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authorLibravatar Linus Torvalds <torvalds@linux-foundation.org>2023-02-21 18:24:12 -0800
committerLibravatar Linus Torvalds <torvalds@linux-foundation.org>2023-02-21 18:24:12 -0800
commit5b7c4cabbb65f5c469464da6c5f614cbd7f730f2 (patch)
treecc5c2d0a898769fd59549594fedb3ee6f84e59a0 /sound/oss/dmasound/dmasound_paula.c
downloadlinux-5b7c4cabbb65f5c469464da6c5f614cbd7f730f2.tar.gz
linux-5b7c4cabbb65f5c469464da6c5f614cbd7f730f2.zip
Merge tag 'net-next-6.3' of git://git.kernel.org/pub/scm/linux/kernel/git/netdev/net-nextgrafted
Pull networking updates from Jakub Kicinski: "Core: - Add dedicated kmem_cache for typical/small skb->head, avoid having to access struct page at kfree time, and improve memory use. - Introduce sysctl to set default RPS configuration for new netdevs. - Define Netlink protocol specification format which can be used to describe messages used by each family and auto-generate parsers. Add tools for generating kernel data structures and uAPI headers. - Expose all net/core sysctls inside netns. - Remove 4s sleep in netpoll if carrier is instantly detected on boot. - Add configurable limit of MDB entries per port, and port-vlan. - Continue populating drop reasons throughout the stack. - Retire a handful of legacy Qdiscs and classifiers. Protocols: - Support IPv4 big TCP (TSO frames larger than 64kB). - Add IP_LOCAL_PORT_RANGE socket option, to control local port range on socket by socket basis. - Track and report in procfs number of MPTCP sockets used. - Support mixing IPv4 and IPv6 flows in the in-kernel MPTCP path manager. - IPv6: don't check net.ipv6.route.max_size and rely on garbage collection to free memory (similarly to IPv4). - Support Penultimate Segment Pop (PSP) flavor in SRv6 (RFC8986). - ICMP: add per-rate limit counters. - Add support for user scanning requests in ieee802154. - Remove static WEP support. - Support minimal Wi-Fi 7 Extremely High Throughput (EHT) rate reporting. - WiFi 7 EHT channel puncturing support (client & AP). BPF: - Add a rbtree data structure following the "next-gen data structure" precedent set by recently added linked list, that is, by using kfunc + kptr instead of adding a new BPF map type. - Expose XDP hints via kfuncs with initial support for RX hash and timestamp metadata. - Add BPF_F_NO_TUNNEL_KEY extension to bpf_skb_set_tunnel_key to better support decap on GRE tunnel devices not operating in collect metadata. - Improve x86 JIT's codegen for PROBE_MEM runtime error checks. - Remove the need for trace_printk_lock for bpf_trace_printk and bpf_trace_vprintk helpers. - Extend libbpf's bpf_tracing.h support for tracing arguments of kprobes/uprobes and syscall as a special case. - Significantly reduce the search time for module symbols by livepatch and BPF. - Enable cpumasks to be used as kptrs, which is useful for tracing programs tracking which tasks end up running on which CPUs in different time intervals. - Add support for BPF trampoline on s390x and riscv64. - Add capability to export the XDP features supported by the NIC. - Add __bpf_kfunc tag for marking kernel functions as kfuncs. - Add cgroup.memory=nobpf kernel parameter option to disable BPF memory accounting for container environments. Netfilter: - Remove the CLUSTERIP target. It has been marked as obsolete for years, and we still have WARN splats wrt races of the out-of-band /proc interface installed by this target. - Add 'destroy' commands to nf_tables. They are identical to the existing 'delete' commands, but do not return an error if the referenced object (set, chain, rule...) did not exist. Driver API: - Improve cpumask_local_spread() locality to help NICs set the right IRQ affinity on AMD platforms. - Separate C22 and C45 MDIO bus transactions more clearly. - Introduce new DCB table to control DSCP rewrite on egress. - Support configuration of Physical Layer Collision Avoidance (PLCA) Reconciliation Sublayer (RS) (802.3cg-2019). Modern version of shared medium Ethernet. - Support for MAC Merge layer (IEEE 802.3-2018 clause 99). Allowing preemption of low priority frames by high priority frames. - Add support for controlling MACSec offload using netlink SET. - Rework devlink instance refcounts to allow