From 09787492537462e3c7b8f67b30ff9704062f97cc Mon Sep 17 00:00:00 2001 From: Abhijeet Kumar Date: Tue, 23 Jan 2018 23:00:51 +0530 Subject: [PATCH 01/31] ALSA: hda: Copying sync power state helper to core The current sync_power_state is local to hda code, moving it core so that other users apart from hda legacy can use it. The helper function ensures the actual state reaches the target state. Signed-off-by: Abhijeet Kumar Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 2 ++ sound/hda/hdac_device.c | 35 +++++++++++++++++++++++++++++++++++ 2 files changed, 37 insertions(+) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 68169e3749de..4c93ff5301bd 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -146,6 +146,8 @@ int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm); bool snd_hdac_check_power_state(struct hdac_device *hdac, hda_nid_t nid, unsigned int target_state); +unsigned int snd_hdac_sync_power_state(struct hdac_device *hdac, + hda_nid_t nid, unsigned int target_state); /** * snd_hdac_read_parm - read a codec parameter * @codec: the codec object diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 06f845e293cb..7ba100bb1c3f 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -3,6 +3,7 @@ */ #include +#include #include #include #include @@ -1064,3 +1065,37 @@ bool snd_hdac_check_power_state(struct hdac_device *hdac, return (state == target_state); } EXPORT_SYMBOL_GPL(snd_hdac_check_power_state); +/** + * snd_hdac_sync_power_state - wait until actual power state matches + * with the target state + * + * @hdac: the HDAC device + * @nid: NID to send the command + * @target_state: target state to check for + * + * Return power state or PS_ERROR if codec rejects GET verb. + */ +unsigned int snd_hdac_sync_power_state(struct hdac_device *codec, + hda_nid_t nid, unsigned int power_state) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(500); + unsigned int state, actual_state, count; + + for (count = 0; count < 500; count++) { + state = snd_hdac_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + if (state & AC_PWRST_ERROR) { + msleep(20); + break; + } + actual_state = (state >> 4) & 0x0f; + if (actual_state == power_state) + break; + if (time_after_eq(jiffies, end_time)) + break; + /* wait until the codec reachs to the target state */ + msleep(1); + } + return state; +} +EXPORT_SYMBOL_GPL(snd_hdac_sync_power_state); From 3b5b899ca67db07a4c4825911072221f99e157e2 Mon Sep 17 00:00:00 2001 From: Abhijeet Kumar Date: Tue, 23 Jan 2018 23:00:52 +0530 Subject: [PATCH 02/31] ALSA: hda: Make use of core codec functions to sync power state Since sync_power_state is moved to core it's better to use the helper function to ensure the actual power state reaches target instead of using the local helper functions already exsisting in hda code. Signed-off-by: Abhijeet Kumar Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 28 +--------------------------- sound/pci/hda/hda_local.h | 6 +++++- 2 files changed, 6 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e018ecbf78a8..5bc3a7468e17 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2702,32 +2702,6 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, } EXPORT_SYMBOL_GPL(snd_hda_codec_set_power_to_all); -/* - * wait until the state is reached, returns the current state - */ -static unsigned int hda_sync_power_state(struct hda_codec *codec, - hda_nid_t fg, - unsigned int power_state) -{ - unsigned long end_time = jiffies + msecs_to_jiffies(500); - unsigned int state, actual_state; - - for (;;) { - state = snd_hda_codec_read(codec, fg, 0, - AC_VERB_GET_POWER_STATE, 0); - if (state & AC_PWRST_ERROR) - break; - actual_state = (state >> 4) & 0x0f; - if (actual_state == power_state) - break; - if (time_after_eq(jiffies, end_time)) - break; - /* wait until the codec reachs to the target state */ - msleep(1); - } - return state; -} - /** * snd_hda_codec_eapd_power_filter - A power filter callback for EAPD * @codec: the HDA codec @@ -2790,7 +2764,7 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, state); snd_hda_codec_set_power_to_all(codec, fg, power_state); } - state = hda_sync_power_state(codec, fg, power_state); + state = snd_hda_sync_power_state(codec, fg, power_state); if (!(state & AC_PWRST_ERROR)) break; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 5b5c324c99b9..321e78baa63c 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -622,7 +622,11 @@ snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid, { return snd_hdac_check_power_state(&codec->core, nid, target_state); } - +static inline bool snd_hda_sync_power_state(struct hda_codec *codec, + hda_nid_t nid, unsigned int target_state) +{ + return snd_hdac_sync_power_state(&codec->core, nid, target_state); +} unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state); From a4463c92db0805581f4dfb700f72533cf25ebd48 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Wed, 14 Feb 2018 00:04:58 +0100 Subject: [PATCH 03/31] ALSA: emu10k1: remove reserved_page The emu10k1-family chips need the first page (index 0) reserved in their page tables for some reason (every emu10k1 driver I've checked does this without much of an explanation). Using the first page for normal samples results in a broken playback. However, we already have a dummy page allocated - so called "silent page" and, in fact, had always been setting it as the first page in the chip page table because an initialization of every entry of the page table to point to a silent page happens after and overwrites the reserved_page allocation. So the only thing remaining to remove the reserved_page allocation is a trivial change to the page allocation logic to ignore the first page entry and start its allocations from the second entry (index 1). Signed-off-by: Maciej S. Szmigiero Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 1 - sound/pci/emu10k1/emu10k1_main.c | 11 ----------- sound/pci/emu10k1/memory.c | 8 ++++++-- 3 files changed, 6 insertions(+), 14 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 4f42affe777c..db32b7de52e0 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1718,7 +1718,6 @@ struct snd_emu10k1 { struct snd_dma_buffer p16v_buffer; struct snd_util_memhdr *memhdr; /* page allocation list */ - struct snd_emu10k1_memblk *reserved_page; /* reserved page */ struct list_head mapped_link_head; struct list_head mapped_order_link_head; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index ccf4415a1c7b..a0a4d31ded53 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1272,12 +1272,6 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) release_firmware(emu->dock_fw); if (emu->irq >= 0) free_irq(emu->irq, emu); - /* remove reserved page */ - if (emu->reserved_page) { - snd_emu10k1_synth_free(emu, - (struct snd_util_memblk *)emu->reserved_page); - emu->reserved_page = NULL; - } snd_util_memhdr_free(emu->memhdr); if (emu->silent_page.area) snd_dma_free_pages(&emu->silent_page); @@ -1993,11 +1987,6 @@ int snd_emu10k1_create(struct snd_card *card, SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; - emu->reserved_page = (struct snd_emu10k1_memblk *) - snd_emu10k1_synth_alloc(emu, 4096); - if (emu->reserved_page) - emu->reserved_page->map_locked = 1; - /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, PAGE_SIZE); silent_page = emu->silent_page.addr << emu->address_mode; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 4f1f69be1865..eaa61024ac7f 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -102,7 +102,7 @@ static void emu10k1_memblk_init(struct snd_emu10k1_memblk *blk) */ static int search_empty_map_area(struct snd_emu10k1 *emu, int npages, struct list_head **nextp) { - int page = 0, found_page = -ENOMEM; + int page = 1, found_page = -ENOMEM; int max_size = npages; int size; struct list_head *candidate = &emu->mapped_link_head; @@ -147,6 +147,10 @@ static int map_memblk(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) page = search_empty_map_area(emu, blk->pages, &next); if (page < 0) /* not found */ return page; + if (page == 0) { + dev_err(emu->card->dev, "trying to map zero (reserved) page\n"); + return -EINVAL; + } /* insert this block in the proper position of mapped list */ list_add_tail(&blk->mapped_link, next); /* append this as a newest block in order list */ @@ -177,7 +181,7 @@ static int unmap_memblk(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) q = get_emu10k1_memblk(p, mapped_link); start_page = q->mapped_page + q->pages; } else - start_page = 0; + start_page = 1; if ((p = blk->mapped_link.next) != &emu->mapped_link_head) { q = get_emu10k1_memblk(p, mapped_link); end_page = q->mapped_page; From 70d0bc7dca918a0add6da817bae70f334fb4cc60 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Wed, 14 Feb 2018 00:05:16 +0100 Subject: [PATCH 04/31] ALSA: emu10k1: use dma_set_mask_and_coherent() We have been calling dma_set_mask() and then dma_set_coherent_mask() with the same value, but there is a dma_set_mask_and_coherent() function that does exactly that so let's use it instead. Signed-off-by: Maciej S. Szmigiero Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index a0a4d31ded53..11e868f569d6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1871,8 +1871,7 @@ int snd_emu10k1_create(struct snd_card *card, emu->address_mode = is_audigy ? 0 : 1; /* set the DMA transfer mask */ emu->dma_mask = emu->address_mode ? EMU10K1_DMA_MASK : AUDIGY_DMA_MASK; - if (dma_set_mask(&pci->dev, emu->dma_mask) < 0 || - dma_set_coherent_mask(&pci->dev, emu->dma_mask) < 0) { + if (dma_set_mask_and_coherent(&pci->dev, emu->dma_mask) < 0) { dev_err(card->dev, "architecture does not support PCI busmaster DMA with mask 0x%lx\n", emu->dma_mask); From 541b9bad169d2422c7c7e3905b105930a1b6eb32 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Wed, 14 Feb 2018 00:06:18 +0100 Subject: [PATCH 05/31] ALSA: emu10k1: add optional debug printouts with DMA addresses When we get a IOMMU page fault for a emu10k1 device it is very hard to discover which of chip many DMA allocations triggered it (since on a IOMMU system the DMA address space is often very different from the CPU one). Let's add optional debug printouts providing this information. These debug printouts are only enabled on an explicit request via the kernel dynamic debug mechanism. Signed-off-by: Maciej S. Szmigiero Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 8 ++++++++ sound/pci/emu10k1/memory.c | 15 +++++++++++---- 2 files changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 11e868f569d6..8decd2a7a404 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1898,6 +1898,9 @@ int snd_emu10k1_create(struct snd_card *card, err = -ENOMEM; goto error; } + dev_dbg(card->dev, "page table address range is %.8lx:%.8lx\n", + (unsigned long)emu->ptb_pages.addr, + (unsigned long)(emu->ptb_pages.addr + emu->ptb_pages.bytes)); emu->page_ptr_table = vmalloc(emu->max_cache_pages * sizeof(void *)); emu->page_addr_table = vmalloc(emu->max_cache_pages * @@ -1912,6 +1915,11 @@ int snd_emu10k1_create(struct snd_card *card, err = -ENOMEM; goto error; } + dev_dbg(card->dev, "silent page range is %.8lx:%.8lx\n", + (unsigned long)emu->silent_page.addr, + (unsigned long)(emu->silent_page.addr + + emu->silent_page.bytes)); + emu->memhdr = snd_util_memhdr_new(emu->max_cache_pages * PAGE_SIZE); if (emu->memhdr == NULL) { err = -ENOMEM; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index eaa61024ac7f..fcb04cbbc9ab 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -34,7 +34,10 @@ * aligned pages in others */ #define __set_ptb_entry(emu,page,addr) \ - (((u32 *)(emu)->ptb_pages.area)[page] = cpu_to_le32(((addr) << (emu->address_mode)) | (page))) + (((__le32 *)(emu)->ptb_pages.area)[page] = \ + cpu_to_le32(((addr) << (emu->address_mode)) | (page))) +#define __get_ptb_entry(emu, page) \ + (le32_to_cpu(((__le32 *)(emu)->ptb_pages.area)[page])) #define UNIT_PAGES (PAGE_SIZE / EMUPAGESIZE) #define MAX_ALIGN_PAGES0 (MAXPAGES0 / UNIT_PAGES) @@ -44,8 +47,7 @@ /* get offset address from aligned page */ #define aligned_page_offset(page) ((page) << PAGE_SHIFT) -#if PAGE_SIZE == 4096 -/* page size == EMUPAGESIZE */ +#if PAGE_SIZE == EMUPAGESIZE && !IS_ENABLED(CONFIG_DYNAMIC_DEBUG) /* fill PTB entrie(s) corresponding to page with addr */ #define set_ptb_entry(emu,page,addr) __set_ptb_entry(emu,page,addr) /* fill PTB entrie(s) corresponding to page with silence pointer */ @@ -58,6 +60,8 @@ static inline void set_ptb_entry(struct snd_emu10k1 *emu, int page, dma_addr_t a page *= UNIT_PAGES; for (i = 0; i < UNIT_PAGES; i++, page++) { __set_ptb_entry(emu, page, addr); + dev_dbg(emu->card->dev, "mapped page %d to entry %.8x\n", page, + (unsigned int)__get_ptb_entry(emu, page)); addr += EMUPAGESIZE; } } @@ -65,9 +69,12 @@ static inline void set_silent_ptb(struct snd_emu10k1 *emu, int page) { int i; page *= UNIT_PAGES; - for (i = 0; i < UNIT_PAGES; i++, page++) + for (i = 0; i < UNIT_PAGES; i++, page++) { /* do not increment ptr */ __set_ptb_entry(emu, page, emu->silent_page.addr); + dev_dbg(emu->card->dev, "mapped silent page %d to entry %.8x\n", + page, (unsigned int)__get_ptb_entry(emu, page)); + } } #endif /* PAGE_SIZE */ From 055e0ae10f509482ca5b140e8e5f4a77e31efdd5 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Wed, 14 Feb 2018 00:06:34 +0100 Subject: [PATCH 06/31] ALSA: emu10k1: make sure synth DMA pages are allocated with DMA functions Commit a5003fc04113 ("[ALSA] emu10k1 - simplify page allocation for synth") switched from using the DMA allocator for synth DMA pages to manually calling alloc_page(). However, this usage has an implicit assumption that the DMA address space for the emu10k1-family chip is the same as the CPU physical address space which is not true for a system with a IOMMU. Since this made the synth part of the driver non-functional on such systems let's effectively revert that commit (while keeping the __synth_free_pages() simplification). Signed-off-by: Maciej S. Szmigiero Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/memory.c | 44 +++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 17 deletions(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index fcb04cbbc9ab..1d0ce7356bbd 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -461,10 +461,19 @@ static void get_single_page_range(struct snd_util_memhdr *hdr, static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, int last_page) { + struct snd_dma_buffer dmab; int page; + dmab.dev.type = SNDRV_DMA_TYPE_DEV; + dmab.dev.dev = snd_dma_pci_data(emu->pci); + for (page = first_page; page <= last_page; page++) { - free_page((unsigned long)emu->page_ptr_table[page]); + if (emu->page_ptr_table[page] == NULL) + continue; + dmab.area = emu->page_ptr_table[page]; + dmab.addr = emu->page_addr_table[page]; + dmab.bytes = PAGE_SIZE; + snd_dma_free_pages(&dmab); emu->page_addr_table[page] = 0; emu->page_ptr_table[page] = NULL; } @@ -476,30 +485,31 @@ static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) { int page, first_page, last_page; + struct snd_dma_buffer dmab; emu10k1_memblk_init(blk); get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - /* first try to allocate from <4GB zone */ - struct page *p = alloc_page(GFP_KERNEL | GFP_DMA32 | - __GFP_NOWARN); - if (!p || (page_to_pfn(p) & ~(emu->dma_mask >> PAGE_SHIFT))) { - if (p) - __free_page(p); - /* try to allocate from <16MB zone */ - p = alloc_page(GFP_ATOMIC | GFP_DMA | - __GFP_NORETRY | /* no OOM-killer */ - __GFP_NOWARN); + if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + PAGE_SIZE, &dmab) < 0) + goto __fail; + if (!is_valid_page(emu, dmab.addr)) { + snd_dma_free_pages(&dmab); + goto __fail; } - if (!p) { - __synth_free_pages(emu, first_page, page - 1); - return -ENOMEM; - } - emu->page_addr_table[page] = page_to_phys(p); - emu->page_ptr_table[page] = page_address(p); + emu->page_addr_table[page] = dmab.addr; + emu->page_ptr_table[page] = dmab.area; } return 0; + +__fail: + /* release allocated pages */ + last_page = page - 1; + __synth_free_pages(emu, first_page, last_page); + + return -ENOMEM; } /* From 04f8773a3e980f60953e7aeb36ec6c2631e11f10 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Wed, 14 Feb 2018 00:07:58 +0100 Subject: [PATCH 07/31] ALSA: emu10k1: add a IOMMU workaround The Audigy 2 CA0102 chip (but most likely others from the emu10k1 family, too) has a problem that from time to time it likes to do few DMA reads a bit beyond its normal allocation and gets very confused if these reads get blocked by a IOMMU. For the first (reserved) page this happens multiple times at every playback, for various synth pages it happens randomly, rarely for PCM playback buffers and the page table memory itself. All these reads seem to follow a similar pattern, observed read offsets beyond the allocation end were 0x00, 0x40, 0x80 and 0xc0 (PCI cache line multiples), so it looks like the device tries to accesses up to 256 extra bytes. As a workaround let's widen these DMA allocations by an extra page if we detect that the device is behind a non-passthrough IOMMU (the DMA memory should be relatively plenty on IOMMU systems). Signed-off-by: Maciej S. Szmigiero Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 3 ++ sound/pci/emu10k1/emu10k1_main.c | 49 ++++++++++++++++++++++++++++---- sound/pci/emu10k1/emupcm.c | 10 ++++++- sound/pci/emu10k1/memory.c | 40 ++++++++++++++++++++++++-- 4 files changed, 93 insertions(+), 9 deletions(-) diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index db32b7de52e0..5ebcc51c0a6a 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1710,6 +1710,7 @@ struct snd_emu10k1 { unsigned int ecard_ctrl; /* ecard control bits */ unsigned int address_mode; /* address mode */ unsigned long dma_mask; /* PCI DMA mask */ + bool iommu_workaround; /* IOMMU workaround needed */ unsigned int delay_pcm_irq; /* in samples */ int max_cache_pages; /* max memory size / PAGE_SIZE */ struct snd_dma_buffer silent_page; /* silent page */ @@ -1877,6 +1878,8 @@ void snd_p16v_resume(struct snd_emu10k1 *emu); /* memory allocation */ struct snd_util_memblk *snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *substream); int snd_emu10k1_free_pages(struct snd_emu10k1 *emu, struct snd_util_memblk *blk); +int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size, + struct snd_dma_buffer *dmab); struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size); int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk); int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 8decd2a7a404..18267de3a269 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -36,6 +36,7 @@ #include #include #include +#include #include #include #include @@ -1758,6 +1759,38 @@ static struct snd_emu_chip_details emu_chip_details[] = { { } /* terminator */ }; +/* + * The chip (at least the Audigy 2 CA0102 chip, but most likely others, too) + * has a problem that from time to time it likes to do few DMA reads a bit + * beyond its normal allocation and gets very confused if these reads get + * blocked by a IOMMU. + * + * This behaviour has been observed for the first (reserved) page + * (for which it happens multiple times at every playback), often for various + * synth pages and sometimes for PCM playback buffers and the page table + * memory itself. + * + * As a workaround let's widen these DMA allocations by an extra page if we + * detect that the device is behind a non-passthrough IOMMU. + */ +static void snd_emu10k1_detect_iommu(struct snd_emu10k1 *emu) +{ + struct iommu_domain *domain; + + emu->iommu_workaround = false; + + if (!iommu_present(emu->card->dev->bus)) + return; + + domain = iommu_get_domain_for_dev(emu->card->dev); + if (domain && domain->type == IOMMU_DOMAIN_IDENTITY) + return; + + dev_notice(emu->card->dev, + "non-passthrough IOMMU detected, widening DMA allocations"); + emu->iommu_workaround = true; +} + int snd_emu10k1_create(struct snd_card *card, struct pci_dev *pci, unsigned short extin_mask, @@ -1770,6 +1803,7 @@ int snd_emu10k1_create(struct snd_card *card, struct snd_emu10k1 *emu; int idx, err; int is_audigy; + size_t page_table_size; unsigned int silent_page; const struct snd_emu_chip_details *c; static struct snd_device_ops ops = { @@ -1867,6 +1901,8 @@ int snd_emu10k1_create(struct snd_card *card, is_audigy = emu->audigy = c->emu10k2_chip; + snd_emu10k1_detect_iommu(emu); + /* set addressing mode */ emu->address_mode = is_audigy ? 