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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/soc/qcom/qdsp6/q6asm-dai.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/soc/qcom/qdsp6/q6asm-dai.c')
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c1327
1 files changed, 1327 insertions, 0 deletions
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
new file mode 100644
index 000000000..5fc8088e6
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -0,0 +1,1327 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <linux/spinlock.h>
+#include <sound/compress_driver.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/of_device.h>
+#include <sound/pcm_params.h>
+#include "q6asm.h"
+#include "q6routing.h"
+#include "q6dsp-errno.h"
+
+#define DRV_NAME "q6asm-fe-dai"
+
+#define PLAYBACK_MIN_NUM_PERIODS 2
+#define PLAYBACK_MAX_NUM_PERIODS 8
+#define PLAYBACK_MAX_PERIOD_SIZE 65536
+#define PLAYBACK_MIN_PERIOD_SIZE 128
+#define CAPTURE_MIN_NUM_PERIODS 2
+#define CAPTURE_MAX_NUM_PERIODS 8
+#define CAPTURE_MAX_PERIOD_SIZE 4096
+#define CAPTURE_MIN_PERIOD_SIZE 320
+#define SID_MASK_DEFAULT 0xF
+
+/* Default values used if user space does not set */
+#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
+#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
+#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
+#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
+
+#define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1)
+#define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
+
+enum stream_state {
+ Q6ASM_STREAM_IDLE = 0,
+ Q6ASM_STREAM_STOPPED,
+ Q6ASM_STREAM_RUNNING,
+};
+
+struct q6asm_dai_rtd {
+ struct snd_pcm_substream *substream;
+ struct snd_compr_stream *cstream;
+ struct snd_codec codec;
+ struct snd_dma_buffer dma_buffer;
+ spinlock_t lock;
+ phys_addr_t phys;
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int periods;
+ unsigned int bytes_sent;
+ unsigned int bytes_received;
+ unsigned int copied_total;
+ uint16_t bits_per_sample;
+ uint16_t source; /* Encoding source bit mask */
+ struct audio_client *audio_client;
+ uint32_t next_track_stream_id;
+ bool next_track;
+ uint32_t stream_id;
+ uint16_t session_id;
+ enum stream_state state;
+ uint32_t initial_samples_drop;
+ uint32_t trailing_samples_drop;
+ bool notify_on_drain;
+};
+
+struct q6asm_dai_data {
+ struct snd_soc_dai_driver *dais;
+ int num_dais;
+ long long int sid;
+};
+
+static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 4,
+ .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS *
+ CAPTURE_MAX_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE,
+ .periods_min = CAPTURE_MIN_NUM_PERIODS,
+ .periods_max = CAPTURE_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS *
+ PLAYBACK_MAX_PERIOD_SIZE),
+ .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
+ .periods_min = PLAYBACK_MIN_NUM_PERIODS,
+ .periods_max = PLAYBACK_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+#define Q6ASM_FEDAI_DRIVER(num) { \
+ .playback = { \
+ .stream_name = "MultiMedia"#num" Playback", \
+ .rates = (SNDRV_PCM_RATE_8000_192000| \
+ SNDRV_PCM_RATE_KNOT), \
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE), \
+ .channels_min = 1, \
+ .channels_max = 8, \
+ .rate_min = 8000, \
+ .rate_max = 192000, \
+ }, \
+ .capture = { \
+ .stream_name = "MultiMedia"#num" Capture", \
+ .rates = (SNDRV_PCM_RATE_8000_48000| \
+ SNDRV_PCM_RATE_KNOT), \
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE), \
+ .channels_min = 1, \
+ .channels_max = 4, \
+ .rate_min = 8000, \
+ .rate_max = 48000, \
+ }, \
+ .name = "MultiMedia"#num, \
+ .id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
+ }
