Merge git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa

* git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa: (22 commits) [ALSA] via82xx - Use DXS_SRC as default for VIA8235/8237/8251 chips [ALSA] hda-codec - Add model entry for ASUS Z62F [ALSA] PCMCIA sound devices shouldn't depend on ISA [ALSA] hda-codec - Fix capture from line-in on VAIO SZ/FE laptops [ALSA] Fix Oops at rmmod with CONFIG_SND_VERBOSE_PROCFS=n [ALSA] PCM core - introduce CONFIG_SND_PCM_XRUN_DEBUG [ALSA] adding __devinitdata to pci_device_id [ALSA] add __devinitdata to all pci_device_id [ALSA] hda-codec - Add codec id for AD1988B codec chip [ALSA] hda-codec - Add model entry for ASUS M9 laptop [ALSA] pcxhr - Fix a compiler warning on 64bit architectures [ALSA] via82xx: tweak VT8251 workaround [ALSA] intel8x0 - Disable ALI5455 SPDIF-input [ALSA] via82xx: add support for VIA VT8251 (AC'97) [ALSA] Fix typos and add information about Jack support to Audiophile-Usb.txt [ALSA] Fix double free in error path of miro driver [ALSA] hda-codec - Add entry for Epox EP-5LDA+ GLi [ALSA] sound/pci/: remove duplicate #include's [ALSA] hda-codec - Use model 'hp' for all HP laptops with AD1981HD [ALSA] continue on IS_ERR from platform device registration ...
LeviMarvin · May 1, 2006 · 494b9ae · 494b9ae
2 parents e0a515b + a769577
commit 494b9ae
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diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt
@@ -1,4 +1,4 @@
-	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.2
+	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.3
 	========================================================
 
 	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>
@@ -22,16 +22,16 @@ The device has 4 audio interfaces, and 2 MIDI ports:
  * Midi In (Mi)
  * Midi Out (Mo)
 
-The internal DAC/ADC has the following caracteristics:
+The internal DAC/ADC has the following characteristics:
 * sample depth of 16 or 24 bits
 * sample rate from 8kHz to 96kHz
-* Two ports can't use different sample depths at the same time.Moreover, the 
+* Two ports can't use different sample depths at the same time. Moreover, the 
 Audiophile USB documentation gives the following Warning: "Please exit any 
 audio application running before switching between bit depths"
 
 Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 
 activated at the same time depending on the audio mode selected:
- * 16-bit/48kHz ==> 4 channels in/ 4 channels out
+ * 16-bit/48kHz ==> 4 channels in/4 channels out
    - Ai+Ao+Di+Do
  * 24-bit/48kHz ==> 4 channels in/2 channels out, 
                     or 2 channels in/4 channels out
@@ -41,8 +41,8 @@ activated at the same time depending on the audio mode selected:
 
 Important facts about the Digital interface:
 --------------------------------------------
- * The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough, 
-though I haven't tested it under linux
+ * The Do port additionally supports surround-encoded AC-3 and DTS passthrough, 
+though I haven't tested it under Linux
    - Note that in this setup only the Do interface can be enabled
  * Apart from recording an audio digital stream, enabling the Di port is a way 
 to synchronize the device to an external sample clock
@@ -60,24 +60,23 @@ synchronization error (for instance sound played at an odd sample rate)
 The Audiophile USB MIDI ports will be automatically supported once the 
 following modules have been loaded:
  * snd-usb-audio
- * snd-seq
  * snd-seq-midi
 
-No additionnal setting is required.
+No additional setting is required.
 
 2.2 - Audio ports
 -----------------
 
 Audio functions of the Audiophile USB device are handled by the snd-usb-audio 
 module. This module can work in a default mode (without any device-specific 
-parameter), or in an advanced mode with the device-specific parameter called 
+parameter), or in an "advanced" mode with the device-specific parameter called 
 "device_setup".
 
 2.2.1 - Default Alsa driver mode
 
-The default behaviour of the snd-usb-audio driver is to parse the device 
+The default behavior of the snd-usb-audio driver is to parse the device 
 capabilities at startup and enable all functions inside the device (including 
-all ports at any sample rates and any sample depths supported). This approach 
+all ports at any supported sample rates and sample depths). This approach 
 has the advantage to let the driver easily switch from sample rates/depths 
 automatically according to the need of the application claiming the device.
 