registration and de-registration under the instance lock. Split the code into multiple files, drop some of the unnecessarily granular locks and factor out common parts of netlink operation handling. - Add TX frame aggregation parameters (for USB drivers). - Add a new attr TCA_EXT_WARN_MSG to report TC (offload) warning messages with notifications for debug. - Allow offloading of UDP NEW connections via act_ct. - Add support for per action HW stats in TC. - Support hardware miss to TC action (continue processing in SW from a specific point in the action chain). - Warn if old Wireless Extension user space interface is used with modern cfg80211/mac80211 drivers. Do not support Wireless Extensions for Wi-Fi 7 devices at all. Everyone should switch to using nl80211 interface instead. - Improve the CAN bit timing configuration. Use extack to return error messages directly to user space, update the SJW handling, including the definition of a new default value that will benefit CAN-FD controllers, by increasing their oscillator tolerance. New hardware / drivers: - Ethernet: - nVidia BlueField-3 support (control traffic driver) - Ethernet support for imx93 SoCs - Motorcomm yt8531 gigabit Ethernet PHY - onsemi NCN26000 10BASE-T1S PHY (with support for PLCA) - Microchip LAN8841 PHY (incl. cable diagnostics and PTP) - Amlogic gxl MDIO mux - WiFi: - RealTek RTL8188EU (rtl8xxxu) - Qualcomm Wi-Fi 7 devices (ath12k) - CAN: - Renesas R-Car V4H Drivers: - Bluetooth: - Set Per Platform Antenna Gain (PPAG) for Intel controllers. - Ethernet NICs: - Intel (1G, igc): - support TSN / Qbv / packet scheduling features of i226 model - Intel (100G, ice): - use GNSS subsystem instead of TTY - multi-buffer XDP support - extend support for GPIO pins to E823 devices - nVidia/Mellanox: - update the shared buffer configuration on PFC commands - implement PTP adjphase function for HW offset control - TC support for Geneve and GRE with VF tunnel offload - more efficient crypto key management method - multi-port eswitch support - Netronome/Corigine: - add DCB IEEE support - support IPsec offloading for NFP3800 - Freescale/NXP (enetc): - support XDP_REDIRECT for XDP non-linear buffers - improve reconfig, avoid link flap and waiting for idle - support MAC Merge layer - Other NICs: - sfc/ef100: add basic devlink support for ef100 - ionic: rx_push mode operation (writing descriptors via MMIO) - bnxt: use the auxiliary bus abstraction for RDMA - r8169: disable ASPM and reset bus in case of tx timeout - cpsw: support QSGMII mode for J721e CPSW9G - cpts: support pulse-per-second output - ngbe: add an mdio bus driver - usbnet: optimize usbnet_bh() by avoiding unnecessary queuing - r8152: handle devices with FW with NCM support - amd-xgbe: support 10Mbps, 2.5GbE speeds and rx-adaptation - virtio-net: support multi buffer XDP - virtio/vsock: replace virtio_vsock_pkt with sk_buff - tsnep: XDP support - Ethernet high-speed switches: - nVidia/Mellanox (mlxsw): - add support for latency TLV (in FW control messages) - Microchip (sparx5): - separate explicit and implicit traffic forwarding rules, make the implicit rules always active - add support for egress DSCP rewrite - IS0 VCAP support (Ingress Classification) - IS2 VCAP filters (protos, L3 addrs, L4 ports, flags, ToS etc.) - ES2 VCAP support (Egress Access Control) - support for Per-Stream Filtering and Policing (802.1Q, 8.6.5.1) - Ethernet embedded switches: - Marvell (mv88e6xxx): - add MAB (port auth) offload support - enable PTP receive for mv88e6390 - NXP (ocelot): - support MAC Merge layer - support for the the vsc7512 internal copper phys - Microchip: - lan9303: convert to PHYLINK - lan966x: support TC flower filter statistics - lan937x: PTP support for KSZ9563/KSZ8563 and LAN937x - lan937x: support Credit Based Shaper configuration - ksz9477: support Energy Efficient Ethernet - other: - qca8k: convert to regmap read/write API, use bulk operations - rswitch: Improve TX timestamp accuracy - Intel WiFi (iwlwifi): - EHT (Wi-Fi 7) rate reporting - STEP equalizer support: transfer some STEP (connection to radio on platforms with integrated wifi) related parameters from the BIOS to the firmware. - Qualcomm 802.11ax WiFi (ath11k): - IPQ5018 support - Fine Timing Measurement (FTM) responder role support - channel 177 support - MediaTek WiFi (mt76): - per-PHY LED support - mt7996: EHT (Wi-Fi 7) support - Wireless Ethernet Dispatch (WED) reset support - switch to using page pool allocator - RealTek WiFi (rtw89): - support new version of Bluetooth co-existance - Mobile: - rmnet: support TX aggregation" * tag 'net-next-6.3' of git://git.kernel.org/pub/scm/linux/kernel/git/netdev/net-next: (1872 commits) page_pool: add a comment explaining the fragment counter usage net: ethtool: fix __ethtool_dev_mm_supported() implementation ethtool: pse-pd: Fix double word in comments xsk: add linux/vmalloc.h to xsk.c sefltests: netdevsim: wait for devlink instance after netns removal selftest: fib_tests: Always cleanup before exit net/mlx5e: Align IPsec ASO result memory to be as required by hardware net/mlx5e: TC, Set CT miss to the specific ct action instance net/mlx5e: Rename CHAIN_TO_REG to MAPPED_OBJ_TO_REG net/mlx5: Refactor tc miss handling to a single function net/mlx5: Kconfig: Make tc offload depend on tc skb extension net/sched: flower: Support hardware miss to tc action net/sched: flower: Move filter handle initialization earlier net/sched: cls_api: Support hardware miss to tc action net/sched: Rename user cookie and act cookie sfc: fix builds without CONFIG_RTC_LIB sfc: clean up some inconsistent indentings net/mlx4_en: Introduce flexible array to silence overflow warning net: lan966x: Fix possible deadlock inside PTP net/ulp: Remove redundant ->clone() test in inet_clone_ulp(). ...
Diffstat (limited to 'sound/oss/dmasound/dmasound_paula.c')
-rw-r--r--sound/oss/dmasound/dmasound_paula.c739
1 files changed, 739 insertions, 0 deletions
diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c
new file mode 100644
index 000000000..23cf8284c
--- /dev/null
+++ b/sound/oss/dmasound/dmasound_paula.c
@@ -0,0 +1,739 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * linux/sound/oss/dmasound/dmasound_paula.c
+ *
+ * Amiga `Paula' DMA Sound Driver
+ *
+ * See linux/sound/oss/dmasound/dmasound_core.c for copyright and credits
+ * prior to 28/01/2001
+ *
+ * 28/01/2001 [0.1] Iain Sandoe
+ * - added versioning
+ * - put in and populated the hardware_afmts field.
+ * [0.2] - put in SNDCTL_DSP_GETCAPS value.
+ * [0.3] - put in constraint on state buffer usage.
+ * [0.4] - put in default hard/soft settings
+*/
+
+
+#include <linux/module.h>
+#include <linux/mm.h>
+#include <linux/init.h>
+#include <linux/ioport.h>
+#include <linux/soundcard.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <linux/uaccess.h>
+#include <asm/setup.h>
+#include <asm/amigahw.h>
+#include <asm/amigaints.h>
+#include <asm/machdep.h>
+
+#include "dmasound.h"
+
+#define DMASOUND_PAULA_REVISION 0
+#define DMASOUND_PAULA_EDITION 4
+
+#define custom amiga_custom
+ /*
+ * The minimum period for audio depends on htotal (for OCS/ECS/AGA)
+ * (Imported from arch/m68k/amiga/amisound.c)
+ */
+
+extern volatile u_short amiga_audio_min_period;
+
+
+ /*
+ * amiga_mksound() should be able to restore the period after beeping
+ * (Imported from arch/m68k/amiga/amisound.c)
+ */
+
+extern u_short amiga_audio_period;
+
+
+ /*
+ * Audio DMA masks
+ */
+
+#define AMI_AUDIO_OFF (DMAF_AUD0 | DMAF_AUD1 | DMAF_AUD2 | DMAF_AUD3)
+#define AMI_AUDIO_8 (DMAF_SETCLR | DMAF_MASTER | DMAF_AUD0 | DMAF_AUD1)
+#define AMI_AUDIO_14 (AMI_AUDIO_8 | DMAF_AUD2 | DMAF_AUD3)
+
+
+ /*
+ * Helper pointers for 16(14)-bit sound
+ */
+
+static int write_sq_block_size_half, write_sq_block_size_quarter;
+
+
+/*** Low level stuff *********************************************************/
+
+
+static void *AmiAlloc(unsigned int size, gfp_t flags);
+static void AmiFree(void *obj, unsigned int size);
+static int AmiIrqInit(void);
+#ifdef MODULE
+static void AmiIrqCleanUp(void);
+#endif
+static void AmiSilence(void);
+static void AmiInit(void);
+static int AmiSetFormat(int format);
+static int AmiSetVolume(int volume);
+static int AmiSetTreble(int treble);
+static void AmiPlayNextFrame(int index);
+static void AmiPlay(void);
+static irqreturn_t AmiInterrupt(int irq, void *dummy);
+
+#ifdef CONFIG_HEARTBEAT
+
+ /*
+ * Heartbeat interferes with sound since the 7 kHz low-pass filter and the
+ * power LED are controlled by the same line.