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+ 88200, 96000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static const struct snd_compr_codec_caps q6asm_compr_caps = {
+ .num_descriptors = 1,
+ .descriptor[0].max_ch = 2,
+ .descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050,
+ 24000, 32000, 44100, 48000, 88200,
+ 96000, 176400, 192000 },
+ .descriptor[0].num_sample_rates = 13,
+ .descriptor[0].bit_rate[0] = 320,
+ .descriptor[0].bit_rate[1] = 128,
+ .descriptor[0].num_bitrates = 2,
+ .descriptor[0].profiles = 0,
+ .descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
+ .descriptor[0].formats = 0,
+};
+
+static void event_handler(uint32_t opcode, uint32_t token,
+ void *payload, void *priv)
+{
+ struct q6asm_dai_rtd *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+
+ switch (opcode) {
+ case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
+ break;
+ case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ break;
+ case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ snd_pcm_period_elapsed(substream);
+ if (prtd->state == Q6ASM_STREAM_RUNNING)
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
+
+ break;
+ }
+ case ASM_CLIENT_EVENT_DATA_READ_DONE:
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ snd_pcm_period_elapsed(substream);
+ if (prtd->state == Q6ASM_STREAM_RUNNING)
+ q6asm_read(prtd->audio_client, prtd->stream_id);
+
+ break;
+ default:
+ break;
+ }
+}
+
+static int q6asm_dai_prepare(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
+ int ret, i;
+
+ pdata = snd_soc_component_get_drvdata(component);
+ if (!pdata)
+ return -EINVAL;
+
+ if (!prtd || !prtd->audio_client) {
+ dev_err(dev, "%s: private data null or audio client freed\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ if (prtd->state) {
+ /* clear the previous setup if any */
+ q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6routing_stream_close(soc_prtd->dai_link->id,
+ substream->stream);
+ }
+
+ ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
+ prtd->phys,
+ (prtd->pcm_size / prtd->periods),
+ prtd->periods);
+
+ if (ret < 0) {
+ dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n",
+ ret);
+ return -ENOMEM;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
+ 0, prtd->bits_per_sample, false);
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = q6asm_open_read(prtd->audio_client, prtd->stream_id,
+ FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
+ }
+
+ if (ret < 0) {
+ dev_err(dev, "%s: q6asm_open_write failed\n", __func__);
+ goto open_err;
+ }
+
+ prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+ ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
+ prtd->session_id, substream->stream);
+ if (ret) {
+ dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);
+ goto routing_err;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_media_format_block_multi_ch_pcm(
+ prtd->audio_client, prtd->stream_id,
+ runtime->rate, runtime->channels, NULL,
+ prtd->bits_per_sample);
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
+ prtd->stream_id,
+ runtime->rate,
+ runtime->channels,
+ prtd->bits_per_sample);
+
+ /* Queue the buffers */
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client, prtd->stream_id);
+
+ }
+ if (ret < 0)
+ dev_info(dev, "%s: CMD Format block failed\n", __func__);
+ else
+ prtd->state = Q6ASM_STREAM_RUNNING;
+
+ return ret;
+
+routing_err:
+ q6asm_cmd(prtd->audio_client, prtd->stream_id, CMD_CLOSE);
+open_err:
+ q6asm_unmap_memory_regions(substream->stream, prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+
+ return ret;
+}
+
+static int q6asm_dai_trigger(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_PAUSE);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_open(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