0 : 1; /* set the DMA transfer mask */ @@ -1893,8 +1929,11 @@ int snd_emu10k1_create(struct snd_card *card, emu->port = pci_resource_start(pci, 0); emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT; - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - (emu->address_mode ? 32 : 16) * 1024, &emu->ptb_pages) < 0) { + + page_table_size = sizeof(u32) * (emu->address_mode ? MAXPAGES1 : + MAXPAGES0); + if (snd_emu10k1_alloc_pages_maybe_wider(emu, page_table_size, + &emu->ptb_pages) < 0) { err = -ENOMEM; goto error; } @@ -1910,8 +1949,8 @@ int snd_emu10k1_create(struct snd_card *card, goto error; } - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), - EMUPAGESIZE, &emu->silent_page) < 0) { + if (snd_emu10k1_alloc_pages_maybe_wider(emu, EMUPAGESIZE, + &emu->silent_page) < 0) { err = -ENOMEM; goto error; } @@ -1995,7 +2034,7 @@ int snd_emu10k1_create(struct snd_card *card, 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; /* Clear silent pages and set up pointers */ - memset(emu->silent_page.area, 0, PAGE_SIZE); + memset(emu->silent_page.area, 0, emu->silent_page.bytes); silent_page = emu->silent_page.addr << emu->address_mode; for (idx = 0; idx < (emu->address_mode ? MAXPAGES1 : MAXPAGES0); idx++) ((u32 *)emu->ptb_pages.area)[idx] = cpu_to_le32(silent_page | idx); diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 2683b9717215..cefe613ef7b7 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -411,12 +411,20 @@ static int snd_emu10k1_playback_hw_params(struct snd_pcm_substream *substream, struct snd_emu10k1 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_emu10k1_pcm *epcm = runtime->private_data; + size_t alloc_size; int err; if ((err = snd_emu10k1_pcm_channel_alloc(epcm, params_channels(hw_params))) < 0) return err; - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) + + alloc_size = params_buffer_bytes(hw_params); + if (emu->iommu_workaround) + alloc_size += EMUPAGESIZE; + err = snd_pcm_lib_malloc_pages(substream, alloc_size); + if (err < 0) return err; + if (emu->iommu_workaround && runtime->dma_bytes >= EMUPAGESIZE) + runtime->dma_bytes -= EMUPAGESIZE; if (err > 0) { /* change */ int mapped; if (epcm->memblk != NULL) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 1d0ce7356bbd..5865f3b90b34 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -377,6 +377,33 @@ int snd_emu10k1_free_pages(struct snd_emu10k1 *emu, struct snd_util_memblk *blk) return snd_emu10k1_synth_free(emu, blk); } +/* + * allocate DMA pages, widening the allocation if necessary + * + * See the comment above snd_emu10k1_detect_iommu() in emu10k1_main.c why + * this might be needed. + * + * If you modify this function check whether __synth_free_pages() also needs + * changes. + */ +int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size, + struct snd_dma_buffer *dmab) +{ + if (emu->iommu_workaround) { + size_t npages = (size + PAGE_SIZE - 1) / PAGE_SIZE; + size_t size_real = npages * PAGE_SIZE; + + /* + * The device has been observed to accesses up to 256 extra + * bytes, but use 1k to be safe. + */ + if (size_real < size + 1024) + size += PAGE_SIZE; + } + + return snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), size, dmab); +} /* * memory allocation using multiple pages (for synth) @@ -472,7 +499,15 @@ static void __synth_free_pages(struct snd_emu10k1 *emu, int first_page, continue; dmab.area = emu->page_ptr_table[page]; dmab.addr = emu->page_addr_table[page]; + + /* + * please keep me in sync with logic in + * snd_emu10k1_alloc_pages_maybe_wider() + */ dmab.bytes = PAGE_SIZE; + if (emu->iommu_workaround) + dmab.bytes *= 2; + snd_dma_free_pages(&dmab); emu->page_addr_table[page] = 0; emu->page_ptr_table[page] = NULL; @@ -491,9 +526,8 @@ static int synth_alloc_pages(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk get_single_page_range(emu->memhdr, blk, &first_page, &last_page); /* allocate kernel pages */ for (page = first_page; page <= last_page; page++) { - if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(emu->pci), - PAGE_SIZE, &dmab) < 0) + if (snd_emu10k1_alloc_pages_maybe_wider(emu, PAGE_SIZE, + &dmab) < 0) goto __fail; if (!is_valid_page(emu, dmab.addr)) { snd_dma_free_pages(&dmab); From 248a380a3c2e7e3363f26c17b6923b430cf6463e Mon Sep 17 00:00:00 2001 From: Matt Ranostay Date: Fri, 16 Feb 2018 21:53:15 -0800 Subject: [PATCH 08/31] ALSA: hda-beep: add SPDX identifiers Add SPDX GPLv2.0+ identifiers and update authors email Signed-off-by: Matt Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 17 ++--------------- sound/pci/hda/hda_beep.h | 17 ++--------------- 2 files changed, 4 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eac..066b5b59c4d7 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -1,22 +1,9 @@ +// SPDX-License-Identifier: GPL-2.0+ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matt Ranostay + * Author: Matt Ranostay * Copyright (c) 2008 Embedded Alley Solutions Inc - * - * This driver 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. - * - * This driver is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 1052ad380e97..d1a6a9c1329a 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -1,22 +1,9 @@ +/* SPDX-License-Identifier: GPL-2.0+ */ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matt Ranostay + * Author: Matt Ranostay * Copyright (c) 2008 Embedded Alley Solutions Inc - * - * This driver 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. - * - * This driver is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #ifndef __SOUND_HDA_BEEP_H From ef21e1750158558d8369067e94d02d285011c37e Mon Sep 17 00:00:00 2001 From: Jaejoong Kim Date: Tue, 27 Feb 2018 11:15:59 +0900 Subject: [PATCH 09/31] ALSA: Use scnprintf() instead of snprintf() for show The show() method should use scnprintf() not snprintf() because snprintf() may returns a value that exceeds its second argument. Signed-off-by: Jaejoong Kim Signed-off-by: Takashi Iwai --- sound/core/init.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 4fa5dd955740..79b4df1c1dc7 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -670,7 +670,7 @@ card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%s\n", card->id); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->id); } static ssize_t @@ -710,7 +710,7 @@ card_number_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%i\n", card->number); + return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); From dd5f313be0d4d7a46f0e82c42aed4fbc784699ea Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 27 Feb 2018 17:29:54 +0000 Subject: [PATCH 10/31] ALSA: control: Fix a bunch of whitespace errors Remove a bunch of trailing whitespace errors. They are fairly annoying if you have your editor set to strip trailing whitespace because you find you've introduced more changes than you were trying to make. Signed-off-by: Richard Fitzgerald Signed-off-by: Takashi Iwai --- sound/core/control.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 8a77620a3854..69734b0eafd0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -105,7 +105,7 @@ static void snd_ctl_empty_read_queue(struct snd_ctl_file * ctl) { unsigned long flags; struct snd_kctl_event *cread; - + spin_lock_irqsave(&ctl->read_lock, flags); while (!list_empty(&ctl->events)) { cread = snd_kctl_event(ctl->events.next); @@ -159,7 +159,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, unsigned long flags; struct snd_ctl_file *ctl; struct snd_kctl_event *ev; - + if (snd_BUG_ON(!card || !id)) return; if (card->shutdown) @@ -213,7 +213,7 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, { unsigned int size; unsigned int idx; - + if (count == 0 || count > MAX_CONTROL_COUNT) return -EINVAL; @@ -238,7 +238,7 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, * @ncontrol: the initialization record * @private_data: the private data to set * - * Allocates a new struct snd_kcontrol instance and initialize from the given + * Allocates a new struct snd_kcontrol instance and initialize from the given * template. When the access field of ncontrol is 0, it's assumed as * READWRITE access. When the count field is 0, it's assumes as one. * @@ -251,7 +251,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, unsigned int count; unsigned int access; int err; - + if (snd_BUG_ON(!ncontrol || !ncontrol->info)) return NULL; @@ -753,7 +753,7 @@ static int snd_ctl_elem_list(struct snd_card *card, struct snd_ctl_elem_id id; unsigned int offset, space, jidx; int err = 0; - + if (copy_from_user(&list, _list, sizeof(list))) return -EFAULT; offset = list.offset; @@ -827,7 +827,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, struct snd_kcontrol_volatile *vd; unsigned int index_offset; int result; - + down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &info->id); if (kctl == NULL) { @@ -992,7 +992,7 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; int result; - + if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); @@ -1020,7 +1020,7 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; int result; - + if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; down_write(&card->controls_rwsem); From 338e17d3f58e9868a12af7deb1edcfb40bd588b2 Mon Sep 17 00:00:00 2001 From: Joey Pabalinas Date: Thu, 1 Mar 2018 04:17:07 -1000 Subject: [PATCH 11/31] ALSA: ice1712: replace strcpy() with strlcpy() Replace unsafe usages of strcpy() to copy the name argument into the sid.name buffer with strlcpy() to guard against possible buffer overflows. Signed-off-by: Joey Pabalinas Suggested-by: Andy Shevchenko Reviewed-by: Andy Shevchenko Signed-off-by: Takashi Iwai --- sound/pci/ice1712/juli.c | 8 ++++---- sound/pci/ice1712/quartet.c | 8 ++++---- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 0dbaccf61f33..21806bab4757 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include @@ -425,10 +426,9 @@ DECLARE_TLV_DB_SCALE(juli_master_db_scale, -6350, 50, 1); static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name) { - struct snd_ctl_elem_id sid; - memset(&sid, 0, sizeof(sid)); - /* FIXME: strcpy is bad. */ - strcpy(sid.name, name); + struct snd_ctl_elem_id sid = {0}; + + strlcpy(sid.name, name, sizeof(sid.name)); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d145b5eb7ff8..5bc836241c97 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -785,10 +786,9 @@ DECLARE_TLV_DB_SCALE(qtet_master_db_scale, -6350, 50, 1); static struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name) { - struct snd_ctl_elem_id sid; - memset(&sid, 0, sizeof(sid)); - /* FIXME: strcpy is bad. */ - strcpy(sid.name, name); + struct snd_ctl_elem_id sid = {0}; + + strlcpy(sid.name, name, sizeof(sid.name)); sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; return snd_ctl_find_id(card, &sid); } From 2e2c177ca84aff092c3c96714b0f6a12900f3946 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Mar 2018 08:26:48 +0100 Subject: [PATCH 12/31] ALSA: vmaster: Propagate slave error In slave_update() of vmaster code ignores the error from the slave get() callback and copies the values. It's not only about the missing error code but also that this may potentially lead to a leak of uninitialized variables when the slave get() don't clear them. This patch fixes slave_update() not to copy the potentially uninitialized values when an error is returned from the slave get() callback, and to propagate the error value properly. Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 8632301489fa..b67de2bb06a2 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -68,10 +68,13 @@ static int slave_update(struct link_slave *slave) return -ENOMEM; uctl->id = slave->slave.id; err = slave->slave.get(&slave->slave, uctl); + if (err < 0) + goto error; for (ch = 0; ch < slave->info.count; ch++) slave->vals[ch] = uctl->value.integer.value[ch]; + error: kfree(uctl); - return 0; + return err < 0 ? err : 0; } /* get the slave ctl info and save the initial values */ From 7a33a02ffb0620b01892c6c6808bb711b3f63e9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Mar 2018 08:32:41 +0100 Subject: [PATCH 13/31] ALSA: vmaster: Zero-clear ctl before calling slave get Use kzalloc() instead of kmalloc() so that we don't need to rely fully on the slave get() callback to clear the control value that might be copied to user-space. Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index b67de2bb06a2..9e96186742d0 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -63,7 +63,7 @@ static int slave_update(struct link_slave *slave) struct snd_ctl_elem_value *uctl; int err, ch; - uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); + uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (!uctl) return -ENOMEM; uctl->id = slave->slave.id; From 0bc66fd3b688ae33ac073de7e79ba1223ecc465c Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Mon, 12 Mar 2018 15:01:00 +0000 Subject: [PATCH 14/31] ALSA: echoaudio: remove redundant initialization of pointer 'pipe' The pointer 'pipe' is being initialized with a value that is never read and it is re-assigned later, hence the initialization is redundant and can be removed. Also remove pointer 'runtime' as it is no longer required. Cleans up clang warning: sound/pci/echoaudio/echoaudio.c:740:20: warning: Value stored to 'pipe' during its initialization is never read Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d68f99e076a8..0935a5c8741f 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -736,8 +736,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct echoaudio *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audiopipe *pipe = runtime->private_data; + struct audiopipe *pipe; int i, err; u32 channelmask = 0; struct snd_pcm_substream *s; From 491f833134ac474434e1c950925c58b2ac13ca72 Mon Sep 17 00:00:00 2001 From: Guneshwor Singh Date: Tue, 13 Mar 2018 16:40:08 +0530 Subject: [PATCH 15/31] ALSA: hda: Add Icelake PCI ID Icelake is a next generation Intel platform. Add PCI ID for it. Signed-off-by: Guneshwor Singh Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 96143df19b21..32acab080781 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2426,6 +2426,9 @@ static const struct pci_device_id azx_ids[] = { /* Cannonlake */ { PCI_DEVICE(0x8086, 0x9dc8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Icelake */ + { PCI_DEVICE(0x8086, 0x34c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, From 5730f9f744cfe20b771adc33f3b476b95d3eebba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Mar 2018 11:18:57 +0100 Subject: [PATCH 16/31] ALSA: pcm: Remove VLA usage A helper function used by snd_pcm_hw_refine() still keeps using VLA for timestamps of hw constraint rules that are non-fixed size. Let's replace the VLA with a simple kmalloc() array. Reference: https://lkml.org/lkml/2018/3/7/621 Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 77ba50ddcf9e..756a9a3884a5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -323,7 +323,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; unsigned int k; - unsigned int rstamps[constrs->rules_num]; + unsigned int *rstamps; unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; unsigned int stamp; struct snd_pcm_hw_rule *r; @@ -331,7 +331,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, struct snd_mask old_mask; struct snd_interval old_interval; bool again; - int changed; + int changed, err = 0; /* * Each application of rule has own sequence number. @@ -339,8 +339,9 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, * Each member of 'rstamps' array represents the sequence number of * recent application of corresponding rule. */ - for (k = 0; k < constrs->rules_num; k++) - rstamps[k] = 0; + rstamps = kcalloc(constrs->rules_num, sizeof(unsigned int), GFP_KERNEL); + if (!rstamps) + return -ENOMEM; /* * Each member of 'vstamps' array represents the sequence number of @@ -398,8 +399,10 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, } changed = r->func(params, r); - if (changed < 0) - return changed; + if (changed < 0) { + err = changed; + goto out; + } /* * When the parameter is changed, notify it to the caller @@ -430,7 +433,9 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, if (again) goto retry; - return 0; + out: + kfree(rstamps); + return err; } static int fixup_unreferenced_params(struct snd_pcm_substream *substream, From 09b9ddfaa18317f463085d602cf5f60a5fa88665 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Mar 2018 11:31:25 +0100 Subject: [PATCH 17/31] ALSA: pcm: Use krealloc() for resizing the rules array Just a minor simplification. Change from kcalloc() shouldn't matter as each array element is fully initialized. Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a83152e7d387..f4a19509cccf 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1129,16 +1129,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, if (constrs->rules_num >= constrs->rules_all) { struct snd_pcm_hw_rule *new; unsigned int new_rules = constrs->rules_all + 16; - new = kcalloc(new_rules, sizeof(*c), GFP_KERNEL); + new = krealloc(constrs->rules, new_rules * sizeof(*c), + GFP_KERNEL); if (!new) { va_end(args); return -ENOMEM; } - if (constrs->rules) { - memcpy(new, constrs->rules, - constrs->rules_num * sizeof(*c)); - kfree(constrs->rules); - } constrs->rules = new; constrs->rules_all = new_rules; } From ceb18f511beeb8b750f027f170eb6d901a082e9a Mon Sep 17 00:00:00 2001 From: Ruslan Bilovol Date: Mon, 19 Mar 2018 03:46:02 +0200 Subject: [PATCH 18/31] ALSA: usb-audio: move audioformat quirks to quirks.c Offload USB audio interface parsing function by moving quirks to a specially designed location (quirks.c) Signed-off-by: Ruslan Bilovol Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 34 ++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 4 ++++ sound/usb/stream.c | 30 +----------------------------- 3 files changed, 39 insertions(+), 29 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ea8f3de92fa4..eeea8e18aea3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1403,3 +1403,37 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return 0; } + +void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, + struct audioformat *fp, + int stream) +{ + switch (chip->usb_id) { + case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ + /* Optoplay sets the sample rate attribute although + * it seems not supporting it in fact. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ + case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + /* doesn't set the sample rate attribute, but supports it */ + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ + case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ + case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ + case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is + an older model 77d:223) */ + /* + * plantronics headset and Griffin iMic have set adaptive-in + * although it's really not... + */ + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; + else + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; + break; + } +} diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index b90c8b7caab5..a80e0ddd0736 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -42,4 +42,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int sample_bytes); +void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, + struct audioformat *fp, + int stream); + #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d1776e5517ff..dbbe854745ab 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -678,35 +678,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) INIT_LIST_HEAD(&fp->list); /* some quirks for attributes here */ - - switch (chip->usb_id) { - case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ - /* Optoplay sets the sample rate attribute although - * it seems not supporting it in fact. - */ - fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ - case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ - case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ - case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ - case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is - an older model 77d:223) */ - /* - * plantronics headset and Griffin iMic have set adaptive-in - * although it's really not... - */ - fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; - else - fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; - break; - } + snd_usb_audioformat_attributes_quirk(chip, fp, stream); /* ok, let's parse further... */ if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream) < 0) { From 9a2fe9b801f585baccf8352d82839dcd54b300cf Mon Sep 17 00:00:00 2001 From: Ruslan Bilovol Date: Wed, 21 Mar 2018 02:03:59 +0200 Subject: [PATCH 19/31] ALSA: usb: initial USB Audio Device Class 3.0 support Recently released USB Audio Class 3.0 specification introduces many significant changes comparing to previous versions, like - new Power Domains, support for LPM/L1 - new Cluster descriptor - changed layout of all class-specific descriptors - new High Capability descriptors - New class-specific String descriptors - new and removed units - additional sources for interrupts - removed Type II Audio Data Formats - ... and many other things (check spec) It also provides backward compatibility through multiple configurations, as well as requires mandatory support for BADD (Basic Audio Device Definition) on each ADC3.0 compliant device This patch adds initial support of UAC3 specification that is enough for Generic I/O Profile (BAOF, BAIF) device support from BADD document. Signed-off-by: Ruslan Bilovol Reviewed-by: Greg Kroah-Hartman Signed-off-by: Takashi Iwai --- include/linux/usb/audio-v2.h | 4 +- include/linux/usb/audio-v3.h | 395 +++++++++++++++++++++++++++++++++ include/uapi/linux/usb/audio.h | 1 + sound/usb/card.c | 7 +- sound/usb/card.h | 2 +- sound/usb/clock.c | 226 +++++++++++++++++-- sound/usb/clock.h | 4 +- sound/usb/format.c | 91 ++++++-- sound/usb/format.h | 6 +- sound/usb/mixer.c | 335 +++++++++++++++++++--------- sound/usb/stream.c | 365 +++++++++++++++++++++++++++--- 11 files changed, 1244 insertions(+), 192 deletions(-) create mode 100644 include/linux/usb/audio-v3.h diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index 3119d0ace7aa..2db83a191e78 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -34,12 +34,12 @@ * */ -static inline bool uac2_control_is_readable(u32 bmControls, u8 control) +static inline bool uac_v2v3_control_is_readable(u32 bmControls, u8 control) { return (bmControls >> (control * 2)) & 0x1; } -static inline bool uac2_control_is_writeable(u32 bmControls, u8 control) +static inline bool uac_v2v3_control_is_writeable(u32 bmControls, u8 control) { return (bmControls >> (control * 2)) & 0x2; } diff --git a/include/linux/usb/audio-v3.h b/include/linux/usb/audio-v3.h new file mode 100644 index 000000000000..a8959aaba0ae --- /dev/null +++ b/include/linux/usb/audio-v3.h @@ -0,0 +1,395 @@ +// SPDX-License-Identifier: GPL-2.0+ +/* + * Copyright (c) 2017 Ruslan Bilovol + * + * This file holds USB constants and structures defined + * by the USB DEVICE CLASS DEFINITION FOR AUDIO DEVICES Release 3.0. + */ + +#ifndef __LINUX_USB_AUDIO_V3_H +#define __LINUX_USB_AUDIO_V3_H + +#include + +/* + * v1.0, v2.0 and v3.0 of this standard have many things in common. For the rest + * of the definitions, please refer to audio.h and audio-v2.h + */ + +/* All High Capability descriptors have these 2 fields at the beginning */ +struct uac3_hc_descriptor_header { + __le16 wLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __le16 wDescriptorID; +} __attribute__ ((packed)); + +/* 4.3.1 CLUSTER DESCRIPTOR HEADER */ +struct uac3_cluster_header_descriptor { + __le16 wLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __le16 wDescriptorID; + __u8 bNrChannels; +} __attribute__ ((packed)); + +/* 4.3.2.1 SEGMENTS */ +struct uac3_cluster_segment_descriptor { + __le16 wLength; + __u8 bSegmentType; + /* __u8[0]; segment-specific data */ +} __attribute__ ((packed)); + +/* 4.3.2.1.1 END SEGMENT */ +struct uac3_cluster_end_segment_descriptor { + __le16 wLength; + __u8 bSegmentType; /* Constant END_SEGMENT */ +} __attribute__ ((packed)); + +/* 4.3.2.1.3.1 INFORMATION SEGMENT */ +struct uac3_cluster_information_segment_descriptor { + __le16 wLength; + __u8 bSegmentType; + __u8 bChPurpose; + __u8 bChRelationship; + __u8 bChGroupID; +} __attribute__ ((packed)); + +/* 4.5.2 CLASS-SPECIFIC AC INTERFACE DESCRIPTOR */ +struct uac3_ac_header_descriptor { + __u8 bLength; /* 10 */ + __u8 bDescriptorType; /* CS_INTERFACE descriptor type */ + __u8 bDescriptorSubtype; /* HEADER descriptor subtype */ + __u8 bCategory; + + /* includes Clock Source, Unit, Terminal, and Power Domain desc. */ + __le16 wTotalLength; + + __le32 bmControls; +} __attribute__ ((packed)); + +/* 4.5.2.1 INPUT TERMINAL DESCRIPTOR */ +struct uac3_input_terminal_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalID; + __le16 wTerminalType; + __u8 bAssocTerminal; + __u8 bCSourceID; + __le32 bmControls; + __le16 wClusterDescrID; + __le16 wExTerminalDescrID; + __le16 wConnectorsDescrID; + __le16 wTerminalDescrStr; +} __attribute__((packed)); + +/* 4.5.2.2 OUTPUT TERMINAL DESCRIPTOR */ +struct uac3_output_terminal_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalID; + __le16 wTerminalType; + __u8 bAssocTerminal; + __u8 bSourceID; + __u8 bCSourceID; + __le32 bmControls; + __le16 wExTerminalDescrID; + __le16 wConnectorsDescrID; + __le16 wTerminalDescrStr; +} __attribute__((packed)); + +/* 4.5.2.7 FEATURE UNIT DESCRIPTOR */ +struct uac3_feature_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bSourceID; + /* bmaControls is actually u32, + * but u8 is needed for the hybrid parser */ + __u8 bmaControls[0]; /* variable length */ + /* wFeatureDescrStr omitted */ +} __attribute__((packed)); + +#define UAC3_DT_FEATURE_UNIT_SIZE(ch) (7 + ((ch) + 1) * 4) + +/* As above, but more useful for defining your own descriptors */ +#define DECLARE_UAC3_FEATURE_UNIT_DESCRIPTOR(ch) \ +struct uac3_feature_unit_descriptor_##ch { \ + __u8 bLength; \ + __u8 bDescriptorType; \ + __u8 bDescriptorSubtype; \ + __u8 bUnitID; \ + __u8 bSourceID; \ + __le32 bmaControls[ch + 1]; \ + __le16 wFeatureDescrStr; \ +} __attribute__ ((packed)) + +/* 4.5.2.12 CLOCK SOURCE DESCRIPTOR */ +struct uac3_clock_source_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bmAttributes; + __le32 bmControls; + __u8 bReferenceTerminal; + __le16 wClockSourceStr; +} __attribute__((packed)); + +/* bmAttribute fields */ +#define UAC3_CLOCK_SOURCE_TYPE_EXT 0x0 +#define UAC3_CLOCK_SOURCE_TYPE_INT 0x1 +#define UAC3_CLOCK_SOURCE_ASYNC (0 << 2) +#define UAC3_CLOCK_SOURCE_SYNCED_TO_SOF (1 << 1) + +/* 4.5.2.13 CLOCK SELECTOR DESCRIPTOR */ +struct uac3_clock_selector_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bNrInPins; + __u8 baCSourceID[]; + /* bmControls and wCSelectorDescrStr omitted */ +} __attribute__((packed)); + +/* 4.5.2.14 CLOCK MULTIPLIER DESCRIPTOR */ +struct uac3_clock_multiplier_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bCSourceID; + __le32 bmControls; + __le16 wCMultiplierDescrStr; +} __attribute__((packed)); + +/* 4.5.2.15 POWER DOMAIN DESCRIPTOR */ +struct uac3_power_domain_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bPowerDomainID; + __le16 waRecoveryTime1; + __le16 waRecoveryTime2; + __u8 bNrEntities; + __u8 baEntityID[]; + /* wPDomainDescrStr omitted */ +} __attribute__((packed)); + +/* As above, but more useful for defining your own descriptors */ +#define DECLARE_UAC3_POWER_DOMAIN_DESCRIPTOR(n) \ +struct uac3_power_domain_descriptor_##n { \ + __u8 bLength; \ + __u8 bDescriptorType; \ + __u8 bDescriptorSubtype; \ + __u8 bPowerDomainID; \ + __le16 waRecoveryTime1; \ + __le16 waRecoveryTime2; \ + __u8 bNrEntities; \ + __u8 baEntityID[n]; \ + __le16 wPDomainDescrStr; \ +} __attribute__ ((packed)) + +/* 4.7.2 CLASS-SPECIFIC AS INTERFACE DESCRIPTOR */ +struct uac3_as_header_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalLink; + __le32 bmControls; + __le16 wClusterDescrID; + __le64 bmFormats; + __u8 bSubslotSize; + __u8 bBitResolution; + __le16 bmAuxProtocols; + __u8 bControlSize; +} __attribute__((packed)); + +#define UAC3_FORMAT_TYPE_I_RAW_DATA (1 << 6) + +/* 4.8.1.2 CLASS-SPECIFIC AS ISOCHRONOUS AUDIO DATA ENDPOINT DESCRIPTOR */ +struct uac3_iso_endpoint_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __le32 bmControls; + __u8 bLockDelayUnits; + __le16 wLockDelay; +} __attribute__((packed)); + +/* 6.1 INTERRUPT DATA MESSAGE */ +struct uac3_interrupt_data_msg { + __u8 bInfo; + __u8 bSourceType; + __le16 wValue; + __le16 wIndex; +} __attribute__((packed)); + +/* A.2 AUDIO AUDIO FUNCTION SUBCLASS CODES */ +#define UAC3_FUNCTION_SUBCLASS_UNDEFINED 0x00 +#define UAC3_FUNCTION_SUBCLASS_FULL_ADC_3_0 0x01 +/* BADD profiles */ +#define UAC3_FUNCTION_SUBCLASS_GENERIC_IO 0x20 +#define UAC3_FUNCTION_SUBCLASS_HEADPHONE 0x21 +#define UAC3_FUNCTION_SUBCLASS_SPEAKER 0x22 +#define UAC3_FUNCTION_SUBCLASS_MICROPHONE 0x23 +#define UAC3_FUNCTION_SUBCLASS_HEADSET 0x24 +#define UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER 0x25 +#define UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE 0x26 + +/* A.7 AUDIO FUNCTION CATEGORY CODES */ +#define UAC3_FUNCTION_SUBCLASS_UNDEFINED 0x00 +#define UAC3_FUNCTION_DESKTOP_SPEAKER 0x01 +#define UAC3_FUNCTION_HOME_THEATER 0x02 +#define UAC3_FUNCTION_MICROPHONE 0x03 +#define UAC3_FUNCTION_HEADSET 0x04 +#define UAC3_FUNCTION_TELEPHONE 0x05 +#define UAC3_FUNCTION_CONVERTER 0x06 +#define UAC3_FUNCTION_SOUND_RECORDER 0x07 +#define UAC3_FUNCTION_IO_BOX 0x08 +#define UAC3_FUNCTION_MUSICAL_INSTRUMENT 0x09 +#define UAC3_FUNCTION_PRO_AUDIO 0x0a +#define UAC3_FUNCTION_AUDIO_VIDEO 0x0b +#define UAC3_FUNCTION_CONTROL_PANEL 0x0c +#define UAC3_FUNCTION_HEADPHONE 0x0d +#define UAC3_FUNCTION_GENERIC_SPEAKER 0x0e +#define UAC3_FUNCTION_HEADSET_ADAPTER 0x0f +#define UAC3_FUNCTION_SPEAKERPHONE 0x10 +#define UAC3_FUNCTION_OTHER 0xff + +/* A.8 AUDIO CLASS-SPECIFIC DESCRIPTOR TYPES */ +#define UAC3_CS_UNDEFINED 0x20 +#define UAC3_CS_DEVICE 0x21 +#define UAC3_CS_CONFIGURATION 0x22 +#define UAC3_CS_STRING 0x23 +#define UAC3_CS_INTERFACE 0x24 +#define UAC3_CS_ENDPOINT 0x25 +#define UAC3_CS_CLUSTER 0x26 + +/* A.10 CLUSTER DESCRIPTOR SEGMENT TYPES */ +#define UAC3_SEGMENT_UNDEFINED 0x00 +#define UAC3_CLUSTER_DESCRIPTION 0x01 +#define UAC3_CLUSTER_VENDOR_DEFINED 0x1F +#define UAC3_CHANNEL_INFORMATION 0x20 +#define UAC3_CHANNEL_AMBISONIC 0x21 +#define UAC3_CHANNEL_DESCRIPTION 0x22 +#define UAC3_CHANNEL_VENDOR_DEFINED 0xFE +#define UAC3_END_SEGMENT 0xFF + +/* A.11 CHANNEL PURPOSE DEFINITIONS */ +#define UAC3_PURPOSE_UNDEFINED 0x00 +#define UAC3_PURPOSE_GENERIC_AUDIO 0x01 +#define UAC3_PURPOSE_VOICE 0x02 +#define UAC3_PURPOSE_SPEECH 0x03 +#define UAC3_PURPOSE_AMBIENT 0x04 +#define UAC3_PURPOSE_REFERENCE 0x05 +#define UAC3_PURPOSE_ULTRASONIC 0x06 +#define UAC3_PURPOSE_VIBROKINETIC 0x07 +#define UAC3_PURPOSE_NON_AUDIO 0xFF + +/* A.12 CHANNEL RELATIONSHIP DEFINITIONS */ +#define UAC3_CH_RELATIONSHIP_UNDEFINED 0x00 +#define UAC3_CH_MONO 0x01 +#define UAC3_CH_LEFT 0x02 +#define UAC3_CH_RIGHT 0x03 +#define UAC3_CH_ARRAY 0x04 +#define UAC3_CH_PATTERN_X 0x20 +#define UAC3_CH_PATTERN_Y 0x21 +#define UAC3_CH_PATTERN_A 0x22 +#define UAC3_CH_PATTERN_B 0x23 +#define UAC3_CH_PATTERN_M 0x24 +#define UAC3_CH_PATTERN_S 0x25 +#define UAC3_CH_FRONT_LEFT 0x80 +#define UAC3_CH_FRONT_RIGHT 0x81 +#define UAC3_CH_FRONT_CENTER 0x82 +#define UAC3_CH_FRONT_LEFT_OF_CENTER 0x83 +#define UAC3_CH_FRONT_RIGHT_OF_CENTER 0x84 +#define UAC3_CH_FRONT_WIDE_LEFT 0x85 +#define UAC3_CH_FRONT_WIDE_RIGHT 0x86 +#define UAC3_CH_SIDE_LEFT 0x87 +#define UAC3_CH_SIDE_RIGHT 0x88 +#define UAC3_CH_SURROUND_ARRAY_LEFT 0x89 +#define UAC3_CH_SURROUND_ARRAY_RIGHT 0x8A +#define UAC3_CH_BACK_LEFT 0x8B +#define UAC3_CH_BACK_RIGHT 0x8C +#define UAC3_CH_BACK_CENTER 0x8D +#define UAC3_CH_BACK_LEFT_OF_CENTER 0x8E +#define UAC3_CH_BACK_RIGHT_OF_CENTER 0x8F +#define UAC3_CH_BACK_WIDE_LEFT 0x90 +#define UAC3_CH_BACK_WIDE_RIGHT 0x91 +#define UAC3_CH_TOP_CENTER 0x92 +#define UAC3_CH_TOP_FRONT_LEFT 0x93 +#define UAC3_CH_TOP_FRONT_RIGHT 0x94 +#define UAC3_CH_TOP_FRONT_CENTER 0x95 +#define UAC3_CH_TOP_FRONT_LOC 0x96 +#define UAC3_CH_TOP_FRONT_ROC 0x97 +#define UAC3_CH_TOP_FRONT_WIDE_LEFT 0x98 +#define UAC3_CH_TOP_FRONT_WIDE_RIGHT 0x99 +#define UAC3_CH_TOP_SIDE_LEFT 0x9A +#define UAC3_CH_TOP_SIDE_RIGHT 0x9B +#define UAC3_CH_TOP_SURR_ARRAY_LEFT 0x9C +#define UAC3_CH_TOP_SURR_ARRAY_RIGHT 0x9D +#define UAC3_CH_TOP_BACK_LEFT 0x9E +#define UAC3_CH_TOP_BACK_RIGHT 0x9F +#define UAC3_CH_TOP_BACK_CENTER 0xA0 +#define UAC3_CH_TOP_BACK_LOC 0xA1 +#define UAC3_CH_TOP_BACK_ROC 0xA2 +#define UAC3_CH_TOP_BACK_WIDE_LEFT 0xA3 +#define UAC3_CH_TOP_BACK_WIDE_RIGHT 0xA4 +#define UAC3_CH_BOTTOM_CENTER 0xA5 +#define UAC3_CH_BOTTOM_FRONT_LEFT 0xA6 +#define UAC3_CH_BOTTOM_FRONT_RIGHT 0xA7 +#define UAC3_CH_BOTTOM_FRONT_CENTER 0xA8 +#define UAC3_CH_BOTTOM_FRONT_LOC 0xA9 +#define UAC3_CH_BOTTOM_FRONT_ROC 0xAA +#define UAC3_CH_BOTTOM_FRONT_WIDE_LEFT 0xAB +#define UAC3_CH_BOTTOM_FRONT_WIDE_RIGHT 0xAC +#define UAC3_CH_BOTTOM_SIDE_LEFT 0xAD +#define UAC3_CH_BOTTOM_SIDE_RIGHT 0xAE +#define UAC3_CH_BOTTOM_SURR_ARRAY_LEFT 0xAF +#define UAC3_CH_BOTTOM_SURR_ARRAY_RIGHT 0xB0 +#define UAC3_CH_BOTTOM_BACK_LEFT 0xB1 +#define UAC3_CH_BOTTOM_BACK_RIGHT 0xB2 +#define UAC3_CH_BOTTOM_BACK_CENTER 0xB3 +#define UAC3_CH_BOTTOM_BACK_LOC 0xB4 +#define UAC3_CH_BOTTOM_BACK_ROC 0xB5 +#define UAC3_CH_BOTTOM_BACK_WIDE_LEFT 0xB6 +#define UAC3_CH_BOTTOM_BACK_WIDE_RIGHT 0xB7 +#define UAC3_CH_LOW_FREQUENCY_EFFECTS 0xB8 +#define UAC3_CH_LFE_LEFT 0xB9 +#define UAC3_CH_LFE_RIGHT 0xBA +#define UAC3_CH_HEADPHONE_LEFT 0xBB +#define UAC3_CH_HEADPHONE_RIGHT 0xBC + +/* A.15 AUDIO CLASS-SPECIFIC AC INTERFACE DESCRIPTOR SUBTYPES */ +/* see audio.h for the rest, which is identical to v1 */ +#define UAC3_EXTENDED_TERMINAL 0x04 +#define UAC3_MIXER_UNIT 0x05 +#define UAC3_SELECTOR_UNIT 0x06 +#define UAC3_FEATURE_UNIT 0x07 +#define UAC3_EFFECT_UNIT 0x08 +#define UAC3_PROCESSING_UNIT 0x09 +#define UAC3_EXTENSION_UNIT 0x0a +#define UAC3_CLOCK_SOURCE 0x0b +#define UAC3_CLOCK_SELECTOR 0x0c +#define UAC3_CLOCK_MULTIPLIER 0x0d +#define UAC3_SAMPLE_RATE_CONVERTER 0x0e +#define UAC3_CONNECTORS 0x0f +#define UAC3_POWER_DOMAIN 0x10 + +/* A.22 AUDIO CLASS-SPECIFIC REQUEST CODES */ +/* see audio-v2.h for the rest, which is identical to v2 */ +#define UAC3_CS_REQ_INTEN 0x04 +#define UAC3_CS_REQ_STRING 0x05 +#define UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR 0x06 + +/* A.23.1 AUDIOCONTROL INTERFACE CONTROL SELECTORS */ +#define UAC3_AC_CONTROL_UNDEFINED 0x00 +#define UAC3_AC_ACTIVE_INTERFACE_CONTROL 0x01 +#define UAC3_AC_POWER_DOMAIN_CONTROL 0x02 + +#endif /* __LINUX_USB_AUDIO_V3_H */ diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h index da3315ed1bcd..3a78e7145689 100644 --- a/include/uapi/linux/usb/audio.h +++ b/include/uapi/linux/usb/audio.h @@ -27,6 +27,7 @@ /* bInterfaceProtocol values to denote the version of the standard used */ #define UAC_VERSION_1 0x00 #define UAC_VERSION_2 0x20 +#define UAC_VERSION_3 0x30 /* A.2 Audio Interface Subclass Codes */ #define USB_SUBCLASS_AUDIOCONTROL 0x01 diff --git a/sound/usb/card.c b/sound/usb/card.c index 8018d56cfecc..4a1c6bb3dfa0 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -7,6 +7,7 @@ * Alan Cox (alan@lxorguk.ukuu.org.uk) * Thomas Sailer (sailer@ife.ee.ethz.ch) * + * Audio Class 3.0 support by Ruslan Bilovol * * 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 @@ -44,6 +45,7 @@ #include #include #include +#include #include #include @@ -281,7 +283,8 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) break; } - case UAC_VERSION_2: { + case UAC_VERSION_2: + case UAC_VERSION_3: { struct usb_interface_assoc_descriptor *assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc; @@ -301,7 +304,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } if (!assoc) { - dev_err(&dev->dev, "Audio class v2 interfaces need an interface association\n"); + dev_err(&dev->dev, "Audio class v2/v3 interfaces need an interface association\n"); return -EINVAL; } diff --git a/sound/usb/card.h b/sound/usb/card.h index ed87cc83eb47..1406292d50ec 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -22,7 +22,7 @@ struct audioformat { unsigned char endpoint; /* endpoint */ unsigned char ep_attr; /* endpoint attributes */ unsigned char datainterval; /* log_2 of data packet interval */ - unsigned char protocol; /* UAC_VERSION_1/2 */ + unsigned char protocol; /* UAC_VERSION_1/2/3 */ unsigned int maxpacksize; /* max. packet size */ unsigned int rates; /* rate bitmasks */ unsigned int rate_min, rate_max; /* min/max rates */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index eb3396ffba4c..25de7fe285d9 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -50,6 +51,22 @@ static struct uac_clock_source_descriptor * return NULL; } +static struct uac3_clock_source_descriptor * + snd_usb_find_clock_source_v3(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac3_clock_source_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC3_CLOCK_SOURCE))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + static struct uac_clock_selector_descriptor * snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface, int clock_id) @@ -69,6 +86,22 @@ static struct uac_clock_selector_descriptor * return NULL; } +static struct uac3_clock_selector_descriptor * + snd_usb_find_clock_selector_v3(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac3_clock_selector_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC3_CLOCK_SELECTOR))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + static struct uac_clock_multiplier_descriptor * snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface, int clock_id) @@ -85,6 +118,22 @@ static struct uac_clock_multiplier_descriptor * return NULL; } +static struct uac3_clock_multiplier_descriptor * + snd_usb_find_clock_multiplier_v3(struct usb_host_interface *ctrl_iface, + int clock_id) +{ + struct uac3_clock_multiplier_descriptor *cs = NULL; + + while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra, + ctrl_iface->extralen, + cs, UAC3_CLOCK_MULTIPLIER))) { + if (cs->bClockID == clock_id) + return cs; + } + + return NULL; +} + static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id) { unsigned char buf; @@ -138,19 +187,33 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } -static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) +static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, + int protocol, + int source_id) { int err; unsigned char data; struct usb_device *dev = chip->dev; - struct uac_clock_source_descriptor *cs_desc = - snd_usb_find_clock_source(chip->ctrl_intf, source_id); + u32 bmControls; - if (!cs_desc) - return 0; + if (protocol == UAC_VERSION_3) { + struct uac3_clock_source_descriptor *cs_desc = + snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); + + if (!cs_desc) + return 0; + bmControls = le32_to_cpu(cs_desc->bmControls); + } else { /* UAC_VERSION_1/2 */ + struct uac_clock_source_descriptor *cs_desc = + snd_usb_find_clock_source(chip->ctrl_intf, source_id); + + if (!cs_desc) + return 0; + bmControls = cs_desc->bmControls; + } /* If a clock source can't tell us whether it's valid, we assume it is */ - if (!uac2_control_is_readable(cs_desc->bmControls, + if (!uac_v2v3_control_is_readable(bmControls, UAC2_CS_CONTROL_CLOCK_VALID - 1)) return 1; @@ -170,9 +233,8 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id) return !!data; } -static int __uac_clock_find_source(struct snd_usb_audio *chip, - int entity_id, unsigned long *visited, - bool validate) +static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, + unsigned long *visited, bool validate) { struct uac_clock_source_descriptor *source; struct uac_clock_selector_descriptor *selector; @@ -191,7 +253,8 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, entity_id)) { + if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_2, + entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", entity_id); @@ -260,6 +323,97 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } +static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, + unsigned long *visited, bool validate) +{ + struct uac3_clock_source_descriptor *source; + struct uac3_clock_selector_descriptor *selector; + struct uac3_clock_multiplier_descriptor *multiplier; + + entity_id &= 0xff; + + if (test_and_set_bit(entity_id, visited)) { + usb_audio_warn(chip, + "%s(): recursive clock topology detected, id %d.\n", + __func__, entity_id); + return -EINVAL; + } + + /* first, see if the ID we're looking for is a clock source already */ + source = snd_usb_find_clock_source_v3(chip->ctrl_intf, entity_id); + if (source) { + entity_id = source->bClockID; + if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_3, + entity_id)) { + usb_audio_err(chip, + "clock source %d is not valid, cannot use\n", + entity_id); + return -ENXIO; + } + return entity_id; + } + + selector = snd_usb_find_clock_selector_v3(chip->ctrl_intf, entity_id); + if (selector) { + int ret, i, cur; + + /* the entity ID we are looking for is a selector. + * find out what it currently selects */ + ret = uac_clock_selector_get_val(chip, selector->bClockID); + if (ret < 0) + return ret; + + /* Selector values are one-based */ + + if (ret > selector->bNrInPins || ret < 1) { + usb_audio_err(chip, + "%s(): selector reported illegal value, id %d, ret %d\n", + __func__, selector->bClockID, ret); + + return -EINVAL; + } + + cur = ret; + ret = __uac3_clock_find_source(chip, selector->baCSourceID[ret - 1], + visited, validate); + if (!validate || ret > 0 || !chip->autoclock) + return ret; + + /* The current clock source is invalid, try others. */ + for (i = 1; i <= selector->bNrInPins; i++) { + int err; + + if (i == cur) + continue; + + ret = __uac3_clock_find_source(chip, selector->baCSourceID[i - 1], + visited, true); + if (ret < 0) + continue; + + err = uac_clock_selector_set_val(chip, entity_id, i); + if (err < 0) + continue; + + usb_audio_info(chip, + "found and selected valid clock source %d\n", + ret); + return ret; + } + + return -ENXIO; + } + + /* FIXME: multipliers only act as pass-thru element for now */ + multiplier = snd_usb_find_clock_multiplier_v3(chip->ctrl_intf, + entity_id); + if (multiplier) + return __uac3_clock_find_source(chip, multiplier->bCSourceID, + visited, validate); + + return -EINVAL; +} + /* * For all kinds of sample rate settings and other device queries, * the clock source (end-leaf) must be used. However, clock selectors, @@ -271,12 +425,22 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, * * Returns the clock source UnitID (>=0) on success, or an error. */ -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id, - bool validate) +int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, + int entity_id, bool validate) { DECLARE_BITMAP(visited, 256); memset(visited, 0, sizeof(visited)); - return __uac_clock_find_source(chip, entity_id, visited, validate); + + switch (protocol) { + case UAC_VERSION_2: + return __uac_clock_find_source(chip, entity_id, visited, + validate); + case UAC_VERSION_3: + return __uac3_clock_find_source(chip, entity_id, visited, + validate); + default: + return -EINVAL; + } } static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, @@ -335,7 +499,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, return 0; } -static int get_sample_rate_v2(struct snd_usb_audio *chip, int iface, +static int get_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, int altsetting, int clock) { struct usb_device *dev = chip->dev; @@ -348,7 +512,7 @@ static int get_sample_rate_v2(struct snd_usb_audio *chip, int iface, snd_usb_ctrl_intf(chip) | (clock << 8), &data, sizeof(data)); if (err < 0) { - dev_warn(&dev->dev, "%d:%d: cannot get freq (v2): err %d\n", + dev_warn(&dev->dev, "%d:%d: cannot get freq (v2/v3): err %d\n", iface, altsetting, err); return 0; } @@ -356,7 +520,7 @@ static int get_sample_rate_v2(struct snd_usb_audio *chip, int iface, return le32_to_cpu(data); } -static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, +static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate) { @@ -365,18 +529,30 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, int err, cur_rate, prev_rate; int clock; bool writeable; - struct uac_clock_source_descriptor *cs_desc; + u32 bmControls; - clock = snd_usb_clock_find_source(chip, fmt->clock, true); + clock = snd_usb_clock_find_source(chip, fmt->protocol, + fmt->clock, true); if (clock < 0) return clock; - prev_rate = get_sample_rate_v2(chip, iface, fmt->altsetting, clock); + prev_rate = get_sample_rate_v2v3(chip, iface, fmt->altsetting, clock); if (prev_rate == rate) return 0; - cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); - writeable = uac2_control_is_writeable(cs_desc->bmControls, UAC2_CS_CONTROL_SAM_FREQ - 1); + if (fmt->protocol == UAC_VERSION_3) { + struct uac3_clock_source_descriptor *cs_desc; + + cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, clock); + bmControls = le32_to_cpu(cs_desc->bmControls); + } else { + struct uac_clock_source_descriptor *cs_desc; + + cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); + bmControls = cs_desc->bmControls; + } + + writeable = uac_v2v3_control_is_writeable(bmControls, UAC2_CS_CONTROL_SAM_FREQ - 1); if (writeable) { data = cpu_to_le32(rate); err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, @@ -386,12 +562,13 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, &data, sizeof(data)); if (err < 0) { usb_audio_err(chip, - "%d:%d: cannot set freq %d (v2): err %d\n", + "%d:%d: cannot set freq %d (v2/v3): err %d\n", iface, fmt->altsetting, rate, err); return err; } - cur_rate = get_sample_rate_v2(chip, iface, fmt->altsetting, clock); + cur_rate = get_sample_rate_v2v3(chip, iface, + fmt->altsetting, clock); } else { cur_rate = prev_rate; } @@ -430,7 +607,8 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, return set_sample_rate_v1(chip, iface, alts, fmt, rate); case UAC_VERSION_2: - return set_sample_rate_v2(chip, iface, alts, fmt, rate); + case UAC_VERSION_3: + return set_sample_rate_v2v3(chip, iface, alts, fmt, rate); } } diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 87557cae1a0b..076e31b79ee0 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -6,7 +6,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate); -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int entity_id, - bool validate); +int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, + int entity_id, bool validate); #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 2c44386e5569..edbe67eeddfa 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include @@ -39,11 +40,11 @@ * @dev: usb device * @fp: audioformat record * @format: the format tag (wFormatTag) - * @fmt: the format type descriptor + * @fmt: the format type descriptor (v1/v2) or AudioStreaming descriptor (v3) */ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - unsigned int format, void *_fmt) + u64 format, void *_fmt) { int sample_width, sample_bytes; u64 pcm_formats = 0; @@ -69,6 +70,18 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format <<= 1; break; } + case UAC_VERSION_3: { + struct uac3_as_header_descriptor *as = _fmt; + + sample_width = as->bBitResolution; + sample_bytes = as->bSubslotSize; + + if (format & UAC3_FORMAT_TYPE_I_RAW_DATA) + pcm_formats |= SNDRV_PCM_FMTBIT_SPECIAL; + + format <<= 1; + break; + } } if ((pcm_formats == 0) && @@ -137,7 +150,7 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, } if (format & ~0x3f) { usb_audio_info(chip, - "%u:%d : unsupported format bits %#x\n", + "%u:%d : unsupported format bits %#llx\n", fp->iface, fp->altsetting, format); } @@ -281,15 +294,16 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, /* * parse the format descriptor and stores the possible sample rates - * on the audioformat table (audio class v2). + * on the audioformat table (audio class v2 and v3). */ -static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, +static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, struct audioformat *fp) { struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0; - int clock = snd_usb_clock_find_source(chip, fp->clock, false); + int clock = snd_usb_clock_find_source(chip, fp->protocol, + fp->clock, false); if (clock < 0) { dev_err(&dev->dev, @@ -368,13 +382,30 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, - struct audioformat *fp, unsigned int format, - struct uac_format_type_i_continuous_descriptor *fmt) + struct audioformat *fp, u64 format, + void *_fmt) { snd_pcm_format_t pcm_format; + unsigned int fmt_type; int ret; - if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { + switch (fp->protocol) { + default: + case UAC_VERSION_1: + case UAC_VERSION_2: { + struct uac_format_type_i_continuous_descriptor *fmt = _fmt; + + fmt_type = fmt->bFormatType; + break; + } + case UAC_VERSION_3: { + /* fp->fmt_type is already set in this case */ + fmt_type = fp->fmt_type; + break; + } + } + + if (fmt_type == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -393,7 +424,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, } fp->formats = pcm_format_to_bits(pcm_format); } else { - fp->formats = parse_audio_format_i_type(chip, fp, format, fmt); + fp->formats = parse_audio_format_i_type(chip, fp, format, _fmt); if (!fp->formats) return -EINVAL; } @@ -405,15 +436,20 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, */ switch (fp->protocol) { default: - case UAC_VERSION_1: + case UAC_VERSION_1: { + struct uac_format_type_i_continuous_descriptor *fmt = _fmt; + fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); break; + } case UAC_VERSION_2: + case UAC_VERSION_3: { /* fp->channels is already set in this case */ - ret = parse_audio_format_rates_v2(chip, fp); + ret = parse_audio_format_rates_v2v3(chip, fp); break; } + } if (fp->channels < 1) { usb_audio_err(chip, @@ -430,7 +466,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, */ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *_fmt) + u64 format, void *_fmt) { int brate, framesize, ret; @@ -445,7 +481,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, break; default: usb_audio_info(chip, - "%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", + "%u:%d : unknown format tag %#llx is detected. processed as MPEG.\n", fp->iface, fp->altsetting, format); fp->formats = SNDRV_PCM_FMTBIT_MPEG; break; @@ -470,7 +506,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, framesize = le16_to_cpu(fmt->wSamplesPerFrame); usb_audio_info(chip, "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); fp->frame_size = framesize; - ret = parse_audio_format_rates_v2(chip, fp); + ret = parse_audio_format_rates_v2v3(chip, fp); break; } } @@ -479,7 +515,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, } int snd_usb_parse_audio_format(struct snd_usb_audio *chip, - struct audioformat *fp, unsigned int format, + struct audioformat *fp, u64 format, struct uac_format_type_i_continuous_descriptor *fmt, int stream) { @@ -520,3 +556,26 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, return 0; } +int snd_usb_parse_audio_format_v3(struct snd_usb_audio *chip, + struct audioformat *fp, + struct uac3_as_header_descriptor *as, + int stream) +{ + u64 format = le64_to_cpu(as->bmFormats); + int err; + + /* + * Type I format bits are D0..D6 + * This test works because type IV is not supported + */ + if (format & 0x7f) + fp->fmt_type = UAC_FORMAT_TYPE_I; + else + fp->fmt_type = UAC_FORMAT_TYPE_III; + + err = parse_audio_format_i(chip, fp, format, as); + if (err < 0) + return err; + + return 0; +} diff --git a/sound/usb/format.h b/sound/usb/format.h index 8c3ff9ce0824..e70171892f32 100644 --- a/sound/usb/format.h +++ b/sound/usb/format.h @@ -3,8 +3,12 @@ #define __USBAUDIO_FORMAT_H int snd_usb_parse_audio_format(struct snd_usb_audio *chip, - struct audioformat *fp, unsigned int format, + struct audioformat *fp, u64 format, struct uac_format_type_i_continuous_descriptor *fmt, int stream); +int snd_usb_parse_audio_format_v3(struct snd_usb_audio *chip, + struct audioformat *fp, + struct uac3_as_header_descriptor *as, + int stream); #endif /* __USBAUDIO_FORMAT_H */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 06b22624ab7a..1c02f7373eda 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -51,6 +51,7 @@ #include #include #include +#include #include #include @@ -189,7 +190,7 @@ static void *find_audio_control_unit(struct mixer_build *state, USB_DT_CS_INTERFACE)) != NULL) { if (hdr->bLength >= 4 && hdr->bDescriptorSubtype >= UAC_INPUT_TERMINAL && - hdr->bDescriptorSubtype <= UAC2_SAMPLE_RATE_CONVERTER && + hdr->bDescriptorSubtype <= UAC3_SAMPLE_RATE_CONVERTER && hdr->bUnitID == unit) return hdr; } @@ -468,9 +469,10 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, validx += cval->idx_off; + if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; - } else { /* UAC_VERSION_2 */ + } else { /* UAC_VERSION_2/3 */ val_len = uac2_ctl_value_size(cval->val_type); /* FIXME */ @@ -723,6 +725,7 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { + int protocol = state->mixer->protocol; int err; void *p1; @@ -730,16 +733,104 @@ static int check_input_term(struct mixer_build *state, int id, while ((p1 = find_audio_control_unit(state, id)) != NULL) { unsigned char *hdr = p1; term->id = id; - switch (hdr[2]) { - case UAC_INPUT_TERMINAL: - if (state->mixer->protocol == UAC_VERSION_1) { - struct uac_input_terminal_descriptor *d = p1; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le16_to_cpu(d->wChannelConfig); - term->name = d->iTerminal; - } else { /* UAC_VERSION_2 */ - struct uac2_input_terminal_descriptor *d = p1; + + if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { + switch (hdr[2]) { + case UAC_INPUT_TERMINAL: + if (protocol == UAC_VERSION_1) { + struct uac_input_terminal_descriptor *d = p1; + + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le16_to_cpu(d->wChannelConfig); + term->name = d->iTerminal; + } else { /* UAC_VERSION_2 */ + struct uac2_input_terminal_descriptor *d = p1; + + /* call recursively to verify that the + * referenced clock entity is valid */ + err = check_input_term(state, d->bCSourceID, term); + if (err < 0) + return err; + + /* save input term properties after recursion, + * to ensure they are not overriden by the + * recursion calls */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; + } + return 0; + case UAC_FEATURE_UNIT: { + /* the header is the same for v1 and v2 */ + struct uac_feature_unit_descriptor *d = p1; + + id = d->bSourceID; + break; /* continue to parse */ + } + case UAC_MIXER_UNIT: { + struct uac_mixer_unit_descriptor *d = p1; + + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->channels = uac_mixer_unit_bNrChannels(d); + term->chconfig = uac_mixer_unit_wChannelConfig(d, protocol); + term->name = uac_mixer_unit_iMixer(d); + return 0; + } + case UAC_SELECTOR_UNIT: + case UAC2_CLOCK_SELECTOR: { + struct uac_selector_unit_descriptor *d = p1; + /* call recursively to retrieve the channel info */ + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->id = id; + term->name = uac_selector_unit_iSelector(d); + return 0; + } + case UAC1_PROCESSING_UNIT: + case UAC1_EXTENSION_UNIT: + /* UAC2_PROCESSING_UNIT_V2 */ + /* UAC2_EFFECT_UNIT */ + case UAC2_EXTENSION_UNIT_V2: { + struct uac_processing_unit_descriptor *d = p1; + + if (protocol == UAC_VERSION_2 && + hdr[2] == UAC2_EFFECT_UNIT) { + /* UAC2/UAC1 unit IDs overlap here in an + * uncompatible way. Ignore this unit for now. + */ + return 0; + } + + if (d->bNrInPins) { + id = d->baSourceID[0]; + break; /* continue to parse */ + } + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->channels = uac_processing_unit_bNrChannels(d); + term->chconfig = uac_processing_unit_wChannelConfig(d, protocol); + term->name = uac_processing_unit_iProcessing(d, protocol); + return 0; + } + case UAC2_CLOCK_SOURCE: { + struct uac_clock_source_descriptor *d = p1; + + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->id = id; + term->name = d->iClockSource; + return 0; + } + default: + return -ENODEV; + } + } else { /* UAC_VERSION_3 */ + switch (hdr[2]) { + case UAC_INPUT_TERMINAL: { + struct uac3_input_terminal_descriptor *d = p1; /* call recursively to verify that the * referenced clock entity is valid */ @@ -752,71 +843,31 @@ static int check_input_term(struct mixer_build *state, int id, * recursion calls */ term->id = id; term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; - } - return 0; - case UAC_FEATURE_UNIT: { - /* the header is the same for v1 and v2 */ - struct uac_feature_unit_descriptor *d = p1; - id = d->bSourceID; - break; /* continue to parse */ - } - case UAC_MIXER_UNIT: { - struct uac_mixer_unit_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ - term->channels = uac_mixer_unit_bNrChannels(d); - term->chconfig = uac_mixer_unit_wChannelConfig(d, state->mixer->protocol); - term->name = uac_mixer_unit_iMixer(d); - return 0; - } - case UAC_SELECTOR_UNIT: - case UAC2_CLOCK_SELECTOR: { - struct uac_selector_unit_descriptor *d = p1; - /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); - if (err < 0) - return err; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ - term->id = id; - term->name = uac_selector_unit_iSelector(d); - return 0; - } - case UAC1_PROCESSING_UNIT: - case UAC1_EXTENSION_UNIT: - /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ - case UAC2_EXTENSION_UNIT_V2: { - struct uac_processing_unit_descriptor *d = p1; - if (state->mixer->protocol == UAC_VERSION_2 && - hdr[2] == UAC2_EFFECT_UNIT) { - /* UAC2/UAC1 unit IDs overlap here in an - * uncompatible way. Ignore this unit for now. - */ + /* REVISIT: UAC3 IT doesn't have channels/cfg */ + term->channels = 0; + term->chconfig = 0; + + term->name = le16_to_cpu(d->wTerminalDescrStr); return 0; } + case UAC3_FEATURE_UNIT: { + struct uac3_feature_unit_descriptor *d = p1; - if (d->bNrInPins) { - id = d->baSourceID[0]; + id = d->bSourceID; break; /* continue to parse */ } - term->type = d->bDescriptorSubtype << 16; /* virtual type */ - term->channels = uac_processing_unit_bNrChannels(d); - term->chconfig = uac_processing_unit_wChannelConfig(d, state->mixer->protocol); - term->name = uac_processing_unit_iProcessing(d, state->mixer->protocol); - return 0; - } - case UAC2_CLOCK_SOURCE: { - struct uac_clock_source_descriptor *d = p1; - term->type = d->bDescriptorSubtype << 16; /* virtual type */ - term->id = id; - term->name = d->iClockSource; - return 0; - } - default: - return -ENODEV; + case UAC3_CLOCK_SOURCE: { + struct uac3_clock_source_descriptor *d = p1; + + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->id = id; + term->name = le16_to_cpu(d->wClockSourceStr); + return 0; + } + default: + return -ENODEV; + } } } return -ENODEV; @@ -1423,7 +1474,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, * The only property of this unit we are interested in is the * clock source validity. If that isn't readable, just bail out. */ - if (!uac2_control_is_readable(hdr->bmControls, + if (!uac_v2v3_control_is_readable(hdr->bmControls, ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) return 0; @@ -1439,7 +1490,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, cval->val_type = USB_MIXER_BOOLEAN; cval->control = UAC2_CS_CONTROL_CLOCK_VALID; - if (uac2_control_is_writeable(hdr->bmControls, + if (uac_v2v3_control_is_writeable(hdr->bmControls, ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); else { @@ -1502,7 +1553,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unitid); return -EINVAL; } - } else { + } else if (state->mixer->protocol == UAC_VERSION_2) { struct uac2_feature_unit_descriptor *ftr = _ftr; if (hdr->bLength < 6) { usb_audio_err(state->chip, @@ -1519,6 +1570,24 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unitid); return -EINVAL; } + } else { /* UAC_VERSION_3 */ + struct uac3_feature_unit_descriptor *ftr = _ftr; + + if (hdr->bLength < 7) { + usb_audio_err(state->chip, + "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } + csize = 4; + channels = (ftr->bLength - 7) / 4 - 1; + bmaControls = ftr->bmaControls; + if (hdr->bLength < 7 + csize) { + usb_audio_err(state->chip, + "unit %u: invalid UAC3_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } } /* parse the source unit */ @@ -1577,7 +1646,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0); } - } else { /* UAC_VERSION_2 */ + } else { /* UAC_VERSION_2/3 */ for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) { unsigned int ch_bits = 0; unsigned int ch_read_only = 0; @@ -1587,9 +1656,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); - if (uac2_control_is_readable(mask, i)) { + if (uac_v2v3_control_is_readable(mask, i)) { ch_bits |= (1 << j); - if (!uac2_control_is_writeable(mask, i)) + if (!uac_v2v3_control_is_writeable(mask, i)) ch_read_only |= (1 << j); } } @@ -1610,9 +1679,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, if (ch_bits & 1) build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only); - if (uac2_control_is_readable(master_bits, i)) + if (uac_v2v3_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, - !uac2_control_is_writeable(master_bits, i)); + !uac_v2v3_control_is_writeable(master_bits, i)); } } @@ -2220,6 +2289,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, static int parse_audio_unit(struct mixer_build *state, int unitid) { unsigned char *p1; + int protocol = state->mixer->protocol; if (test_and_set_bit(unitid, state->unitbitmap)) return 0; /* the unit already visited */ @@ -2230,36 +2300,61 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return -EINVAL; } - switch (p1[2]) { - case UAC_INPUT_TERMINAL: - return 0; /* NOP */ - case UAC_MIXER_UNIT: - return parse_audio_mixer_unit(state, unitid, p1); - case UAC2_CLOCK_SOURCE: - return parse_clock_source_unit(state, unitid, p1); - case UAC_SELECTOR_UNIT: - case UAC2_CLOCK_SELECTOR: - return parse_audio_selector_unit(state, unitid, p1); - case UAC_FEATURE_UNIT: - return parse_audio_feature_unit(state, unitid, p1); - case UAC1_PROCESSING_UNIT: - /* UAC2_EFFECT_UNIT has the same value */ - if (state->mixer->protocol == UAC_VERSION_1) - return parse_audio_processing_unit(state, unitid, p1); - else - return 0; /* FIXME - effect units not implemented yet */ - case UAC1_EXTENSION_UNIT: - /* UAC2_PROCESSING_UNIT_V2 has the same value */ - if (state->mixer->protocol == UAC_VERSION_1) + if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { + switch (p1[2]) { + case UAC_INPUT_TERMINAL: + return 0; /* NOP */ + case UAC_MIXER_UNIT: + return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); + case UAC_SELECTOR_UNIT: + case UAC2_CLOCK_SELECTOR: + return parse_audio_selector_unit(state, unitid, p1); + case UAC_FEATURE_UNIT: + return parse_audio_feature_unit(state, unitid, p1); + case UAC1_PROCESSING_UNIT: + /* UAC2_EFFECT_UNIT has the same value */ + if (protocol == UAC_VERSION_1) + return parse_audio_processing_unit(state, unitid, p1); + else + return 0; /* FIXME - effect units not implemented yet */ + case UAC1_EXTENSION_UNIT: + /* UAC2_PROCESSING_UNIT_V2 has the same value */ + if (protocol == UAC_VERSION_1) + return parse_audio_extension_unit(state, unitid, p1); + else /* UAC_VERSION_2 */ + return parse_audio_processing_unit(state, unitid, p1); + case UAC2_EXTENSION_UNIT_V2: return parse_audio_extension_unit(state, unitid, p1); - else /* UAC_VERSION_2 */ + default: + usb_audio_err(state->chip, + "unit %u: unexpected type 0x%02x\n", unitid, p1[2]); + return -EINVAL; + } + } else { /* UAC_VERSION_3 */ + switch (p1[2]) { + case UAC_INPUT_TERMINAL: + return 0; /* NOP */ + case UAC3_MIXER_UNIT: + return parse_audio_mixer_unit(state, unitid, p1); + case UAC3_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); + case UAC3_CLOCK_SELECTOR: + return parse_audio_selector_unit(state, unitid, p1); + case UAC3_FEATURE_UNIT: + return parse_audio_feature_unit(state, unitid, p1); + case UAC3_EFFECT_UNIT: + return 0; /* FIXME - effect units not implemented yet */ + case UAC3_PROCESSING_UNIT: return parse_audio_processing_unit(state, unitid, p1); - case UAC2_EXTENSION_UNIT_V2: - return parse_audio_extension_unit(state, unitid, p1); - default: - usb_audio_err(state->chip, - "unit %u: unexpected type 0x%02x\n", unitid, p1[2]); - return -EINVAL; + case UAC3_EXTENSION_UNIT: + return parse_audio_extension_unit(state, unitid, p1); + default: + usb_audio_err(state->chip, + "unit %u: unexpected type 0x%02x\n", unitid, p1[2]); + return -EINVAL; + } } } @@ -2330,7 +2425,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) err = parse_audio_unit(&state, desc->bSourceID); if (err < 0 && err != -EINVAL) return err; - } else { /* UAC_VERSION_2 */ + } else if (mixer->protocol == UAC_VERSION_2) { struct uac2_output_terminal_descriptor *desc = p; if (desc->bLength < sizeof(*desc)) @@ -2351,6 +2446,27 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0 && err != -EINVAL) return err; + } else { /* UAC_VERSION_3 */ + struct uac3_output_terminal_descriptor *desc = p; + + if (desc->bLength < sizeof(*desc)) + continue; /* invalid descriptor? */ + /* mark terminal ID as visited */ + set_bit(desc->bTerminalID, state.unitbitmap); + state.oterm.id = desc->bTerminalID; + state.oterm.type = le16_to_cpu(desc->wTerminalType); + state.oterm.name = le16_to_cpu(desc->wTerminalDescrStr); + err = parse_audio_unit(&state, desc->bSourceID); + if (err < 0 && err != -EINVAL) + return err; + + /* + * For UAC3, use the same approach to also add the + * clock selectors + */ + err = parse_audio_unit(&state, desc->bCSourceID); + if (err < 0 && err != -EINVAL) + return err; } } @@ -2597,6 +2713,9 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, case UAC_VERSION_2: mixer->protocol = UAC_VERSION_2; break; + case UAC_VERSION_3: + mixer->protocol = UAC_VERSION_3; + break; } if ((err = snd_usb_mixer_controls(mixer)) < 0 || diff --git a/sound/usb/stream.c b/sound/usb/stream.c index dbbe854745ab..6a8f5843334e 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include @@ -311,6 +312,153 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, return chmap; } +/* UAC3 device stores channels information in Cluster Descriptors */ +static struct +snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor + *cluster) +{ + unsigned int channels = cluster->bNrChannels; + struct snd_pcm_chmap_elem *chmap; + void *p = cluster; + int len, c; + + if (channels > ARRAY_SIZE(chmap->map)) + return NULL; + + chmap = kzalloc(sizeof(*chmap), GFP_KERNEL); + if (!chmap) + return NULL; + + len = le16_to_cpu(cluster->wLength); + c = 0; + p += sizeof(struct uac3_cluster_header_descriptor); + + while (((p - (void *)cluster) < len) && (c < channels)) { + struct uac3_cluster_segment_descriptor *cs_desc = p; + u16 cs_len; + u8 cs_type; + + cs_len = le16_to_cpu(cs_desc->wLength); + cs_type = cs_desc->bSegmentType; + + if (cs_type == UAC3_CHANNEL_INFORMATION) { + struct uac3_cluster_information_segment_descriptor *is = p; + unsigned char map; + + /* + * TODO: this conversion is not complete, update it + * after adding UAC3 values to asound.h + */ + switch (is->bChPurpose) { + case UAC3_CH_MONO: + map = SNDRV_CHMAP_MONO; + break; + case UAC3_CH_LEFT: + case UAC3_CH_FRONT_LEFT: + case UAC3_CH_HEADPHONE_LEFT: + map = SNDRV_CHMAP_FL; + break; + case UAC3_CH_RIGHT: + case UAC3_CH_FRONT_RIGHT: + case UAC3_CH_HEADPHONE_RIGHT: + map = SNDRV_CHMAP_FR; + break; + case UAC3_CH_FRONT_CENTER: + map = SNDRV_CHMAP_FC; + break; + case UAC3_CH_FRONT_LEFT_OF_CENTER: + map = SNDRV_CHMAP_FLC; + break; + case UAC3_CH_FRONT_RIGHT_OF_CENTER: + map = SNDRV_CHMAP_FRC; + break; + case UAC3_CH_SIDE_LEFT: + map = SNDRV_CHMAP_SL; + break; + case UAC3_CH_SIDE_RIGHT: + map = SNDRV_CHMAP_SR; + break; + case UAC3_CH_BACK_LEFT: + map = SNDRV_CHMAP_RL; + break; + case UAC3_CH_BACK_RIGHT: + map = SNDRV_CHMAP_RR; + break; + case UAC3_CH_BACK_CENTER: + map = SNDRV_CHMAP_RC; + break; + case UAC3_CH_BACK_LEFT_OF_CENTER: + map = SNDRV_CHMAP_RLC; + break; + case UAC3_CH_BACK_RIGHT_OF_CENTER: + map = SNDRV_CHMAP_RRC; + break; + case UAC3_CH_TOP_CENTER: + map = SNDRV_CHMAP_TC; + break; + case UAC3_CH_TOP_FRONT_LEFT: + map = SNDRV_CHMAP_TFL; + break; + case UAC3_CH_TOP_FRONT_RIGHT: + map = SNDRV_CHMAP_TFR; + break; + case UAC3_CH_TOP_FRONT_CENTER: + map = SNDRV_CHMAP_TFC; + break; + case UAC3_CH_TOP_FRONT_LOC: + map = SNDRV_CHMAP_TFLC; + break; + case UAC3_CH_TOP_FRONT_ROC: + map = SNDRV_CHMAP_TFRC; + break; + case UAC3_CH_TOP_SIDE_LEFT: + map = SNDRV_CHMAP_TSL; + break; + case UAC3_CH_TOP_SIDE_RIGHT: + map = SNDRV_CHMAP_TSR; + break; + case UAC3_CH_TOP_BACK_LEFT: + map = SNDRV_CHMAP_TRL; + break; + case UAC3_CH_TOP_BACK_RIGHT: + map = SNDRV_CHMAP_TRR; + break; + case UAC3_CH_TOP_BACK_CENTER: + map = SNDRV_CHMAP_TRC; + break; + case UAC3_CH_BOTTOM_CENTER: + map = SNDRV_CHMAP_BC; + break; + case UAC3_CH_LOW_FREQUENCY_EFFECTS: + map = SNDRV_CHMAP_LFE; + break; + case UAC3_CH_LFE_LEFT: + map = SNDRV_CHMAP_LLFE; + break; + case UAC3_CH_LFE_RIGHT: + map = SNDRV_CHMAP_RLFE; + break; + case UAC3_CH_RELATIONSHIP_UNDEFINED: + default: + map = SNDRV_CHMAP_UNKNOWN; + break; + } + chmap->map[c++] = map; + } + p += cs_len; + } + + if (channels < c) + pr_err("%s: channel number mismatch\n", __func__); + + chmap->channels = channels; + + for (; c < channels; c++) + chmap->map[c] = SNDRV_CHMAP_UNKNOWN; + + return chmap; +} + /* * add this endpoint to the chip instance. * if a stream with the same endpoint already exists, append to it. @@ -461,10 +609,11 @@ snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface, return NULL; } -static struct uac2_output_terminal_descriptor * - snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, - int terminal_id) +static void * +snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface, + int terminal_id) { + /* OK to use with both UAC2 and UAC3 */ struct uac2_output_terminal_descriptor *term = NULL; while ((term = snd_usb_find_csint_desc(ctrl_iface->extra, @@ -484,10 +633,12 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; int i, altno, err, stream; - unsigned int format = 0, num_channels = 0; + u64 format = 0; + unsigned int num_channels = 0; struct audioformat *fp = NULL; int num, protocol, clock = 0; - struct uac_format_type_i_continuous_descriptor *fmt; + struct uac_format_type_i_continuous_descriptor *fmt = NULL; + struct snd_pcm_chmap_elem *chmap_v3 = NULL; unsigned int chconfig; dev = chip->dev; @@ -624,38 +775,158 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) iface_no, altno, as->bTerminalLink); continue; } - } - /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); - if (!fmt) { + case UAC_VERSION_3: { + struct uac3_input_terminal_descriptor *input_term; + struct uac3_output_terminal_descriptor *output_term; + struct uac3_as_header_descriptor *as; + struct uac3_cluster_header_descriptor *cluster; + struct uac3_hc_descriptor_header hc_header; + u16 cluster_id, wLength; + + as = snd_usb_find_csint_desc(alts->extra, + alts->extralen, + NULL, UAC_AS_GENERAL); + + if (!as) { + dev_err(&dev->dev, + "%u:%d : UAC_AS_GENERAL descriptor not found\n", + iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + dev_err(&dev->dev, + "%u:%d : invalid UAC_AS_GENERAL desc\n", + iface_no, altno); + continue; + } + + cluster_id = le16_to_cpu(as->wClusterDescrID); + if (!cluster_id) { + dev_err(&dev->dev, + "%u:%d : no cluster descriptor\n", + iface_no, altno); + continue; + } + + /* + * Get number of channels and channel map through + * High Capability Cluster Descriptor + * + * First step: get High Capability header and + * read size of Cluster Descriptor + */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), + UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + cluster_id, + snd_usb_ctrl_intf(chip), + &hc_header, sizeof(hc_header)); + if (err < 0) + return err; + else if (err != sizeof(hc_header)) { + dev_err(&dev->dev, + "%u:%d : can't get High Capability descriptor\n", + iface_no, altno); + return -EIO; + } + + /* + * Second step: allocate needed amount of memory + * and request Cluster Descriptor + */ + wLength = le16_to_cpu(hc_header.wLength); + cluster = kzalloc(wLength, GFP_KERNEL); + if (!cluster) + return -ENOMEM; + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), + UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + cluster_id, + snd_usb_ctrl_intf(chip), + cluster, wLength); + if (err < 0) { + kfree(cluster); + return err; + } else if (err != wLength) { + dev_err(&dev->dev, + "%u:%d : can't get Cluster Descriptor\n", + iface_no, altno); + kfree(cluster); + return -EIO; + } + + num_channels = cluster->bNrChannels; + chmap_v3 = convert_chmap_v3(cluster); + + kfree(cluster); + + format = le64_to_cpu(as->bmFormats); + + /* lookup the terminal associated to this interface + * to extract the clock */ + input_term = snd_usb_find_input_terminal_descriptor( + chip->ctrl_intf, + as->bTerminalLink); + + if (input_term) { + clock = input_term->bCSourceID; + break; + } + + output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + as->bTerminalLink); + if (output_term) { + clock = output_term->bCSourceID; + break; + } + dev_err(&dev->dev, - "%u:%d : no UAC_FORMAT_TYPE desc\n", - iface_no, altno); + "%u:%d : bogus bTerminalLink %d\n", + iface_no, altno, as->bTerminalLink); continue; } - if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || - ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { - dev_err(&dev->dev, - "%u:%d : invalid UAC_FORMAT_TYPE desc\n", - iface_no, altno); - continue; } - /* - * Blue Microphones workaround: The last altsetting is identical - * with the previous one, except for a larger packet size, but - * is actually a mislabeled two-channel setting; ignore it. - */ - if (fmt->bNrChannels == 1 && - fmt->bSubframeSize == 2 && - altno == 2 && num == 3 && - fp && fp->altsetting == 1 && fp->channels == 1 && - fp->formats == SNDRV_PCM_FMTBIT_S16_LE && - protocol == UAC_VERSION_1 && - le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == + if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { + /* get format type */ + fmt = snd_usb_find_csint_desc(alts->extra, + alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) { + dev_err(&dev->dev, + "%u:%d : no UAC_FORMAT_TYPE desc\n", + iface_no, altno); + continue; + } + if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) + || ((protocol == UAC_VERSION_2) && + (fmt->bLength < 6))) { + dev_err(&dev->dev, + "%u:%d : invalid UAC_FORMAT_TYPE desc\n", + iface_no, altno); + continue; + } + + /* + * Blue Microphones workaround: The last altsetting is + * identical with the previous one, except for a larger + * packet size, but is actually a mislabeled two-channel + * setting; ignore it. + */ + if (fmt->bNrChannels == 1 && + fmt->bSubframeSize == 2 && + altno == 2 && num == 3 && + fp && fp->altsetting == 1 && fp->channels == 1 && + fp->formats == SNDRV_PCM_FMTBIT_S16_LE && + protocol == UAC_VERSION_1 && + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == fp->maxpacksize * 2) - continue; + continue; + } fp = kzalloc(sizeof(*fp), GFP_KERNEL); if (!fp) @@ -681,17 +952,39 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) snd_usb_audioformat_attributes_quirk(chip, fp, stream); /* ok, let's parse further... */ - if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream) < 0) { - kfree(fp->rate_table); - kfree(fp); - fp = NULL; - continue; + if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { + if (snd_usb_parse_audio_format(chip, fp, format, + fmt, stream) < 0) { + kfree(fp->rate_table); + kfree(fp); + fp = NULL; + continue; + } + } else { + struct uac3_as_header_descriptor *as; + + as = snd_usb_find_csint_desc(alts->extra, + alts->extralen, + NULL, UAC_AS_GENERAL); + + if (snd_usb_parse_audio_format_v3(chip, fp, as, + stream) < 0) { + kfree(fp->rate_table); + kfree(fp); + fp = NULL; + continue; + } } /* Create chmap */ if (fp->channels != num_channels) chconfig = 0; - fp->chmap = convert_chmap(fp->channels, chconfig, protocol); + + if (protocol == UAC_VERSION_3) + fp->chmap = chmap_v3; + else + fp->chmap = convert_chmap(fp->channels, chconfig, + protocol); dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); err = snd_usb_add_audio_stream(chip, stream, fp); From 21e9b3e931f78497b19b1f8f3d59d19412c1a28f Mon Sep 17 00:00:00 2001 From: Andrew Chant Date: Thu, 22 Mar 2018 14:39:55 -0700 Subject: [PATCH 20/31] ALSA: usb-audio: fix uac control query argument This patch fixes code readability and should have no functional change. Correct uac control query functions to account for the 1-based indexing of USB Audio Class control identifiers. The function parameter, u8 control, should be the constant defined in audio-v2.h to identify the control to be checked for readability or writeability. This patch fixes all callers that had adjusted, and makes explicit the mapping between audio_feature_info[] array index and the associated control identifier. Signed-off-by: Andrew Chant Signed-off-by: Takashi Iwai --- include/linux/usb/audio-v2.h | 4 +-- sound/usb/clock.c | 5 +-- sound/usb/mixer.c | 69 ++++++++++++++++++++++-------------- 3 files changed, 48 insertions(+), 30 deletions(-) diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index 2db83a191e78..aaafecf073ff 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -36,12 +36,12 @@ static inline bool uac_v2v3_control_is_readable(u32 bmControls, u8 control) { - return (bmControls >> (control * 2)) & 0x1; + return (bmControls >> ((control - 1) * 2)) & 0x1; } static inline bool uac_v2v3_control_is_writeable(u32 bmControls, u8 control) { - return (bmControls >> (control * 2)) & 0x2; + return (bmControls >> ((control - 1) * 2)) & 0x2; } /* 4.7.2 Class-Specific AC Interface Descriptor */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 25de7fe285d9..ab39ccb974c6 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -214,7 +214,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, /* If a clock source can't tell us whether it's valid, we assume it is */ if (!uac_v2v3_control_is_readable(bmControls, - UAC2_CS_CONTROL_CLOCK_VALID - 1)) + UAC2_CS_CONTROL_CLOCK_VALID)) return 1; err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, @@ -552,7 +552,8 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, bmControls = cs_desc->bmControls; } - writeable = uac_v2v3_control_is_writeable(bmControls, UAC2_CS_CONTROL_SAM_FREQ - 1); + writeable = uac_v2v3_control_is_writeable(bmControls, + UAC2_CS_CONTROL_SAM_FREQ); if (writeable) { data = cpu_to_le32(rate); err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 1c02f7373eda..3075ac50a391 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -879,26 +879,27 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { + int control; const char *name; int type; /* data type for uac1 */ int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, - { "Volume", USB_MIXER_S16, -1 }, - { "Tone Control - Bass", USB_MIXER_S8, -1 }, - { "Tone Control - Mid", USB_MIXER_S8, -1 }, - { "Tone Control - Treble", USB_MIXER_S8, -1 }, - { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, - { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, - { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, - { "Loudness", USB_MIXER_BOOLEAN, -1 }, + { UAC_FU_MUTE, "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { UAC_FU_VOLUME, "Volume", USB_MIXER_S16, -1 }, + { UAC_FU_BASS, "Tone Control - Bass", USB_MIXER_S8, -1 }, + { UAC_FU_MID, "Tone Control - Mid", USB_MIXER_S8, -1 }, + { UAC_FU_TREBLE, "Tone Control - Treble", USB_MIXER_S8, -1 }, + { UAC_FU_GRAPHIC_EQUALIZER, "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemented yet */ + { UAC_FU_AUTOMATIC_GAIN, "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { UAC_FU_DELAY, "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { UAC_FU_BASS_BOOST, "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { UAC_FU_LOUDNESS, "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16, -1 }, - { "Input Gain Pad Control", USB_MIXER_S16, -1 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, + { UAC2_FU_INPUT_GAIN, "Input Gain Control", USB_MIXER_S16, -1 }, + { UAC2_FU_INPUT_GAIN_PAD, "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { UAC2_FU_PHASE_INVERTER, "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1293,6 +1294,17 @@ static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, strlcpy(kctl->id.name, "Headphone", sizeof(kctl->id.name)); } +static struct usb_feature_control_info *get_feature_control_info(int control) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(audio_feature_info); ++i) { + if (audio_feature_info[i].control == control) + return &audio_feature_info[i]; + } + return NULL; +} + static void build_feature_ctl(struct mixer_build *state, void *raw_desc, unsigned int ctl_mask, int control, struct usb_audio_term *iterm, int unitid, @@ -1308,8 +1320,6 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, const struct usbmix_name_map *map; unsigned int range; - control++; /* change from zero-based to 1-based value */ - if (control == UAC_FU_GRAPHIC_EQUALIZER) { /* FIXME: not supported yet */ return; @@ -1325,7 +1335,10 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - ctl_info = &audio_feature_info[control-1]; + + ctl_info = get_feature_control_info(control); + if (!ctl_info) + return; if (state->mixer->protocol == UAC_VERSION_1) cval->val_type = ctl_info->type; else /* UAC_VERSION_2 */ @@ -1475,7 +1488,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, * clock source validity. If that isn't readable, just bail out. */ if (!uac_v2v3_control_is_readable(hdr->bmControls, - ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + UAC2_CS_CONTROL_CLOCK_VALID)) return 0; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1491,7 +1504,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, cval->control = UAC2_CS_CONTROL_CLOCK_VALID; if (uac_v2v3_control_is_writeable(hdr->bmControls, - ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + UAC2_CS_CONTROL_CLOCK_VALID)) kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); else { cval->master_readonly = 1; @@ -1625,6 +1638,8 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, /* check all control types */ for (i = 0; i < 10; i++) { unsigned int ch_bits = 0; + int control = audio_feature_info[i].control; + for (j = 0; j < channels; j++) { unsigned int mask; @@ -1640,25 +1655,26 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, * (for ease of programming). */ if (ch_bits & 1) - build_feature_ctl(state, _ftr, ch_bits, i, + build_feature_ctl(state, _ftr, ch_bits, control, &iterm, unitid, 0); if (master_bits & (1 << i)) - build_feature_ctl(state, _ftr, 0, i, &iterm, - unitid, 0); + build_feature_ctl(state, _ftr, 0, control, + &iterm, unitid, 0); } } else { /* UAC_VERSION_2/3 */ for (i = 0; i < ARRAY_SIZE(audio_feature_info); i++) { unsigned int ch_bits = 0; unsigned int ch_read_only = 0; + int control = audio_feature_info[i].control; for (j = 0; j < channels; j++) { unsigned int mask; mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); - if (uac_v2v3_control_is_readable(mask, i)) { + if (uac_v2v3_control_is_readable(mask, control)) { ch_bits |= (1 << j); - if (!uac_v2v3_control_is_writeable(mask, i)) + if (!uac_v2v3_control_is_writeable(mask, control)) ch_read_only |= (1 << j); } } @@ -1677,11 +1693,12 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, * (for ease of programming). */ if (ch_bits & 1) - build_feature_ctl(state, _ftr, ch_bits, i, + build_feature_ctl(state, _ftr, ch_bits, control, &iterm, unitid, ch_read_only); - if (uac_v2v3_control_is_readable(master_bits, i)) + if (uac_v2v3_control_is_readable(master_bits, control)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, - !uac_v2v3_control_is_writeable(master_bits, i)); + !uac_v2v3_control_is_writeable(master_bits, + control)); } } From 71426535f49fe6034d0e0db77608b91a0c1a022d Mon Sep 17 00:00:00 2001 From: Nobutaka Okabe Date: Fri, 23 Mar 2018 19:18:22 +0900 Subject: [PATCH 21/31] ALSA: usb-audio: Add native DSD support for Luxman DA-06 Add native DSD support quirk for Luxman DA-06 DAC, by adding the PID/VID 1852:5065. Rename "is_marantz_denon_dac()" function to "is_itf_usb_dsd_2alts_dac()" to cover broader device family sharing the same USB audio implementation(*). For the same reason, rename "is_teac_dsd_dac()" function to "is_itf_usb_dsd_3alts_dac()". (*) These devices have the same USB controller "ITF-USB DSD", supplied by INTERFACE Co., Ltd. "ITF-USB DSD" USB controller has two patterns, Pattern 1. (2 altsets version) - Altset 0: for control - Altset 1: for stream (S32) - Altset 2: for stream (S32, DSD_U32) Pattern 2. (3 altsets version) - Altset 0: for control - Altset 1: for stream (S16) - Altset 2: for stream (S32) - Altset 3: for stream (S32, DSD_U32) "is_itf_usb_dsd_2alts_dac()" returns true, if the DAC has "Pattern 1" USB controller, and "is_itf_usb_dsd_3alts_dac()" returns true, if "Pattern2". Signed-off-by: Nobutaka Okabe Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6aa0ebd4a5a2..dec86df3f47a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1149,24 +1149,27 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) return false; } -/* Marantz/Denon USB DACs need a vendor cmd to switch +/* ITF-USB DSD based DACs need a vendor cmd to switch * between PCM and native DSD mode + * (2 altsets version) */ -static bool is_marantz_denon_dac(unsigned int id) +static bool is_itf_usb_dsd_2alts_dac(unsigned int id) { switch (id) { case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + case USB_ID(0x1852, 0x5065): /* Luxman DA-06 */ return true; } return false; } -/* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch - * between PCM/DOP and native DSD mode +/* ITF-USB DSD based DACs need a vendor cmd to switch + * between PCM and native DSD mode + * (3 altsets version) */ -static bool is_teac_dsd_dac(unsigned int id) +static bool is_itf_usb_dsd_3alts_dac(unsigned int id) { switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ @@ -1183,7 +1186,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct usb_device *dev = subs->dev; int err; - if (is_marantz_denon_dac(subs->stream->chip->usb_id)) { + if (is_itf_usb_dsd_2alts_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1204,7 +1207,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_teac_dsd_dac(subs->stream->chip->usb_id)) { + } else if (is_itf_usb_dsd_3alts_dac(subs->stream->chip->usb_id)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1300,10 +1303,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Marantz/Denon devices with USB DAC functionality need a delay + /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ - if (is_marantz_denon_dac(chip->usb_id) + if (is_itf_usb_dsd_2alts_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); @@ -1390,14 +1393,14 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; } - /* Denon/Marantz devices with USB DAC functionality */ - if (is_marantz_denon_dac(chip->usb_id)) { + /* ITF-USB DSD based DACs (2 altsets version) */ + if (is_itf_usb_dsd_2alts_dac(chip->usb_id)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } - /* TEAC devices with USB DAC functionality */ - if (is_teac_dsd_dac(chip->usb_id)) { + /* ITF-USB DSD based DACs (3 altsets version) */ + if (is_itf_usb_dsd_3alts_dac(chip->usb_id)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } From 74dc71f83e500b2c6bb0d01a947e1b7231456dd8 Mon Sep 17 00:00:00 2001 From: Nobutaka Okabe Date: Fri, 23 Mar 2018 19:20:00 +0900 Subject: [PATCH 22/31] ALSA: usb-audio: FIX native DSD support for TEAC UD-501 DAC There are two versions of TEAC UD-501, the normal version and the vendor updated version(UD-501V2). They have the same VID/PID, but the num of the altsetting is different, UD-501 has 2 altsets for stream, and UD-501V2 has 3. So, add the logic to distinguish them by the Product Name, not by the PID. Signed-off-by: Nobutaka Okabe Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 37 +++++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 10 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index dec86df3f47a..da73405c3115 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1153,14 +1153,23 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) * between PCM and native DSD mode * (2 altsets version) */ -static bool is_itf_usb_dsd_2alts_dac(unsigned int id) +static bool is_itf_usb_dsd_2alts_dac(struct snd_usb_audio *chip) { - switch (id) { + char *product = chip->dev->product; /* DAC product name */ + + switch (chip->usb_id) { case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ case USB_ID(0x1852, 0x5065): /* Luxman DA-06 */ return true; + case USB_ID(0x0644, 0x8043): + /* TEAC UD-501 */ + if (product && strcmp("UD-501", product)) { + return true; + } else { + return false; + } } return false; } @@ -1169,13 +1178,21 @@ static bool is_itf_usb_dsd_2alts_dac(unsigned int id) * between PCM and native DSD mode * (3 altsets version) */ -static bool is_itf_usb_dsd_3alts_dac(unsigned int id) +static bool is_itf_usb_dsd_3alts_dac(struct snd_usb_audio *chip) { - switch (id) { - case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ + char *product = chip->dev->product; /* DAC product name */ + + switch (chip->usb_id) { case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ case USB_ID(0x0644, 0x804a): /* TEAC UD-301 */ return true; + case USB_ID(0x0644, 0x8043): + /* TEAC UD-501V2/UD-503/NT-503 */ + if (product && !strcmp("UD-501", product)) { + return true; + } else { + return false; + } } return false; } @@ -1186,7 +1203,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct usb_device *dev = subs->dev; int err; - if (is_itf_usb_dsd_2alts_dac(subs->stream->chip->usb_id)) { + if (is_itf_usb_dsd_2alts_dac(subs->stream->chip)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1207,7 +1224,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_itf_usb_dsd_3alts_dac(subs->stream->chip->usb_id)) { + } else if (is_itf_usb_dsd_3alts_dac(subs->stream->chip)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1306,7 +1323,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ - if (is_itf_usb_dsd_2alts_dac(chip->usb_id) + if (is_itf_usb_dsd_2alts_dac(chip) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); @@ -1394,13 +1411,13 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* ITF-USB DSD based DACs (2 altsets version) */ - if (is_itf_usb_dsd_2alts_dac(chip->usb_id)) { + if (is_itf_usb_dsd_2alts_dac(chip)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } /* ITF-USB DSD based DACs (3 altsets version) */ - if (is_itf_usb_dsd_3alts_dac(chip->usb_id)) { + if (is_itf_usb_dsd_3alts_dac(chip)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } From f3b906d720e429a3acd95f8cbd4fda366fb1659d Mon Sep 17 00:00:00 2001 From: Nobutaka Okabe Date: Fri, 23 Mar 2018 19:21:13 +0900 Subject: [PATCH 23/31] ALSA: usb-audio: Integrate native DSD support for ITF-USB based DACs. Integrate the native DSD support quirk codes of "ITF-USB DSD" based DACs. Now, "is_itf_usb_dsd_2alts_dac()" and "is_itf_usb_dsd_3alts_dac()" is integrated into one function "is_itf_usb_dsd_dac()". So, remove the logic to distinguish UD-501 and UD-501V2 by the "Product Name". The integration is possible by changing the following two functions. - snd_usb_select_mode_quirk(): Change the determination condition of the DSD mode switch command, from the altset number being used, to the audio format being played. Actually, this operation is same as playback using ASIO driver in Windows environment. - snd_usb_interface_dsd_format_quirk(): To which altset supports native DSD is determined by the number of altsets. Previously, it's a constant "2" or "3". Signed-off-by: Nobutaka Okabe Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 84 +++++++++++++--------------------------------- 1 file changed, 23 insertions(+), 61 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index da73405c3115..acbeb52f6fd6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1151,48 +1151,18 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) /* ITF-USB DSD based DACs need a vendor cmd to switch * between PCM and