Initial Contribution
msm-2.6.38: tag AU_LINUX_ANDROID_GINGERBREAD.02.03.04.00.142
Signed-off-by: Bryan Huntsman <bryanh@codeaurora.org>
diff --git a/arch/arm/mach-msm/qdsp5/audio_in.c b/arch/arm/mach-msm/qdsp5/audio_in.c
new file mode 100644
index 0000000..6fc5d6b
--- /dev/null
+++ b/arch/arm/mach-msm/qdsp5/audio_in.c
@@ -0,0 +1,996 @@
+/* arch/arm/mach-msm/qdsp5/audio_in.c
+ *
+ * pcm audio input device
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 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.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/fs.h>
+#include <linux/miscdevice.h>
+#include <linux/uaccess.h>
+#include <linux/kthread.h>
+#include <linux/wait.h>
+#include <linux/dma-mapping.h>
+
+#include <linux/delay.h>
+
+#include <linux/msm_audio_aac.h>
+
+#include <asm/atomic.h>
+#include <asm/ioctls.h>
+#include <mach/msm_adsp.h>
+#include <mach/msm_rpcrouter.h>
+
+#include "audmgr.h"
+
+#include <mach/qdsp5/qdsp5audpreproccmdi.h>
+#include <mach/qdsp5/qdsp5audpreprocmsg.h>
+#include <mach/qdsp5/qdsp5audreccmdi.h>
+#include <mach/qdsp5/qdsp5audrecmsg.h>
+#include <mach/debug_mm.h>
+
+/* FRAME_NUM must be a power of two */
+#define FRAME_NUM (8)
+#define FRAME_SIZE (2052 * 2)
+#define MONO_DATA_SIZE (2048)
+#define STEREO_DATA_SIZE (MONO_DATA_SIZE * 2)
+#define DMASZ (FRAME_SIZE * FRAME_NUM)
+
+struct buffer {
+ void *data;
+ uint32_t size;
+ uint32_t read;
+ uint32_t addr;
+};
+
+struct audio_in {
+ struct buffer in[FRAME_NUM];
+
+ spinlock_t dsp_lock;
+
+ atomic_t in_bytes;
+
+ struct mutex lock;
+ struct mutex read_lock;
+ wait_queue_head_t wait;
+
+ 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 bit_rate; /* bit rate for AAC */
+ uint32_t record_quality; /* record quality (bits/sample/channel)
+ for AAC*/
+ uint32_t buffer_cfg_ioctl; /* to allow any one of buffer set ioctl */
+ 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;
+
+ struct audmgr audmgr;
+
+ /* data allocated for various buffers */
+ char *data;
+ dma_addr_t phys;
+
+ int opened;
+ int enabled;
+ int running;
+ int stopped; /* set when stopped, cleared on flush */
+
+ /* audpre settings */
+ int tx_agc_enable;
+ audpreproc_cmd_cfg_agc_params tx_agc_cfg;
+ int ns_enable;
+ 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 */
+ int iir_enable;
+ audpreproc_cmd_cfg_iir_tuning_filter_params iir_cfg;
+};
+
+static int audio_in_dsp_enable(struct audio_in *audio, int enable);
+static int audio_in_encoder_config(struct audio_in *audio);
+static int audio_dsp_read_buffer(struct audio_in *audio, uint32_t read_cnt);
+static void audio_flush(struct audio_in *audio);
+static int audio_dsp_set_tx_agc(struct audio_in *audio);
+static int audio_dsp_set_ns(struct audio_in *audio);
+static int audio_dsp_set_iir(struct audio_in *audio);
+
+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_11025;
+ }
+}
+
+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_11025;
+ }
+}
+
+static unsigned convert_samp_index(unsigned index)
+{
+ switch (index) {
+ case RPC_AUD_DEF_SAMPLE_RATE_48000: return 48000;