+ struct q6asm_dai_rtd *prtd;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
+ int ret = 0;
+ int stream_id;
+
+ stream_id = cpu_dai->driver->id;
+
+ pdata = snd_soc_component_get_drvdata(component);
+ if (!pdata) {
+ dev_err(dev, "Drv data not found ..\n");
+ return -EINVAL;
+ }
+
+ prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(dev,
+ (q6asm_cb)event_handler, prtd, stream_id,
+ LEGACY_PCM_MODE);
+ if (IS_ERR(prtd->audio_client)) {
+ dev_info(dev, "%s: Could not allocate memory\n", __func__);
+ ret = PTR_ERR(prtd->audio_client);
+ kfree(prtd);
+ return ret;
+ }
+
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ runtime->hw = q6asm_dai_hardware_playback;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ runtime->hw = q6asm_dai_hardware_capture;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ dev_info(dev, "snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
+ PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
+ if (ret < 0) {
+ dev_err(dev, "constraint for buffer bytes min max ret = %d\n",
+ ret);
+ }
+ }
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret < 0) {
+ dev_err(dev, "constraint for period bytes step ret = %d\n",
+ ret);
+ }
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret < 0) {
+ dev_err(dev, "constraint for buffer bytes step ret = %d\n",
+ ret);
+ }
+
+ runtime->private_data = prtd;
+
+ snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
+
+ runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
+
+
+ if (pdata->sid < 0)
+ prtd->phys = substream->dma_buffer.addr;
+ else
+ prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
+
+ return 0;
+}
+
+static int q6asm_dai_close(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->audio_client) {
+ if (prtd->state)
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
+
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ }
+ q6routing_stream_close(soc_prtd->dai_link->id,
+ substream->stream);
+ kfree(prtd);
+ return 0;
+}
+
+static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int q6asm_dai_hw_params(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ prtd->pcm_size = params_buffer_bytes(params);
+ prtd->periods = params_periods(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ prtd->bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ prtd->bits_per_sample = 24;
+ break;
+ }
+
+ return 0;
+}
+
+static void compress_event_handler(uint32_t opcode, uint32_t token,
+ void *payload, void *priv)
+{
+ struct q6asm_dai_rtd *prtd = priv;
+ struct snd_compr_stream *substream = prtd->cstream;
+ unsigned long flags;
+ u32 wflags = 0;
+ uint64_t avail;
+ uint32_t bytes_written, bytes_to_write;
+ bool is_last_buffer = false;
+
+ switch (opcode) {
+ case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (!prtd->bytes_sent) {
+ q6asm_stream_remove_initial_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->initial_samples_drop);
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ prtd->pcm_count, 0, 0, 0);
+ prtd->bytes_sent += prtd->pcm_count;
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (prtd->notify_on_drain) {
+ if (substream->partial_drain) {
+ /*
+ * Close old stream and make it stale, switch
+ * the active stream now!
+ */
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id,
+ CMD_CLOSE);
+ /*
+ * vaild stream ids start from 1, So we are
+ * toggling this between 1 and 2.
+ */
+ prtd->stream_id = (prtd->stream_id == 1 ? 2 : 1);
+ }
+
+ snd_compr_drain_notify(prtd->cstream);
+ prtd->notify_on_drain = false;
+
+ } else {
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ case ASM_CLIENT_EVENT_DATA_WRITE_DONE:
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ bytes_written = token >> ASM_WRITE_TOKEN_LEN_SHIFT;
+ prtd->copied_total += bytes_written;
+ snd_compr_fragment_elapsed(substream);
+
+ if (prtd->state != Q6ASM_STREAM_RUNNING) {
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+
+ avail = prtd->bytes_received - prtd->bytes_sent;
+ if (avail > prtd->pcm_count) {
+ bytes_to_write = prtd->pcm_count;
+ } else {
+ if (substream->partial_drain || prtd->notify_on_drain)
+ is_last_buffer = true;
+ bytes_to_write = avail;
+ }
+
+ if (bytes_to_write) {
+ if (substream->partial_drain && is_last_buffer) {
+ wflags |= ASM_LAST_BUFFER_FLAG;
+ q6asm_stream_remove_trailing_silence(prtd->audio_client,
+ prtd->stream_id,
+ prtd->trailing_samples_drop);
+ }
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ bytes_to_write, 0, 0, wflags);
+
+ prtd->bytes_sent += bytes_to_write;
+ }
+
+ if (prtd->notify_on_drain && is_last_buffer)
+ q6asm_cmd_nowait(prtd->audio_client,
+ prtd->stream_id, CMD_EOS);
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ default:
+ break;
+ }
+}
+
+static int q6asm_dai_compr_open(struct snd_soc_component *component,
+ struct snd_compr_stream *stream)
+{
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
+ struct q6asm_dai_rtd *prtd;
+ int stream_id, size, ret;
+
+ stream_id = cpu_dai->driver->id;
+ pdata = snd_soc_component_get_drvdata(component);
+ if (!pdata) {
+ dev_err(dev, "Drv data not found ..\n");
+ return -EINVAL;
+ }
+
+ prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
+ if (!prtd)
+ return -ENOMEM;
+
+ /* DSP expects stream id from 1 */
+ prtd->stream_id = 1;
+
+ prtd->cstream = stream;
+ prtd->audio_client = q6asm_audio_client_alloc(dev,
+ (q6asm_cb)compress_event_handler,
+ prtd, stream_id, LEGACY_PCM_MODE);
+ if (IS_ERR(prtd->audio_client)) {
+ dev_err(dev, "Could not allocate memory\n");
+ ret = PTR_ERR(prtd->audio_client);
+ goto free_prtd;
+ }
+
+ size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE *
+ COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+ &prtd->dma_buffer);
+ if (ret) {
+ dev_err(dev, "Cannot allocate buffer(s)\n");
+ goto free_client;
+ }
+
+ if (pdata->sid < 0)
+ prtd->phys = prtd->dma_buffer.addr;
+ else
+ prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
+
+ snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
+ spin_lock_init(&prtd->lock);
+ runtime->private_data = prtd;
+
+ return 0;
+
+free_client:
+ q6asm_audio_client_free(prtd->audio_client);
+free_prtd:
+ kfree(prtd);
+
+ return ret;
+}
+
+static int q6asm_dai_compr_free(struct snd_soc_component *component,
+ struct snd_compr_stream *stream)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+
+ if (prtd->audio_client) {
+ if (prtd->state) {
+ q6asm_cmd(prtd->audio_client, prtd->stream_id,
+ CMD_CLOSE);
+ if (prtd->next_track_stream_id) {
+ q6asm_cmd(prtd->audio_client,
+ prtd->next_track_stream_id,
+ CMD_CLOSE);
+ }
+ }
+
+ snd_dma_free_pages(&prtd->dma_buffer);
+ q6asm_unmap_memory_regions(stream->direction,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ }
+ q6routing_stream_close(rtd->dai_link->id, stream->direction);
+ kfree(prtd);
+
+ return 0;
+}
+
+static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_codec *codec,
+ int stream_id)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct q6asm_flac_cfg flac_cfg;
+ struct q6asm_wma_cfg wma_cfg;
+ struct q6asm_alac_cfg alac_cfg;
+ struct q6asm_ape_cfg ape_cfg;
+ unsigned int wma_v9 = 0;
+ struct device *dev = component->dev;
+ int ret;
+ union snd_codec_options *codec_options;
+ struct snd_dec_flac *flac;
+ struct snd_dec_wma *wma;
+ struct snd_dec_alac *alac;
+ struct snd_dec_ape *ape;
