diff options
Diffstat (limited to 'recipes/linux/linux-replicant/htcdream/msm_snd.patch')
| -rw-r--r-- | recipes/linux/linux-replicant/htcdream/msm_snd.patch | 3160 |
1 files changed, 3160 insertions, 0 deletions
diff --git a/recipes/linux/linux-replicant/htcdream/msm_snd.patch b/recipes/linux/linux-replicant/htcdream/msm_snd.patch new file mode 100644 index 0000000000..cdc354d001 --- /dev/null +++ b/recipes/linux/linux-replicant/htcdream/msm_snd.patch @@ -0,0 +1,3160 @@ +commit 92945ccd0921e940fc2675dc394141db0f7c1386 +Author: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org> +Date: Sat Oct 17 23:37:16 2009 +0200 + + Sound: MSM soc : imported alsa for the MSM from codeaurora + + I had to make two little change to make it compile: + snd_ep = msm_rpc_connect_compatible(snd_rpc_ids.prog, + became: + snd_ep = msm_rpc_connect(snd_rpc_ids.prog, + and I also changed snd_rpc_ids.vers(with ifdefs) + to a known and working magick number: + + it still has serious runtime problems such as: + *can produce kernel oops under theses conditions: + start alsamixer and if the second bar is on 0 or 4 it can play music with aplay + increase the routing alsamixer bar to the max + decrease the routing bar to 4 or less + then It may have a null pointer problem + That bug could be because it tries to route to route to speakers and handset + at the same time(SND_DEVICE_HEADSET_AND_SPEAKER in android) + that is to say it could be the same bug than here: + http://gitorious.org/replicant/msm7k/commit/370d37a088368ca8cc478e76c928a1ce6589495e + but I need time to verify that + *can pannick if you send things to /dev/dsp when the oss emulation is activated + *only aplay works(mplayer,gstreamer don't work) + for mplayer an ioctl didn't return...so it get stuck before playing + I traced it until + rc = wait_event_interruptible(the_locks.read_wait,(prtd->in_count > 0)|| prtd->stopped); + in msm-pcm.c + +diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig +index ef025c6..4b1a48f 100644 +--- a/sound/soc/Kconfig ++++ b/sound/soc/Kconfig +@@ -32,6 +32,7 @@ source "sound/soc/omap/Kconfig" + source "sound/soc/pxa/Kconfig" + source "sound/soc/s3c24xx/Kconfig" + source "sound/soc/sh/Kconfig" ++source "sound/soc/msm/Kconfig" + + # Supported codecs + source "sound/soc/codecs/Kconfig" +diff --git a/sound/soc/Makefile b/sound/soc/Makefile +index 86a9b1f..ea754e5 100644 +--- a/sound/soc/Makefile ++++ b/sound/soc/Makefile +@@ -11,3 +11,4 @@ obj-$(CONFIG_SND_SOC) += omap/ + obj-$(CONFIG_SND_SOC) += pxa/ + obj-$(CONFIG_SND_SOC) += s3c24xx/ + obj-$(CONFIG_SND_SOC) += sh/ ++obj-$(CONFIG_SND_SOC) += msm/ +diff --git a/sound/soc/msm/Kconfig b/sound/soc/msm/Kconfig +new file mode 100644 +index 0000000..7df1a40 +--- /dev/null ++++ b/sound/soc/msm/Kconfig +@@ -0,0 +1,37 @@ ++menu "MSM SoC Audio support" ++ ++config SND_MSM_SOC ++ tristate "SoC Audio for the MSM series chips" ++ depends on ARCH_MSM_ARM11 && SND_SOC ++ select MSM_ADSP ++ help ++ To add support for ALSA PCM driver for MSM board. ++ ++config SND_QSD_SOC ++ tristate "SoC Audio for the QSD8x50 chip" ++ depends on ARCH_QSD8X50 && SND_SOC && QSD_AUDIO ++ default y ++ help ++ To add support for ALSA PCM driver for QSD8k board. ++ ++ ++config SND_MSM_DAI_SOC ++ tristate "SoC CPU/CODEC DAI for the MSM chip" ++ depends on SND_MSM_SOC || SND_QSD_SOC ++ help ++ To add support for ALSA PCM driver for MSM board. ++ ++config SND_MSM_SOC_MSM7K ++ tristate "SoC Audio support for MSM7K" ++ depends on SND_MSM_SOC ++ help ++ To add support for SoC audio on msm7k for msm72x1 or msm7x27 ++ ++config SND_QSD_SOC_QSD8K ++ tristate "SoC Audio support for QSD8K" ++ depends on SND_QSD_SOC ++ help ++ To add support for SoC audio on qsd8k. ++ ++ ++endmenu +diff --git a/sound/soc/msm/Makefile b/sound/soc/msm/Makefile +new file mode 100644 +index 0000000..fbfce6d +--- /dev/null ++++ b/sound/soc/msm/Makefile +@@ -0,0 +1,17 @@ ++# MSM CPU/CODEC DAI Support ++snd-soc-msm-dai-objs := msm-dai.o ++obj-$(CONFIG_SND_MSM_DAI_SOC) += snd-soc-msm-dai.o ++ ++# MSM Platform Support ++snd-soc-msm-objs := msm-pcm.o msm7k-pcm.o ++obj-$(CONFIG_SND_MSM_SOC) += snd-soc-msm.o ++ ++snd-soc-qsd-objs := qsd8k-pcm.o ++obj-$(CONFIG_SND_QSD_SOC) += snd-soc-qsd.o ++ ++# MSM Machine Support ++snd-soc-msm7k-objs := msm7201.o ++obj-$(CONFIG_SND_MSM_SOC_MSM7K) += snd-soc-msm7k.o ++ ++snd-soc-qsd8k-objs := qsd8k.o ++obj-$(CONFIG_SND_QSD_SOC_QSD8K) += snd-soc-qsd8k.o +diff --git a/sound/soc/msm/msm-dai.c b/sound/soc/msm/msm-dai.c +new file mode 100644 +index 0000000..564e7fe +--- /dev/null ++++ b/sound/soc/msm/msm-dai.c +@@ -0,0 +1,143 @@ ++/* sound/soc/msm/msm-dai.c ++ * ++ * Copyright (C) 2008 Google, Inc. ++ * Copyright (C) 2008 HTC Corporation ++ * Copyright (c) 2008-2009, Code Aurora Forum. All rights reserved. ++ * ++ * Derived from msm-pcm.c and msm7201.c. ++ * ++ * This software is licensed under the terms of the GNU General Public ++ * License version 2, as published by the Free Software Foundation, and ++ * may be copied, distributed, and modified under those terms. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/device.h> ++#include <linux/delay.h> ++#include <linux/clk.h> ++#include <linux/platform_device.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++#include "msm-pcm.h" ++ ++struct snd_soc_dai msm_dais[] = { ++{ ++ .name = "CODEC_DAI", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .rates = USE_RATE, ++ .rate_min = USE_RATE_MIN, ++ .rate_max = USE_RATE_MAX, ++ .formats = USE_FORMATS, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .rate_min = USE_RATE_MIN, ++ .rates = USE_RATE, ++ .formats = USE_FORMATS, ++ }, ++}, ++{ ++ .name = "CPU_DAI", ++ .id = 0, ++ .playback = { ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .rates = USE_RATE, ++ .rate_min = USE_RATE_MIN, ++ .rate_max = USE_RATE_MAX, ++ .formats = USE_FORMATS, ++ }, ++ .capture = { ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .rate_min = USE_RATE_MIN, ++ .rates = USE_RATE, ++ .formats = USE_FORMATS, ++ }, ++}, ++}; ++EXPORT_SYMBOL_GPL(msm_dais); ++ ++int msm_pcm_probe(struct platform_device *devptr) ++{ ++ struct snd_card *card; ++ struct snd_soc_codec *codec; ++ int ret; ++ ++ struct snd_soc_device *socdev = platform_get_drvdata(devptr); ++ ++ printk(KERN_ERR "msm_soc: create pcms\n"); ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ codec->name = "MSM-CARD"; ++ codec->owner = THIS_MODULE; ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) { ++ printk(KERN_ERR "msm_soc: failed to create pcms\n"); ++ goto __nopcm; ++ } ++ ++ card = socdev->codec->card; ++ ++ ret = snd_soc_init_card(socdev); ++ if (ret < 0) { ++ printk(KERN_ERR "msm_soc: failed to register card\n"); ++ goto __nodev; ++ } ++ ++ return 0; ++ ++__nodev: ++ snd_soc_free_pcms(socdev); ++__nopcm: ++ kfree(codec); ++ return ret; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_msm = { ++ .probe = msm_pcm_probe, ++}; ++EXPORT_SYMBOL_GPL(soc_codec_dev_msm); ++ ++ ++static int __init msm_dai_init(void) ++{ ++ return snd_soc_register_dais(msm_dais, ARRAY_SIZE(msm_dais)); ++} ++ ++static void __exit msm_dai_exit(void) ++{ ++ snd_soc_unregister_dais(msm_dais, ARRAY_SIZE(msm_dais)); ++} ++ ++module_init(msm_dai_init); ++module_exit(msm_dai_exit); ++ ++/* Module information */ ++MODULE_DESCRIPTION("MSM Codec/Cpu Dai driver"); ++MODULE_LICENSE("GPL v2"); +diff --git a/sound/soc/msm/msm-pcm.c b/sound/soc/msm/msm-pcm.c +new file mode 100644 +index 0000000..90e200d +--- /dev/null ++++ b/sound/soc/msm/msm-pcm.c +@@ -0,0 +1,643 @@ ++/* sound/soc/msm/msm-pcm.c ++ * ++ * Copyright (C) 2008 Google, Inc. ++ * Copyright (C) 2008 HTC Corporation ++ * Copyright (c) 2008-2009, Code Aurora Forum. All rights reserved. ++ * ++ * This software is licensed under the terms of the GNU General Public ++ * License version 2, as published by the Free Software Foundation, and ++ * may be copied, distributed, and modified under those terms. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++ ++#include <linux/init.h> ++#include <linux/err.h> ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/time.h> ++#include <linux/wait.h> ++#include <linux/platform_device.h> ++#include <sound/core.h> ++#include <sound/soc.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <asm/dma.h> ++#include <linux/dma-mapping.h> ++ ++#include "msm-pcm.h" ++ ++#define MAX_DATA_SIZE 496 ++#define AUDPP_ALSA_DECODER (-1) ++ ++#define DB_TABLE_INDEX (50) ++ ++#define audio_send_queue_recbs(prtd, cmd, len) \ ++ msm_adsp_write(prtd->audrec, QDSP_uPAudRecBitStreamQueue, cmd, len) ++#define audio_send_queue_rec(prtd, cmd, len) \ ++ msm_adsp_write(prtd->audrec, QDSP_uPAudRecCmdQueue, cmd, len) ++ ++int intcnt; ++static int audio_dsp_send_buffer(struct msm_audio *prtd, ++ unsigned idx, unsigned len); ++ ++struct audio_frame { ++ uint16_t count_low; ++ uint16_t count_high; ++ uint16_t bytes; ++ uint16_t unknown; ++ unsigned char samples[]; ++} __attribute__ ((packed)); ++ ++/* Table contains dB to raw value mapping */ ++static const unsigned decoder_db_table[] = { ++ ++ 31 , /* -50 dB */ ++ 35 , 39 , 44 , 50 , 56 , ++ 63 , 70 , 79 , 89 , 99 , ++ 112 , 125 , 141 , 158 , 177 , ++ 199 , 223 , 251 , 281 , 316 , ++ 354 , 398 , 446 , 501 , 562 , ++ 630 , 707 , 794 , 891 , 999 , ++ 1122 , 1258 , 1412 , 1584 , 1778 , ++ 1995 , 2238 , 2511 , 2818 , 3162 , ++ 3548 , 3981 , 4466 , 5011 , 5623 , ++ 6309 , 7079 , 7943 , 8912 , 10000 , ++ 11220 , 12589 , 14125 , 15848 , 17782 , ++ 19952 , 22387 , 25118 , 28183 , 31622 , ++ 35481 , 39810 , 44668 , 50118 , 56234 , ++ 63095 , 70794 , 79432 , 89125 , 100000 , ++ 112201 , 125892 , 141253 , 158489 , 177827 , ++ 199526 , 223872 , 251188 , 281838 , 316227 , ++ 354813 , 398107 , 446683 , 501187 , 562341 , ++ 630957 , 707945 , 794328 , 891250 , 1000000 , ++ 1122018 , 1258925 , 1412537 , 1584893 , 1778279 , ++ 1995262 , 2238721 , 2511886 , 2818382 , 3162277 , ++ 3548133 /* 51 dB */ ++ ++}; ++ ++static unsigned compute_db_raw(int db) ++{ ++ unsigned reg_val = 0; /* Computed result for correspondent db */ ++ /* Check if the given db is out of range */ ++ if (db <= MIN_DB) ++ return 0; ++ else if (db > MAX_DB) ++ db = MAX_DB; /* If db is too high then set to max */ ++ reg_val = decoder_db_table[DB_TABLE_INDEX+db]; ++ return reg_val; ++} ++ ++int msm_audio_volume_update(unsigned id, ++ int volume, int pan) ++{ ++ unsigned vol_raw; ++ ++ vol_raw = compute_db_raw(volume); ++ printk(KERN_INFO "volume: %8x vol_raw: %8x \n", volume, vol_raw); ++ return audpp_set_volume_and_pan(id, vol_raw, pan); ++} ++EXPORT_SYMBOL(msm_audio_volume_update); ++ ++void alsa_dsp_event(void *data, unsigned id, uint16_t *msg) ++{ ++ struct msm_audio *prtd = data; ++ struct buffer *frame; ++ unsigned long flag; ++ ++ switch (id) { ++ case AUDPP_MSG_STATUS_MSG: ++ break; ++ case AUDPP_MSG_SPA_BANDS: ++ break; ++ case AUDPP_MSG_HOST_PCM_INTF_MSG:{ ++ unsigned id = msg[2]; ++ unsigned idx = msg[3] - 1; ++ if (id != AUDPP_MSG_HOSTPCM_ID_ARM_RX) { ++ printk(KERN_ERR "bogus id\n"); ++ break; ++ } ++ if (idx > 1) { ++ printk(KERN_ERR "bogus buffer idx\n"); ++ break; ++ } ++ /* Update with actual sent buffer size */ ++ if (prtd->out[idx].used != BUF_INVALID_LEN) ++ prtd->pcm_irq_pos += prtd->out[idx].used; ++ ++ if (prtd->pcm_irq_pos > prtd->pcm_size) ++ prtd->pcm_irq_pos = prtd->pcm_count; ++ ++ if (prtd->ops->playback) ++ prtd->ops->playback(prtd); ++ ++ spin_lock_irqsave(&the_locks.write_dsp_lock, flag); ++ if (prtd->running) { ++ prtd->out[idx].used = 0; ++ frame = prtd->out + prtd->out_tail; ++ if (frame->used) { ++ audio_dsp_send_buffer(prtd, ++ prtd->out_tail, ++ frame->used); ++ prtd->out_tail ^= 1; ++ } else { ++ prtd->out_needed++; ++ } ++ wake_up(&the_locks.write_wait); ++ } ++ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag); ++ break; ++ } ++ case AUDPP_MSG_PCMDMAMISSED: ++ printk(KERN_ERR "alsa_dsp_event: PCMDMAMISSED %d\n", msg[0]); ++ break; ++ case AUDPP_MSG_CFG_MSG: ++ if (msg[0] == AUDPP_MSG_ENA_ENA) { ++ prtd->out_needed = 0; ++ prtd->running = 1; ++ audio_dsp_out_enable(prtd, 1); ++ } else if (msg[0] == AUDPP_MSG_ENA_DIS) { ++ prtd->running = 0; ++ } else { ++ printk(KERN_ERR "alsa_dsp_event:CFG_MSG=%d\n", msg[0]); ++ } ++ break; ++ case EVENT_MSG_ID: ++ printk(KERN_INFO"alsa_dsp_event: arm9 event\n"); ++ break; ++ default: ++ printk(KERN_ERR "alsa_dsp_event: UNKNOWN (%d)\n", id); ++ } ++} ++ ++void alsa_audpre_dsp_event(void *data, unsigned id, size_t len, ++ void (*getevent) (void *ptr, size_t len)) ++{ ++ uint16_t msg[MAX_DATA_SIZE/2]; ++ ++ if (len > MAX_DATA_SIZE) { ++ printk(KERN_ERR"audpre: event too large(%d bytes)\n", len); ++ return; ++ } ++ getevent(msg, len); ++ ++ switch (id) { ++ case AUDPREPROC_MSG_CMD_CFG_DONE_MSG: ++ break; ++ case AUDPREPROC_MSG_ERROR_MSG_ID: ++ printk(KERN_ERR "audpre: err_index %d\n", msg[0]); ++ break; ++ case EVENT_MSG_ID: ++ printk(KERN_INFO"audpre: arm9 event\n"); ++ break; ++ default: ++ printk(KERN_ERR "audpre: unknown event %d\n", id); ++ } ++} ++ ++void audrec_dsp_event(void *data, unsigned id, size_t len, ++ void (*getevent) (void *ptr, size_t len)) ++{ ++ struct msm_audio *prtd = data; ++ unsigned long flag; ++ uint16_t msg[MAX_DATA_SIZE/2]; ++ ++ if (len > MAX_DATA_SIZE) { ++ printk(KERN_ERR"audrec: event/msg too large(%d bytes)\n", len); ++ return; ++ } ++ getevent(msg, len); ++ ++ switch (id) { ++ case AUDREC_MSG_CMD_CFG_DONE_MSG: ++ if (msg[0] & AUDREC_MSG_CFG_DONE_TYPE_0_UPDATE) { ++ if (msg[0] & AUDREC_MSG_CFG_DONE_TYPE_0_ENA) ++ audrec_encoder_config(prtd); ++ else ++ prtd->running = 0; ++ } ++ break; ++ case AUDREC_MSG_CMD_AREC_PARAM_CFG_DONE_MSG:{ ++ prtd->running = 1; ++ break; ++ } ++ case AUDREC_MSG_FATAL_ERR_MSG: ++ printk(KERN_ERR "audrec: ERROR %x\n", msg[0]); ++ break; ++ case AUDREC_MSG_PACKET_READY_MSG: ++ alsa_get_dsp_frames(prtd); ++ ++intcnt; ++ if (prtd->channel_mode == 1) { ++ spin_lock_irqsave(&the_locks.read_dsp_lock, flag); ++ prtd->pcm_irq_pos += prtd->pcm_count; ++ if (prtd->pcm_irq_pos >= prtd->pcm_size) ++ prtd->pcm_irq_pos = 0; ++ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag); ++ ++ if (prtd->ops->capture) ++ prtd->ops->capture(prtd); ++ } else if ((prtd->channel_mode == 0) && (intcnt % 2 == 0)) { ++ spin_lock_irqsave(&the_locks.read_dsp_lock, flag); ++ prtd->pcm_irq_pos += prtd->pcm_count; ++ if (prtd->pcm_irq_pos >= prtd->pcm_size) ++ prtd->pcm_irq_pos = 0; ++ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag); ++ if (prtd->ops->capture) ++ prtd->ops->capture(prtd); ++ } ++ break; ++ case EVENT_MSG_ID: ++ printk(KERN_INFO"audrec: arm9 event\n"); ++ break; ++ default: ++ printk(KERN_ERR "audrec: unknown event %d\n", id); ++ } ++} ++ ++struct msm_adsp_ops aud_pre_adsp_ops = { ++ .event = alsa_audpre_dsp_event, ++}; ++ ++struct msm_adsp_ops aud_rec_adsp_ops = { ++ .event = audrec_dsp_event, ++}; ++ ++int alsa_adsp_configure(struct msm_audio *prtd) ++{ ++ int ret, i; ++ ++ if (prtd->dir == SNDRV_PCM_STREAM_PLAYBACK) { ++ prtd->data = prtd->playback_substream->dma_buffer.area; ++ prtd->phys = prtd->playback_substream->dma_buffer.addr; ++ } ++ if (prtd->dir == SNDRV_PCM_STREAM_CAPTURE) { ++ prtd->data = prtd->capture_substream->dma_buffer.area; ++ prtd->phys = prtd->capture_substream->dma_buffer.addr; ++ } ++ if (!prtd->data) { ++ ret = -ENOMEM; ++ goto err1; ++ } ++ ++ ret = audmgr_open(&prtd->audmgr); ++ if (ret) ++ goto err2; ++ if (prtd->dir == SNDRV_PCM_STREAM_PLAYBACK) { ++ prtd->out_buffer_size = PLAYBACK_DMASZ; ++ prtd->out_sample_rate = 44100; ++ prtd->out_channel_mode = AUDPP_CMD_PCM_INTF_STEREO_V; ++ prtd->out_weight = 100; ++ ++ prtd->out[0].data = prtd->data + 0; ++ prtd->out[0].addr = prtd->phys + 0; ++ prtd->out[0].size = BUFSZ; ++ prtd->out[1].data = prtd->data + BUFSZ; ++ prtd->out[1].addr = prtd->phys + BUFSZ; ++ prtd->out[1].size = BUFSZ; ++ } ++ if (prtd->dir == SNDRV_PCM_STREAM_CAPTURE) { ++ prtd->samp_rate = RPC_AUD_DEF_SAMPLE_RATE_44100; ++ prtd->samp_rate_index = AUDREC_CMD_SAMP_RATE_INDX_44100; ++ prtd->channel_mode = AUDREC_CMD_STEREO_MODE_STEREO; ++ prtd->buffer_size = STEREO_DATA_SIZE; ++ prtd->type = AUDREC_CMD_TYPE_0_INDEX_WAV; ++ prtd->tx_agc_cfg.cmd_id = AUDPREPROC_CMD_CFG_AGC_PARAMS; ++ prtd->ns_cfg.cmd_id = AUDPREPROC_CMD_CFG_NS_PARAMS; ++ prtd->iir_cfg.cmd_id = ++ AUDPREPROC_CMD_CFG_IIR_TUNING_FILTER_PARAMS; ++ ++ ret = msm_adsp_get("AUDPREPROCTASK", ++ &prtd->audpre, &aud_pre_adsp_ops, prtd); ++ if (ret) ++ goto err3; ++ ret = msm_adsp_get("AUDRECTASK", ++ &prtd->audrec, &aud_rec_adsp_ops, prtd); ++ if (ret) { ++ msm_adsp_put(prtd->audpre); ++ goto err3; ++ } ++ prtd->dsp_cnt = 0; ++ prtd->in_head = 0; ++ prtd->in_tail = 0; ++ prtd->in_count = 0; ++ for (i = 0; i < FRAME_NUM; i++) { ++ prtd->in[i].size = 0; ++ prtd->in[i].read = 0; ++ } ++ } ++ ++ return 0; ++ ++err3: ++ audmgr_close(&prtd->audmgr); ++ ++err2: ++ prtd->data = NULL; ++err1: ++ return ret; ++} ++EXPORT_SYMBOL(alsa_adsp_configure); ++ ++int alsa_audio_configure(struct msm_audio *prtd) ++{ ++ struct audmgr_config cfg; ++ int rc; ++ ++ if (prtd->enabled) ++ return 0; ++ ++ /* refuse to start if we're not ready with first buffer */ ++ if (!prtd->out[0].used) ++ return -EIO; ++ ++ cfg.tx_rate = 0; ++ cfg.rx_rate = RPC_AUD_DEF_SAMPLE_RATE_48000; ++ cfg.def_method = RPC_AUD_DEF_METHOD_HOST_PCM; ++ cfg.codec = RPC_AUD_DEF_CODEC_PCM; ++ cfg.snd_method = RPC_SND_METHOD_MIDI; ++ rc = audmgr_enable(&prtd->audmgr, &cfg); ++ if (rc < 0) ++ return rc; ++ ++ if (audpp_enable(AUDPP_ALSA_DECODER, alsa_dsp_event, prtd)) { ++ printk(KERN_ERR "audio: audpp_enable() failed\n"); ++ audmgr_disable(&prtd->audmgr); ++ return -ENODEV; ++ } ++ ++ prtd->enabled = 1; ++ return 0; ++} ++EXPORT_SYMBOL(alsa_audio_configure); ++ ++ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf, ++ size_t count, loff_t *pos) ++{ ++ unsigned long flag; ++ const char __user *start = buf; ++ struct buffer *frame; ++ size_t xfer; ++ int rc = 0; ++ ++ mutex_lock(&the_locks.write_lock); ++ while (count > 0) { ++ frame = prtd->out + prtd->out_head; ++ rc = wait_event_interruptible(the_locks.write_wait, ++ (frame->used == 0) ++ || (prtd->stopped)); ++ if (rc < 0) ++ break; ++ if (prtd->stopped) { ++ rc = -EBUSY; ++ break; ++ } ++ xfer = count > frame->size ? frame->size : count; ++ if (copy_from_user(frame->data, buf, xfer)) { ++ rc = -EFAULT; ++ break; ++ } ++ frame->used = xfer; ++ prtd->out_head ^= 1; ++ count -= xfer; ++ buf += xfer; ++ ++ spin_lock_irqsave(&the_locks.write_dsp_lock, flag); ++ frame = prtd->out + prtd->out_tail; ++ if (frame->used && prtd->out_needed) { ++ audio_dsp_send_buffer(prtd, prtd->out_tail, ++ frame->used); ++ prtd->out_tail ^= 1; ++ prtd->out_needed--; ++ } ++ spin_unlock_irqrestore(&the_locks.write_dsp_lock, flag); ++ } ++ mutex_unlock(&the_locks.write_lock); ++ if (buf > start) ++ return buf - start; ++ return rc; ++} ++EXPORT_SYMBOL(alsa_send_buffer); ++ ++int alsa_audio_disable(struct msm_audio *prtd) ++{ ++ if (prtd->enabled) { ++ mutex_lock(&the_locks.lock); ++ prtd->enabled = 0; ++ audio_dsp_out_enable(prtd, 0); ++ wake_up(&the_locks.write_wait); ++ audpp_disable(AUDPP_ALSA_DECODER, prtd); ++ audmgr_disable(&prtd->audmgr); ++ prtd->out_needed = 0; ++ mutex_unlock(&the_locks.lock); ++ } ++ return 0; ++} ++EXPORT_SYMBOL(alsa_audio_disable); ++ ++int alsa_audrec_disable(struct msm_audio *prtd) ++{ ++ if (prtd->enabled) { ++ mutex_lock(&the_locks.lock); ++ prtd->enabled = 0; ++ alsa_rec_dsp_enable(prtd, 0); ++ wake_up(&the_locks.read_wait); ++ msm_adsp_disable(prtd->audpre); ++ msm_adsp_disable(prtd->audrec); ++ audmgr_disable(&prtd->audmgr); ++ prtd->out_needed = 0; ++ prtd->opened = 0; ++ mutex_unlock(&the_locks.lock); ++ } ++ return 0; ++} ++EXPORT_SYMBOL(alsa_audrec_disable); ++ ++static int audio_dsp_read_buffer(struct msm_audio *prtd, uint32_t read_cnt) ++{ ++ audrec_cmd_packet_ext_ptr cmd; ++ ++ memset(&cmd, 0, sizeof(cmd)); ++ cmd.cmd_id = AUDREC_CMD_PACKET_EXT_PTR; ++ /* Both WAV and AAC use AUDREC_CMD_TYPE_0 */ ++ cmd.type = AUDREC_CMD_TYPE_0; ++ cmd.curr_rec_count_msw = read_cnt >> 16; ++ cmd.curr_rec_count_lsw = read_cnt; ++ ++ return audio_send_queue_recbs(prtd, &cmd, sizeof(cmd)); ++} ++ ++int audrec_encoder_config(struct msm_audio *prtd) ++{ ++ audrec_cmd_arec0param_cfg cmd; ++ uint16_t *data = (void *)prtd->data; ++ unsigned n; ++ ++ memset(&cmd, 0, sizeof(cmd)); ++ cmd.cmd_id = AUDREC_CMD_AREC0PARAM_CFG; ++ cmd.ptr_to_extpkt_buffer_msw = prtd->phys >> 16; ++ cmd.ptr_to_extpkt_buffer_lsw = prtd->phys; ++ cmd.buf_len = FRAME_NUM; /* Both WAV and AAC use 8 frames */ ++ cmd.samp_rate_index = prtd->samp_rate_index; ++ /* 0 for mono, 1 for stereo */ ++ cmd.stereo_mode = prtd->channel_mode; ++ cmd.rec_quality = 0x1C00; ++ ++ /* prepare buffer pointers: ++ * Mono: 1024 samples + 4 halfword header ++ * Stereo: 2048 samples + 4 halfword header ++ */ ++ ++ for (n = 0; n < FRAME_NUM; n++) { ++ prtd->in[n].data = data + 4; ++ data += (4 + (prtd->channel_mode ? 