+ case RPC_AUD_DEF_SAMPLE_RATE_44100: return 44100;
+ case RPC_AUD_DEF_SAMPLE_RATE_32000: return 32000;
+ case RPC_AUD_DEF_SAMPLE_RATE_24000: return 24000;
+ case RPC_AUD_DEF_SAMPLE_RATE_22050: return 22050;
+ case RPC_AUD_DEF_SAMPLE_RATE_16000: return 16000;
+ case RPC_AUD_DEF_SAMPLE_RATE_12000: return 12000;
+ case RPC_AUD_DEF_SAMPLE_RATE_11025: return 11025;
+ case RPC_AUD_DEF_SAMPLE_RATE_8000: return 8000;
+ default: return 11025;
+ }
+}
+
+/* must be called with audio->lock held */
+static int audio_in_enable(struct audio_in *audio)
+{
+ struct audmgr_config cfg;
+ int rc;
+
+ if (audio->enabled)
+ return 0;
+
+ cfg.tx_rate = audio->samp_rate;
+ cfg.rx_rate = RPC_AUD_DEF_SAMPLE_RATE_NONE;
+ cfg.def_method = RPC_AUD_DEF_METHOD_RECORD;
+ if (audio->type == AUDREC_CMD_TYPE_0_INDEX_WAV)
+ cfg.codec = RPC_AUD_DEF_CODEC_PCM;
+ else
+ cfg.codec = RPC_AUD_DEF_CODEC_AAC;
+ cfg.snd_method = RPC_SND_METHOD_MIDI;
+
+ rc = audmgr_enable(&audio->audmgr, &cfg);
+ if (rc < 0)
+ return rc;
+
+ if (msm_adsp_enable(audio->audpre)) {
+ MM_ERR("msm_adsp_enable(audpre) failed\n");
+ return -ENODEV;
+ }
+ if (msm_adsp_enable(audio->audrec)) {
+ MM_ERR("msm_adsp_enable(audrec) failed\n");
+ return -ENODEV;
+ }
+
+ audio->enabled = 1;
+ audio_in_dsp_enable(audio, 1);
+
+ return 0;
+}
+
+/* must be called with audio->lock held */
+static int audio_in_disable(struct audio_in *audio)
+{
+ if (audio->enabled) {
+ audio->enabled = 0;
+
+ audio_in_dsp_enable(audio, 0);
+
+ wake_up(&audio->wait);
+
+ msm_adsp_disable(audio->audrec);
+ msm_adsp_disable(audio->audpre);
+ audmgr_disable(&audio->audmgr);
+ }
+ return 0;
+}
+
+/* ------------------- dsp --------------------- */
+static void audpre_dsp_event(void *data, unsigned id, size_t len,
+ void (*getevent)(void *ptr, size_t len))
+{
+ uint16_t msg[2];
+ getevent(msg, sizeof(msg));
+
+ switch (id) {
+ case AUDPREPROC_MSG_CMD_CFG_DONE_MSG:
+ MM_INFO("type %d, status_flag %d\n", msg[0], msg[1]);
+ break;
+ case AUDPREPROC_MSG_ERROR_MSG_ID:
+ MM_INFO("err_index %d\n", msg[0]);
+ break;
+ case ADSP_MESSAGE_ID:
+ MM_DBG("Received ADSP event: module enable(audpreproctask)\n");
+ break;
+ default:
+ MM_ERR("unknown event %d\n", id);
+ }
+}
+
+struct audio_frame {
+ uint16_t count_low;
+ uint16_t count_high;
+ uint16_t bytes;
+ uint16_t unknown;
+ unsigned char samples[];
+} __attribute__((packed));
+
+static void audio_in_get_dsp_frames(struct audio_in *audio)
+{
+ struct audio_frame *frame;
+ uint32_t index;
+ unsigned long flags;
+
+ index = audio->in_head;
+
+ /* XXX check for bogus frame size? */
+
+ frame = (void *) (((char *)audio->in[index].data) -
+ sizeof(*frame));
+ spin_lock_irqsave(&audio->dsp_lock, flags);
+ audio->in[index].size = frame->bytes;
+
+ audio->in_head = (audio->in_head + 1) & (FRAME_NUM - 1);
+
+ /* If overflow, move the tail index foward. */
+ if (audio->in_head == audio->in_tail)
+ audio->in_tail = (audio->in_tail + 1) & (FRAME_NUM - 1);
+ else
+ audio->in_count++;
+
+ audio_dsp_read_buffer(audio, audio->dsp_cnt++);