+
+ codec_options = &(prtd->codec.options);
+
+ memcpy(&prtd->codec, codec, sizeof(*codec));
+
+ switch (codec->id) {
+ case SND_AUDIOCODEC_FLAC:
+
+ memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg));
+ flac = &codec_options->flac_d;
+
+ flac_cfg.ch_cfg = codec->ch_in;
+ flac_cfg.sample_rate = codec->sample_rate;
+ flac_cfg.stream_info_present = 1;
+ flac_cfg.sample_size = flac->sample_size;
+ flac_cfg.min_blk_size = flac->min_blk_size;
+ flac_cfg.max_blk_size = flac->max_blk_size;
+ flac_cfg.max_frame_size = flac->max_frame_size;
+ flac_cfg.min_frame_size = flac->min_frame_size;
+
+ ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+ stream_id,
+ &flac_cfg);
+ if (ret < 0) {
+ dev_err(dev, "FLAC CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_WMA:
+ wma = &codec_options->wma_d;
+
+ memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg));
+
+ wma_cfg.sample_rate = codec->sample_rate;
+ wma_cfg.num_channels = codec->ch_in;
+ wma_cfg.bytes_per_sec = codec->bit_rate / 8;
+ wma_cfg.block_align = codec->align;
+ wma_cfg.bits_per_sample = prtd->bits_per_sample;
+ wma_cfg.enc_options = wma->encoder_option;
+ wma_cfg.adv_enc_options = wma->adv_encoder_option;
+ wma_cfg.adv_enc_options2 = wma->adv_encoder_option2;
+
+ if (wma_cfg.num_channels == 1)
+ wma_cfg.channel_mask = 4; /* Mono Center */
+ else if (wma_cfg.num_channels == 2)
+ wma_cfg.channel_mask = 3; /* Stereo FL/FR */
+ else
+ return -EINVAL;
+
+ /* check the codec profile */
+ switch (codec->profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ wma_cfg.fmtag = 0x161;
+ wma_v9 = 1;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10:
+ wma_cfg.fmtag = 0x166;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ wma_cfg.fmtag = 0x162;
+ break;
+
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ wma_cfg.fmtag = 0x163;
+ break;
+
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ wma_cfg.fmtag = 0x167;
+ break;
+
+ default:
+ dev_err(dev, "Unknown WMA profile:%x\n",
+ codec->profile);
+ return -EIO;
+ }
+
+ if (wma_v9)
+ ret = q6asm_stream_media_format_block_wma_v9(
+ prtd->audio_client, stream_id,
+ &wma_cfg);
+ else
+ ret = q6asm_stream_media_format_block_wma_v10(
+ prtd->audio_client, stream_id,
+ &wma_cfg);
+ if (ret < 0) {
+ dev_err(dev, "WMA9 CMD failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_ALAC:
+ memset(&alac_cfg, 0x0, sizeof(alac_cfg));
+ alac = &codec_options->alac_d;
+
+ alac_cfg.sample_rate = codec->sample_rate;
+ alac_cfg.avg_bit_rate = codec->bit_rate;
+ alac_cfg.bit_depth = prtd->bits_per_sample;
+ alac_cfg.num_channels = codec->ch_in;
+
+ alac_cfg.frame_length = alac->frame_length;
+ alac_cfg.pb = alac->pb;
+ alac_cfg.mb = alac->mb;
+ alac_cfg.kb = alac->kb;
+ alac_cfg.max_run = alac->max_run;
+ alac_cfg.compatible_version = alac->compatible_version;
+ alac_cfg.max_frame_bytes = alac->max_frame_bytes;
+
+ switch (codec->ch_in) {
+ case 1:
+ alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO;
+ break;
+ case 2:
+ alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_STEREO;
+ break;
+ }
+ ret = q6asm_stream_media_format_block_alac(prtd->audio_client,
+ stream_id,
+ &alac_cfg);
+ if (ret < 0) {
+ dev_err(dev, "ALAC CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ case SND_AUDIOCODEC_APE:
+ memset(&ape_cfg, 0x0, sizeof(ape_cfg));
+ ape = &codec_options->ape_d;
+
+ ape_cfg.sample_rate = codec->sample_rate;
+ ape_cfg.num_channels = codec->ch_in;
+ ape_cfg.bits_per_sample = prtd->bits_per_sample;
+
+ ape_cfg.compatible_version = ape->compatible_version;
+ ape_cfg.compression_level = ape->compression_level;
+ ape_cfg.format_flags = ape->format_flags;
+ ape_cfg.blocks_per_frame = ape->blocks_per_frame;