@@ -114,9 +113,9 @@ gain).
 For people having this problem, the snd-usb-audio module has a new module 
 parameter called "device_setup".
 
-2.2.2.1 - Initializing the working mode of the Audiohile USB
+2.2.2.1 - Initializing the working mode of the Audiophile USB
 
-As far as the Audiohile USB device is concerned, this value let the user 
+As far as the Audiophile USB device is concerned, this value let the user 
 specify:
  * the sample depth
  * the sample rate
@@ -174,20 +173,20 @@ The parameter can be given:
 
 IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:
 -------------------------------------------
- * You may need to _first_ intialize the module with the correct device_setup 
+ * You may need to _first_ initialize the module with the correct device_setup 
    parameter and _only_after_ turn on the Audiophile USB device
  * This is especially true when switching the sample depth:
-   - first trun off the device
-   - de-register the snd-usb-audio module
-   - change the device_setup parameter (by either manually reprobing the module 
-     or changing modprobe.conf)
+   - first turn off the device
+   - de-register the snd-usb-audio module (modprobe -r)
+   - change the device_setup parameter by changing the device_setup
+     option in /etc/modprobe.conf 
    - turn on the device
 
 2.2.2.3 - Audiophile USB's device_setup structure
 
 If you want to understand the device_setup magic numbers for the Audiophile 
 USB, you need some very basic understanding of binary computation. However, 
-this is not required to use the parameter and you may skip thi section.
+this is not required to use the parameter and you may skip this section.
 
 The device_setup is one byte long and its structure is the following:
 
@@ -231,11 +230,11 @@ Caution:
 
 2.2.3 -  USB implementation details for this device
 
-You may safely skip this section if you're not interrested in driver 
+You may safely skip this section if you're not interested in driver 
 development.
 
-This section describes some internals aspect of the device and summarize the 
-data I got by usb-snooping the windows and linux drivers.
+This section describes some internal aspects of the device and summarize the 
+data I got by usb-snooping the windows and Linux drivers.
 
 The M-Audio Audiophile USB has 7 USB Interfaces:
 a "USB interface":
@@ -277,9 +276,9 @@ Here is a short description of the AltSettings capabilities:
   - 16-bit depth, 8-48kHz sample mode
   - Synch playback (Do), audio format type III IEC1937_AC-3
 
-In order to ensure a correct intialization of the device, the driver 
+In order to ensure a correct initialization of the device, the driver 
 _must_know_ how the device will be used:
- * if DTS is choosen, only Interface 2 with AltSet nb.6 must be
+ * if DTS is chosen, only Interface 2 with AltSet nb.6 must be
    registered
  * if 96KHz only AltSets nb.1 of each interface must be selected
  * if samples are using 24bits/48KHz then AltSet 2 must me used if
@@ -290,7 +289,7 @@ _must_know_ how the device will be used:
    is not connected
 
 When device_setup is given as a parameter to the snd-usb-audio module, the 
-parse_audio_enpoint function uses a quirk called 
+parse_audio_endpoints function uses a quirk called 
 "audiophile_skip_setting_quirk" in order to prevent AltSettings not 
 corresponding to device_setup from being registered in the driver.
 
@@ -317,9 +316,8 @@ However you may see the following warning message:
 using the "default" ALSA device. This is less efficient than it could be. 
 Consider using a hardware device instead rather than using the plug layer."
 
-
 3.2 - Patching alsa to use direct pcm device
--------------------------------------------
+--------------------------------------------
 A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. 
 However it has not been included in the CVS tree.
 
@@ -331,3 +329,32 @@ After having applied the patch you can run jackd with the following command
 line:
   % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1
 