+ spin_unlock_irqrestore(&audio->dsp_lock, flags);
+
+ wake_up(&audio->wait);
+}
+
+static void audrec_dsp_event(void *data, unsigned id, size_t len,
+ void (*getevent)(void *ptr, size_t len))
+{
+ struct audio_in *audio = data;
+ uint16_t msg[3];
+ getevent(msg, sizeof(msg));
+
+ 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) {
+ MM_INFO("CFG ENABLED\n");
+ audio_in_encoder_config(audio);
+ } else {
+ MM_INFO("CFG SLEEP\n");
+ audio->running = 0;
+ audio->tx_agc_enable = 0;
+ audio->ns_enable = 0;
+ audio->iir_enable = 0;
+ }
+ } else {
+ MM_INFO("CMD_CFG_DONE %x\n", msg[0]);
+ }
+ break;
+ case AUDREC_MSG_CMD_AREC_PARAM_CFG_DONE_MSG: {
+ MM_INFO("PARAM CFG DONE\n");
+ audio->running = 1;
+ audio_dsp_set_tx_agc(audio);
+ audio_dsp_set_ns(audio);
+ audio_dsp_set_iir(audio);
+ break;
+ }
+ case AUDREC_MSG_FATAL_ERR_MSG:
+ MM_ERR("ERROR %x\n", msg[0]);
+ break;
+ case AUDREC_MSG_PACKET_READY_MSG:
+/* REC_DBG("type %x, count %d", msg[0], (msg[1] | (msg[2] << 16))); */
+ audio_in_get_dsp_frames(audio);
+ break;
+ case ADSP_MESSAGE_ID:
+ MM_DBG("Received ADSP event: module \
+ enable/disable(audrectask)\n");
+ break;
+ default:
+ MM_ERR("unknown event %d\n", id);
+ }
+}
+
+struct msm_adsp_ops audpre_adsp_ops = {
+ .event = audpre_dsp_event,
+};
+
+struct msm_adsp_ops audrec_adsp_ops = {
+ .event = audrec_dsp_event,
+};
+
+
+#define audio_send_queue_pre(audio, cmd, len) \
+ msm_adsp_write(audio->audpre, QDSP_uPAudPreProcCmdQueue, cmd, len)
+#define audio_send_queue_recbs(audio, cmd, len) \
+ msm_adsp_write(audio->audrec, QDSP_uPAudRecBitStreamQueue, cmd, len)
+#define audio_send_queue_rec(audio, cmd, len) \
+ msm_adsp_write(audio->audrec, \
+ QDSP_uPAudRecCmdQueue, cmd, len)
+
+/* Convert Bit Rate to Record Quality field of DSP */
+static unsigned int bitrate_to_record_quality(unsigned int sample_rate,
+ unsigned int channel, unsigned int bit_rate) {
+ unsigned int temp;
+
+ temp = sample_rate * channel;
+ MM_DBG(" sample rate * channel = %d \n", temp);
+ /* To represent in Q12 fixed format */
+ temp = (bit_rate * 4096) / temp;
+ MM_DBG(" Record Quality = 0x%8x \n", temp);
+ return temp;
+}
+
+static int audio_dsp_set_tx_agc(struct audio_in *audio)
+{
+ audpreproc_cmd_cfg_agc_params cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+
+ audio->tx_agc_cfg.cmd_id = AUDPREPROC_CMD_CFG_AGC_PARAMS;
+ if (audio->tx_agc_enable) {
+ /* cmd.tx_agc_param_mask = 0xFE00 from sample code */
+ audio->tx_agc_cfg.tx_agc_param_mask =
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_COMP_SLOPE) |
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_COMP_TH) |
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_EXP_SLOPE) |
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_EXP_TH) |
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_COMP_AIG_FLAG) |
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_COMP_STATIC_GAIN) |
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_TX_AGC_ENA_FLAG);
+ audio->tx_agc_cfg.tx_agc_enable_flag =
+ AUDPREPROC_CMD_TX_AGC_ENA_FLAG_ENA;