native DSD mode - * (2 altsets version) */ -static bool is_itf_usb_dsd_2alts_dac(struct snd_usb_audio *chip) +static bool is_itf_usb_dsd_dac(unsigned int id) { - char *product = chip->dev->product; /* DAC product name */ - - switch (chip->usb_id) { + switch (id) { case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ case USB_ID(0x1852, 0x5065): /* Luxman DA-06 */ - return true; - case USB_ID(0x0644, 0x8043): - /* TEAC UD-501 */ - if (product && strcmp("UD-501", product)) { - return true; - } else { - return false; - } - } - return false; -} - -/* ITF-USB DSD based DACs need a vendor cmd to switch - * between PCM and native DSD mode - * (3 altsets version) - */ -static bool is_itf_usb_dsd_3alts_dac(struct snd_usb_audio *chip) -{ - char *product = chip->dev->product; /* DAC product name */ - - switch (chip->usb_id) { + case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-501V2/UD-503/NT-503 */ case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ case USB_ID(0x0644, 0x804a): /* TEAC UD-301 */ return true; - case USB_ID(0x0644, 0x8043): - /* TEAC UD-501V2/UD-503/NT-503 */ - if (product && !strcmp("UD-501", product)) { - return true; - } else { - return false; - } } return false; } @@ -1203,7 +1173,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct usb_device *dev = subs->dev; int err; - if (is_itf_usb_dsd_2alts_dac(subs->stream->chip)) { + if (is_itf_usb_dsd_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1213,37 +1183,26 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, mdelay(20); /* Delay needed after setting the interface */ - switch (fmt->altsetting) { - case 2: /* DSD mode requested */ - case 1: /* PCM mode requested */ - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, - USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, - fmt->altsetting - 1, 1, NULL, 0); - if (err < 0) - return err; - break; - } - mdelay(20); - } else if (is_itf_usb_dsd_3alts_dac(subs->stream->chip)) { /* Vendor mode switch cmd is required. */ - switch (fmt->altsetting) { - case 3: /* DSD mode (DSD_U32) requested */ + if (fmt->formats & SNDRV_PCM_FMTBIT_DSD_U32_BE) { + /* DSD mode (DSD_U32) requested */ err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, 1, 1, NULL, 0); if (err < 0) return err; - break; - case 2: /* PCM or DOP mode (S32) requested */ - case 1: /* PCM mode (S16) requested */ + } else { + /* PCM or DOP mode (S32) requested */ + /* PCM mode (S16) requested */ err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0, USB_DIR_OUT|USB_TYPE_VENDOR|USB_RECIP_INTERFACE, 0, 1, NULL, 0); if (err < 0) return err; - break; + } + mdelay(20); } return 0; } @@ -1323,7 +1282,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ - if (is_itf_usb_dsd_2alts_dac(chip) + if (is_itf_usb_dsd_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); @@ -1349,6 +1308,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, struct audioformat *fp, unsigned int sample_bytes) { + struct usb_interface *iface; + /* Playback Designs */ if (USB_ID_VENDOR(chip->usb_id) == 0x23ba) { switch (fp->altsetting) { @@ -1410,15 +1371,16 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; } - /* ITF-USB DSD based DACs (2 altsets version) */ - if (is_itf_usb_dsd_2alts_dac(chip)) { - if (fp->altsetting == 2) - return SNDRV_PCM_FMTBIT_DSD_U32_BE; - } + /* ITF-USB DSD based DACs */ + if (is_itf_usb_dsd_dac(chip->usb_id)) { + iface = usb_ifnum_to_if(chip->dev, fp->iface); - /* ITF-USB DSD based DACs (3 altsets version) */ - if (is_itf_usb_dsd_3alts_dac(chip)) { - if (fp->altsetting == 3) + /* Altsetting 2 support native DSD if the num of altsets is + * three (0-2), + * Altsetting 3 support native DSD if the num of altsets is + * four (0-3). + */ + if (fp->altsetting == iface->num_altsetting - 1) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } From 02a5d6925cd34c3b774bdb8eefb057c40a30e870 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Mar 2018 18:10:14 +0100 Subject: [PATCH 24/31] ALSA: pcm: Avoid potential races between OSS ioctls and read/write Although we apply the params_lock mutex to the whole read and write operations as well as snd_pcm_oss_change_params(), we may still face some races. First off, the params_lock is taken inside the read and write loop. This is intentional for avoiding the too long locking, but it allows the in-between parameter change, which might lead to invalid pointers. We check the readiness of the stream and set up via snd_pcm_oss_make_ready() at the beginning of read and write, but it's called only once, by assuming that it remains ready in the rest. Second, many ioctls that may change the actual parameters (i.e. setting runtime->oss.params=1) aren't protected, hence they can be processed in a half-baked state. This patch is an attempt to plug these holes. The stream readiness check is moved inside the read/write inner loop, so that the stream is always set up in a proper state before further processing. Also, each ioctl that may change the parameter is wrapped with the params_lock for avoiding the races. The issues were triggered by syzkaller in a few different scenarios, particularly the one below appearing as GPF in loopback_pos_update. Reported-by: syzbot+c4227aec125487ec3efa@syzkaller.appspotmail.com Cc: Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 134 +++++++++++++++++++++++++++++++-------- 1 file changed, 106 insertions(+), 28 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 02298c9c6020..f8bfee8022e0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -823,8 +823,8 @@ static int choose_rate(struct snd_pcm_substream *substream, return snd_pcm_hw_param_near(substream, params, SNDRV_PCM_HW_PARAM_RATE, best_rate, NULL); } -static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, - bool trylock) +/* call with params_lock held */ +static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hw_params *params, *sparams; @@ -838,11 +838,8 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, const struct snd_mask *sformat_mask; struct snd_mask mask; - if (trylock) { - if (!(mutex_trylock(&runtime->oss.params_lock))) - return -EAGAIN; - } else if (mutex_lock_interruptible(&runtime->oss.params_lock)) - return -ERESTARTSYS; + if (!runtime->oss.params) + return 0; sw_params = kzalloc(sizeof(*sw_params), GFP_KERNEL); params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); @@ -1068,6 +1065,23 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, kfree(sw_params); kfree(params); kfree(sparams); + return err; +} + +/* this one takes the lock by itself */ +static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, + bool trylock) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + if (trylock) { + if (!(mutex_trylock(&runtime->oss.params_lock))) + return -EAGAIN; + } else if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + + err = snd_pcm_oss_change_params_locked(substream); mutex_unlock(&runtime->oss.params_lock); return err; } @@ -1096,11 +1110,14 @@ static int snd_pcm_oss_get_active_substream(struct snd_pcm_oss_file *pcm_oss_fil return 0; } +/* call with params_lock held */ static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream) { int err; struct snd_pcm_runtime *runtime = substream->runtime; + if (!runtime->oss.prepare) + return 0; err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_PREPARE, NULL); if (err < 0) { pcm_dbg(substream->pcm, @@ -1120,14 +1137,35 @@ static int snd_pcm_oss_make_ready(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime; int err; - if (substream == NULL) - return 0; runtime = substream->runtime; if (runtime->oss.params) { err = snd_pcm_oss_change_params(substream, false); if (err < 0) return err; } + if (runtime->oss.prepare) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + err = snd_pcm_oss_prepare(substream); + mutex_unlock(&runtime->oss.params_lock); + if (err < 0) + return err; + } + return 0; +} + +/* call with params_lock held */ +static int snd_pcm_oss_make_ready_locked(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + int err; + + runtime = substream->runtime; + if (runtime->oss.params) { + err = snd_pcm_oss_change_params_locked(substream); + if (err < 0) + return err; + } if (runtime->oss.prepare) { err = snd_pcm_oss_prepare(substream); if (err < 0) @@ -1332,13 +1370,14 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha if (atomic_read(&substream->mmap_count)) return -ENXIO; - if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) - return tmp; while (bytes > 0) { if (mutex_lock_interruptible(&runtime->oss.params_lock)) { tmp = -ERESTARTSYS; break; } + tmp = snd_pcm_oss_make_ready_locked(substream); + if (tmp < 0) + goto err; if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { tmp = bytes; if (tmp + runtime->oss.buffer_used > runtime->oss.period_bytes) @@ -1439,13 +1478,14 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use if (atomic_read(&substream->mmap_count)) return -ENXIO; - if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) - return tmp; while (bytes > 0) { if (mutex_lock_interruptible(&runtime->oss.params_lock)) { tmp = -ERESTARTSYS; break; } + tmp = snd_pcm_oss_make_ready_locked(substream); + if (tmp < 0) + goto err; if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { if (runtime->oss.buffer_used == 0) { tmp = snd_pcm_oss_read2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1); @@ -1501,10 +1541,12 @@ static int snd_pcm_oss_reset(struct snd_pcm_oss_file *pcm_oss_file) continue; runtime = substream->runtime; snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); + mutex_lock(&runtime->oss.params_lock); runtime->oss.prepare = 1; runtime->oss.buffer_used = 0; runtime->oss.prev_hw_ptr_period = 0; runtime->oss.period_ptr = 0; + mutex_unlock(&runtime->oss.params_lock); } return 0; } @@ -1590,9 +1632,10 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) goto __direct; if ((err = snd_pcm_oss_make_ready(substream)) < 0) return err; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); - mutex_lock(&runtime->oss.params_lock); if (runtime->oss.buffer_used > 0) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "sync: buffer_used\n"); @@ -1643,7 +1686,9 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) substream->f_flags = saved_f_flags; if (err < 0) return err; + mutex_lock(&runtime->oss.params_lock); runtime->oss.prepare = 1; + mutex_unlock(&runtime->oss.params_lock); } substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; @@ -1654,8 +1699,10 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); if (err < 0) return err; + mutex_lock(&runtime->oss.params_lock); runtime->oss.buffer_used = 0; runtime->oss.prepare = 1; + mutex_unlock(&runtime->oss.params_lock); } return 0; } @@ -1674,10 +1721,13 @@ static int snd_pcm_oss_set_rate(struct snd_pcm_oss_file *pcm_oss_file, int rate) rate = 1000; else if (rate > 192000) rate = 192000; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (runtime->oss.rate != rate) { runtime->oss.params = 1; runtime->oss.rate = rate; } + mutex_unlock(&runtime->oss.params_lock); } return snd_pcm_oss_get_rate(pcm_oss_file); } @@ -1705,10 +1755,13 @@ static int snd_pcm_oss_set_channels(struct snd_pcm_oss_file *pcm_oss_file, unsig if (substream == NULL) continue; runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (runtime->oss.channels != channels) { runtime->oss.params = 1; runtime->oss.channels = channels; } + mutex_unlock(&runtime->oss.params_lock); } return snd_pcm_oss_get_channels(pcm_oss_file); } @@ -1794,10 +1847,13 @@ static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int for if (substream == NULL) continue; runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (runtime->oss.format != format) { runtime->oss.params = 1; runtime->oss.format = format; } + mutex_unlock(&runtime->oss.params_lock); } } return snd_pcm_oss_get_format(pcm_oss_file); @@ -1817,8 +1873,6 @@ static int snd_pcm_oss_set_subdivide1(struct snd_pcm_substream *substream, int s { struct snd_pcm_runtime *runtime; - if (substream == NULL) - return 0; runtime = substream->runtime; if (subdivide == 0) { subdivide = runtime->oss.subdivision; @@ -1842,9 +1896,16 @@ static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int for (idx = 1; idx >= 0; --idx) { struct snd_pcm_substream *substream = pcm_oss_file->streams[idx]; + struct snd_pcm_runtime *runtime; + if (substream == NULL) continue; - if ((err = snd_pcm_oss_set_subdivide1(substream, subdivide)) < 0) + runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + err = snd_pcm_oss_set_subdivide1(substream, subdivide); + mutex_unlock(&runtime->oss.params_lock); + if (err < 0) return err; } return err; @@ -1854,8 +1915,6 @@ static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsign { struct snd_pcm_runtime *runtime; - if (substream == NULL) - return 0; runtime = substream->runtime; if (runtime->oss.subdivision || runtime->oss.fragshift) return -EINVAL; @@ -1875,9 +1934,16 @@ static int snd_pcm_oss_set_fragment(struct snd_pcm_oss_file *pcm_oss_file, unsig for (idx = 1; idx >= 0; --idx) { struct snd_pcm_substream *substream = pcm_oss_file->streams[idx]; + struct snd_pcm_runtime *runtime; + if (substream == NULL) continue; - if ((err = snd_pcm_oss_set_fragment1(substream, val)) < 0) + runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + err = snd_pcm_oss_set_fragment1(substream, val); + mutex_unlock(&runtime->oss.params_lock); + if (err < 0) return err; } return err; @@ -1961,6 +2027,9 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr } if (psubstream) { runtime = psubstream->runtime; + cmd = 0; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (trigger & PCM_ENABLE_OUTPUT) { if (runtime->oss.trigger) goto _skip1; @@ -1978,13 +2047,19 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr cmd = SNDRV_PCM_IOCTL_DROP; runtime->oss.prepare = 1; } - err = snd_pcm_kernel_ioctl(psubstream, cmd, NULL); - if (err < 0) - return err; - } _skip1: + mutex_unlock(&runtime->oss.params_lock); + if (cmd) { + err = snd_pcm_kernel_ioctl(psubstream, cmd, NULL); + if (err < 0) + return err; + } + } if (csubstream) { runtime = csubstream->runtime; + cmd = 0; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (trigger & PCM_ENABLE_INPUT) { if (runtime->oss.trigger) goto _skip2; @@ -1999,11 +2074,14 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr cmd = SNDRV_PCM_IOCTL_DROP; runtime->oss.prepare = 1; } - err = snd_pcm_kernel_ioctl(csubstream, cmd, NULL); - if (err < 0) - return err; - } _skip2: + mutex_unlock(&runtime->oss.params_lock); + if (cmd) { + err = snd_pcm_kernel_ioctl(csubstream, cmd, NULL); + if (err < 0) + return err; + } + } return 0; } From 40cab6e88cb0b6c56d3f30b7491a20e803f948f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Mar 2018 08:03:26 +0100 Subject: [PATCH 25/31] ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams OSS PCM stream management isn't modal but it allows ioctls issued at any time for changing the parameters. In the previous hardening patch ("ALSA: pcm: Avoid potential races between OSS ioctls and read/write"), we covered these races and prevent the corruption by protecting the concurrent accesses via params_lock mutex. However, this means that some ioctls that try to change the stream parameter (e.g. channels or format) would be blocked until the read/write finishes, and it may take really long. Basically changing the parameter while reading/writing is an invalid operation, hence it's even more user-friendly from the API POV if it returns -EBUSY in such a situation. This patch adds such checks in the relevant ioctls with the addition of read/write access refcount. Cc: Signed-off-by: Takashi Iwai --- include/sound/pcm_oss.h | 1 + sound/core/oss/pcm_oss.c | 36 +++++++++++++++++++++++++++--------- 2 files changed, 28 insertions(+), 9 deletions(-) diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index 760c969d885d..12bbf8c81112 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -57,6 +57,7 @@ struct snd_pcm_oss_runtime { char *buffer; /* vmallocated period */ size_t buffer_used; /* used length from period buffer */ struct mutex params_lock; + atomic_t rw_ref; /* concurrent read/write accesses */ #ifdef CONFIG_SND_PCM_OSS_PLUGINS struct snd_pcm_plugin *plugin_first; struct snd_pcm_plugin *plugin_last; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f8bfee8022e0..a9082f219561 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1370,6 +1370,7 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha if (atomic_read(&substream->mmap_count)) return -ENXIO; + atomic_inc(&runtime->oss.rw_ref); while (bytes > 0) { if (mutex_lock_interruptible(&runtime->oss.params_lock)) { tmp = -ERESTARTSYS; @@ -1433,6 +1434,7 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha } tmp = 0; } + atomic_dec(&runtime->oss.rw_ref); return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } @@ -1478,6 +1480,7 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use if (atomic_read(&substream->mmap_count)) return -ENXIO; + atomic_inc(&runtime->oss.rw_ref); while (bytes > 0) { if (mutex_lock_interruptible(&runtime->oss.params_lock)) { tmp = -ERESTARTSYS; @@ -1526,6 +1529,7 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use } tmp = 0; } + atomic_dec(&runtime->oss.rw_ref); return xfer > 0 ? (snd_pcm_sframes_t)xfer : tmp; } @@ -1632,8 +1636,11 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) goto __direct; if ((err = snd_pcm_oss_make_ready(substream)) < 0) return err; - if (mutex_lock_interruptible(&runtime->oss.params_lock)) + atomic_inc(&runtime->oss.rw_ref); + if (mutex_lock_interruptible(&runtime->oss.params_lock)) { + atomic_dec(&runtime->oss.rw_ref); return -ERESTARTSYS; + } format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); if (runtime->oss.buffer_used > 0) { @@ -1645,10 +1652,8 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) runtime->oss.buffer + runtime->oss.buffer_used, size); err = snd_pcm_oss_sync1(substream, runtime->oss.period_bytes); - if (err < 0) { - mutex_unlock(&runtime->oss.params_lock); - return err; - } + if (err < 0) + goto unlock; } else if (runtime->oss.period_ptr > 0) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "sync: period_ptr\n"); @@ -1658,10 +1663,8 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) runtime->oss.buffer, size * 8 / width); err = snd_pcm_oss_sync1(substream, size); - if (err < 0) { - mutex_unlock(&runtime->oss.params_lock); - return err; - } + if (err < 0) + goto unlock; } /* * The ALSA's period might be a bit large than OSS one. @@ -1675,7 +1678,11 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) else if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED) snd_pcm_lib_writev(substream, NULL, size); } +unlock: mutex_unlock(&runtime->oss.params_lock); + atomic_dec(&runtime->oss.rw_ref); + if (err < 0) + return