+ */
+
+static void (*saved_heartbeat)(int) = NULL;
+
+static inline void disable_heartbeat(void)
+{
+ if (mach_heartbeat) {
+ saved_heartbeat = mach_heartbeat;
+ mach_heartbeat = NULL;
+ }
+ AmiSetTreble(dmasound.treble);
+}
+
+static inline void enable_heartbeat(void)
+{
+ if (saved_heartbeat)
+ mach_heartbeat = saved_heartbeat;
+}
+#else /* !CONFIG_HEARTBEAT */
+#define disable_heartbeat() do { } while (0)
+#define enable_heartbeat() do { } while (0)
+#endif /* !CONFIG_HEARTBEAT */
+
+
+/*** Mid level stuff *********************************************************/
+
+static void AmiMixerInit(void);
+static int AmiMixerIoctl(u_int cmd, u_long arg);
+static int AmiWriteSqSetup(void);
+static int AmiStateInfo(char *buffer, size_t space);
+
+
+/*** Translations ************************************************************/
+
+/* ++TeSche: radically changed for new expanding purposes...
+ *
+ * These two routines now deal with copying/expanding/translating the samples
+ * from user space into our buffer at the right frequency. They take care about
+ * how much data there's actually to read, how much buffer space there is and
+ * to convert samples into the right frequency/encoding. They will only work on
+ * complete samples so it may happen they leave some bytes in the input stream
+ * if the user didn't write a multiple of the current sample size. They both
+ * return the number of bytes they've used from both streams so you may detect
+ * such a situation. Luckily all programs should be able to cope with that.
+ *
+ * I think I've optimized anything as far as one can do in plain C, all
+ * variables should fit in registers and the loops are really short. There's
+ * one loop for every possible situation. Writing a more generalized and thus
+ * parameterized loop would only produce slower code. Feel free to optimize
+ * this in assembler if you like. :)
+ *
+ * I think these routines belong here because they're not yet really hardware
+ * independent, especially the fact that the Falcon can play 16bit samples
+ * only in stereo is hardcoded in both of them!
+ *
+ * ++geert: split in even more functions (one per format)
+ */
+
+
+ /*
+ * Native format
+ */
+
+static ssize_t ami_ct_s8(const u_char __user *userPtr, size_t userCount,
+ u_char frame[], ssize_t *frameUsed, ssize_t frameLeft)
+{
+ ssize_t count, used;
+
+ if (!dmasound.soft.stereo) {
+ void *p = &frame[*frameUsed];
+ count = min_t(unsigned long, userCount, frameLeft) & ~1;
+ used = count;
+ if (copy_from_user(p, userPtr, count))
+ return -EFAULT;
+ } else {
+ u_char *left = &frame[*frameUsed>>1];
+ u_char *right = left+write_sq_block_size_half;
+ count = min_t(unsigned long, userCount, frameLeft)>>1 & ~1;
+ used = count*2;
+ while (count > 0) {
+ if (get_user(*left++, userPtr++)
+ || get_user(*right++, userPtr++))
+ return -EFAULT;
+ count--;
+ }
+ }
+ *frameUsed += used;
+ return used;
+}
+
+
+ /*
+ * Copy and convert 8 bit data
+ */
+
+#define GENERATE_AMI_CT8(funcname, convsample) \
+static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
+ u_char frame[], ssize_t *frameUsed, \
+ ssize_t frameLeft) \
+{ \
+ ssize_t count, used; \
+ \
+ if (!dmasound.soft.stereo) { \
+ u_char *p = &frame[*frameUsed]; \