+ /* cmd.param_mask = 0xFFF0 from sample code */
+ audio->tx_agc_cfg.param_mask =
+ (1 << AUDPREPROC_CMD_PARAM_MASK_RMS_TAY) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_RELEASEK) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_DELAY) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_ATTACKK) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_LEAKRATE_SLOW) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_LEAKRATE_FAST) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_AIG_RELEASEK) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_AIG_MIN) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_AIG_MAX) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_LEAK_UP) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_LEAK_DOWN) |
+ (1 << AUDPREPROC_CMD_PARAM_MASK_AIG_ATTACKK);
+ } else {
+ audio->tx_agc_cfg.tx_agc_param_mask =
+ (1 << AUDPREPROC_CMD_TX_AGC_PARAM_MASK_TX_AGC_ENA_FLAG);
+ audio->tx_agc_cfg.tx_agc_enable_flag =
+ AUDPREPROC_CMD_TX_AGC_ENA_FLAG_DIS;
+ }
+ cmd = audio->tx_agc_cfg;
+
+ return audio_send_queue_pre(audio, &cmd, sizeof(cmd));
+}
+
+static int audio_enable_tx_agc(struct audio_in *audio, int enable)
+{
+ if (audio->tx_agc_enable != enable) {
+ audio->tx_agc_enable = enable;
+ if (audio->running)
+ audio_dsp_set_tx_agc(audio);
+ }
+ return 0;
+}
+
+static int audio_dsp_set_ns(struct audio_in *audio)
+{
+ audpreproc_cmd_cfg_ns_params cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+
+ audio->ns_cfg.cmd_id = AUDPREPROC_CMD_CFG_NS_PARAMS;
+
+ if (audio->ns_enable) {
+ /* cmd.ec_mode_new is fixed as 0x0064 when enable
+ * from sample code */
+ audio->ns_cfg.ec_mode_new =
+ AUDPREPROC_CMD_EC_MODE_NEW_NS_ENA |
+ AUDPREPROC_CMD_EC_MODE_NEW_HB_ENA |
+ AUDPREPROC_CMD_EC_MODE_NEW_VA_ENA;
+ } else {
+ audio->ns_cfg.ec_mode_new =
+ AUDPREPROC_CMD_EC_MODE_NEW_NLMS_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_DES_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_NS_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_CNI_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_NLES_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_HB_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_VA_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_PCD_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_FEHI_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_NEHI_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_NLPP_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_FNE_DIS |
+ AUDPREPROC_CMD_EC_MODE_NEW_PRENLMS_DIS;
+ }
+ cmd = audio->ns_cfg;
+
+ return audio_send_queue_pre(audio, &cmd, sizeof(cmd));
+}
+
+static int audio_enable_ns(struct audio_in *audio, int enable)
+{
+ if (audio->ns_enable != enable) {
+ audio->ns_enable = enable;
+ if (audio->running)
+ audio_dsp_set_ns(audio);
+ }
+ return 0;
+}
+
+static int audio_dsp_set_iir(struct audio_in *audio)
+{
+ audpreproc_cmd_cfg_iir_tuning_filter_params cmd;
+
+ memset(&cmd, 0, sizeof(cmd));
+
+ audio->iir_cfg.cmd_id = AUDPREPROC_CMD_CFG_IIR_TUNING_FILTER_PARAMS;
+
+ if (audio->iir_enable)