2048 : 1024)); ++ } ++ ++ return audio_send_queue_rec(prtd, &cmd, sizeof(cmd)); ++} ++ ++int audio_dsp_out_enable(struct msm_audio *prtd, int yes) ++{ ++ audpp_cmd_pcm_intf cmd; ++ memset(&cmd, 0, sizeof(cmd)); ++ cmd.cmd_id = AUDPP_CMD_PCM_INTF_2; ++ cmd.object_num = AUDPP_CMD_PCM_INTF_OBJECT_NUM; ++ cmd.config = AUDPP_CMD_PCM_INTF_CONFIG_CMD_V; ++ cmd.intf_type = AUDPP_CMD_PCM_INTF_RX_ENA_ARMTODSP_V; ++ ++ if (yes) { ++ cmd.write_buf1LSW = prtd->out[0].addr; ++ cmd.write_buf1MSW = prtd->out[0].addr >> 16; ++ cmd.write_buf1_len = 0; ++ cmd.write_buf2LSW = prtd->out[1].addr; ++ cmd.write_buf2MSW = prtd->out[1].addr >> 16; ++ cmd.write_buf2_len = prtd->out[1].used; ++ cmd.arm_to_rx_flag = AUDPP_CMD_PCM_INTF_ENA_V; ++ cmd.weight_decoder_to_rx = prtd->out_weight; ++ cmd.weight_arm_to_rx = 1; ++ cmd.partition_number_arm_to_dsp = 0; ++ cmd.sample_rate = prtd->out_sample_rate; ++ cmd.channel_mode = prtd->out_channel_mode; ++ } ++ return audpp_send_queue2(&cmd, sizeof(cmd)); ++} ++ ++int alsa_buffer_read(struct msm_audio *prtd, void __user *buf, ++ size_t count, loff_t *pos) ++{ ++ unsigned long flag; ++ void *data; ++ uint32_t index; ++ uint32_t size; ++ int rc = 0; ++ ++ mutex_lock(&the_locks.read_lock); ++ while (count > 0) { ++ rc = wait_event_interruptible(the_locks.read_wait, ++ (prtd->in_count > 0) ++ || prtd->stopped); ++ if (rc < 0) ++ break; ++ ++ if (prtd->stopped) { ++ rc = -EBUSY; ++ break; ++ } ++ ++ index = prtd->in_tail; ++ data = (uint8_t *) prtd->in[index].data; ++ size = prtd->in[index].size; ++ if (count >= size) { ++ if (copy_to_user(buf, data, size)) { ++ rc = -EFAULT; ++ break; ++ } ++ spin_lock_irqsave(&the_locks.read_dsp_lock, flag); ++ if (index != prtd->in_tail) { ++ /* overrun: data is invalid, we need to retry */ ++ spin_unlock_irqrestore(&the_locks.read_dsp_lock, ++ flag); ++ continue; ++ } ++ prtd->in[index].size = 0; ++ prtd->in_tail = (prtd->in_tail + 1) & (FRAME_NUM - 1); ++ prtd->in_count--; ++ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag); ++ count -= size; ++ buf += size; ++ } else { ++ break; ++ } ++ } ++ mutex_unlock(&the_locks.read_lock); ++ return rc; ++} ++EXPORT_SYMBOL(alsa_buffer_read); ++ ++static int audio_dsp_send_buffer(struct msm_audio *prtd, ++ unsigned idx, unsigned len) ++{ ++ audpp_cmd_pcm_intf_send_buffer cmd; ++ cmd.cmd_id = AUDPP_CMD_PCM_INTF_2; ++ cmd.host_pcm_object = AUDPP_CMD_PCM_INTF_OBJECT_NUM; ++ cmd.config = AUDPP_CMD_PCM_INTF_BUFFER_CMD_V; ++ cmd.intf_type = AUDPP_CMD_PCM_INTF_RX_ENA_ARMTODSP_V; ++ cmd.dsp_to_arm_buf_id = 0; ++ cmd.arm_to_dsp_buf_id = idx + 1; ++ cmd.arm_to_dsp_buf_len = len; ++ return audpp_send_queue2(&cmd, sizeof(cmd)); ++} ++ ++int alsa_rec_dsp_enable(struct msm_audio *prtd, int enable) ++{ ++ audrec_cmd_cfg cmd; ++ ++ memset(&cmd, 0, sizeof(cmd)); ++ cmd.cmd_id = AUDREC_CMD_CFG; ++ cmd.type_0 = enable ? AUDREC_CMD_TYPE_0_ENA : AUDREC_CMD_TYPE_0_DIS; ++ cmd.type_0 |= (AUDREC_CMD_TYPE_0_UPDATE | prtd->type); ++ cmd.type_1 = 0; ++ ++ return audio_send_queue_rec(prtd, &cmd, sizeof(cmd)); ++} ++EXPORT_SYMBOL(alsa_rec_dsp_enable); ++ ++void alsa_get_dsp_frames(struct msm_audio *prtd) ++{ ++ struct audio_frame *frame; ++ uint32_t index = 0; ++ unsigned long flag; ++ ++ if (prtd->type == AUDREC_CMD_TYPE_0_INDEX_WAV) { ++ index = prtd->in_head; ++ ++ frame = ++ (void *)(((char *)prtd->in[index].data) - sizeof(*frame)); ++ ++ spin_lock_irqsave(&the_locks.read_dsp_lock, flag); ++ prtd->in[index].size = frame->bytes; ++ ++ prtd->in_head = (prtd->in_head + 1) & (FRAME_NUM - 1); ++ ++ /* If overflow, move the tail index foward. */ ++ if (prtd->in_head == prtd->in_tail) ++ prtd->in_tail = (prtd->in_tail + 1) & (FRAME_NUM - 1); ++ else ++ prtd->in_count++; ++ ++ audio_dsp_read_buffer(prtd, prtd->dsp_cnt++); ++ spin_unlock_irqrestore(&the_locks.read_dsp_lock, flag); ++ ++ wake_up(&the_locks.read_wait); ++ } else { ++ /* TODO AAC not supported yet. */ ++ } ++} ++EXPORT_SYMBOL(alsa_get_dsp_frames); +diff --git a/sound/soc/msm/msm-pcm.h b/sound/soc/msm/msm-pcm.h +new file mode 100644 +index 0000000..7563ef0 +--- /dev/null ++++ b/sound/soc/msm/msm-pcm.h +@@ -0,0 +1,200 @@ ++/* sound/soc/msm/msm-pcm.h ++ * ++ * Copyright (C) 2008 Google, Inc. ++ * Copyright (C) 2008 HTC Corporation ++ * Copyright (c) 2008-2009, Code Aurora Forum. All rights reserved. ++ * ++ * This software is licensed under the terms of the GNU General Public ++ * License version 2, as published by the Free Software Foundation, and ++ * may be copied, distributed, and modified under those terms. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++#ifndef _MSM_PCM_H ++#define _MSM_PCM_H ++ ++ ++#include <mach/qdsp5/qdsp5audppcmdi.h> ++#include <mach/qdsp5/qdsp5audppmsg.h> ++#include <mach/qdsp5/qdsp5audreccmdi.h> ++#include <mach/qdsp5/qdsp5audrecmsg.h> ++#include <mach/qdsp5/qdsp5audpreproccmdi.h> ++#include <mach/qdsp5/qdsp5audpreprocmsg.h> ++ ++#include <../arch/arm/mach-msm/qdsp5/adsp.h> ++#include <../arch/arm/mach-msm/qdsp5/audmgr.h> ++ ++ ++#define FRAME_NUM (8) ++#define FRAME_SIZE (2052 * 2) ++#define MONO_DATA_SIZE (2048) ++#define STEREO_DATA_SIZE (MONO_DATA_SIZE * 2) ++#define CAPTURE_DMASZ (FRAME_SIZE * FRAME_NUM) ++ ++#define BUFSZ (960 * 5) ++#define PLAYBACK_DMASZ (BUFSZ * 2) ++ ++#define MSM_PLAYBACK_DEFAULT_VOLUME 0 /* 0dB */ ++#define MSM_PLAYBACK_DEFAULT_PAN 0 ++ ++#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE ++#define USE_CHANNELS_MIN 1 ++#define USE_CHANNELS_MAX 2 ++/* Support unconventional sample rates 12000, 24000 as well */ ++#define USE_RATE \ ++ (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) ++#define USE_RATE_MIN 8000 ++#define USE_RATE_MAX 48000 ++#define MAX_BUFFER_PLAYBACK_SIZE \ ++ (4800*4) ++/* 2048 frames (Mono), 1024 frames (Stereo) */ ++#define CAPTURE_SIZE 4096 ++#define MAX_BUFFER_CAPTURE_SIZE (4096*4) ++#define MAX_PERIOD_SIZE BUFSZ ++#define USE_PERIODS_MAX 1024 ++#define USE_PERIODS_MIN 1 ++ ++ ++#define MAX_DB (16) ++#define MIN_DB (-50) ++#define PCMPLAYBACK_DECODERID 5 ++ ++/* 0xFFFFFFFF Indicates not to be used for audio data copy */ ++#define BUF_INVALID_LEN 0xFFFFFFFF ++ ++extern int copy_count; ++extern int intcnt; ++ ++struct msm_volume { ++ bool update; ++ int volume; /* Volume parameter, in dB Scale */ ++ int pan; ++}; ++ ++struct buffer { ++ void *data; ++ unsigned size; ++ unsigned used; ++ unsigned addr; ++}; ++ ++struct buffer_rec { ++ void *data; ++ unsigned int size; ++ unsigned int read; ++ unsigned int addr; ++}; ++ ++struct audio_locks { ++ struct mutex lock; ++ struct mutex write_lock; ++ struct mutex read_lock; ++ spinlock_t read_dsp_lock; ++ spinlock_t write_dsp_lock; ++ spinlock_t mixer_lock; ++ wait_queue_head_t read_wait; ++ wait_queue_head_t write_wait; ++}; ++ ++extern struct audio_locks the_locks; ++ ++struct msm_audio_event_callbacks { ++ /* event is called from interrupt context when a message ++ * arrives from the DSP. ++ */ ++ void (*playback)(void *); ++ void (*capture)(void *); ++}; ++ ++ ++struct msm_audio { ++ struct buffer out[2]; ++ struct buffer_rec in[8]; ++ ++ uint8_t out_head; ++ uint8_t out_tail; ++ uint8_t out_needed; /* number of buffers the dsp is waiting for */ ++ atomic_t out_bytes; ++ ++ /* configuration to use on next enable */ ++ uint32_t out_sample_rate; ++ uint32_t out_channel_mode; ++ uint32_t out_weight; ++ uint32_t out_buffer_size; ++ ++ struct audmgr audmgr; ++ struct snd_pcm_substream *playback_substream; ++ struct snd_pcm_substream *capture_substream; ++ ++ /* data allocated for various buffers */ ++ char *data; ++ dma_addr_t phys; ++ ++ unsigned int pcm_size; ++ unsigned int pcm_count; ++ unsigned int pcm_irq_pos; /* IRQ position */ ++ unsigned int pcm_buf_pos; /* position in buffer */ ++ ++ struct msm_adsp_module *audpre; ++ struct msm_adsp_module *audrec; ++ ++ /* configuration to use on next enable */ ++ uint32_t samp_rate; ++ uint32_t channel_mode; ++ uint32_t buffer_size; /* 2048 for mono, 4096 for stereo */ ++ uint32_t type; /* 0 for PCM ,1 for AAC */ ++ uint32_t dsp_cnt; ++ uint32_t in_head; /* next buffer dsp will write */ ++ uint32_t in_tail; /* next buffer read() will read */ ++ uint32_t in_count; /* number of buffers available to read() */ ++ ++ unsigned short samp_rate_index; ++ ++ /* audpre settings */ ++ audpreproc_cmd_cfg_agc_params tx_agc_cfg; ++ audpreproc_cmd_cfg_ns_params ns_cfg; ++ /* For different sample rate, the coeff might be different. * ++ * All the coeff should be passed from user space */ ++ audpreproc_cmd_cfg_iir_tuning_filter_params iir_cfg; ++ ++ struct msm_audio_event_callbacks *ops; ++ ++ int dir; ++ int opened; ++ int enabled; ++ int running; ++ int stopped; /* set when stopped, cleared on flush */ ++}; ++ ++ ++ ++/* platform data */ ++extern int audio_dsp_out_enable(struct msm_audio *prtd, int yes); ++extern struct snd_soc_platform msm_soc_platform; ++extern struct snd_soc_dai msm_dais[2]; ++extern struct snd_soc_codec_device soc_codec_dev_msm; ++ ++int audrec_encoder_config(struct msm_audio *prtd); ++extern void alsa_get_dsp_frames(struct msm_audio *prtd); ++extern int alsa_rec_dsp_enable(struct msm_audio *prtd, int enable); ++extern int alsa_audrec_disable(struct msm_audio *prtd); ++extern int alsa_audio_configure(struct msm_audio *prtd); ++extern int alsa_audio_disable(struct msm_audio *prtd); ++extern int alsa_adsp_configure(struct msm_audio *prtd); ++extern int alsa_buffer_read(struct msm_audio *prtd, void __user *buf, ++ size_t count, loff_t *pos); ++ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf, ++ size_t count, loff_t *pos); ++int msm_audio_volume_update(unsigned id, ++ int volume, int pan); ++extern struct audio_locks the_locks; ++extern struct msm_volume msm_vol_ctl; ++ ++#endif /*_MSM_PCM_H*/ +diff --git a/sound/soc/msm/msm7201.c b/sound/soc/msm/msm7201.c +new file mode 100644 +index 0000000..977fbac +--- /dev/null ++++ b/sound/soc/msm/msm7201.c +@@ -0,0 +1,337 @@ ++/* linux/sound/soc/msm/msm7201.c ++ * ++ * Copyright (c) 2008-2009, Code Aurora Forum. All rights reserved. ++ * ++ * All source code in this file is licensed under the following license except ++ * where indicated. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License version 2 as published ++ * by the Free Software Foundation. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++#include <linux/init.h> ++#include <linux/err.h> ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/time.h> ++#include <linux/wait.h> ++#include <linux/platform_device.h> ++#include <sound/core.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/pcm.h> ++#include <sound/tlv.h> ++#include <sound/initval.h> ++#include <sound/control.h> ++#include <asm/dma.h> ++#include <linux/dma-mapping.h> ++ ++#include "msm-pcm.h" ++#include <asm/mach-types.h> ++#include <mach/msm_rpcrouter.h> ++ ++static struct msm_rpc_endpoint *snd_ep; ++ ++struct msm_snd_rpc_ids { ++ unsigned long prog; ++ unsigned long vers; ++ unsigned long rpc_set_snd_device; ++ int device; ++}; ++ ++static struct msm_snd_rpc_ids snd_rpc_ids; ++ ++static struct platform_device *msm_audio_snd_device; ++ ++static int snd_msm_volume_info(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; /* Volume Param, in dB */ ++ uinfo->value.integer.min = MIN_DB; ++ uinfo->value.integer.max = MAX_DB; ++ return 0; ++} ++ ++static int snd_msm_volume_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ spin_lock_irq(&the_locks.mixer_lock); ++ ucontrol->value.integer.value[0] = msm_vol_ctl.volume; ++ spin_unlock_irq(&the_locks.mixer_lock); ++ return 0; ++} ++ ++static int snd_msm_volume_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ int change; ++ int volume; ++ ++ volume = ucontrol->value.integer.value[0]; ++ spin_lock_irq(&the_locks.mixer_lock); ++ change = (msm_vol_ctl.volume != volume); ++ if (change) { ++ msm_vol_ctl.update = 1; ++ msm_vol_ctl.volume = volume; ++ } ++ spin_unlock_irq(&the_locks.mixer_lock); ++ return change; ++} ++ ++static int snd_msm_device_info(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; /* Device */ ++ ++ /* ++ * The number of devices supported is 26 (0 to 25) ++ */ ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = 25; ++ return 0; ++} ++ ++static int snd_msm_device_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = (uint32_t)snd_rpc_ids.device; ++ return 0; ++} ++ ++int msm_snd_init_rpc_ids(void) ++{ ++ snd_rpc_ids.prog = 0x30000002; ++#ifdef CONFIG_MSM_AMSS_VERSION_6225 ++ //TODO: complete for other versions ++ snd_rpc_ids.vers = 0xaa2b1a44; ++#else ++ //seem a new magich number...not in arch/arm/mach-msm and it seem to be for a new amss version ++ snd_rpc_ids.vers = 0x00020001; ++#endif ++ /* ++ * The magic number 2 corresponds to the rpc call ++ * index for snd_set_device ++ */ ++ snd_rpc_ids.rpc_set_snd_device = 2; ++ return 0; ++} ++ ++int msm_snd_rpc_connect(void) ++{ ++ if (snd_ep) { ++ printk(KERN_INFO "%s: snd_ep already connected\n", __func__); ++ return 0; ++ } ++ ++ /* Initialize rpc ids */ ++ if (msm_snd_init_rpc_ids()) { ++ printk(KERN_ERR "%s: snd rpc ids initialization failed\n" ++ , __func__); ++ return -ENODATA; ++ } ++ ++ snd_ep = msm_rpc_connect(snd_rpc_ids.prog, ++ snd_rpc_ids.vers, 0); ++ if (IS_ERR(snd_ep)) { ++ printk(KERN_ERR "%s: failed (compatible VERS = %ld)\n", ++ __func__, snd_rpc_ids.vers); ++ snd_ep = NULL; ++ return -EAGAIN; ++ } ++ return 0; ++} ++ ++int msm_snd_rpc_close(void) ++{ ++ int rc = 0; ++ ++ if (IS_ERR(snd_ep)) { ++ printk(KERN_ERR "%s: snd handle unavailable, rc = %ld\n", ++ __func__, PTR_ERR(snd_ep)); ++ return -EAGAIN; ++ } ++ ++ rc = msm_rpc_close(snd_ep); ++ snd_ep = NULL; ++ ++ if (rc < 0) { ++ printk(KERN_ERR "%s: close rpc failed! rc = %d\n", ++ __func__, rc); ++ return -EAGAIN; ++ } else ++ printk(KERN_INFO "rpc close success\n"); ++ ++ return rc; ++} ++ ++static int snd_msm_device_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ int rc = 0; ++ struct snd_start_req { ++ struct rpc_request_hdr hdr; ++ uint32_t rpc_snd_device; ++ uint32_t snd_mute_ear_mute; ++ uint32_t snd_mute_mic_mute; ++ uint32_t callback_ptr; ++ uint32_t client_data; ++ } req; ++ ++ snd_rpc_ids.device = (int)ucontrol->value.integer.value[0]; ++ req.hdr.type = 0; ++ req.hdr.rpc_vers = 2; ++ ++ req.rpc_snd_device = cpu_to_be32(snd_rpc_ids.device); ++ req.snd_mute_ear_mute = cpu_to_be32(1); ++ req.snd_mute_mic_mute = cpu_to_be32(0); ++ req.callback_ptr = -1; ++ req.client_data = cpu_to_be32(0); ++ ++ req.hdr.prog = snd_rpc_ids.prog; ++ req.hdr.vers = snd_rpc_ids.vers; ++ ++ rc = msm_rpc_call(snd_ep, snd_rpc_ids.rpc_set_snd_device , ++ &req, sizeof(req), 5 * HZ); ++ ++ if (rc < 0) { ++ printk(KERN_ERR "%s: snd rpc call failed! rc = %d\n", ++ __func__, rc); ++ } else ++ printk(KERN_INFO "snd device connected \n"); ++ ++ return rc; ++} ++ ++/* Supported range -50dB to 18dB */ ++static const DECLARE_TLV_DB_LINEAR(db_scale_linear, -5000, 1800); ++ ++#define MSM_EXT(xname, xindex, fp_info, fp_get, fp_put, addr) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ ++ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ ++ .name = xname, .index = xindex, \ ++ .info = fp_info,\ ++ .get = fp_get, .put = fp_put, \ ++ .private_value = addr, \ ++} ++ ++#define MSM_EXT_TLV(xname, xindex, fp_info, fp_get, fp_put, addr, tlv_array) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ ++ .access = (SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ ++ SNDRV_CTL_ELEM_ACCESS_READWRITE), \ ++ .name = xname, .index = xindex, \ ++ .info = fp_info,\ ++ .get = fp_get, .put = fp_put, .tlv.p = tlv_array, \ ++ .private_value = addr, \ ++} ++ ++static struct snd_kcontrol_new snd_msm_controls[] = { ++ MSM_EXT_TLV("PCM Playback Volume", 0, snd_msm_volume_info, \ ++ snd_msm_volume_get, snd_msm_volume_put, 0, db_scale_linear), ++ MSM_EXT("device", 1, snd_msm_device_info, snd_msm_device_get, \ ++ snd_msm_device_put, 0), ++}; ++ ++static int msm_new_mixer(struct snd_card *card) ++{ ++ unsigned int idx; ++ int err; ++ ++ printk(KERN_ERR "msm_soc:ALSA MSM Mixer Setting"); ++ strcpy(card->mixername, "MSM Mixer"); ++ for (idx = 0; idx < ARRAY_SIZE(snd_msm_controls); idx++) { ++ err = snd_ctl_add(card, ++ snd_ctl_new1(&snd_msm_controls[idx], NULL)); ++ if (err < 0) ++ return err; ++ } ++ return 0; ++} ++ ++static int msm_soc_dai_init(struct snd_soc_codec *codec) ++{ ++ ++ int ret = 0; ++ ret = msm_new_mixer(codec->card); ++ if (ret < 0) { ++ printk(KERN_ERR "msm_soc:ALSA MSM Mixer Fail"); ++ } ++ ++ return ret; ++} ++ ++ ++static struct snd_soc_dai_link msm_dai = { ++ .name = "ASOC", ++ .stream_name = "ASOC", ++ .codec_dai = &msm_dais[0], ++ .cpu_dai = &msm_dais[1], ++ .init = msm_soc_dai_init, ++}; ++ ++struct snd_soc_card snd_soc_card_msm = { ++ .name = "msm-audio", ++ .dai_link = &msm_dai, ++ .num_links = 1, ++ .platform = &msm_soc_platform, ++}; ++ ++/* msm_audio audio subsystem */ ++static struct snd_soc_device msm_audio_snd_devdata = { ++ .card = &snd_soc_card_msm, ++ .codec_dev = &soc_codec_dev_msm, ++}; ++ ++ ++static int __init msm_audio_init(void) ++{ ++ int ret; ++ ++ msm_audio_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!msm_audio_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(msm_audio_snd_device, &msm_audio_snd_devdata); ++ msm_audio_snd_devdata.dev = &msm_audio_snd_device->dev; ++ ret = platform_device_add(msm_audio_snd_device); ++ if (ret) { ++ platform_device_put(msm_audio_snd_device); ++ return ret; ++ } ++ mutex_init(&the_locks.lock); ++ mutex_init(&the_locks.write_lock); ++ mutex_init(&the_locks.read_lock); ++ spin_lock_init(&the_locks.read_dsp_lock); ++ spin_lock_init(&the_locks.write_dsp_lock); ++ spin_lock_init(&the_locks.mixer_lock); ++ init_waitqueue_head(&the_locks.write_wait); ++ init_waitqueue_head(&the_locks.read_wait); ++ msm_vol_ctl.volume = MSM_PLAYBACK_DEFAULT_VOLUME; ++ msm_vol_ctl.pan = MSM_PLAYBACK_DEFAULT_PAN; ++ ++ ret = msm_snd_rpc_connect(); ++ ++ return ret; ++} ++ ++static void __exit msm_audio_exit(void) ++{ ++ msm_snd_rpc_close(); ++ platform_device_unregister(msm_audio_snd_device); ++} ++ ++module_init(msm_audio_init); ++module_exit(msm_audio_exit); ++ ++MODULE_DESCRIPTION("PCM module"); ++MODULE_LICENSE("GPL v2"); +diff --git a/sound/soc/msm/msm7k-pcm.c b/sound/soc/msm/msm7k-pcm.c +new file mode 100644 +index 0000000..38e8283 +--- /dev/null ++++ b/sound/soc/msm/msm7k-pcm.c +@@ -0,0 +1,574 @@ ++/* linux/sound/soc/msm/msm7k-pcm.c ++ * ++ * Copyright (c) 2008-2009, Code Aurora Forum. All rights reserved. ++ * ++ * All source code in this file is licensed under the following license except ++ * where indicated. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License version 2 as published ++ * by the Free Software Foundation. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++ ++ ++#include <linux/init.h> ++#include <linux/err.h> ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/time.h> ++#include <linux/wait.h> ++#include <linux/platform_device.h> ++#include <sound/core.