+3.2 - Getting 2 input and/or output interfaces in Jack
+------------------------------------------------------
+
+As you can see, starting the Jack server this way will only enable 1 stereo
+input (Di or Ai) and 1 stereo output (Ao or Do).
+
+This is due to the following restrictions:
+* Jack can only open one capture device and one playback device at a time
+* The Audiophile USB is seen as 2 (or three) Alsa devices: hw:1,0, hw:1,1
+  (and optionally hw:1,2)
+If you want to get Ai+Di and/or Ao+Do support with Jack, you would need to
+combine the Alsa devices into one logical "complex" device.
+
+If you want to give it a try, I recommend reading the information from
+this page: http://www.sound-man.co.uk/linuxaudio/ice1712multi.html
+It is related to another device (ice1712) but can be adapted to suit
+the Audiophile USB.
+
+Enabling multiple Audiophile USB interfaces for Jackd will certainly require:
+* patching Jack with the previously mentioned "Big Endian" patch
+* patching Jackd with the MMAP_COMPLEX patch (see the ice1712 page)
+* patching the alsa-lib/src/pcm/pcm_multi.c file (see the ice1712 page)
+* define a multi device (combination of hw:1,0 and hw:1,1) in your .asoundrc
+  file 
+* start jackd with this device
+
+I had no success in testing this for now, but this may be due to my OS
+configuration. If you have any success with this kind of setup, please
+drop me an email.
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -1172,7 +1172,7 @@
   }        
 
   /* PCI IDs */
-  static struct pci_device_id snd_mychip_ids[] = {
+  static struct pci_device_id snd_mychip_ids[] __devinitdata = {
           { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
             PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
           ....
@@ -1565,7 +1565,7 @@
         <informalexample>
           <programlisting>
 <![CDATA[
-  static struct pci_device_id snd_mychip_ids[] = {
+  static struct pci_device_id snd_mychip_ids[] __devinitdata = {
           { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
             PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
           ....

diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c
@@ -616,7 +616,7 @@ static struct snd_kcontrol_new snd_cx88_capture_volume = {
  * Only boards with eeprom and byte 1 at eeprom=1 have it
  */
 
-static struct pci_device_id cx88_audio_pci_tbl[] = {
+static struct pci_device_id cx88_audio_pci_tbl[] __devinitdata = {
 	{0x14f1,0x8801,PCI_ANY_ID,PCI_ANY_ID,0,0,0},
 	{0x14f1,0x8811,PCI_ANY_ID,PCI_ANY_ID,0,0,0},
 	{0, }

diff --git a/include/sound/pcm.h b/include/sound/pcm.h
@@ -374,12 +374,14 @@ struct snd_pcm_substream {
 	/* -- OSS things -- */
 	struct snd_pcm_oss_substream oss;
 #endif
+#ifdef CONFIG_SND_VERBOSE_PROCFS
 	struct snd_info_entry *proc_root;
 	struct snd_info_entry *proc_info_entry;
 	struct snd_info_entry *proc_hw_params_entry;
 	struct snd_info_entry *proc_sw_params_entry;
 	struct snd_info_entry *proc_status_entry;
 	struct snd_info_entry *proc_prealloc_entry;
+#endif
 	/* misc flags */
 	unsigned int no_mmap_ctrl: 1;
 	unsigned int hw_opened: 1;
@@ -400,12 +402,14 @@ struct snd_pcm_str {
 	struct snd_pcm_oss_stream oss;
 #endif
 	struct snd_pcm_file *files;
+#ifdef CONFIG_SND_VERBOSE_PROCFS
 	struct snd_info_entry *proc_root;
 	struct snd_info_entry *proc_info_entry;
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
 	unsigned int xrun_debug;	/* 0 = disabled, 1 = verbose, 2 = stacktrace */
 	struct snd_info_entry *proc_xrun_debug_entry;
 #endif
+#endif
 };
 
 struct snd_pcm {

diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h
@@ -75,7 +75,9 @@ struct snd_pcm_oss_substream {
 struct snd_pcm_oss_stream {
 	struct snd_pcm_oss_setup *setup_list;	/* setup list */
 	struct mutex setup_mutex;
+#ifdef CONFIG_SND_VERBOSE_PROCFS
 	struct snd_info_entry *proc_entry;
+#endif
 };
 
 struct snd_pcm_oss {

diff --git a/sound/core/Kconfig b/sound/core/Kconfig
@@ -142,7 +142,7 @@ config SND_SUPPORT_OLD_API
 
 config SND_VERBOSE_PROCFS
 	bool "Verbose procfs contents"
-	depends on SND
+	depends on SND && PROC_FS
 	default y
 	help
 	  Say Y here to include code for verbose procfs contents (provides
@@ -171,3 +171,13 @@ config SND_DEBUG_DETECT
 	help
 	  Say Y here to enable extra-verbose log messages printed when
 	  detecting devices.
+
+config SND_PCM_XRUN_DEBUG
+	bool "Enable PCM ring buffer overrun/underrun debugging"
+	default n
+	depends on SND_DEBUG && SND_VERBOSE_PROCFS
+	help
+	  Say Y to enable the PCM ring buffer overrun/underrun debugging.
+	  It is usually not required, but if you have trouble with
+	  sound clicking when system is loaded, it may help to determine
+	  the process or driver which causes the scheduling gaps.
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
@@ -1242,6 +1242,8 @@ static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int for
 