+ count = min_t(size_t, userCount, frameLeft) & ~1; \
+ used = count; \
+ while (count > 0) { \
+ u_char data; \
+ if (get_user(data, userPtr++)) \
+ return -EFAULT; \
+ *p++ = convsample(data); \
+ count--; \
+ } \
+ } else { \
+ u_char *left = &frame[*frameUsed>>1]; \
+ u_char *right = left+write_sq_block_size_half; \
+ count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
+ used = count*2; \
+ while (count > 0) { \
+ u_char data; \
+ if (get_user(data, userPtr++)) \
+ return -EFAULT; \
+ *left++ = convsample(data); \
+ if (get_user(data, userPtr++)) \
+ return -EFAULT; \
+ *right++ = convsample(data); \
+ count--; \
+ } \
+ } \
+ *frameUsed += used; \
+ return used; \
+}
+
+#define AMI_CT_ULAW(x) (dmasound_ulaw2dma8[(x)])
+#define AMI_CT_ALAW(x) (dmasound_alaw2dma8[(x)])
+#define AMI_CT_U8(x) ((x) ^ 0x80)
+
+GENERATE_AMI_CT8(ami_ct_ulaw, AMI_CT_ULAW)
+GENERATE_AMI_CT8(ami_ct_alaw, AMI_CT_ALAW)
+GENERATE_AMI_CT8(ami_ct_u8, AMI_CT_U8)
+
+
+ /*
+ * Copy and convert 16 bit data
+ */
+
+#define GENERATE_AMI_CT_16(funcname, convsample) \
+static ssize_t funcname(const u_char __user *userPtr, size_t userCount, \
+ u_char frame[], ssize_t *frameUsed, \
+ ssize_t frameLeft) \
+{ \
+ const u_short __user *ptr = (const u_short __user *)userPtr; \
+ ssize_t count, used; \
+ u_short data; \
+ \
+ if (!dmasound.soft.stereo) { \
+ u_char *high = &frame[*frameUsed>>1]; \
+ u_char *low = high+write_sq_block_size_half; \
+ count = min_t(size_t, userCount, frameLeft)>>1 & ~1; \
+ used = count*2; \
+ while (count > 0) { \
+ if (get_user(data, ptr++)) \
+ return -EFAULT; \
+ data = convsample(data); \
+ *high++ = data>>8; \
+ *low++ = (data>>2) & 0x3f; \
+ count--; \
+ } \
+ } else { \
+ u_char *lefth = &frame[*frameUsed>>2]; \
+ u_char *leftl = lefth+write_sq_block_size_quarter; \
+ u_char *righth = lefth+write_sq_block_size_half; \
+ u_char *rightl = righth+write_sq_block_size_quarter; \
+ count = min_t(size_t, userCount, frameLeft)>>2 & ~1; \
+ used = count*4; \
+ while (count > 0) { \
+ if (get_user(data, ptr++)) \
+ return -EFAULT; \
+ data = convsample(data); \
+ *lefth++ = data>>8; \
+ *leftl++ = (data>>2) & 0x3f; \
+ if (get_user(data, ptr++)) \
+ return -EFAULT; \
+ data = convsample(data); \
+ *righth++ = data>>8; \
+ *rightl++ = (data>>2) & 0x3f; \
+ count--; \
+ } \
+ } \
+ *frameUsed += used; \
+ return used; \
+}
+
+#define AMI_CT_S16BE(x) (x)
+#define AMI_CT_U16BE(x) ((x) ^ 0x8000)
+#define AMI_CT_S16LE(x) (le2be16((x)))
+#define AMI_CT_U16LE(x) (le2be16((x)) ^ 0x8000)
+
+GENERATE_AMI_CT_16(ami_ct_s16be, AMI_CT_S16BE)
+GENERATE_AMI_CT_16(ami_ct_u16be, AMI_CT_U16BE)
+GENERATE_AMI_CT_16(ami_ct_s16le, AMI_CT_S16LE)
+GENERATE_AMI_CT_16(ami_ct_u16le, AMI_CT_U16LE)
+
+
+static TRANS transAmiga = {
+ .ct_ulaw = ami_ct_ulaw,
+ .ct_alaw = ami_ct_alaw,
+ .ct_s8 = ami_ct_s8,
+ .ct_u8 = ami_ct_u8,
+ .ct_s16be = ami_ct_s16be,
+ .ct_u16be = ami_ct_u16be,
+ .ct_s16le = ami_ct_s16le,
+ .ct_u16le = ami_ct_u16le,
+};
+
+/*** Low level stuff *********************************************************/
+
+static inline void StopDMA(void)
+{
+ custom.aud[0].audvol = custom.aud[1].audvol = 0;
+ custom.aud[2].audvol = custom.aud[3].audvol = 0;
+ custom.dmacon = AMI_AUDIO_OFF;
+ enable_heartbeat();
+}