+ ape_cfg.final_frame_blocks = ape->final_frame_blocks;
+ ape_cfg.total_frames = ape->total_frames;
+ ape_cfg.seek_table_present = ape->seek_table_present;
+
+ ret = q6asm_stream_media_format_block_ape(prtd->audio_client,
+ stream_id,
+ &ape_cfg);
+ if (ret < 0) {
+ dev_err(dev, "APE CMD Format block failed:%d\n", ret);
+ return -EIO;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static int q6asm_dai_compr_set_params(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_params *params)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = stream->private_data;
+ int dir = stream->direction;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = component->dev;
+ int ret;
+
+ pdata = snd_soc_component_get_drvdata(component);
+ if (!pdata)
+ return -EINVAL;
+
+ if (!prtd || !prtd->audio_client) {
+ dev_err(dev, "private data null or audio client freed\n");
+ return -EINVAL;
+ }
+
+ prtd->periods = runtime->fragments;
+ prtd->pcm_count = runtime->fragment_size;
+ prtd->pcm_size = runtime->fragments * runtime->fragment_size;
+ prtd->bits_per_sample = 16;
+
+ if (dir == SND_COMPRESS_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id,
+ params->codec.profile, prtd->bits_per_sample,
+ true);
+
+ if (ret < 0) {
+ dev_err(dev, "q6asm_open_write failed\n");
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return ret;
+ }
+ }
+
+ prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+ ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE,
+ prtd->session_id, dir);
+ if (ret) {
+ dev_err(dev, "Stream reg failed ret:%d\n", ret);
+ return ret;
+ }
+
+ ret = __q6asm_dai_compr_set_codec_params(component, stream,
+ &params->codec,
+ prtd->stream_id);
+ if (ret) {
+ dev_err(dev, "codec param setup failed ret:%d\n", ret);
+ return ret;
+ }
+
+ ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
+ (prtd->pcm_size / prtd->periods),
+ prtd->periods);
+
+ if (ret < 0) {
+ dev_err(dev, "Buffer Mapping failed ret:%d\n", ret);
+ return -ENOMEM;
+ }
+
+ prtd->state = Q6ASM_STREAM_RUNNING;
+
+ return 0;
+}
+
+static int q6asm_dai_compr_set_metadata(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_metadata *metadata)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ int ret = 0;
+
+ switch (metadata->key) {
+ case SNDRV_COMPRESS_ENCODER_PADDING:
+ prtd->trailing_samples_drop = metadata->value[0];
+ break;
+ case SNDRV_COMPRESS_ENCODER_DELAY:
+ prtd->initial_samples_drop = metadata->value[0];
+ if (prtd->next_track_stream_id) {
+ ret = q6asm_open_write(prtd->audio_client,
+ prtd->next_track_stream_id,
+ prtd->codec.id,
+ prtd->codec.profile,
+ prtd->bits_per_sample,
+ true);
+ if (ret < 0) {
+ dev_err(component->dev, "q6asm_open_write failed\n");
+ return ret;
+ }
+ ret = __q6asm_dai_compr_set_codec_params(component, stream,
+ &prtd->codec,
+ prtd->next_track_stream_id);
+ if (ret < 0) {
+ dev_err(component->dev, "q6asm_open_write failed\n");
+ return ret;
+ }
+
+ ret = q6asm_stream_remove_initial_silence(prtd->audio_client,
+ prtd->next_track_stream_id,
+ prtd->initial_samples_drop);
+ prtd->next_track_stream_id = 0;
+
+ }
+
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
+ struct snd_compr_stream *stream, int cmd)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = q6asm_run_nowait(prtd->audio_client, prtd->stream_id,
+ 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = q6asm_cmd_nowait(prtd->audio_client, prtd->stream_id,
+ CMD_PAUSE);
+ break;
+ case SND_COMPR_TRIGGER_NEXT_TRACK:
+ prtd->next_track = true;
+ prtd->next_track_stream_id = (prtd->stream_id == 1 ? 2 : 1);
+ break;
+ case SND_COMPR_TRIGGER_DRAIN:
+ case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+ prtd->notify_on_drain = true;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_tstamp *tstamp)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ tstamp->copied_total = prtd->copied_total;
+ tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return 0;
+}
+
+static int q6asm_compr_copy(struct snd_soc_component *component,
+ struct snd_compr_stream *stream, char __user *buf,
+ size_t count)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ unsigned long flags;
+ u32 wflags = 0;
+ int avail, bytes_in_flight = 0;
+ void *dstn;
+ size_t copy;
+ u32 app_pointer;
+ u32 bytes_received;
+
+ bytes_received = prtd->bytes_received;
+
+ /**
+ * Make sure that next track data pointer is aligned at 32 bit boundary
+ * This is a Mandatory requirement from DSP data buffers alignment
+ */
+ if (prtd->next_track)
+ bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
+
+ app_pointer = bytes_received/prtd->pcm_size;
+ app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
+ dstn = prtd->dma_buffer.area + app_pointer;
+
+ if (count < prtd->pcm_size - app_pointer) {
+ if (copy_from_user(dstn, buf, count))
+ return -EFAULT;
+ } else {
+ copy = prtd->pcm_size - app_pointer;
+ if (copy_from_user(dstn, buf, copy))
+ return -EFAULT;
+ if (copy_from_user(prtd->dma_buffer.area, buf + copy,
+ count - copy))
+ return -EFAULT;
+ }
+
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ bytes_in_flight = prtd->bytes_received - prtd->copied_total;
+
+ if (prtd->next_track) {
+ prtd->next_track = false;
+ prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
+ prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
+ }
+
+ prtd->bytes_received = bytes_received + count;
+
+ /* Kick off the data to dsp if its starving!! */
+ if (prtd->state == Q6ASM_STREAM_RUNNING && (bytes_in_flight == 0)) {
+ uint32_t bytes_to_write = prtd->pcm_count;
+
+ avail = prtd->bytes_received - prtd->bytes_sent;
+
+ if (avail < prtd->pcm_count)
+ bytes_to_write = avail;
+
+ q6asm_write_async(prtd->audio_client, prtd->stream_id,
+ bytes_to_write, 0, 0, wflags);
+ prtd->bytes_sent += bytes_to_write;
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return count;
+}
+
+static int q6asm_dai_compr_mmap(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct vm_area_struct *vma)
+{
+ struct snd_compr_runtime *runtime = stream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct device *dev = component->dev;
+
+ return dma_mmap_coherent(dev, vma,
+ prtd->dma_buffer.area, prtd->dma_buffer.addr,
+ prtd->dma_buffer.bytes);
+}
+
+static int q6asm_dai_compr_get_caps(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_caps *caps)
+{
+ caps->direction = SND_COMPRESS_PLAYBACK;
+ caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
+ caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
+ caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
+ caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+ caps->num_codecs = 5;
+ caps->codecs[0] = SND_AUDIOCODEC_MP3;
+ caps->codecs[1] = SND_AUDIOCODEC_FLAC;
+ caps->codecs[2] = SND_AUDIOCODEC_WMA;
+ caps->codecs[3] = SND_AUDIOCODEC_ALAC;
+ caps->codecs[4] = SND_AUDIOCODEC_APE;
+
+ return 0;
+}
+
+static int q6asm_dai_compr_get_codec_caps(struct snd_soc_component *component,
+ struct snd_compr_stream *stream,
+ struct snd_compr_codec_caps *codec)
+{
+ switch (codec->codec) {
+ case SND_AUDIOCODEC_MP3:
+ *codec = q6asm_compr_caps;
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+static const struct snd_compress_ops q6asm_dai_compress_ops = {
+ .open = q6asm_dai_compr_open,
+ .free = q6asm_dai_compr_free,
+ .set_params = q6asm_dai_compr_set_params,
+ .set_metadata = q6asm_dai_compr_set_metadata,
+ .pointer = q6asm_dai_compr_pointer,