 	if (format != AFMT_QUERY) {
 		formats = snd_pcm_oss_get_formats(pcm_oss_file);
+		if (formats < 0)
+			return formats;
 		if (!(formats & format))
 			format = AFMT_U8;
 		for (idx = 1; idx >= 0; --idx) {
@@ -2212,7 +2214,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area)
 	return 0;
 }
 
-#ifdef CONFIG_PROC_FS
+#ifdef CONFIG_SND_VERBOSE_PROCFS
 /*
  *  /proc interface
  */
@@ -2366,10 +2368,10 @@ static void snd_pcm_oss_proc_done(struct snd_pcm *pcm)
 		}
 	}
 }
-#else /* !CONFIG_PROC_FS */
+#else /* !CONFIG_SND_VERBOSE_PROCFS */
 #define snd_pcm_oss_proc_init(pcm)
 #define snd_pcm_oss_proc_done(pcm)
-#endif /* CONFIG_PROC_FS */
+#endif /* CONFIG_SND_VERBOSE_PROCFS */
 
 /*
  *  ENTRY functions

diff --git a/sound/core/pcm.c b/sound/core/pcm.c
@@ -142,7 +142,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card,
 	return -ENOIOCTLCMD;
 }
 
-#if defined(CONFIG_PROC_FS) && defined(CONFIG_SND_VERBOSE_PROCFS)
+#ifdef CONFIG_SND_VERBOSE_PROCFS
 
 #define STATE(v) [SNDRV_PCM_STATE_##v] = #v
 #define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v
@@ -436,7 +436,7 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry,
 	snd_iprintf(buffer, "appl_ptr    : %ld\n", runtime->control->appl_ptr);
 }
 
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
 static void snd_pcm_xrun_debug_read(struct snd_info_entry *entry,
 				    struct snd_info_buffer *buffer)
 {
@@ -480,7 +480,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr)
 	}
 	pstr->proc_info_entry = entry;
 
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
 	if ((entry = snd_info_create_card_entry(pcm->card, "xrun_debug",
 						pstr->proc_root)) != NULL) {
 		entry->c.text.read_size = 64;
@@ -501,7 +501,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr)
 
 static int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr)
 {
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
 	if (pstr->proc_xrun_debug_entry) {
 		snd_info_unregister(pstr->proc_xrun_debug_entry);
 		pstr->proc_xrun_debug_entry = NULL;
@@ -599,12 +599,12 @@ static int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream)
 	}
 	return 0;
 }
-#else /* !CONFIG_PROC_FS */
+#else /* !CONFIG_SND_VERBOSE_PROCFS */
 static inline int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) { return 0; }
 static inline int snd_pcm_stream_proc_done(struct snd_pcm_str *pstr) { return 0; }
 static inline int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) { return 0; }
 static inline int snd_pcm_substream_proc_done(struct snd_pcm_substream *substream) { return 0; }
-#endif /* CONFIG_PROC_FS */
+#endif /* CONFIG_SND_VERBOSE_PROCFS */
 
 /**
  * snd_pcm_new_stream - create a new PCM stream

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
@@ -130,7 +130,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram
 static void xrun(struct snd_pcm_substream *substream)
 {
 	snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
 	if (substream->pstr->xrun_debug) {
 		snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n",
 			   substream->pcm->card->number,
@@ -204,7 +204,7 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs
 	delta = hw_ptr_interrupt - new_hw_ptr;
 	if (delta > 0) {
 		if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) {
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
 			if (runtime->periods > 1 && substream->pstr->xrun_debug) {
 				snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2);
 				if (substream->pstr->xrun_debug > 1)
@@ -249,7 +249,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
 	delta = old_hw_ptr - new_hw_ptr;
 	if (delta > 0) {
 		if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) {
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
 			if (runtime->periods > 2 && substream->pstr->xrun_debug) {
 				snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2);
 				if (substream->pstr->xrun_debug > 1)