+
+static void *AmiAlloc(unsigned int size, gfp_t flags)
+{
+ return amiga_chip_alloc((long)size, "dmasound [Paula]");
+}
+
+static void AmiFree(void *obj, unsigned int size)
+{
+ amiga_chip_free (obj);
+}
+
+static int __init AmiIrqInit(void)
+{
+ /* turn off DMA for audio channels */
+ StopDMA();
+
+ /* Register interrupt handler. */
+ if (request_irq(IRQ_AMIGA_AUD0, AmiInterrupt, 0, "DMA sound",
+ AmiInterrupt))
+ return 0;
+ return 1;
+}
+
+#ifdef MODULE
+static void AmiIrqCleanUp(void)
+{
+ /* turn off DMA for audio channels */
+ StopDMA();
+ /* release the interrupt */
+ free_irq(IRQ_AMIGA_AUD0, AmiInterrupt);
+}
+#endif /* MODULE */
+
+static void AmiSilence(void)
+{
+ /* turn off DMA for audio channels */
+ StopDMA();
+}
+
+
+static void AmiInit(void)
+{
+ int period, i;
+
+ AmiSilence();
+
+ if (dmasound.soft.speed)
+ period = amiga_colorclock/dmasound.soft.speed-1;
+ else
+ period = amiga_audio_min_period;
+ dmasound.hard = dmasound.soft;
+ dmasound.trans_write = &transAmiga;
+
+ if (period < amiga_audio_min_period) {
+ /* we would need to squeeze the sound, but we won't do that */
+ period = amiga_audio_min_period;
+ } else if (period > 65535) {
+ period = 65535;
+ }
+ dmasound.hard.speed = amiga_colorclock/(period+1);
+
+ for (i = 0; i < 4; i++)
+ custom.aud[i].audper = period;
+ amiga_audio_period = period;
+}
+
+
+static int AmiSetFormat(int format)
+{
+ int size;
+
+ /* Amiga sound DMA supports 8bit and 16bit (pseudo 14 bit) modes */
+
+ switch (format) {
+ case AFMT_QUERY:
+ return dmasound.soft.format;
+ case AFMT_MU_LAW:
+ case AFMT_A_LAW:
+ case AFMT_U8:
+ case AFMT_S8:
+ size = 8;
+ break;
+ case AFMT_S16_BE:
+ case AFMT_U16_BE:
+ case AFMT_S16_LE:
+ case AFMT_U16_LE:
+ size = 16;
+ break;
+ default: /* :-) */
+ size = 8;
+ format = AFMT_S8;
+ }
+
+ dmasound.soft.format = format;
+ dmasound.soft.size = size;
+ if (dmasound.minDev == SND_DEV_DSP) {
+ dmasound.dsp.format = format;
+ dmasound.dsp.size = dmasound.soft.size;
+ }
+ AmiInit();
+
+ return format;
+}
+
+
+#define VOLUME_VOXWARE_TO_AMI(v) \
+ (((v) < 0) ? 0 : ((v) > 100) ? 64 : ((v) * 64)/100)
+#define VOLUME_AMI_TO_VOXWARE(v) ((v)*100/64)
+
+static int AmiSetVolume(int volume)
+{
+ dmasound.volume_left = VOLUME_VOXWARE_TO_AMI(volume & 0xff);
+ custom.aud[0].audvol = dmasound.volume_left;
+ dmasound.volume_right = VOLUME_VOXWARE_TO_AMI((volume & 0xff00) >> 8);
+ custom.aud[1].audvol = dmasound.volume_right;
+ if (dmasound.hard.size == 16) {
+ if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
+ custom.aud[2].audvol = 1;
+ custom.aud[3].audvol = 1;
+ } else {
+ custom.aud[2].audvol = 0;
+ custom.aud[3].audvol = 0;
+ }
+ }
+ return VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
+ (VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
+}
+
+static int AmiSetTreble(int treble)
+{
+ dmasound.treble = treble;
+ if (treble < 50)
+ ciaa.pra &= ~0x02;
+ else
+ ciaa.pra |= 0x02;
+ return treble;
+}
+
+
+#define AMI_PLAY_LOADED 1
+#define AMI_PLAY_PLAYING 2
+#define AMI_PLAY_MASK 3
+
+
+static void AmiPlayNextFrame(int index)
+{
+ u_char *start, *ch0, *ch1, *ch2, *ch3;
+ u_long size;
+
+ /* used by AmiPlay() if all doubts whether there really is something
+ * to be played are already wiped out.