+ .trigger = q6asm_dai_compr_trigger,
+ .get_caps = q6asm_dai_compr_get_caps,
+ .get_codec_caps = q6asm_dai_compr_get_codec_caps,
+ .mmap = q6asm_dai_compr_mmap,
+ .copy = q6asm_compr_copy,
+};
+
+static int q6asm_dai_pcm_new(struct snd_soc_component *component,
+ struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ size_t size = q6asm_dai_hardware_playback.buffer_bytes_max;
+
+ return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV,
+ component->dev, size);
+}
+
+static const struct snd_soc_dapm_widget q6asm_dapm_widgets[] = {
+ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, SND_SOC_NOPM, 0, 0),
+};
+
+static const struct snd_soc_component_driver q6asm_fe_dai_component = {
+ .name = DRV_NAME,
+ .open = q6asm_dai_open,
+ .hw_params = q6asm_dai_hw_params,
+ .close = q6asm_dai_close,
+ .prepare = q6asm_dai_prepare,
+ .trigger = q6asm_dai_trigger,
+ .pointer = q6asm_dai_pointer,
+ .pcm_construct = q6asm_dai_pcm_new,
+ .compress_ops = &q6asm_dai_compress_ops,
+ .dapm_widgets = q6asm_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(q6asm_dapm_widgets),
+ .legacy_dai_naming = 1,
+};
+
+static struct snd_soc_dai_driver q6asm_fe_dais_template[] = {
+ Q6ASM_FEDAI_DRIVER(1),
+ Q6ASM_FEDAI_DRIVER(2),
+ Q6ASM_FEDAI_DRIVER(3),
+ Q6ASM_FEDAI_DRIVER(4),
+ Q6ASM_FEDAI_DRIVER(5),
+ Q6ASM_FEDAI_DRIVER(6),
+ Q6ASM_FEDAI_DRIVER(7),
+ Q6ASM_FEDAI_DRIVER(8),
+};
+
+static int of_q6asm_parse_dai_data(struct device *dev,
+ struct q6asm_dai_data *pdata)
+{
+ struct snd_soc_dai_driver *dai_drv;
+ struct snd_soc_pcm_stream empty_stream;
+ struct device_node *node;
+ int ret, id, dir, idx = 0;
+
+
+ pdata->num_dais = of_get_child_count(dev->of_node);
+ if (!pdata->num_dais) {
+ dev_err(dev, "No dais found in DT\n");
+ return -EINVAL;
+ }
+
+ pdata->dais = devm_kcalloc(dev, pdata->num_dais, sizeof(*dai_drv),
+ GFP_KERNEL);
+ if (!pdata->dais)
+ return -ENOMEM;
+
+ memset(&empty_stream, 0, sizeof(empty_stream));
+
+ for_each_child_of_node(dev->of_node, node) {
+ ret = of_property_read_u32(node, "reg", &id);
+ if (ret || id >= MAX_SESSIONS || id < 0) {
+ dev_err(dev, "valid dai id not found:%d\n", ret);
+ continue;
+ }
+
+ dai_drv = &pdata->dais[idx++];
+ *dai_drv = q6asm_fe_dais_template[id];
+
+ ret = of_property_read_u32(node, "direction", &dir);
+ if (ret)
+ continue;
+
+ if (dir == Q6ASM_DAI_RX)
+ dai_drv->capture = empty_stream;
+ else if (dir == Q6ASM_DAI_TX)
+ dai_drv->playback = empty_stream;
+
+ if (of_property_read_bool(node, "is-compress-dai"))
+ dai_drv->compress_new = snd_soc_new_compress;
+ }
+
+ return 0;
+}
+
+static int q6asm_dai_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct device_node *node = dev->of_node;
+ struct of_phandle_args args;
+ struct q6asm_dai_data *pdata;
+ int rc;
+
+ pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata)
+ return -ENOMEM;
+
+ rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
+ if (rc < 0)
+ pdata->sid = -1;
+ else
+ pdata->sid = args.args[0] & SID_MASK_DEFAULT;
+
+ dev_set_drvdata(dev, pdata);
+
+ rc = of_q6asm_parse_dai_data(dev, pdata);
+ if (rc)
+ return rc;
+
+ return devm_snd_soc_register_component(dev, &q6asm_fe_dai_component,
+ pdata->dais, pdata->num_dais);
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id q6asm_dai_device_id[] = {
+ { .compatible = "qcom,q6asm-dais" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6asm_dai_device_id);
+#endif
+
+static struct platform_driver q6asm_dai_platform_driver = {
+ .driver = {
+ .name = "q6asm-dai",
+ .of_match_table = of_match_ptr(q6asm_dai_device_id),
+ },
+ .probe = q6asm_dai_probe,
+};
+module_platform_driver(q6asm_dai_platform_driver);
+
+MODULE_DESCRIPTION("Q6ASM dai driver");
+MODULE_LICENSE("GPL v2");