From 926a01ce1ef5e27281af0270e4476979c0522954 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/version.h b/include/sound/version.h index 22939142dd23..1f5d4872d623 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v1.2.3 From 6c941c8556dd9269be621cd8159fc60e955a91b3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/version.h b/include/sound/version.h index 22939142dd23..1f5d4872d623 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v1.2.3 From 681b84e17747e1c208e8e1acc54cc5e612da84d1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:00 +0100 Subject: sound: pcm: add vmalloc buffer helper functions There are now five copies of the code to allocate a PCM buffer using vmalloc(). Add a sixth in the core so that the others can be removed. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a79f16b..0ad2d28f2360 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -905,6 +905,44 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags); +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream); +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset); +#if 0 /* for kernel-doc */ +/** + * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * Allocates the PCM substream buffer using vmalloc(), i.e., the memory is + * contiguous in kernel virtual space, but not in physical memory. Use this + * if the buffer is accessed by kernel code but not by device DMA. + * + * Returns 1 if the buffer was changed, 0 if not changed, or a negative error + * code. + */ +static int snd_pcm_lib_alloc_vmalloc_buffer + (struct snd_pcm_substream *substream, size_t size); +/** + * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses + * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory. + */ +static int snd_pcm_lib_alloc_vmalloc_32_buffer + (struct snd_pcm_substream *substream, size_t size); +#endif +#define snd_pcm_lib_alloc_vmalloc_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO) +#define snd_pcm_lib_alloc_vmalloc_32_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO) + #ifdef CONFIG_SND_DMA_SGBUF /* * SG-buffer handling -- cgit v1.2.3 From ad8decb7f5dfd556e4a8400e37b127cd20d8e4c5 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 20 Dec 2009 19:01:50 +0100 Subject: ALSA: jazz16: Add support for Media Vision Jazz16 chipset This is one of Sound Blaster Pro compatible chipsets which is supported by Linux OSS driver and was missing native supoort for ALSA. The Jazz16 audio codec is Crystal CS4216 which is capable of playback and recording up to 48 kHz stereo. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/sb.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/sb.h b/include/sound/sb.h index 4e62ee1e4115..95353542256a 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ -- cgit v1.2.3 From 41116e926cb92292fa4fcbe888ae8133fa0038e6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 09:00:14 +0100 Subject: ALSA: cs46xx - Fix suspend/resume with new DSP Fix the basic suspend/resume of snd-cs46xx drivers with new DSP. References: https://bugzilla.redhat.com/show_bug.cgi?id=498287 https://bugzilla.redhat.com/show_bug.cgi?id=160751 Tested-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- include/sound/cs46xx_dsp_spos.h | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h index 7c44667e79a6..49b03c9e5e55 100644 --- a/include/sound/cs46xx_dsp_spos.h +++ b/include/sound/cs46xx_dsp_spos.h @@ -118,9 +118,11 @@ struct dsp_scb_descriptor { struct snd_info_entry *proc_info; int ref_count; - spinlock_t lock; - int deleted; + u16 volume[2]; + unsigned int deleted :1; + unsigned int updated :1; + unsigned int volume_set :1; }; struct dsp_task_descriptor { -- cgit v1.2.3 From 4757968dbff3d43f373f08de973014a9bd41ef0a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 28 Dec 2009 16:15:03 +0100 Subject: ALSA: Release v1.0.22.1 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/version.h b/include/sound/version.h index 1f5d4872d623..7fed23442db8 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.22" +#define CONFIG_SND_VERSION "1.0.22.1" #define CONFIG_SND_DATE "" -- cgit v1.2.3 From 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 11:47:57 +0100 Subject: ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions In some debug cases, it might be usefull to see previous ring buffer positions to determine position problems from the lowlevel drivers. Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a79f16b..4e18a6dbe690 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -262,6 +262,8 @@ struct snd_pcm_hw_constraint_list { unsigned int mask; }; +struct snd_pcm_hwptr_log; + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -340,6 +342,10 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif + +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + struct snd_pcm_hwptr_log *hwptr_log; +#endif }; struct snd_pcm_group { /* keep linked substreams */ -- cgit v1.2.3 From f240406babfe1526998e10583ea5eccc2676a433 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 5 Jan 2010 17:19:34 +0100 Subject: ALSA: pcm_lib - cleanup & merge hw_ptr update functions Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them. The main change is hw_ptr_interrupt variable removal to simplify code logic. This variable can be computed directly from hw_ptr. Ensure that updated hw_ptr is not lower than previous one (it was possible with old code in some obscure situations when interrupt was delayed or the lowlevel driver returns wrong ring buffer position value). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 1 - include/sound/pcm_oss.h | 2 +- 2 files changed, 1 insertion(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 4e18a6dbe690..fe1b131842be 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,7 +271,6 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ - snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index cc4e226f35fd..760c969d885d 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -61,7 +61,7 @@ struct snd_pcm_oss_runtime { struct snd_pcm_plugin *plugin_first; struct snd_pcm_plugin *plugin_last; #endif - unsigned int prev_hw_ptr_interrupt; + unsigned int prev_hw_ptr_period; }; struct snd_pcm_oss_file { -- cgit v1.2.3 From 1250932e48d3b698415b1f04775433cf1da688d6 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 7 Jan 2010 15:36:31 +0100 Subject: ALSA: pcm_lib - optimize wake_up() calls for PCM I/O As noted by pl bossart , the PCM I/O routines (snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls until all samples are not processed. Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index fe1b131842be..e26fb3c58037 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,6 +311,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ + unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ wait_queue_head_t sleep; struct fasync_struct *fasync; @@ -839,6 +840,8 @@ void snd_pcm_set_sync(struct snd_pcm_substream *substream); int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg); +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); int snd_pcm_playback_xrun_check(struct snd_pcm_substream *substream); int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream); -- cgit v1.2.3 From d1458279bf9c575a52fd22818ca19c463f380aba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:16:52 +0100 Subject: ALSA: Add snd_pci_quirk_lookup_id() Added a new function to look up a quirk entry with the given PCI SSID instead of a pci device pointer. This can be used when the searched ID is overridden for debugging or such a purpose. Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 +++ 1 file changed, 3 insertions(+) (limited to 'include') diff --git a/include/sound/core.h b/include/sound/core.h index a61499c22b0b..89e0ac17f44a 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -458,5 +458,8 @@ struct snd_pci_quirk { const struct snd_pci_quirk * snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); +const struct snd_pci_quirk * +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list); #endif /* __SOUND_CORE_H */ -- cgit v1.2.3 From c32d977b8157bf67cdf47729ce7dd054a26eb534 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:58:57 +0100 Subject: ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffers pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd need non-cached behavior more or less, even for the intermediate ring- buffers. Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1d4ca2aae50d..aabf48bb8ee6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1021,6 +1021,10 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area); +#define snd_pcm_lib_mmap_vmalloc snd_pcm_lib_mmap_noncached + static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) { *max = dma < 4 ? 64 * 1024 : 128 * 1024; -- cgit v1.2.3 From c91a988dc6551c66418690e36b2a23cdb0255da8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 10:32:15 +0100 Subject: ALSA: pcm_core: Fix wake_up() optimization This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e26fb3c58037..3bc9bca771ec 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,8 +311,9 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ - wait_queue_head_t sleep; + unsigned int twake: 1; /* do transfer (!poll) wakeup */ + wait_queue_head_t sleep; /* poll sleep */ + wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; /* -- private section -- */ -- cgit v1.2.3 From e7636925789b042ff9d98c51d48392e8c5549480 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 26 Jan 2010 17:08:24 +0100 Subject: ALSA: pcm_lib - return back hw_ptr_interrupt Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3bc9bca771ec..13bc83ca35fb 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,6 +271,7 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ -- cgit v1.2.3 From 28e1b773083d349d5223f586a39fa30f5d0f1c36 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:09 +0100 Subject: ALSA: usbaudio: parse USB descriptors with structs In preparation of support for v2.0 audio class, use the structs from linux/usb/audio.h and add some new ones to describe the fields that are actually parsed by the descriptor decoders. Also, factor out code from usb_create_streams(). This makes it easier to adopt the new iteration logic needed for v2.0. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) (limited to 'include') diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index eaf9dffe0a01..44f82d8e09c5 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -81,7 +81,7 @@ /* Terminal Control Selectors */ /* 4.3.2 Class-Specific AC Interface Descriptor */ -struct uac_ac_header_descriptor { +struct uac_ac_header_descriptor_v1 { __u8 bLength; /* 8 + n */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* UAC_MS_HEADER */ @@ -95,7 +95,7 @@ struct uac_ac_header_descriptor { /* As above, but more useful for defining your own descriptors: */ #define DECLARE_UAC_AC_HEADER_DESCRIPTOR(n) \ -struct uac_ac_header_descriptor_##n { \ +struct uac_ac_header_descriptor_v1_##n { \ __u8 bLength; \ __u8 bDescriptorType; \ __u8 bDescriptorSubtype; \ @@ -131,7 +131,7 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 /* 4.3.2.2 Output Terminal Descriptor */ -struct uac_output_terminal_descriptor { +struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ __u8 bDescriptorType; /* CS_INTERFACE descriptor type */ __u8 bDescriptorSubtype; /* OUTPUT_TERMINAL descriptor subtype */ @@ -171,7 +171,7 @@ struct uac_feature_unit_descriptor_##ch { \ } __attribute__ ((packed)) /* 4.5.2 Class-Specific AS Interface Descriptor */ -struct uac_as_header_descriptor { +struct uac_as_header_descriptor_v1 { __u8 bLength; /* in bytes: 7 */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* AS_GENERAL */ @@ -232,6 +232,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +/* Formats - Audio Data Format Type I Codes */ + +struct uac_format_type_ii_discrete_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __le16 wMaxBitRate; + __le16 wSamplesPerFrame; + __u8 bSamFreqType; + __u8 tSamFreq[][3]; +} __attribute__((packed)); + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 @@ -253,6 +266,17 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_FILL_MAX 0x80 /* A.10.2 Feature Unit Control Selectors */ + +struct uac_feature_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bSourceID; + __u8 bControlSize; + __u8 controls[0]; /* variable length */ +} __attribute__((packed)); + #define UAC_FU_CONTROL_UNDEFINED 0x00 #define UAC_MUTE_CONTROL 0x01 #define UAC_VOLUME_CONTROL 0x02 -- cgit v1.2.3 From 8fee4aff8c89c229593b76a6ab172a9cad24b412 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:10 +0100 Subject: ALSA: usbaudio: introduce new types for audio class v2 This patch adds some definitions for audio class v2. Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have different numerical representations in both standards, so there is need for a _V1 add-on now. usbmixer.c is changed accordingly. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 57 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) (limited to 'include') diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 44f82d8e09c5..fb1a97bf943d 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -25,6 +25,9 @@ #define USB_SUBCLASS_AUDIOSTREAMING 0x02 #define USB_SUBCLASS_MIDISTREAMING 0x03 +#define UAC_VERSION_1 0x00 +#define UAC_VERSION_2 0x20 + /* A.5 Audio Class-Specific AC Interface Descriptor Subtypes */ #define UAC_HEADER 0x01 #define UAC_INPUT_TERMINAL 0x02 @@ -180,6 +183,19 @@ struct uac_as_header_descriptor_v1 { __le16 wFormatTag; /* The Audio Data Format */ } __attribute__ ((packed)); +struct uac_as_header_descriptor_v2 { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalLink; + __u8 bmControls; + __u8 bFormatType; + __u32 bmFormats; + __u8 bNrChannels; + __u32 bmChannelConfig; + __u8 iChannelNames; +} __attribute__((packed)); + #define UAC_DT_AS_HEADER_SIZE 7 /* Formats - A.1.1 Audio Data Format Type I Codes */ @@ -232,6 +248,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +struct uac_format_type_i_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bSubslotSize; + __u8 bFormatType; + __u8 bBitResolution; + __u8 bHeaderLength; + __u8 bControlSize; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - Audio Data Format Type I Codes */ struct uac_format_type_ii_discrete_descriptor { @@ -245,11 +274,26 @@ struct uac_format_type_ii_discrete_descriptor { __u8 tSamFreq[][3]; } __attribute__((packed)); +struct