++++++++++++++++++++++++++++++++---------- pulse/pcm_pulse.c | 90 +++++++++++++++++++++++----------------------------- 2 files changed, 95 insertions(+), 67 deletions(-) diff --git a/pulse/ctl_pulse.c b/pulse/ctl_pulse.c index 2caa29b..879f260 100644 --- a/pulse/ctl_pulse.c +++ b/pulse/ctl_pulse.c @@ -150,9 +150,13 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p) return -EBADFD; + err = pulse_check_connection(ctl->p); + if (err < 0) + return err; + o = pa_context_get_sink_info_by_name(ctl->p->context, ctl->sink, sink_info_cb, ctl); if (o) { @@ -182,17 +186,27 @@ static int pulse_update_volume(snd_ctl_pulse_t * ctl) static int pulse_elem_count(snd_ctl_ext_t * ext) { snd_ctl_pulse_t *ctl = ext->private_data; - int count = 0; + int count = 0, err; assert(ctl); + if (!ctl->p || !ctl->p->mainloop) + return -EBADFD; + pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) { + count = err; + goto finish; + } + if (ctl->source) count += 2; if (ctl->sink) count += 2; +finish: pa_threaded_mainloop_unlock(ctl->p->mainloop); return count; @@ -202,16 +216,21 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, snd_ctl_elem_id_t * id) { snd_ctl_pulse_t *ctl = ext->private_data; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + if (ctl->source) { if (offset == 0) snd_ctl_elem_id_set_name(id, SOURCE_VOL_NAME); @@ -220,14 +239,19 @@ static int pulse_elem_list(snd_ctl_ext_t * ext, unsigned int offset, } else offset += 2; + err = 0; + +finish: pa_threaded_mainloop_unlock(ctl->p->mainloop); - if (offset == 2) - snd_ctl_elem_id_set_name(id, SINK_VOL_NAME); - else if (offset == 3) - snd_ctl_elem_id_set_name(id, SINK_MUTE_NAME); + if (err >= 0) { + if (offset == 2) + snd_ctl_elem_id_set_name(id, SINK_VOL_NAME); + else if (offset == 3) + snd_ctl_elem_id_set_name(id, SINK_MUTE_NAME); + } - return 0; + return err; } static snd_ctl_ext_key_t pulse_find_elem(snd_ctl_ext_t * ext, @@ -266,7 +290,7 @@ static int pulse_get_attribute(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -319,7 +343,7 @@ static int pulse_read_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -371,7 +395,7 @@ static int pulse_write_integer(snd_ctl_ext_t * ext, snd_ctl_ext_key_t key, assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -476,7 +500,7 @@ static void pulse_subscribe_events(snd_ctl_ext_t * ext, int subscribe) assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return; pa_threaded_mainloop_lock(ctl->p->mainloop); @@ -491,17 +515,23 @@ static int pulse_read_event(snd_ctl_ext_t * ext, snd_ctl_elem_id_t * id, { snd_ctl_pulse_t *ctl = ext->private_data; int offset; - int err = -EAGAIN; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); - if (!ctl->updated || !ctl->subscribed) + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + + if (!ctl->updated || !ctl->subscribed) { + err = -EAGAIN; goto finish; + } if (ctl->source) offset = 2; @@ -540,20 +570,28 @@ static int pulse_ctl_poll_revents(snd_ctl_ext_t * ext, struct pollfd *pfd, unsigned short *revents) { snd_ctl_pulse_t *ctl = ext->private_data; - int err = 0; + int err; assert(ctl); - if (!ctl->p || !ctl->p->mainloop || !ctl->p->context) + if (!ctl->p || !ctl->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(ctl->p->mainloop); + err = pulse_check_connection(ctl->p); + if (err < 0) + goto finish; + if (ctl->updated) *revents = POLLIN; else *revents = 0; + err = 0; + +finish: + pa_threaded_mainloop_unlock(ctl->p->mainloop); return err; diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index a625f55..98983f8 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -135,11 +135,15 @@ static int update_active(snd_pcm_pulse_t *pcm) { if (!pcm->p) return -EBADFD; - ret = check_active(pcm); + ret = check_stream(pcm); if (ret < 0) - return ret; + goto finish; + + ret = check_active(pcm); - if (ret > 0) +finish: + + if (ret != 0) /* On error signal the caller, too */ pulse_poll_activate(pcm->p); else pulse_poll_deactivate(pcm->p); @@ -199,12 +203,12 @@ static int pulse_start(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -244,12 +248,12 @@ static int pulse_stop(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -291,12 +295,12 @@ static int pulse_drain(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -328,7 +332,7 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; if (io->state == SND_PCM_STATE_XRUN) @@ -339,12 +343,7 @@ static snd_pcm_sframes_t pulse_pointer(snd_pcm_ioplug_t * io) pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -379,18 +378,13 @@ static int pulse_delay(snd_pcm_ioplug_t * io, snd_pcm_sframes_t * delayp) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - err = -EBADFD; - goto finish; - } - for (;;) { - err = pulse_check_connection(pcm->p); + err = check_stream(pcm); if (err < 0) goto finish; @@ -433,17 +427,12 @@ static snd_pcm_sframes_t pulse_write(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -493,17 +482,12 @@ static snd_pcm_sframes_t pulse_read(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (!pcm->stream) { - ret = -EBADFD; - goto finish; - } - - ret = pulse_check_connection(pcm->p); + ret = check_stream(pcm); if (ret < 0) goto finish; @@ -624,13 +608,16 @@ static int pulse_pcm_poll_revents(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - err = check_active(pcm); + err = check_stream(pcm); + if (err < 0) + goto finish; + err = check_active(pcm); if (err < 0) goto finish; @@ -655,7 +642,7 @@ static int pulse_prepare(snd_pcm_ioplug_t * io) assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -757,7 +744,7 @@ static int pulse_hw_params(snd_pcm_ioplug_t * io, assert(pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); @@ -856,23 +843,26 @@ static int pulse_pause(snd_pcm_ioplug_t * io, int enable) { snd_pcm_pulse_t *pcm = io->private_data; int err = 0; + pa_operation *o; assert (pcm); - if (!pcm->p) + if (!pcm->p || !pcm->p->mainloop) return -EBADFD; pa_threaded_mainloop_lock(pcm->p->mainloop); - if (pcm->stream) { - pa_operation *o; - o = pa_stream_cork(pcm->stream, enable, NULL, NULL); - if (o) - pa_operation_unref(o); - else - err = -EIO; - } + err = check_stream(pcm); + if (err < 0) + goto finish; + + o = pa_stream_cork(pcm->stream, enable, NULL, NULL); + if (o) + pa_operation_unref(o); + else + err = -EIO; +finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); return err; -- 1.6.4 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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