+ /* cmd.active_flag is 0xFFFF from sample code but 0x0001 here */
+ audio->iir_cfg.active_flag = AUDPREPROC_CMD_IIR_ACTIVE_FLAG_ENA;
+ else
+ audio->iir_cfg.active_flag = AUDPREPROC_CMD_IIR_ACTIVE_FLAG_DIS;
+
+ cmd = audio->iir_cfg;
+
+ return audio_send_queue_pre(audio, &cmd, sizeof(cmd));
+}
+
+static int audio_enable_iir(struct audio_in *audio, int enable)
+{
+ if (audio->iir_enable != enable) {
+ audio->iir_enable = enable;
+ if (audio->running)
+ audio_dsp_set_iir(audio);
+ }
+ return 0;
+}
+
+static int audio_in_dsp_enable(struct audio_in *audio, 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 | audio->type);
+ cmd.type_1 = 0;
+
+ return audio_send_queue_rec(audio, &cmd, sizeof(cmd));
+}
+
+static int audio_in_encoder_config(struct audio_in *audio)
+{
+ audrec_cmd_arec0param_cfg cmd;
+ uint16_t *data = (void *) audio->data;
+ unsigned n;
+
+ memset(&cmd, 0, sizeof(cmd));
+ cmd.cmd_id = AUDREC_CMD_AREC0PARAM_CFG;
+ cmd.ptr_to_extpkt_buffer_msw = audio->phys >> 16;
+ cmd.ptr_to_extpkt_buffer_lsw = audio->phys;
+ cmd.buf_len = FRAME_NUM; /* Both WAV and AAC use 8 frames */
+ cmd.samp_rate_index = audio->samp_rate_index;
+ cmd.stereo_mode = audio->channel_mode; /* 0 for mono, 1 for stereo */
+
+ /* cmd.rec_quality is based on user set bit rate / sample rate /
+ * channel
+ */
+ cmd.rec_quality = audio->record_quality;
+
+ /* prepare buffer pointers:
+ * Mono: 1024 samples + 4 halfword header
+ * Stereo: 2048 samples + 4 halfword header
+ * AAC
+ * Mono/Stere: 768 + 4 halfword header
+ */
+ for (n = 0; n < FRAME_NUM; n++) {
+ audio->in[n].data = data + 4;
+ if (audio->type == AUDREC_CMD_TYPE_0_INDEX_WAV)
+ data += (4 + (audio->channel_mode ? 2048 : 1024));
+ else if (audio->type == AUDREC_CMD_TYPE_0_INDEX_AAC)
+ data += (4 + 768);
+ }
+
+ return audio_send_queue_rec(audio, &cmd, sizeof(cmd));
+}
+
+static int audio_dsp_read_buffer(struct audio_in *audio, 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(audio, &cmd, sizeof(cmd));
+}
+
+/* ------------------- device --------------------- */
+
+static void audio_flush(struct audio_in *audio)
+{
+ int i;
+
+ audio->dsp_cnt = 0;
+ audio->in_head = 0;
+ audio->in_tail = 0;
+ audio->in_count = 0;
+ for (i = 0; i < FRAME_NUM; i++) {
+ audio->in[i].size = 0;
+ audio->in[i].read = 0;
+ }
+}
+
+static long audio_in_ioctl(struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct audio_in *audio = file->private_data;
+ int rc;
+
+ if (cmd == AUDIO_GET_STATS) {
+ struct msm_audio_stats stats;
+ stats.byte_count = atomic_read(&audio->in_bytes);
+ if (copy_to_user((void *) arg, &stats, sizeof(stats)))
+ return -EFAULT;
+ return 0;
+ }
+
+ mutex_lock(&audio->lock);
+ switch (cmd) {
+ case AUDIO_START:
+ rc = audio_in_enable(audio);
+ break;
+ case AUDIO_STOP:
+ rc = audio_in_disable(audio);
+ audio->stopped = 1;
+ break;
+ case AUDIO_FLUSH:
+ if (audio->stopped) {
+ /* Make sure we're stopped and we wake any threads
+ * that might be blocked holding the read_lock.
+ * While audio->stopped read threads will always
+ * exit immediately.