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/control.h> ++#include <asm/dma.h> ++#include <linux/dma-mapping.h> ++ ++#include "msm-pcm.h" ++ ++#define SND_DRIVER "snd_msm" ++#define MAX_PCM_DEVICES SNDRV_CARDS ++#define MAX_PCM_SUBSTREAMS 1 ++ ++struct snd_msm { ++ struct snd_card *card; ++ struct snd_pcm *pcm; ++}; ++ ++int copy_count; ++ ++struct audio_locks the_locks; ++EXPORT_SYMBOL(the_locks); ++struct msm_volume msm_vol_ctl; ++EXPORT_SYMBOL(msm_vol_ctl); ++ ++ ++static unsigned convert_dsp_samp_index(unsigned index) ++{ ++ switch (index) { ++ case 48000: ++ return AUDREC_CMD_SAMP_RATE_INDX_48000; ++ case 44100: ++ return AUDREC_CMD_SAMP_RATE_INDX_44100; ++ case 32000: ++ return AUDREC_CMD_SAMP_RATE_INDX_32000; ++ case 24000: ++ return AUDREC_CMD_SAMP_RATE_INDX_24000; ++ case 22050: ++ return AUDREC_CMD_SAMP_RATE_INDX_22050; ++ case 16000: ++ return AUDREC_CMD_SAMP_RATE_INDX_16000; ++ case 12000: ++ return AUDREC_CMD_SAMP_RATE_INDX_12000; ++ case 11025: ++ return AUDREC_CMD_SAMP_RATE_INDX_11025; ++ case 8000: ++ return AUDREC_CMD_SAMP_RATE_INDX_8000; ++ default: ++ return AUDREC_CMD_SAMP_RATE_INDX_44100; ++ } ++} ++ ++static unsigned convert_samp_rate(unsigned hz) ++{ ++ switch (hz) { ++ case 48000: ++ return RPC_AUD_DEF_SAMPLE_RATE_48000; ++ case 44100: ++ return RPC_AUD_DEF_SAMPLE_RATE_44100; ++ case 32000: ++ return RPC_AUD_DEF_SAMPLE_RATE_32000; ++ case 24000: ++ return RPC_AUD_DEF_SAMPLE_RATE_24000; ++ case 22050: ++ return RPC_AUD_DEF_SAMPLE_RATE_22050; ++ case 16000: ++ return RPC_AUD_DEF_SAMPLE_RATE_16000; ++ case 12000: ++ return RPC_AUD_DEF_SAMPLE_RATE_12000; ++ case 11025: ++ return RPC_AUD_DEF_SAMPLE_RATE_11025; ++ case 8000: ++ return RPC_AUD_DEF_SAMPLE_RATE_8000; ++ default: ++ return RPC_AUD_DEF_SAMPLE_RATE_44100; ++ } ++} ++ ++static struct snd_pcm_hardware msm_pcm_playback_hardware = { ++ .info = SNDRV_PCM_INFO_INTERLEAVED, ++ .formats = USE_FORMATS, ++ .rates = USE_RATE, ++ .rate_min = USE_RATE_MIN, ++ .rate_max = USE_RATE_MAX, ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .buffer_bytes_max = MAX_BUFFER_PLAYBACK_SIZE, ++ .period_bytes_min = 64, ++ .period_bytes_max = MAX_PERIOD_SIZE, ++ .periods_min = USE_PERIODS_MIN, ++ .periods_max = USE_PERIODS_MAX, ++ .fifo_size = 0, ++}; ++ ++static struct snd_pcm_hardware msm_pcm_capture_hardware = { ++ .info = SNDRV_PCM_INFO_INTERLEAVED, ++ .formats = USE_FORMATS, ++ .rates = USE_RATE, ++ .rate_min = USE_RATE_MIN, ++ .rate_max = USE_RATE_MAX, ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .buffer_bytes_max = MAX_BUFFER_CAPTURE_SIZE, ++ .period_bytes_min = CAPTURE_SIZE, ++ .period_bytes_max = CAPTURE_SIZE, ++ .periods_min = USE_PERIODS_MIN, ++ .periods_max = USE_PERIODS_MAX, ++ .fifo_size = 0, ++}; ++ ++/* Conventional and unconventional sample rate supported */ ++static unsigned int supported_sample_rates[] = { ++ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 ++}; ++ ++static struct snd_pcm_hw_constraint_list constraints_sample_rates = { ++ .count = ARRAY_SIZE(supported_sample_rates), ++ .list = supported_sample_rates, ++ .mask = 0, ++}; ++ ++static void playback_event_handler(void *data) ++{ ++ struct msm_audio *prtd = data; ++ snd_pcm_period_elapsed(prtd->playback_substream); ++} ++ ++static void capture_event_handler(void *data) ++{ ++ struct msm_audio *prtd = data; ++ snd_pcm_period_elapsed(prtd->capture_substream); ++} ++ ++static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd = runtime->private_data; ++ ++ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream); ++ prtd->pcm_count = snd_pcm_lib_period_bytes(substream); ++ prtd->pcm_irq_pos = 0; ++ prtd->pcm_buf_pos = 0; ++ ++ /* rate and channels are sent to audio driver */ ++ prtd->out_sample_rate = runtime->rate; ++ prtd->out_channel_mode = runtime->channels; ++ ++ return 0; ++} ++ ++static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd = runtime->private_data; ++ struct audmgr_config cfg; ++ int rc; ++ ++ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream); ++ prtd->pcm_count = snd_pcm_lib_period_bytes(substream); ++ prtd->pcm_irq_pos = 0; ++ prtd->pcm_buf_pos = 0; ++ ++ /* rate and channels are sent to audio driver */ ++ prtd->samp_rate = convert_samp_rate(runtime->rate); ++ prtd->samp_rate_index = convert_dsp_samp_index(runtime->rate); ++ prtd->channel_mode = (runtime->channels - 1); ++ prtd->buffer_size = prtd->channel_mode ? STEREO_DATA_SIZE : \ ++ MONO_DATA_SIZE; ++ ++ if (prtd->enabled == 1) ++ return 0; ++ ++ prtd->type = AUDREC_CMD_TYPE_0_INDEX_WAV; ++ ++ cfg.tx_rate = convert_samp_rate(runtime->rate); ++ cfg.rx_rate = RPC_AUD_DEF_SAMPLE_RATE_NONE; ++ cfg.def_method = RPC_AUD_DEF_METHOD_RECORD; ++ cfg.codec = RPC_AUD_DEF_CODEC_PCM; ++ cfg.snd_method = RPC_SND_METHOD_MIDI; ++ ++ rc = audmgr_enable(&prtd->audmgr, &cfg); ++ if (rc < 0) ++ return rc; ++ ++ if (msm_adsp_enable(prtd->audpre)) { ++ audmgr_disable(&prtd->audmgr); ++ return -ENODEV; ++ } ++ if (msm_adsp_enable(prtd->audrec)) { ++ msm_adsp_disable(prtd->audpre); ++ audmgr_disable(&prtd->audmgr); ++ return -ENODEV; ++ } ++ prtd->enabled = 1; ++ alsa_rec_dsp_enable(prtd, 1); ++ ++ return 0; ++} ++ ++static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ case SNDRV_PCM_TRIGGER_RESUME: ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ break; ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ break; ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static snd_pcm_uframes_t ++msm_pcm_playback_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *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 msm_pcm_capture_copy(struct snd_pcm_substream *substream, ++ int channel, snd_pcm_uframes_t hwoff, void __user *buf, ++ snd_pcm_uframes_t frames) ++{ ++ int rc = 0, rc1 = 0, rc2 = 0; ++ int fbytes = 0; ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd = substream->runtime->private_data; ++ ++ int monofbytes = 0; ++ char *bufferp = NULL; ++ ++ fbytes = frames_to_bytes(runtime, frames); ++ monofbytes = fbytes / 2; ++ if (runtime->channels == 2) { ++ rc = alsa_buffer_read(prtd, buf, fbytes, NULL); ++ } else { ++ bufferp = buf; ++ rc1 = alsa_buffer_read(prtd, bufferp, monofbytes, NULL); ++ bufferp = buf + monofbytes ; ++ rc2 = alsa_buffer_read(prtd, bufferp, monofbytes, NULL); ++ rc = rc1 + rc2; ++ } ++ prtd->pcm_buf_pos += fbytes; ++ return rc; ++} ++ ++static snd_pcm_uframes_t ++msm_pcm_capture_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd = runtime->private_data; ++ ++ return bytes_to_frames(runtime, (prtd->pcm_irq_pos)); ++} ++ ++static int msm_pcm_capture_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd = runtime->private_data; ++ ++ alsa_audrec_disable(prtd); ++ audmgr_close(&prtd->audmgr); ++ msm_adsp_put(prtd->audrec); ++ msm_adsp_put(prtd->audpre); ++ kfree(prtd); ++ ++ return 0; ++} ++ ++struct msm_audio_event_callbacks snd_msm_audio_ops = { ++ .playback = playback_event_handler, ++ .capture = capture_event_handler, ++}; ++ ++static int msm_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd; ++ int ret = 0; ++ ++ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL); ++ if (prtd == NULL) { ++ ret = -ENOMEM; ++ return ret; ++ } ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ msm_vol_ctl.update = 1; /* Update Volume, with Cached value */ ++ runtime->hw = msm_pcm_playback_hardware; ++ prtd->dir = SNDRV_PCM_STREAM_PLAYBACK; ++ prtd->playback_substream = substream; ++ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ++ runtime->hw = msm_pcm_capture_hardware; ++ prtd->dir = SNDRV_PCM_STREAM_CAPTURE; ++ prtd->capture_substream = substream; ++ } ++ ret = snd_pcm_hw_constraint_list(runtime, 0, ++ SNDRV_PCM_HW_PARAM_RATE, ++ &constraints_sample_rates); ++ if (ret < 0) ++ goto out; ++ /* 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) ++ goto out; ++ ++ prtd->ops = &snd_msm_audio_ops; ++ prtd->out[0].used = BUF_INVALID_LEN; ++ prtd->out_head = 1; /* point to second buffer on startup */ ++ runtime->private_data = prtd; ++ ++ ret = alsa_adsp_configure(prtd); ++ if (ret) ++ goto out; ++ copy_count = 0; ++ return 0; ++ ++ out: ++ kfree(prtd); ++ return ret; ++} ++ ++static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a, ++ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames) ++{ ++ int rc = 1; ++ int fbytes = 0; ++ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd = runtime->private_data; ++ ++ fbytes = frames_to_bytes(runtime, frames); ++ rc = alsa_send_buffer(prtd, buf, fbytes, NULL); ++ ++copy_count; ++ prtd->pcm_buf_pos += fbytes; ++ if (copy_count == 1) { ++ mutex_lock(&the_locks.lock); ++ alsa_audio_configure(prtd); ++ mutex_unlock(&the_locks.lock); ++ } ++ if ((prtd->running) && (msm_vol_ctl.update)) { ++ rc = msm_audio_volume_update(PCMPLAYBACK_DECODERID, ++ msm_vol_ctl.volume, msm_vol_ctl.pan); ++ msm_vol_ctl.update = 0; ++ } ++ ++ return rc; ++} ++ ++static int msm_pcm_playback_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct msm_audio *prtd = runtime->private_data; ++ ++ alsa_audio_disable(prtd); ++ audmgr_close(&prtd->audmgr); ++ kfree(prtd); ++ ++ return 0; ++} ++ ++ ++static int msm_pcm_copy(struct snd_pcm_substream *substream, int a, ++ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames) ++{ ++ int ret = 0; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames); ++ return ret; ++} ++ ++static int msm_pcm_close(struct snd_pcm_substream *substream) ++{ ++ int ret = 0; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = msm_pcm_playback_close(substream); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = msm_pcm_capture_close(substream); ++ return ret; ++} ++static int msm_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ int ret = 0; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = msm_pcm_playback_prepare(substream); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = msm_pcm_capture_prepare(substream); ++ return ret; ++} ++ ++static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream) ++{ ++ snd_pcm_uframes_t ret = 0; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = msm_pcm_playback_pointer(substream); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = msm_pcm_capture_pointer(substream); ++ return ret; ++} ++ ++int msm_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ ++ if (substream->pcm->device & 1) { ++ runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED; ++ runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED; ++ } ++ return 0; ++ ++} ++ ++static struct snd_pcm_ops msm_pcm_ops = { ++ .open = msm_pcm_open, ++ .copy = msm_pcm_copy, ++ .hw_params = msm_pcm_hw_params, ++ .close = msm_pcm_close, ++ .ioctl = snd_pcm_lib_ioctl, ++ .prepare = msm_pcm_prepare, ++ .trigger = msm_pcm_trigger, ++ .pointer = msm_pcm_pointer, ++}; ++ ++ ++ ++static int msm_pcm_remove(struct platform_device *devptr) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(devptr); ++ snd_soc_free_pcms(socdev); ++ kfree(socdev->codec); ++ platform_set_drvdata(devptr, NULL); ++ return 0; ++} ++ ++static int pcm_preallocate_dma_buffer(struct snd_pcm *pcm, ++ int stream) ++{ ++ struct snd_pcm_substream *substream = pcm->streams[stream].substream; ++ struct snd_dma_buffer *buf = &substream->dma_buffer; ++ size_t size; ++ if (!stream) ++ size = PLAYBACK_DMASZ; ++ else ++ size = CAPTURE_DMASZ; ++ ++ buf->dev.type = SNDRV_DMA_TYPE_DEV; ++ buf->dev.dev = pcm->card->dev; ++ buf->private_data = NULL; ++ buf->area = dma_alloc_coherent(pcm->card->dev, size, ++ &buf->addr, GFP_KERNEL); ++ if (!buf->area) ++ return -ENOMEM; ++ ++ buf->bytes = size; ++ return 0; ++} ++ ++static void msm_pcm_free_dma_buffers(struct snd_pcm *pcm) ++{ ++ struct snd_pcm_substream *substream; ++ struct snd_dma_buffer *buf; ++ int stream; ++ ++ for (stream = 0; stream < 2; stream++) { ++ substream = pcm->streams[stream].substream; ++ if (!substream) ++ continue; ++ ++ buf = &substream->dma_buffer; ++ if (!buf->area) ++ continue; ++ ++ dma_free_coherent(pcm->card->dev, buf->bytes, ++ buf->area, buf->addr); ++ buf->area = NULL; ++ } ++} ++ ++static int msm_pcm_new(struct snd_card *card, ++ struct snd_soc_dai *codec_dai, ++ struct snd_pcm *pcm) ++{ ++ int ret; ++ if (!card->dev->coherent_dma_mask) ++ card->dev->coherent_dma_mask = DMA_32BIT_MASK; ++ ++ if (codec_dai->playback.channels_min) { ++ ret = pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_PLAYBACK); ++ if (ret) ++ return ret; ++ } ++ ++ if (codec_dai->capture.channels_min) { ++ ret = pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_CAPTURE); ++ if (ret) ++ msm_pcm_free_dma_buffers(pcm); ++ } ++ return ret; ++} ++ ++struct snd_soc_platform msm_soc_platform = { ++ .name = "msm-audio", ++ .remove = msm_pcm_remove, ++ .pcm_ops = &msm_pcm_ops, ++ .pcm_new = msm_pcm_new, ++ .pcm_free = msm_pcm_free_dma_buffers, ++}; ++EXPORT_SYMBOL(msm_soc_platform); ++ ++static int __init msm_soc_platform_init(void) ++{ ++ return snd_soc_register_platform(&msm_soc_platform); ++} ++module_init(msm_soc_platform_init); ++ ++static void __exit msm_soc_platform_exit(void) ++{ ++ snd_soc_unregister_platform(&msm_soc_platform); ++} ++module_exit(msm_soc_platform_exit); ++ ++MODULE_DESCRIPTION("PCM module platform driver"); ++MODULE_LICENSE("GPL v2"); +diff --git a/sound/soc/msm/qsd-pcm.h b/sound/soc/msm/qsd-pcm.h +new file mode 100644 +index 0000000..6d919c4 +--- /dev/null ++++ b/sound/soc/msm/qsd-pcm.h +@@ -0,0 +1,97 @@ ++/* linux/sound/soc/msm/qsd-pcm.h ++ * ++ * Copyright (C) 2008 Google, Inc. ++ * Copyright (C) 2008 HTC Corporation ++ * Copyright (c) 2008-2009, Code Aurora Forum. All rights reserved. ++ * ++ * This software is licensed under the terms of the GNU General Public ++ * License version 2, as published by the Free Software Foundation, and ++ * may be copied, distributed, and modified under those terms. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++#ifndef _QSD_PCM_H ++#define _QSD_PCM_H ++ ++#include <linux/msm_audio.h> ++ ++#include <mach/qdsp6/msm8k_ard.h> ++#include <mach/qdsp6/msm8k_cad_write_pcm_format.h> ++#include <mach/qdsp6/msm8k_cad_devices.h> ++#include <mach/qdsp6/msm8k_cad.h> ++#include <mach/qdsp6/msm8k_cad_ioctl.h> ++#include <mach/qdsp6/msm8k_cad_volume.h> ++ ++extern void register_cb(void *); ++ ++#define USE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) ++#define USE_CHANNELS_MIN 1 ++#define USE_CHANNELS_MAX 2 ++#define USE_RATE (SNDRV_PCM_RATE_8000_48000 \ ++ | SNDRV_PCM_RATE_CONTINUOUS) ++#define USE_RATE_MIN 8000 ++#define USE_RATE_MAX 48000 ++#define MAX_BUFFER_SIZE (4096*4) ++#define MAX_PERIOD_SIZE 4096 ++#define MIN_PERIOD_SIZE 40 ++#define USE_PERIODS_MAX 1024 ++#define USE_PERIODS_MIN 1 ++ ++struct audio_locks { ++ struct mutex lock; ++ struct mutex mixer_lock; ++}; ++ ++struct qsd_ctl { ++ uint16_t tx_volume; /* Volume parameter */ ++ uint16_t rx_volume; /* Volume parameter */ ++ int32_t strm_volume; /* stream volume*/ ++ uint16_t update; ++ int16_t pan; ++ uint16_t capture_device; /* Device parameter */ ++ uint16_t playback_device; /* Device parameter */ ++ uint16_t tx_mute; /* Mute parameter */ ++ uint16_t rx_mute; /* Mute parameter */ ++}; ++ ++extern struct audio_locks the_locks; ++extern struct snd_pcm_ops qsd_pcm_ops; ++ ++struct qsd_audio { ++ struct snd_pcm_substream *playback_substream; ++ struct snd_pcm_substream *capture_substream; ++ ++ /* data allocated for various buffers */ ++ char *data; ++ dma_addr_t phys; ++ ++ unsigned int pcm_size; ++ unsigned int pcm_count; ++ unsigned int pcm_irq_pos; /* IRQ position */ ++ unsigned int pcm_buf_pos; /* position in buffer */ ++ ++ int dir; ++ int opened; ++ int enabled; ++ int running; ++ int stopped; /* set when stopped, cleared on flush */ ++ ++ struct cad_open_struct_type cos; ++ uint32_t cad_w_handle; ++ struct cad_buf_struct_type cbs; ++}; ++ ++extern struct qsd_ctl qsd_glb_ctl; ++ ++extern struct snd_soc_dai msm_dais[2]; ++extern struct snd_soc_codec_device soc_codec_dev_msm; ++extern struct snd_soc_platform qsd_soc_platform; ++ ++#endif /*_QSD_PCM_H*/ +diff --git a/sound/soc/msm/qsd8k-pcm.c b/sound/soc/msm/qsd8k-pcm.c +new file mode 100644 +index 0000000..afba42d +--- /dev/null ++++ b/sound/soc/msm/qsd8k-pcm.c +@@ -0,0 +1,618 @@ ++/* linux/sound/soc/msm/qsd8k-pcm.c ++ * ++ * Copyright (c) 2009, Code Aurora Forum. All rights reserved. ++ * ++ * All source code in this file is licensed under the following license except ++ * where indicated. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License version 2 as published ++ * by the Free Software Foundation. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++#include <linux/init.h> ++#include <linux/err.h> ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/time.h> ++#include <linux/wait.h> ++#include <linux/platform_device.h> ++#include <sound/core.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/control.h> ++#include <asm/dma.h> ++#include <linux/dma-mapping.h> ++ ++#include "qsd-pcm.h" ++ ++struct snd_pcm_runtime *runtime_dummy; ++static int rc = 1; ++ ++#define SND_DRIVER "snd_qsd" ++#define MAX_PCM_DEVICES SNDRV_CARDS ++#define MAX_PCM_SUBSTREAMS 1 ++ ++struct snd_qsd { ++ struct snd_card *card; ++ struct snd_pcm *pcm; ++}; ++ ++struct qsd_ctl qsd_glb_ctl; ++EXPORT_SYMBOL(qsd_glb_ctl); ++struct audio_locks the_locks; ++EXPORT_SYMBOL(the_locks); ++ ++static unsigned convert_dsp_samp_index(unsigned index) ++{ ++ switch (index) { ++ case 48000: ++ return 3; ++ case 44100: ++ return 4; ++ case 32000: ++ return 5; ++ case 24000: ++ return 6; ++ case 22050: ++ return 7; ++ case 16000: ++ return 8; ++ case 12000: ++ return 9; ++ case 11025: ++ return 10; ++ case 8000: ++ return 11; ++ default: ++ return 3; ++ } ++} ++ ++static struct snd_pcm_hardware qsd_pcm_playback_hardware = { ++ .info = SNDRV_PCM_INFO_INTERLEAVED, ++ .formats = USE_FORMATS, ++ .rates = USE_RATE, ++ .rate_min = USE_RATE_MIN, ++ .rate_max = USE_RATE_MAX, ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .buffer_bytes_max = MAX_BUFFER_SIZE, ++ .period_bytes_min = MIN_PERIOD_SIZE, ++ .period_bytes_max = MAX_PERIOD_SIZE, ++ .periods_min = USE_PERIODS_MIN, ++ .periods_max = USE_PERIODS_MAX, ++ .fifo_size = 0, ++}; ++ ++static struct snd_pcm_hardware qsd_pcm_capture_hardware = { ++ .info = SNDRV_PCM_INFO_INTERLEAVED, ++ .formats = USE_FORMATS, ++ .rates = USE_RATE, ++ .rate_min = USE_RATE_MIN, ++ .rate_max = USE_RATE_MAX, ++ .channels_min = USE_CHANNELS_MIN, ++ .channels_max = USE_CHANNELS_MAX, ++ .buffer_bytes_max = MAX_BUFFER_SIZE, ++ .period_bytes_min = MIN_PERIOD_SIZE, ++ .period_bytes_max = MAX_PERIOD_SIZE, ++ .periods_min = USE_PERIODS_MIN, ++ .periods_max = USE_PERIODS_MAX, ++ .fifo_size = 0, ++}; ++ ++int qsd_audio_volume_update(struct qsd_audio *prtd) ++{ ++ ++ int rc = 0; ++ struct cad_flt_cfg_strm_vol cad_strm_volume; ++ struct cad_filter_struct flt; ++ ++ printk(KERN_INFO "qsd_audio_volume_update: updating volume"); ++ memset(&cad_strm_volume, 0, sizeof(struct cad_flt_cfg_strm_vol)); ++ memset(&flt, 0, sizeof(struct cad_filter_struct)); ++ ++ cad_strm_volume.volume = qsd_glb_ctl.strm_volume; ++ flt.filter_type = CAD_DEVICE_FILTER_TYPE_VOL; ++ flt.format_block = &cad_strm_volume; ++ flt.cmd = CAD_FILTER_CONFIG_STREAM_VOLUME; ++ flt.format_block_len = sizeof(struct cad_flt_cfg_strm_vol); ++ ++ rc = cad_ioctl(prtd->cad_w_handle, ++ CAD_IOCTL_CMD_SET_STREAM_FILTER_CONFIG, ++ &flt, ++ sizeof(struct cad_filter_struct)); ++ if (rc) ++ printk(KERN_ERR "cad_ioctl() set volume failed\n"); ++ return rc; ++} ++ ++static int qsd_pcm_playback_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd = runtime->private_data; ++ ++ struct cad_stream_device_struct_type cad_stream_dev; ++ struct cad_stream_info_struct_type cad_stream_info; ++ struct cad_write_pcm_format_struct_type cad_write_pcm_fmt; ++ u32 stream_device[1]; ++ ++ if (prtd->enabled) ++ return 0; ++ ++ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream); ++ prtd->pcm_count = snd_pcm_lib_period_bytes(substream); ++ prtd->pcm_irq_pos = 0; ++ prtd->pcm_buf_pos = 0; ++ ++ cad_stream_info.app_type = CAD_STREAM_APP_PLAYBACK; ++ cad_stream_info.priority = 0; ++ cad_stream_info.buf_mem_type = CAD_STREAM_BUF_MEM_HEAP; ++ cad_stream_info.ses_buf_max_size = prtd->pcm_count; ++ ++ mutex_lock(&the_locks.lock); ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_SET_STREAM_INFO, ++ &cad_stream_info, ++ sizeof(struct cad_stream_info_struct_type)); ++ mutex_unlock(&the_locks.lock); ++ if (rc) ++ printk(KERN_ERR "cad ioctl failed\n"); ++ ++ stream_device[0] = CAD_HW_DEVICE_ID_DEFAULT_RX ; ++ cad_stream_dev.device = (u32 *) &stream_device[0]; ++ cad_stream_dev.device_len = 1; ++ mutex_lock(&the_locks.lock); ++ ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_SET_STREAM_DEVICE, ++ &cad_stream_dev, ++ sizeof(struct cad_stream_device_struct_type)); ++ mutex_unlock(&the_locks.lock); ++ if (rc) ++ printk(KERN_ERR "cad ioctl failed\n"); ++ ++ cad_write_pcm_fmt.us_ver_id = CAD_WRITE_PCM_VERSION_10; ++ cad_write_pcm_fmt.pcm.us_sample_rate = ++ convert_dsp_samp_index(runtime->rate); ++ cad_write_pcm_fmt.pcm.us_channel_config = runtime->channels; ++ cad_write_pcm_fmt.pcm.us_width = 1; ++ cad_write_pcm_fmt.pcm.us_sign = 0; ++ ++ mutex_lock(&the_locks.lock); ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_SET_STREAM_CONFIG, ++ &cad_write_pcm_fmt, ++ sizeof(struct cad_write_pcm_format_struct_type)); ++ mutex_unlock(&the_locks.lock); ++ if (rc) ++ printk(KERN_ERR "cad ioctl failed\n"); ++ ++ mutex_lock(&the_locks.lock); ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_STREAM_START, ++ NULL, 0); ++ mutex_unlock(&the_locks.lock); ++ if (rc) ++ printk(KERN_ERR "cad ioctl failed\n"); ++ else { ++ prtd->enabled = 1; ++ mutex_lock(&the_locks.mixer_lock); ++ qsd_glb_ctl.update = 1; /* Update Volume, with Cached value */ ++ mutex_unlock(&the_locks.mixer_lock); ++ } ++ return rc; ++} ++ ++static int qsd_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ case SNDRV_PCM_TRIGGER_RESUME: ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ break; ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ break; ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static snd_pcm_uframes_t ++qsd_pcm_playback_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *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)); ++} ++ ++void alsa_event_cb_playback(void) ++{ ++ if (runtime_dummy) { ++ struct qsd_audio *prtd = runtime_dummy->private_data; ++ prtd->pcm_irq_pos += prtd->pcm_count; ++ snd_pcm_period_elapsed(prtd->playback_substream); ++ } ++} ++void alsa_event_cb_capture(u32 event, void *evt_packet, ++ u32 evt_packet_len, void *client_data) ++{ ++ struct qsd_audio *prtd = client_data; ++ prtd->pcm_irq_pos += prtd->pcm_count; ++ snd_pcm_period_elapsed(prtd->capture_substream); ++} ++ ++ ++static int qsd_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd; ++ struct cad_event_struct_type alsa_event; ++ int ret = 0; ++ ++ prtd = kzalloc(sizeof(struct qsd_audio), GFP_KERNEL); ++ if (prtd == NULL) { ++ ret = -ENOMEM; ++ return ret; ++ } ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ runtime_dummy = runtime; ++ printk(KERN_INFO "Stream = SNDRV_PCM_STREAM_PLAYBACK\n"); ++ runtime->hw = qsd_pcm_playback_hardware; ++ prtd->dir = SNDRV_PCM_STREAM_PLAYBACK; ++ prtd->playback_substream = substream; ++ prtd->cos.op_code = CAD_OPEN_OP_WRITE; ++ } else { ++ printk(KERN_INFO "Stream = SNDRV_PCM_STREAM_CAPTURE\n"); ++ runtime->hw = qsd_pcm_capture_hardware; ++ prtd->dir = SNDRV_PCM_STREAM_CAPTURE; ++ prtd->capture_substream = substream; ++ prtd->cos.op_code = CAD_OPEN_OP_READ; ++ } ++ ++ /* 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) { ++ kfree(prtd); ++ return ret; ++ } ++ ++ runtime->private_data = prtd; ++ ++ prtd->cos.format = CAD_FORMAT_PCM; ++ ++ mutex_lock(&the_locks.lock); ++ prtd->cad_w_handle = cad_open(&prtd->cos); ++ mutex_unlock(&the_locks.lock); ++ ++ mutex_lock(&the_locks.lock); ++ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ++ alsa_event.callback = &alsa_event_cb_capture; ++ alsa_event.client_data = prtd; ++ ++ ret = cad_ioctl(prtd->cad_w_handle, ++ CAD_IOCTL_CMD_SET_STREAM_EVENT_LSTR, ++ &alsa_event, sizeof(struct cad_event_struct_type)); ++ if (ret) { ++ mutex_unlock(&the_locks.lock); ++ cad_close(prtd->cad_w_handle); ++ kfree(prtd); ++ return ret; ++ } ++ } else ++ register_cb(&alsa_event_cb_playback); ++ mutex_unlock(&the_locks.lock); ++ prtd->enabled = 0; ++ ++ return 0; ++} ++ ++static int qsd_pcm_playback_copy(struct snd_pcm_substream *substream, int a, ++ snd_pcm_uframes_t hwoff, void __user *buf, ++ snd_pcm_uframes_t frames) ++{ ++ int fbytes = 0; ++ size_t xfer; ++ int rc; ++ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd = runtime->private_data; ++ ++ fbytes = frames_to_bytes(runtime, frames); ++ prtd->cbs.buffer = (void *)buf; ++ prtd->cbs.phys_addr = 0; ++ prtd->cbs.max_size = fbytes; ++ prtd->cbs.actual_size = fbytes; ++ ++ prtd->pcm_buf_pos += fbytes; ++ mutex_lock(&the_locks.lock); ++ xfer = cad_write(prtd->cad_w_handle, &prtd->cbs); ++ mutex_unlock(&the_locks.lock); ++ ++ mutex_lock(&the_locks.mixer_lock); ++ if (qsd_glb_ctl.update) { ++ rc = qsd_audio_volume_update(prtd); ++ qsd_glb_ctl.update = 0; ++ } ++ mutex_unlock(&the_locks.mixer_lock); ++ ++ return 0; ++} ++ ++static int qsd_pcm_playback_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd = runtime->private_data; ++ ++ mutex_lock(&the_locks.lock); ++ cad_close(prtd->cad_w_handle); ++ mutex_unlock(&the_locks.lock); ++ prtd->enabled = 0; ++ ++ /* ++ * TODO: Deregister the async callback handler. ++ * Currently cad provides no interface to do so. ++ */ ++ register_cb(NULL); ++ kfree(prtd); ++ ++ return 0; ++} ++ ++static int qsd_pcm_capture_copy(struct snd_pcm_substream *substream, int a, ++ snd_pcm_uframes_t hwoff, void __user *buf, ++ snd_pcm_uframes_t frames) ++{ ++ int fbytes = 0; ++ size_t xfer; ++ int rc = 0; ++ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd = runtime->private_data; ++ ++ fbytes = frames_to_bytes(runtime, frames); ++ fbytes = fbytes; ++ ++ prtd->cbs.buffer = (void *)buf; ++ prtd->cbs.phys_addr = 0; ++ prtd->cbs.max_size = fbytes; ++ prtd->cbs.actual_size = fbytes; ++ ++ mutex_lock(&the_locks.lock); ++ xfer = cad_read(prtd->cad_w_handle, &prtd->cbs); ++ mutex_unlock(&the_locks.lock); ++ ++ prtd->pcm_buf_pos += fbytes; ++ mutex_lock(&the_locks.mixer_lock); ++ if (qsd_glb_ctl.update) { ++ rc = qsd_audio_volume_update(prtd); ++ qsd_glb_ctl.update = 0; ++ } ++ mutex_unlock(&the_locks.mixer_lock); ++ ++ if (xfer < fbytes) ++ return -EIO; ++ ++ return rc; ++} ++ ++static snd_pcm_uframes_t ++qsd_pcm_capture_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd = runtime->private_data; ++ ++ return bytes_to_frames(runtime, (prtd->pcm_irq_pos)); ++} ++ ++static int qsd_pcm_capture_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd = runtime->private_data; ++ ++ mutex_lock(&the_locks.lock); ++ cad_close(prtd->cad_w_handle); ++ mutex_unlock(&the_locks.lock); ++ ++ /* ++ * TODO: Deregister the async callback handler. ++ * Currently cad provides no interface to do so. ++ */ ++ kfree(prtd); ++ ++ return 0; ++} ++ ++static int qsd_pcm_capture_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct qsd_audio *prtd = runtime->private_data; ++ int rc = 0; ++ ++ struct cad_stream_device_struct_type cad_stream_dev; ++ struct cad_stream_info_struct_type cad_stream_info; ++ struct cad_write_pcm_format_struct_type cad_write_pcm_fmt; ++ u32 stream_device[1]; ++ ++ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream); ++ prtd->pcm_count = snd_pcm_lib_period_bytes(substream); ++ prtd->pcm_irq_pos = 0; ++ prtd->pcm_buf_pos = 0; ++ ++ cad_stream_info.app_type = CAD_STREAM_APP_RECORD; ++ cad_stream_info.priority = 0; ++ cad_stream_info.buf_mem_type = CAD_STREAM_BUF_MEM_HEAP; ++ cad_stream_info.ses_buf_max_size = prtd->pcm_count; ++ ++ if (prtd->enabled) ++ return 0; ++ ++ mutex_lock(&the_locks.lock); ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_SET_STREAM_INFO, ++ &cad_stream_info, ++ sizeof(struct cad_stream_info_struct_type)); ++ if (rc) { ++ mutex_unlock(&the_locks.lock); ++ return rc; ++ } ++ ++ stream_device[0] = CAD_HW_DEVICE_ID_DEFAULT_TX ; ++ cad_stream_dev.device = (u32 *) &stream_device[0]; ++ cad_stream_dev.device_len = 1; ++ ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_SET_STREAM_DEVICE, ++ &cad_stream_dev, ++ sizeof(struct cad_stream_device_struct_type)); ++ if (rc) { ++ mutex_unlock(&the_locks.lock); ++ return rc; ++ } ++ ++ cad_write_pcm_fmt.us_ver_id = CAD_WRITE_PCM_VERSION_10; ++ cad_write_pcm_fmt.pcm.us_sample_rate = ++ convert_dsp_samp_index(runtime->rate); ++ cad_write_pcm_fmt.pcm.us_channel_config = runtime->channels; ++ cad_write_pcm_fmt.pcm.us_width = 1; ++ cad_write_pcm_fmt.pcm.us_sign = 0; ++ ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_SET_STREAM_CONFIG, ++ &cad_write_pcm_fmt, ++ sizeof(struct cad_write_pcm_format_struct_type)); ++ if (rc) { ++ mutex_unlock(&the_locks.lock); ++ return rc; ++ } ++ rc = cad_ioctl(prtd->cad_w_handle, CAD_IOCTL_CMD_STREAM_START, ++ NULL, 0); ++ mutex_unlock(&the_locks.lock); ++ if (!rc) ++ prtd->enabled = 1; ++ return rc; ++} ++ ++ ++static int qsd_pcm_copy(struct snd_pcm_substream *substream, int a, ++ snd_pcm_uframes_t hwoff, void __user *buf, ++ snd_pcm_uframes_t frames) ++{ ++ int ret = 0; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = qsd_pcm_playback_copy(substream, a, hwoff, buf, frames); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = qsd_pcm_capture_copy(substream, a, hwoff, buf, frames); ++ return ret; ++} ++ ++static int qsd_pcm_close(struct snd_pcm_substream *substream) ++{ ++ int ret = 0; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = qsd_pcm_playback_close(substream); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = qsd_pcm_capture_close(substream); ++ return ret; ++} ++static int qsd_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ int ret = 0; ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = qsd_pcm_playback_prepare(substream); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = qsd_pcm_capture_prepare(substream); ++ return ret; ++} ++ ++static snd_pcm_uframes_t qsd_pcm_pointer(struct snd_pcm_substream *substream) ++{ ++ snd_pcm_uframes_t ret = 0; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ ret = qsd_pcm_playback_pointer(substream); ++ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ ret = qsd_pcm_capture_pointer(substream); ++ return ret; ++} ++ ++int qsd_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ ++ if (substream->pcm->device & 1) { ++ runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED; ++ runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED; ++ } ++ return 0; ++} ++ ++struct snd_pcm_ops qsd_pcm_ops = { ++ .open = qsd_pcm_open, ++ .copy = qsd_pcm_copy, ++ .hw_params = qsd_pcm_hw_params, ++ .close = qsd_pcm_close, ++ .ioctl = snd_pcm_lib_ioctl, ++ .prepare = qsd_pcm_prepare, ++ .trigger = qsd_pcm_trigger, ++ .pointer = qsd_pcm_pointer, ++}; ++EXPORT_SYMBOL_GPL(qsd_pcm_ops); ++ ++static int qsd_pcm_remove(struct platform_device *devptr) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(devptr); ++ snd_soc_free_pcms(socdev); ++ kfree(socdev->codec); ++ platform_set_drvdata(devptr, NULL); ++ return 0; ++} ++ ++static int qsd_pcm_new(struct snd_card *card, ++ struct snd_soc_dai *codec_dai, ++ struct snd_pcm *pcm) ++{ ++ if (!card->dev->coherent_dma_mask) ++ card->dev->coherent_dma_mask = DMA_32BIT_MASK; ++ ++ return 0; ++} ++ ++struct snd_soc_platform qsd_soc_platform = { ++ .name = "qsd-audio", ++ .remove = qsd_pcm_remove, ++ .pcm_ops = &qsd_pcm_ops, ++ .pcm_new = qsd_pcm_new, ++}; ++EXPORT_SYMBOL(qsd_soc_platform); ++ ++static int __init qsd_soc_platform_init(void) ++{ ++ return snd_soc_register_platform(&qsd_soc_platform); ++} ++module_init(qsd_soc_platform_init); ++ ++static void __exit qsd_soc_platform_exit(void) ++{ ++ snd_soc_unregister_platform(&qsd_soc_platform); ++} ++module_exit(qsd_soc_platform_exit); ++ ++MODULE_DESCRIPTION("PCM module platform driver"); ++MODULE_LICENSE("GPL v2"); +diff --git a/sound/soc/msm/qsd8k.c b/sound/soc/msm/qsd8k.c +new file mode 100644 +index 0000000..979fde7 +--- /dev/null ++++ b/sound/soc/msm/qsd8k.c +@@ -0,0 +1,382 @@ ++/* linux/sound/soc/msm/qsd8k.c ++ * ++ * Copyright (c) 2009, Code Aurora Forum. All rights reserved. ++ * ++ * All source code in this file is licensed under the following license except ++ * where indicated. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License version 2 as published ++ * by the Free Software Foundation. ++ * ++ * This program 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, you can find it at http://www.fsf.org. ++ */ ++ ++#include <linux/init.h> ++#include <linux/err.h> ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/time.h> ++#include <linux/wait.h> ++#include <linux/platform_device.h> ++#include <sound/core.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/control.h> ++#include <asm/dma.h> ++#include <linux/dma-mapping.h> ++ ++#include "qsd-pcm.h" ++ ++static struct platform_device *qsd_audio_snd_device; ++ ++static int snd_qsd_route_info(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; /* Device */ ++ uinfo->value.integer.min = CAD_HW_DEVICE_ID_HANDSET_MIC; ++ uinfo->value.integer.max = CAD_HW_DEVICE_ID_DEFAULT_RX; ++ return 0; ++} ++ ++static int snd_qsd_route_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = ++ (uint32_t) qsd_glb_ctl.playback_device; ++ ucontrol->value.integer.value[1] = ++ (uint32_t) qsd_glb_ctl.capture_device; ++ return 0; ++} ++ ++static int snd_get_device_type(int device) ++{ ++ switch (device) { ++ case CAD_HW_DEVICE_ID_HANDSET_MIC: ++ case CAD_HW_DEVICE_ID_HEADSET_MIC: ++ case CAD_HW_DEVICE_ID_BT_SCO_MIC: ++ case CAD_HW_DEVICE_ID_DEFAULT_TX: ++ return CAD_TX_DEVICE; ++ case CAD_HW_DEVICE_ID_HANDSET_SPKR: ++ case CAD_HW_DEVICE_ID_HEADSET_SPKR_MONO: ++ case CAD_HW_DEVICE_ID_HEADSET_SPKR_STEREO: ++ case CAD_HW_DEVICE_ID_SPKR_PHONE_MIC: ++ case CAD_HW_DEVICE_ID_SPKR_PHONE_MONO: ++ case CAD_HW_DEVICE_ID_SPKR_PHONE_STEREO: ++ case CAD_HW_DEVICE_ID_BT_SCO_SPKR: ++ case CAD_HW_DEVICE_ID_BT_A2DP_SPKR: ++ case CAD_HW_DEVICE_ID_DEFAULT_RX: ++ return CAD_RX_DEVICE; ++ default: ++ return -ENODEV; ++ } ++} ++ ++static int snd_qsd_route_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ int rc = 0; ++ int device, direction; ++ ++ device = ucontrol->value.integer.value[0]; ++ direction = snd_get_device_type(device); ++ ++ if (direction < 0) ++ return direction; ++ ++ rc = audio_switch_device(device); ++ if (rc < 0) { ++ printk(KERN_ERR "audio_switch_device failed\n"); ++ return rc; ++ } ++ ++ if (CAD_RX_DEVICE == direction) ++ qsd_glb_ctl.playback_device = device; ++ else /* CAD_TX_DEVICE */ ++ qsd_glb_ctl.capture_device = device; ++ ++ return 0; ++} ++ ++static int snd_vol_info(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; /* Volume */ ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = 100; ++ return 0; ++} ++ ++static int snd_rx_vol_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = (uint32_t) qsd_glb_ctl.rx_volume; ++ return 0; ++} ++ ++static int snd_rx_vol_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct msm_vol_info vi; ++ int rc = 0; ++ ++ vi.vol = ucontrol->value.integer.value[0]; ++ vi.path = CAD_RX_DEVICE; ++ ++ rc = audio_set_device_volume_path(&vi); ++ ++ if (rc) ++ printk(KERN_ERR "audio_set_device_volume failed\n"); ++ else ++ qsd_glb_ctl.rx_volume = ucontrol->value.integer.value[0]; ++ ++ return rc; ++} ++ ++static int snd_tx_vol_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = (uint32_t) qsd_glb_ctl.tx_volume; ++ return 0; ++} ++ ++static int snd_tx_vol_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct msm_vol_info vi; ++ int rc = 0; ++ ++ vi.vol = ucontrol->value.integer.value[0]; ++ vi.path = CAD_TX_DEVICE; ++ ++ rc = audio_set_device_volume_path(&vi); ++ ++ if (rc) ++ printk(KERN_ERR "audio_set_device_volume failed\n"); ++ else ++ qsd_glb_ctl.tx_volume = ucontrol->value.integer.value[0]; ++ ++ return rc; ++} ++ ++static int snd_tx_mute_info(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; /* MUTE */ ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = 1; ++ return 0; ++} ++ ++static int snd_tx_mute_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = (uint32_t) qsd_glb_ctl.tx_mute; ++ return 0; ++} ++ ++static int snd_tx_mute_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ int rc = 0; ++ struct msm_mute_info m; ++ ++ m.path = CAD_TX_DEVICE; ++ m.mute = ucontrol->value.integer.value[0]; ++ ++ rc = audio_set_device_mute(&m); ++ if (rc) ++ printk(KERN_ERR "Capture device mute failed\n"); ++ else ++ qsd_glb_ctl.tx_mute = ucontrol->value.integer.value[0]; ++ return rc; ++} ++ ++static int snd_rx_mute_info(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; /* MUTE */ ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = 1; ++ return 0; ++} ++ ++static int snd_rx_mute_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = (uint32_t) qsd_glb_ctl.rx_mute; ++ return 0; ++} ++ ++static int snd_rx_mute_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ int rc = 0; ++ struct msm_mute_info m; ++ ++ m.path = CAD_RX_DEVICE; ++ m.mute = ucontrol->value.integer.value[0]; ++ ++ rc = audio_set_device_mute(&m); ++ if (rc) ++ printk(KERN_ERR "Playback device mute failed\n"); ++ else ++ qsd_glb_ctl.rx_mute = ucontrol->value.integer.value[0]; ++ return rc; ++} ++ ++static int snd_strm_vol_info(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; /* Volume Param, in gain */ ++ uinfo->value.integer.min = CAD_STREAM_MIN_GAIN; ++ uinfo->value.integer.max = CAD_STREAM_MAX_GAIN; ++ return 0; ++} ++ ++static int snd_strm_vol_get(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = qsd_glb_ctl.strm_volume; ++ return 0; ++} ++ ++static int snd_strm_vol_put(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ int change; ++ int volume; ++ ++ if (ucontrol->value.integer.value[0] > CAD_STREAM_MAX_GAIN) ++ ucontrol->value.integer.value[0] = CAD_STREAM_MAX_GAIN; ++ if (ucontrol->value.integer.value[0] < CAD_STREAM_MIN_GAIN) ++ ucontrol->value.integer.value[0] = CAD_STREAM_MIN_GAIN; ++ ++ volume = ucontrol->value.integer.value[0]; ++ change = (qsd_glb_ctl.strm_volume != volume); ++ mutex_lock(&the_locks.mixer_lock); ++ if (change) { ++ qsd_glb_ctl.strm_volume = volume; ++ qsd_glb_ctl.update = 1; ++ } ++ mutex_unlock(&the_locks.mixer_lock); ++ return 0; ++} ++ ++#define QSD_EXT(xname, xindex, fp_info, fp_get, fp_put, addr) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ ++ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ ++ .name = xname, .index = xindex, \ ++ .info = fp_info,\ ++ .get = fp_get, .put = fp_put, \ ++ .private_value = addr, \ ++} ++ ++static struct snd_kcontrol_new snd_qsd_controls[] = { ++ QSD_EXT("Master Route", 1, snd_qsd_route_info, \ ++ snd_qsd_route_get, snd_qsd_route_put, 0), ++ QSD_EXT("Master Volume Playback", 2, snd_vol_info, \ ++ snd_rx_vol_get, snd_rx_vol_put, 0), ++ QSD_EXT("Master Volume Capture", 3, snd_vol_info, \ ++ snd_tx_vol_get, snd_tx_vol_put, 0), ++ QSD_EXT("Master Mute Playback", 4, snd_rx_mute_info, \ ++ snd_rx_mute_get, snd_rx_mute_put, 0), ++ QSD_EXT("Master Mute Capture", 5, snd_tx_mute_info, \ ++ snd_tx_mute_get, snd_tx_mute_put, 0), ++ QSD_EXT("Stream Volume", 6, snd_strm_vol_info, \ ++ snd_strm_vol_get, snd_strm_vol_put, 0), ++}; ++ ++static int qsd_new_mixer(struct snd_card *card) ++{ ++ unsigned int idx; ++ int err; ++ ++ strcpy(card->mixername, "MSM Mixer"); ++ for (idx = 0; idx < ARRAY_SIZE(snd_qsd_controls); idx++) { ++ err = snd_ctl_add(card, ++ snd_ctl_new1(&snd_qsd_controls[idx], NULL)); ++ if (err < 0) ++ return err; ++ } ++ return 0; ++} ++ ++static int qsd_soc_dai_init(struct snd_soc_codec *codec) ++{ ++ ++ int ret = 0; ++ ret = qsd_new_mixer(codec->card); ++ if (ret < 0) { ++ printk(KERN_ERR "msm_soc:ALSA MSM Mixer Fail"); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_dai_link qsd_dai = { ++ .name = "ASOC", ++ .stream_name = "ASOC", ++ .codec_dai = &msm_dais[0], ++ .cpu_dai = &msm_dais[1], ++ .init = qsd_soc_dai_init, ++}; ++ ++struct snd_soc_card snd_soc_card_qsd = { ++ .name = "qsd-audio", ++ .dai_link = &qsd_dai, ++ .num_links = 1, ++ .platform = &qsd_soc_platform, ++}; ++ ++/* qsd_audio audio subsystem */ ++static struct snd_soc_device qsd_audio_snd_devdata = { ++ .card = &snd_soc_card_qsd, ++ .codec_dev = &soc_codec_dev_msm, ++}; ++ ++static int __init qsd_audio_init(void) ++{ ++ int ret; ++ ++ qsd_audio_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!qsd_audio_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(qsd_audio_snd_device, &qsd_audio_snd_devdata); ++ qsd_audio_snd_devdata.dev = &qsd_audio_snd_device->dev; ++ ret = platform_device_add(qsd_audio_snd_device); ++ if (ret) { ++ platform_device_put(qsd_audio_snd_device); ++ return ret; ++ } ++ mutex_init(&the_locks.lock); ++ mutex_init(&the_locks.mixer_lock); ++ ++ return ret; ++} ++ ++static void __exit qsd_audio_exit(void) ++{ ++ kfree(qsd_audio_snd_devdata.codec_dev); ++ platform_device_unregister(qsd_audio_snd_device); ++} ++ ++module_init(qsd_audio_init); ++module_exit(qsd_audio_exit); ++ ++MODULE_DESCRIPTION("PCM module"); ++MODULE_LICENSE("GPL v2"); |