err; /* * finish sync: drain the buffer */ @@ -1723,6 +1730,8 @@ static int snd_pcm_oss_set_rate(struct snd_pcm_oss_file *pcm_oss_file, int rate) rate = 192000; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; + if (atomic_read(&runtime->oss.rw_ref)) + return -EBUSY; if (runtime->oss.rate != rate) { runtime->oss.params = 1; runtime->oss.rate = rate; @@ -1757,6 +1766,8 @@ static int snd_pcm_oss_set_channels(struct snd_pcm_oss_file *pcm_oss_file, unsig runtime = substream->runtime; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; + if (atomic_read(&runtime->oss.rw_ref)) + return -EBUSY; if (runtime->oss.channels != channels) { runtime->oss.params = 1; runtime->oss.channels = channels; @@ -1847,6 +1858,8 @@ static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int for if (substream == NULL) continue; runtime = substream->runtime; + if (atomic_read(&runtime->oss.rw_ref)) + return -EBUSY; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; if (runtime->oss.format != format) { @@ -1901,6 +1914,8 @@ static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int if (substream == NULL) continue; runtime = substream->runtime; + if (atomic_read(&runtime->oss.rw_ref)) + return -EBUSY; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; err = snd_pcm_oss_set_subdivide1(substream, subdivide); @@ -1939,6 +1954,8 @@ static int snd_pcm_oss_set_fragment(struct snd_pcm_oss_file *pcm_oss_file, unsig if (substream == NULL) continue; runtime = substream->runtime; + if (atomic_read(&runtime->oss.rw_ref)) + return -EBUSY; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; err = snd_pcm_oss_set_fragment1(substream, val); @@ -2333,6 +2350,7 @@ static void snd_pcm_oss_init_substream(struct snd_pcm_substream *substream, runtime->oss.maxfrags = 0; runtime->oss.subdivision = 0; substream->pcm_release = snd_pcm_oss_release_substream; + atomic_set(&runtime->oss.rw_ref, 0); } static int snd_pcm_oss_release_file(struct snd_pcm_oss_file *pcm_oss_file) From 5a222e8494524d04cc8de63726644759e5b132f7 Mon Sep 17 00:00:00 2001 From: Andrew Chant Date: Fri, 23 Mar 2018 19:25:23 -0700 Subject: [PATCH 26/31] ALSA: usb-audio: UAC2 jack detection This implements UAC2 jack detection support, presenting jack status as a boolean read-only mono mixer. The presence of any channel in the UAC2_TE_CONNECTOR control for a terminal will result in the mixer saying the jack is connected. Mixer naming follows the convention in sound/core/ctljack.c, terminating the mixer with " Jack". For additional clues as to which jack is being presented, the name is prefixed with " - Input Jack" or " - Output Jack" depending on if it's an input or output terminal. This is required because terminal names are ambiguous between inputs and outputs and often duplicated - Bidirectional terminal types (0x400 -> 0x4FF) "... may be used separately for input only or output only. These types require two Terminal descriptors. Both have the same type." (quote from "USB Device Class Definition for Terminal Types") Since bidirectional terminal types are common for headphone adapters, this distinguishes between two otherwise identically-named jack controls. Tested with a UAC2 audio device with connector control capability. Signed-off-by: Andrew Chant Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 98 ++++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 97 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 3075ac50a391..97ed2155205d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1235,6 +1235,21 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, return changed; } +/* get the current value from a mixer element */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + int val, err; + + err = snd_usb_get_cur_mix_value(cval, 0, 0, &val); + if (err < 0) + return filter_error(cval, err); + val = (val != 0); + ucontrol->value.integer.value[0] = val; + return 0; +} + static struct snd_kcontrol_new usb_feature_unit_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", /* will be filled later manually */ @@ -1252,6 +1267,16 @@ static const struct snd_kcontrol_new usb_feature_unit_ctl_ro = { .put = NULL, }; +/* A UAC connector mixer control */ +static struct snd_kcontrol_new usb_connector_ctl_ro = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", /* will be filled later manually */ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = snd_ctl_boolean_mono_info, + .get = mixer_ctl_connector_get, + .put = NULL, +}; + /* * This symbol is exported in order to allow the mixer quirks to * hook up to the standard feature unit control mechanism @@ -1464,6 +1489,54 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static void get_connector_control_name(struct mixer_build *state, + struct usb_audio_term *term, + bool is_input, char *name, int name_size) +{ + int name_len = get_term_name(state, term, name, name_size, 0); + + if (name_len == 0) + strlcpy(name, "Unknown", name_size); + + /* + * sound/core/ctljack.c has a convention of naming jack controls + * by ending in " Jack". Make it slightly more useful by + * indicating Input or Output after the terminal name. + */ + if (is_input) + strlcat(name, " - Input Jack", name_size); + else + strlcat(name, " - Output Jack", name_size); +} + +/* Build a mixer control for a UAC connector control (jack-detect) */ +static void build_connector_control(struct mixer_build *state, + struct usb_audio_term *term, bool is_input) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *cval; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return; + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, term->id); + cval->control = UAC2_TE_CONNECTOR; + cval->val_type = USB_MIXER_BOOLEAN; + cval->channels = 1; /* report true if any channel is connected */ + cval->min = 0; + cval->max = 1; + kctl = snd_ctl_new1(&usb_connector_ctl_ro, cval); + if (!kctl) { + usb_audio_err(state->chip, "cannot malloc kcontrol\n"); + kfree(cval); + return; + } + get_connector_control_name(state, term, is_input, kctl->id.name, + sizeof(kctl->id.name)); + kctl->private_free = snd_usb_mixer_elem_free; + snd_usb_mixer_add_control(&cval->head, kctl); +} + static int parse_clock_source_unit(struct mixer_build *state, int unitid, void *_ftr) { @@ -1770,6 +1843,23 @@ static void build_mixer_unit_ctl(struct mixer_build *state, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_audio_input_terminal(struct mixer_build *state, int unitid, + void *raw_desc) +{ + struct usb_audio_term iterm; + struct uac2_input_terminal_descriptor *d = raw_desc; + + check_input_term(state, d->bTerminalID, &iterm); + if (state->mixer->protocol == UAC_VERSION_2) { + /* Check for jack detection. */ + if (uac_v2v3_control_is_readable(d->bmControls, + UAC2_TE_CONNECTOR)) { + build_connector_control(state, &iterm, true); + } + } + return 0; +} + /* * parse a mixer unit */ @@ -2320,7 +2410,7 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) { switch (p1[2]) { case UAC_INPUT_TERMINAL: - return 0; /* NOP */ + return parse_audio_input_terminal(state, unitid, p1); case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); case UAC2_CLOCK_SOURCE: @@ -2463,6 +2553,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0 && err != -EINVAL) return err; + + if (uac_v2v3_control_is_readable(desc->bmControls, + UAC2_TE_CONNECTOR)) { + build_connector_control(&state, &state.oterm, + false); + } } else { /* UAC_VERSION_3 */ struct uac3_output_terminal_descriptor *desc = p; From 568fa7e087ef98bc85b5aa31ea7c9252c1305c1f Mon Sep 17 00:00:00 2001 From: Andrew Chant Date: Fri, 23 Mar 2018 19:25:24 -0700 Subject: [PATCH 27/31] ALSA: usb-audio: update clock valid control Make the "clock valid" control a global control instead of a mixer so that it doesn't appear in mixer applications. Additionally, remove the check for writeability prohibited by spec, and Use common code to read the control value. Tested with a UAC2 Audio device that presents a clock validity control. The control still shows up in /proc usbmixer but not in alsamixer. Signed-off-by: Andrew Chant Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 97ed2155205d..04dab6f65535 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1235,9 +1235,9 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, return changed; } -/* get the current value from a mixer element */ -static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* get the boolean value from the master channel of a UAC control */ +static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; int val, err; @@ -1267,13 +1267,16 @@ static const struct snd_kcontrol_new usb_feature_unit_ctl_ro = { .put = NULL, }; -/* A UAC connector mixer control */ -static struct snd_kcontrol_new usb_connector_ctl_ro = { +/* + * A control which shows the boolean value from reading a UAC control on + * the master channel. + */ +static struct snd_kcontrol_new usb_bool_master_control_ctl_ro = { .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = "", /* will be filled later manually */ .access = SNDRV_CTL_ELEM_ACCESS_READ, .info = snd_ctl_boolean_mono_info, - .get = mixer_ctl_connector_get, + .get = mixer_ctl_master_bool_get, .put = NULL, }; @@ -1520,12 +1523,18 @@ static void build_connector_control(struct mixer_build *state, if (!cval) return; snd_usb_mixer_elem_init_std(&cval->head, state->mixer, term->id); + /* + * The first byte from reading the UAC2_TE_CONNECTOR control returns the + * number of channels connected. This boolean ctl will simply report + * if any channels are connected or not. + * (Audio20_final.pdf Table 5-10: Connector Control CUR Parameter Block) + */ cval->control = UAC2_TE_CONNECTOR; cval->val_type = USB_MIXER_BOOLEAN; cval->channels = 1; /* report true if any channel is connected */ cval->min = 0; cval->max = 1; - kctl = snd_ctl_new1(&usb_connector_ctl_ro, cval); + kctl = snd_ctl_new1(&usb_bool_master_control_ctl_ro, cval); if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); @@ -1576,13 +1585,9 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid, cval->val_type = USB_MIXER_BOOLEAN; cval->control = UAC2_CS_CONTROL_CLOCK_VALID; - if (uac_v2v3_control_is_writeable(hdr->bmControls, - UAC2_CS_CONTROL_CLOCK_VALID)) - kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); - else { - cval->master_readonly = 1; - kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); - } + cval->master_readonly = 1; + /* From UAC2 5.2.5.1.2 "Only the get request is supported." */ + kctl = snd_ctl_new1(&usb_bool_master_control_ctl_ro, cval); if (!kctl) { kfree(cval); From 306a4f3ca7f3c7dfa473ebd19d66e40e59d99734 Mon Sep 17 00:00:00 2001 From: Robert Rosengren Date: Mon, 26 Mar 2018 07:24:49 +0200 Subject: [PATCH 28/31] ALSA: aloop: Mark paused device as inactive Show paused ALSA aloop device as inactive, i.e. the control "PCM Slave Active" set as false. Notification sent upon state change. This makes it possible for client capturing from aloop device to know if data is expected. Without it the client expects data even if playback is paused. Signed-off-by: Robert Rosengren Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 1063a4377502..58e349fc893f 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -296,6 +296,8 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause |= stream; loopback_timer_stop(dpcm); spin_unlock(&cable->lock); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: @@ -304,6 +306,8 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause &= ~stream; loopback_timer_start(dpcm); spin_unlock(&cable->lock); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); break; default: return -EINVAL; @@ -892,9 +896,11 @@ static int loopback_active_get(struct snd_kcontrol *kcontrol, [kcontrol->id.subdevice][kcontrol->id.device ^ 1]; unsigned int val = 0; - if (cable != NULL) - val = (cable->running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) ? - 1 : 0; + if (cable != NULL) { + unsigned int running = cable->running ^ cable->pause; + + val = (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) ? 1 : 0; + } ucontrol->value.integer.value[0] = val; return 0; } From f6d297df4dd47ef949540e4a201230d0c5308325 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Mar 2018 14:32:23 +0200 Subject: [PATCH 29/31] ALSA: pcm: Fix mutex unbalance in OSS emulation ioctls The previous fix 40cab6e88cb0 ("ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams") introduced some mutex unbalance; the check of runtime->oss.rw_ref was inserted in a wrong place after the mutex lock. This patch fixes the inconsistency by rewriting with the helper functions to lock/unlock parameters with the stream check. Fixes: 40cab6e88cb0 ("ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams") Reported-by: Dan Carpenter Cc: Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 67 +++++++++++++++++++++++++--------------- 1 file changed, 42 insertions(+), 25 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index a9082f219561..f19b4586e33b 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -823,6 +823,23 @@ static int choose_rate(struct snd_pcm_substream *substream, return snd_pcm_hw_param_near(substream, params, SNDRV_PCM_HW_PARAM_RATE, best_rate, NULL); } +/* parameter locking: returns immediately if tried during streaming */ +static int lock_params(struct snd_pcm_runtime *runtime) +{ + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + if (atomic_read(&runtime->oss.rw_ref)) { + mutex_unlock(&runtime->oss.params_lock); + return -EBUSY; + } + return 0; +} + +static void unlock_params(struct snd_pcm_runtime *runtime) +{ + mutex_unlock(&runtime->oss.params_lock); +} + /* call with params_lock held */ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) { @@ -1721,6 +1738,8 @@ static int snd_pcm_oss_set_rate(struct snd_pcm_oss_file *pcm_oss_file, int rate) for (idx = 1; idx >= 0; --idx) { struct snd_pcm_substream *substream = pcm_oss_file->streams[idx]; struct snd_pcm_runtime *runtime; + int err; + if (substream == NULL) continue; runtime = substream->runtime; @@ -1728,15 +1747,14 @@ static int snd_pcm_oss_set_rate(struct snd_pcm_oss_file *pcm_oss_file, int rate) rate = 1000; else if (rate > 192000) rate = 192000; - if (mutex_lock_interruptible(&runtime->oss.params_lock)) - return -ERESTARTSYS; - if (atomic_read(&runtime->oss.rw_ref)) - return -EBUSY; + err = lock_params(runtime); + if (err < 0) + return err; if (runtime->oss.rate != rate) { runtime->oss.params = 1; runtime->oss.rate = rate; } - mutex_unlock(&runtime->oss.params_lock); + unlock_params(runtime); } return snd_pcm_oss_get_rate(pcm_oss_file); } @@ -1761,18 +1779,19 @@ static int snd_pcm_oss_set_channels(struct snd_pcm_oss_file *pcm_oss_file, unsig for (idx = 1; idx >= 0; --idx) { struct snd_pcm_substream *substream = pcm_oss_file->streams[idx]; struct snd_pcm_runtime *runtime; + int err; + if (substream == NULL) continue; runtime = substream->runtime; - if (mutex_lock_interruptible(&runtime->oss.params_lock)) - return -ERESTARTSYS; - if (atomic_read(&runtime->oss.rw_ref)) - return -EBUSY; + err = lock_params(runtime); + if (err < 0) + return err; if (runtime->oss.channels != channels) { runtime->oss.params = 1; runtime->oss.channels = channels; } - mutex_unlock(&runtime->oss.params_lock); + unlock_params(runtime); } return snd_pcm_oss_get_channels(pcm_oss_file); } @@ -1845,6 +1864,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int format) { int formats, idx; + int err; if (format != AFMT_QUERY) { formats = snd_pcm_oss_get_formats(pcm_oss_file); @@ -1858,15 +1878,14 @@ static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int for if (substream == NULL) continue; runtime = substream->runtime; - if (atomic_read(&runtime->oss.rw_ref)) - return -EBUSY; - if (mutex_lock_interruptible(&runtime->oss.params_lock)) - return -ERESTARTSYS; + err = lock_params(runtime); + if (err < 0) + return err; if (runtime->oss.format != format) { runtime->oss.params = 1; runtime->oss.format = format; } - mutex_unlock(&runtime->oss.params_lock); + unlock_params(runtime); } } return snd_pcm_oss_get_format(pcm_oss_file); @@ -1914,12 +1933,11 @@ static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int if (substream == NULL) continue; runtime = substream->runtime; - if (atomic_read(&runtime->oss.rw_ref)) - return -EBUSY; - if (mutex_lock_interruptible(&runtime->oss.params_lock)) - return -ERESTARTSYS; + err = lock_params(runtime); + if (err < 0) + return err; err = snd_pcm_oss_set_subdivide1(substream, subdivide); - mutex_unlock(&runtime->oss.params_lock); + unlock_params(runtime); if (err < 0) return err; } @@ -1954,12 +1972,11 @@ static int snd_pcm_oss_set_fragment(struct snd_pcm_oss_file *pcm_oss_file, unsig if (substream == NULL) continue; runtime = substream->runtime; - if (atomic_read(&runtime->oss.rw_ref)) - return -EBUSY; - if (mutex_lock_interruptible(&runtime->oss.params_lock)) - return -ERESTARTSYS; + err = lock_params(runtime); + if (err < 0) + return err; err = snd_pcm_oss_set_fragment1(substream, val); - mutex_unlock(&runtime->oss.params_lock); + unlock_params(runtime); if (err < 0) return err; } From 623760257b39632ffc5051fc88167b546dcfa791 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 27 Mar 2018 15:30:01 +0100 Subject: [PATCH 30/31] ALSA: usb-audio: fix memory leak on cval With the current exit return path of the ctl_info allocation failure cval is not being freed resulting in a memory leak. Fix this by kfree'ing it on the return. Detected by CoverityScan, CID#1466878 ("Resource Leak") Fixes: 21e9b3e931f7 ("ALSA: usb-audio: fix uac control query argument") Signed-off-by: Colin Ian King Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 04dab6f65535..301ad61ed426 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1365,8 +1365,10 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->cmask = ctl_mask; ctl_info = get_feature_control_info(control); - if (!ctl_info) + if (!ctl_info) { + kfree(cval); return; + } if (state->mixer->protocol == UAC_VERSION_1) cval->val_type = ctl_info->type; else /* UAC_VERSION_2 */ From b44d419b98fae759b4f746186b1d1c8d01d962f2 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 29 Mar 2018 12:03:19 +0300 Subject: [PATCH 31/31] ALSA: usb-audio: silence a static checker warning We recently made "format" a u64 variable so now static checkers complain that this shift will wrap around if format is more than 31. I don't think it makes a difference for runtime, but it's simple to silence the warning. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/usb/format.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index edbe67eeddfa..49e7ec6d2399 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -55,7 +55,7 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; - format = 1 << format; + format = 1ULL << format; break; }