+ */
+ wake_up(&audio->wait);
+ mutex_lock(&audio->read_lock);
+ audio_flush(audio);
+ mutex_unlock(&audio->read_lock);
+ }
+ case AUDIO_SET_CONFIG: {
+ struct msm_audio_config cfg;
+ /* The below code is to make mutual exclusive between
+ * AUDIO_SET_CONFIG and AUDIO_SET_STREAM_CONFIG.
+ * Allow any one IOCTL.
+ */
+ if (audio->buffer_cfg_ioctl == AUDIO_SET_STREAM_CONFIG) {
+ rc = -EINVAL;
+ break;
+ }
+ if (copy_from_user(&cfg, (void *) arg, sizeof(cfg))) {
+ rc = -EFAULT;
+ break;
+ }
+ if (cfg.channel_count == 1) {
+ cfg.channel_count = AUDREC_CMD_STEREO_MODE_MONO;
+ } else if (cfg.channel_count == 2) {
+ cfg.channel_count = AUDREC_CMD_STEREO_MODE_STEREO;
+ } else {
+ rc = -EINVAL;
+ break;
+ }
+
+ if (cfg.type == 0) {
+ cfg.type = AUDREC_CMD_TYPE_0_INDEX_WAV;
+ } else if (cfg.type == 1) {
+ cfg.type = AUDREC_CMD_TYPE_0_INDEX_AAC;
+ } else {
+ rc = -EINVAL;
+ break;
+ }
+ audio->samp_rate = convert_samp_rate(cfg.sample_rate);
+ audio->samp_rate_index =
+ convert_dsp_samp_index(cfg.sample_rate);
+ audio->channel_mode = cfg.channel_count;
+ audio->buffer_size =
+ audio->channel_mode ? STEREO_DATA_SIZE
+ : MONO_DATA_SIZE;
+ audio->type = cfg.type;
+ audio->buffer_cfg_ioctl = AUDIO_SET_CONFIG;
+ rc = 0;
+ break;
+ }
+ case AUDIO_GET_CONFIG: {
+ struct msm_audio_config cfg;
+ cfg.buffer_size = audio->buffer_size;
+ cfg.buffer_count = FRAME_NUM;
+ cfg.sample_rate = convert_samp_index(audio->samp_rate);
+ if (audio->channel_mode == AUDREC_CMD_STEREO_MODE_MONO)
+ cfg.channel_count = 1;
+ else
+ cfg.channel_count = 2;
+ if (audio->type == AUDREC_CMD_TYPE_0_INDEX_WAV)
+ cfg.type = 0;
+ else
+ cfg.type = 1;
+ cfg.unused[0] = 0;
+ cfg.unused[1] = 0;
+ cfg.unused[2] = 0;
+ if (copy_to_user((void *) arg, &cfg, sizeof(cfg)))
+ rc = -EFAULT;
+ else
+ rc = 0;
+ break;
+ }
+ case AUDIO_GET_STREAM_CONFIG: {
+ struct msm_audio_stream_config cfg;
+ cfg.buffer_size = audio->buffer_size;
+ cfg.buffer_count = FRAME_NUM;
+ if (copy_to_user((void *)arg, &cfg, sizeof(cfg)))
+ rc = -EFAULT;
+ else
+ rc = 0;
+ break;
+ }
+ case AUDIO_SET_STREAM_CONFIG: {
+ struct msm_audio_stream_config cfg;
+ /* The below code is to make mutual exclusive between
+ * AUDIO_SET_CONFIG and AUDIO_SET_STREAM_CONFIG.
+ * Allow any one IOCTL.
