diff options
65 files changed, 196 insertions, 160 deletions
diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index 4692c8e77dc..b535c2a198f 100644 --- 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 index 68eeebc17ff..1faf76383ba 100644 --- 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 index f9d87b86492..320b3d9384b 100644 --- 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 index df70e7592ab..373425895fa 100644 --- 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 index 39df2baca18..c854647b6f3 100644 --- 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 index 8efc1b12f3a..4262a1c8773 100644 --- 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 index c5978d6c608..ac990bf0b48 100644 --- 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 index 122e10a61ab..84b00038236 100644 --- 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 index 230a940d00b..eedc6cb038b 100644 --- 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) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index a0119ae67dc..428f8c169ee 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -100,8 +100,10 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { snd_pcm_lib_preallocate_dma_free(substream); +#ifdef CONFIG_SND_VERBOSE_PROCFS snd_info_unregister(substream->proc_prealloc_entry); substream->proc_prealloc_entry = NULL; +#endif return 0; } @@ -124,7 +126,7 @@ int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm) return 0; } -#ifdef CONFIG_PROC_FS +#ifdef CONFIG_SND_VERBOSE_PROCFS /* * read callback for prealloc proc file * @@ -203,9 +205,9 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) substream->proc_prealloc_entry = entry; } -#else /* !CONFIG_PROC_FS */ +#else /* !CONFIG_SND_VERBOSE_PROCFS */ #define preallocate_info_init(s) -#endif +#endif /* CONFIG_SND_VERBOSE_PROCFS */ /* * pre-allocate the buffer and create a proc file for the substream diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index e35fd5779a9..ae0df549fac 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -675,10 +675,8 @@ static int __init alsa_card_dummy_init(void) continue; device = platform_device_register_simple(SND_DUMMY_DRIVER, i, NULL, 0); - if (IS_ERR(device)) { - err = PTR_ERR(device); - goto errout; - } + if (IS_ERR(device)) + continue; devices[i] = device; cards++; } @@ -686,14 +684,10 @@ static int __init alsa_card_dummy_init(void) #ifdef MODULE printk(KERN_ERR "Dummy soundcard not found or device busy\n"); #endif - err = -ENODEV; - goto errout; + snd_dummy_unregister_all(); + return -ENODEV; } return 0; - - errout: - snd_dummy_unregister_all(); - return err; } static void __exit alsa_card_dummy_exit(void) diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 9ea3059a706..da7ef26995c 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -251,10 +251,8 @@ static int __init alsa_card_mpu401_init(void) #endif device = platform_device_register_simple(SND_MPU401_DRIVER, i, NULL, 0); - if (IS_ERR(device)) { - err = PTR_ERR(device); - goto errout; - } + if (IS_ERR(device)) + continue; platform_devices[i] = device; snd_mpu401_devices++; } @@ -266,14 +264,10 @@ static int __init alsa_card_mpu401_init(void) #ifdef MODULE printk(KERN_ERR "MPU-401 device not found or device busy\n"); #endif - err = -ENODEV; - goto errout; + snd_mpu401_unregister_all(); + return -ENODEV; } return 0; - - errout: - snd_mpu401_unregister_all(); - return err;< |