uac_format_type_ii_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __u16 wMaxBitRate; + __u16 wSamplesPerFrame; + __u8 bHeaderLength; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 #define UAC_FORMAT_TYPE_II 0x2 #define UAC_FORMAT_TYPE_III 0x3 +#define UAC_EXT_FORMAT_TYPE_I 0x81 +#define UAC_EXT_FORMAT_TYPE_II 0x82 +#define UAC_EXT_FORMAT_TYPE_III 0x83 struct uac_iso_endpoint_descriptor { __u8 bLength; /* in bytes: 7 */ @@ -265,6 +309,19 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_PITCH_CONTROL 0x02 #define UAC_EP_CS_ATTR_FILL_MAX 0x80 +/* Audio class v2.0: CLOCK_SOURCE descriptor */ + +struct uac_clock_source_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bmAttributes; + __u8 bmControls; + __u8 bAssocTerminal; + __u8 iClockSource; +} __attribute__((packed)); + /* A.10.2 Feature Unit Control Selectors */ struct uac_feature_unit_descriptor { -- cgit v1.2.3 From de48c7bc6f93c6c8e0be8612c9d72a2dc92eaa01 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:13 +0100 Subject: ALSA: usbaudio: consolidate header files Use the definitions from linux/usb/audio.h all over the ALSA USB audio driver and add some missing definitions there as well. Use the endpoint attribute macros from linux/usb/ch9 and remove the own things from sound/usb/usbaudio.h. Now things are also nicely prefixed which makes understanding the code easier. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index fb1a97bf943d..6bb293684eb8 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -35,8 +35,17 @@ #define UAC_MIXER_UNIT 0x04 #define UAC_SELECTOR_UNIT 0x05 #define UAC_FEATURE_UNIT 0x06 -#define UAC_PROCESSING_UNIT 0x07 -#define UAC_EXTENSION_UNIT 0x08 +#define UAC_PROCESSING_UNIT_V1 0x07 +#define UAC_EXTENSION_UNIT_V1 0x08 + +/* UAC v2.0 types */ +#define UAC_EFFECT_UNIT 0x07 +#define UAC_PROCESSING_UNIT_V2 0x08 +#define UAC_EXTENSION_UNIT_V2 0x09 +#define UAC_CLOCK_SOURCE 0x0a +#define UAC_CLOCK_SELECTOR 0x0b +#define UAC_CLOCK_MULTIPLIER 0x0c +#define UAC_SAMPLE_RATE_CONVERTER 0x0d /* A.6 Audio Class-Specific AS Interface Descriptor Subtypes */ #define UAC_AS_GENERAL 0x01 @@ -69,6 +78,10 @@ #define UAC_GET_STAT 0xff +/* Audio class v2.0 handles all the parameter calls differently */ +#define UAC2_CS_CUR 0x01 +#define UAC2_CS_RANGE 0x02 + /* MIDI - A.1 MS Class-Specific Interface Descriptor Subtypes */ #define UAC_MS_HEADER 0x01 #define UAC_MIDI_IN_JACK 0x02 @@ -133,6 +146,10 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_MICROPHONE_ARRAY 0x205 #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 +/* Terminals - control selectors */ + +#define UAC_TERMINAL_CS_COPY_PROTECT_CONTROL 0x01 + /* 4.3.2.2 Output Terminal Descriptor */ struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ @@ -263,6 +280,9 @@ struct uac_format_type_i_ext_descriptor { /* Formats - Audio Data Format Type I Codes */ +#define UAC_FORMAT_TYPE_II_MPEG 0x1001 +#define UAC_FORMAT_TYPE_II_AC3 0x1002 + struct uac_format_type_ii_discrete_descriptor { __u8 bLength; __u8 bDescriptorType; @@ -285,6 +305,13 @@ struct uac_format_type_ii_ext_descriptor { __u8 bSideBandProtocol; } __attribute__((packed)); +/* type III */ +#define UAC_FORMAT_TYPE_III_IEC1937_AC3 0x2001 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG1_LAYER1 0x2002 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_NOEXT 0x2003 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_EXT 0x2004 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER1_LS 0x2005 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER23_LS 0x2006 /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 -- cgit v1.2.3 From b30477d5e2961bfd90ad4146c517871ca8a6bebc Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 3 Mar 2010 11:05:55 +0100 Subject: ALSA: timer - pass real event in snd_timer_notify1() to instance callback Do not use hardcoded SNDRV_TIMER_EVENT_START value. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f57bb92eb5a..098595500632 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -544,7 +544,7 @@ struct snd_rawmidi_status { * Timer section - /dev/snd/timer */ -#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 5) +#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6) enum { SNDRV_TIMER_CLASS_NONE = -1, -- cgit v1.2.3 From 2b9ddcb8b2ce6a44f0f969000f16b016caa64294 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 4 Mar 2010 19:46:18 +0100 Subject: ALSA: usb/audio.h: Fix field order Signed-off-by: Daniel Mack Cc: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 6bb293684eb8..4d3e450e2b03 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -269,8 +269,8 @@ struct uac_format_type_i_ext_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; - __u8 bSubslotSize; __u8 bFormatType; + __u8 bSubslotSize; __u8 bBitResolution; __u8 bHeaderLength; __u8 bControlSize; -- cgit v1.2.3