+ */
+ start = write_sq.buffers[write_sq.front];
+ size = (write_sq.count == index ? write_sq.rear_size
+ : write_sq.block_size)>>1;
+
+ if (dmasound.hard.stereo) {
+ ch0 = start;
+ ch1 = start+write_sq_block_size_half;
+ size >>= 1;
+ } else {
+ ch0 = start;
+ ch1 = start;
+ }
+
+ disable_heartbeat();
+ custom.aud[0].audvol = dmasound.volume_left;
+ custom.aud[1].audvol = dmasound.volume_right;
+ if (dmasound.hard.size == 8) {
+ custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
+ custom.aud[0].audlen = size;
+ custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
+ custom.aud[1].audlen = size;
+ custom.dmacon = AMI_AUDIO_8;
+ } else {
+ size >>= 1;
+ custom.aud[0].audlc = (u_short *)ZTWO_PADDR(ch0);
+ custom.aud[0].audlen = size;
+ custom.aud[1].audlc = (u_short *)ZTWO_PADDR(ch1);
+ custom.aud[1].audlen = size;
+ if (dmasound.volume_left == 64 && dmasound.volume_right == 64) {
+ /* We can play pseudo 14-bit only with the maximum volume */
+ ch3 = ch0+write_sq_block_size_quarter;
+ ch2 = ch1+write_sq_block_size_quarter;
+ custom.aud[2].audvol = 1; /* we are being affected by the beeps */
+ custom.aud[3].audvol = 1; /* restoring volume here helps a bit */
+ custom.aud[2].audlc = (u_short *)ZTWO_PADDR(ch2);
+ custom.aud[2].audlen = size;
+ custom.aud[3].audlc = (u_short *)ZTWO_PADDR(ch3);
+ custom.aud[3].audlen = size;
+ custom.dmacon = AMI_AUDIO_14;
+ } else {
+ custom.aud[2].audvol = 0;
+ custom.aud[3].audvol = 0;
+ custom.dmacon = AMI_AUDIO_8;
+ }
+ }
+ write_sq.front = (write_sq.front+1) % write_sq.max_count;
+ write_sq.active |= AMI_PLAY_LOADED;
+}
+
+
+static void AmiPlay(void)
+{
+ int minframes = 1;
+
+ custom.intena = IF_AUD0;
+
+ if (write_sq.active & AMI_PLAY_LOADED) {
+ /* There's already a frame loaded */
+ custom.intena = IF_SETCLR | IF_AUD0;
+ return;
+ }
+
+ if (write_sq.active & AMI_PLAY_PLAYING)
+ /* Increase threshold: frame 1 is already being played */
+ minframes = 2;
+
+ if (write_sq.count < minframes) {
+ /* Nothing to do */
+ custom.intena = IF_SETCLR | IF_AUD0;
+ return;
+ }
+
+ if (write_sq.count <= minframes &&
+ write_sq.rear_size < write_sq.block_size && !write_sq.syncing) {
+ /* hmmm, the only existing frame is not
+ * yet filled and we're not syncing?
+ */
+ custom.intena = IF_SETCLR | IF_AUD0;
+ return;
+ }
+
+ AmiPlayNextFrame(minframes);
+
+ custom.intena = IF_SETCLR | IF_AUD0;
+}
+
+
+static irqreturn_t AmiInterrupt(int irq, void *dummy)
+{
+ int minframes = 1;
+
+ custom.intena = IF_AUD0;
+
+ if (!write_sq.active) {
+ /* Playing was interrupted and sq_reset() has already cleared
+ * the sq variables, so better don't do anything here.
+ */
+ WAKE_UP(write_sq.sync_queue);
+ return IRQ_HANDLED;
+ }
+
+ if (write_sq.active & AMI_PLAY_PLAYING) {
+ /* We've just finished a frame */
+ write_sq.count--;
+ WAKE_UP(write_sq.action_queue);
+ }
+
+ if (write_sq.active & AMI_PLAY_LOADED)
+ /* Increase threshold: frame 1 is already being played */
+ minframes = 2;
+
+ /* Shift the flags */
+ write_sq.active = (write_sq.active<<1) & AMI_PLAY_MASK;
+
+ if (!write_sq.active)
+ /* No frame is playing, disable audio DMA */
+ StopDMA();
+
+ custom.intena = IF_SETCLR | IF_AUD0;
+
+ if (write_sq.count >= minframes)
+ /* Try to play the next frame */
+ AmiPlay();
+
+ if (!write_sq.active)
+ /* Nothing to play anymore.
+ Wake up a process waiting for audio output to drain. */
+ WAKE_UP(write_sq.sync_queue);
+ return IRQ_HANDLED;
+}
+
+/*** Mid level stuff *********************************************************/
+
+
+/*
+ * /dev/mixer abstraction
+ */
+
+static void __init AmiMixerInit(void)
+{
+ dmasound.volume_left = 64;
+ dmasound.volume_right = 64;
+ custom.aud[0].audvol = dmasound.volume_left;
+ custom.aud[3].audvol = 1; /* For pseudo 14bit */
+ custom.aud[1].audvol = dmasound.volume_right;
+ custom.aud[2].audvol = 1; /* For pseudo 14bit */
+ dmasound.treble = 50;
+}
+
+static int AmiMixerIoctl(u_int cmd, u_long arg)
+{
+ int data;
+ switch (cmd) {
+ case SOUND_MIXER_READ_DEVMASK:
+ return IOCTL_OUT(arg, SOUND_MASK_VOLUME | SOUND_MASK_TREBLE);
+ case SOUND_MIXER_READ_RECMASK:
+ return IOCTL_OUT(arg, 0);
+ case SOUND_MIXER_READ_STEREODEVS:
+ return IOCTL_OUT(arg, SOUND_MASK_VOLUME);
+ case SOUND_MIXER_READ_VOLUME:
+ return IOCTL_OUT(arg,
+ VOLUME_AMI_TO_VOXWARE(dmasound.volume_left) |
+ VOLUME_AMI_TO_VOXWARE(dmasound.volume_right) << 8);
+ case SOUND_MIXER_WRITE_VOLUME:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_volume(data));
+ case SOUND_MIXER_READ_TREBLE:
+ return IOCTL_OUT(arg, dmasound.treble);
+ case SOUND_MIXER_WRITE_TREBLE:
+ IOCTL_IN(arg, data);
+ return IOCTL_OUT(arg, dmasound_set_treble(data));
+ }
+ return -EINVAL;
+}
+
+
+static int AmiWriteSqSetup(void)
+{
+ write_sq_block_size_half = write_sq.block_size>>1;
+ write_sq_block_size_quarter = write_sq_block_size_half>>1;
+ return 0;
+}
+
+
+static int AmiStateInfo(char *buffer, size_t space)
+{
+ int len = 0;
+ len += sprintf(buffer+len, "\tsound.volume_left = %d [0...64]\n",
+ dmasound.volume_left);
+ len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
+ dmasound.volume_right);
+ if (len >= space) {
+ printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
+ len = space ;
+ }
+ return len;
+}
+
+
+/*** Machine definitions *****************************************************/
+
+static SETTINGS def_hard = {
+ .format = AFMT_S8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 8000
+} ;
+
+static SETTINGS def_soft = {
+ .format = AFMT_U8,
+ .stereo = 0,
+ .size = 8,
+ .speed = 8000
+} ;
+
+static MACHINE machAmiga = {
+ .name = "Amiga",
+ .name2 = "AMIGA",
+ .owner = THIS_MODULE,
+ .dma_alloc = AmiAlloc,
+ .dma_free = AmiFree,
+ .irqinit = AmiIrqInit,
+#ifdef MODULE
+ .irqcleanup = AmiIrqCleanUp,
+#endif /* MODULE */
+ .init = AmiInit,
+ .silence = AmiSilence,
+ .setFormat = AmiSetFormat,
+ .setVolume = AmiSetVolume,
+ .setTreble = AmiSetTreble,
+ .play = AmiPlay,
+ .mixer_init = AmiMixerInit,
+ .mixer_ioctl = AmiMixerIoctl,
+ .write_sq_setup = AmiWriteSqSetup,
+ .state_info = AmiStateInfo,
+ .min_dsp_speed = 8000,
+ .version = ((DMASOUND_PAULA_REVISION<<8) | DMASOUND_PAULA_EDITION),
+ .hardware_afmts = (AFMT_S8 | AFMT_S16_BE), /* h'ware-supported formats *only* here */
+ .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */
+};
+
+
+/*** Config & Setup **********************************************************/
+
+
+static int __init amiga_audio_probe(struct platform_device *pdev)
+{
+ dmasound.mach = machAmiga;
+ dmasound.mach.default_hard = def_hard ;
+ dmasound.mach.default_soft = def_soft ;
+ return dmasound_init();
+}
+
+static int __exit amiga_audio_remove(struct platform_device *pdev)
+{
+ dmasound_deinit();
+ return 0;
+}
+
+static struct platform_driver amiga_audio_driver = {
+ .remove = __exit_p(amiga_audio_remove),
+ .driver = {
+ .name = "amiga-audio",
+ },
+};
+
+module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe);
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:amiga-audio");