+ */
+ if (audio->buffer_cfg_ioctl == AUDIO_SET_CONFIG) {
+ rc = -EINVAL;
+ break;
+ }
+ if (copy_from_user(&cfg, (void *)arg, sizeof(cfg))) {
+ rc = -EFAULT;
+ break;
+ } else
+ rc = 0;
+ audio->buffer_size = cfg.buffer_size;
+ /* The IOCTL is only of AAC, set the encoder as AAC */
+ audio->type = 1;
+ audio->buffer_cfg_ioctl = AUDIO_SET_STREAM_CONFIG;
+ break;
+ }
+ case AUDIO_GET_AAC_ENC_CONFIG: {
+ struct msm_audio_aac_enc_config cfg;
+ if (audio->channel_mode == AUDREC_CMD_STEREO_MODE_MONO)
+ cfg.channels = 1;
+ else
+ cfg.channels = 2;
+ cfg.sample_rate = convert_samp_index(audio->samp_rate);
+ cfg.bit_rate = audio->bit_rate;
+ cfg.stream_format = AUDIO_AAC_FORMAT_RAW;
+ if (copy_to_user((void *)arg, &cfg, sizeof(cfg)))
+ rc = -EFAULT;
+ else
+ rc = 0;
+ break;
+ }
+ case AUDIO_SET_AAC_ENC_CONFIG: {
+ struct msm_audio_aac_enc_config cfg;
+ unsigned int record_quality;
+ if (copy_from_user(&cfg, (void *)arg, sizeof(cfg))) {
+ rc = -EFAULT;
+ break;
+ }
+ if (cfg.stream_format != AUDIO_AAC_FORMAT_RAW) {
+ MM_ERR("unsupported AAC format\n");
+ rc = -EINVAL;
+ break;
+ }
+ record_quality = bitrate_to_record_quality(cfg.sample_rate,
+ cfg.channels, cfg.bit_rate);
+ /* Range of Record Quality Supported by DSP, Q12 format */
+ if ((record_quality < 0x800) || (record_quality > 0x4000)) {
+ MM_ERR("Unsupported bit rate \n");
+ rc = -EINVAL;
+ break;
+ }
+ if (cfg.channels == 1) {
+ cfg.channels = AUDREC_CMD_STEREO_MODE_MONO;
+ } else if (cfg.channels == 2) {
+ cfg.channels = AUDREC_CMD_STEREO_MODE_STEREO;
+ } else {
+ rc = -EINVAL;
+ break;
+ }
+ audio->samp_rate = convert_samp_rate(cfg.sample_rate);
+ audio->samp_rate_index =
+ convert_dsp_samp_index(cfg.sample_rate);
+ audio->channel_mode = cfg.channels;
+ audio->bit_rate = cfg.bit_rate;
+ audio->record_quality = record_quality;
+ MM_DBG(" Record Quality = 0x%8x \n", audio->record_quality);
+ rc = 0;
+ break;
+ }
+ default:
+ rc = -EINVAL;
+ }
+ mutex_unlock(&audio->lock);
+ return rc;
+}
+
+static ssize_t audio_in_read(struct file *file,
+ char __user *buf,
+ size_t count, loff_t *pos)
+{
+ struct audio_in *audio = file->private_data;
+ unsigned long flags;
+ const char __user *start = buf;
+ void *data;
+ uint32_t index;
+ uint32_t size;
+ int rc = 0;
+
+ mutex_lock(&audio->read_lock);
+ while (count > 0) {
+ rc = wait_event_interruptible(
+ audio->wait, (audio->in_count > 0) || audio->stopped);
+ if (rc < 0)
+ break;
+
+ if (audio->stopped && !audio->in_count) {
+ rc = 0;/* End of File */
+ break;
+ }
+
+ index = audio->in_tail;
+ data = (uint8_t *) audio->in[index].data;
+ size = audio->in[index].size;
+ if (count >= size) {
+ /* order the reads on the buffer */
+ dma_coherent_post_ops();
+ if (copy_to_user(buf, data, size)) {
+ rc = -EFAULT;
+ break;
+ }
+ spin_lock_irqsave(&audio->dsp_lock, flags);
+ if (index != audio->in_tail) {
+ /* overrun -- data is invalid and we need to retry */
+ spin_unlock_irqrestore(&audio->dsp_lock, flags);
+ continue;
+ }
+ audio->in[index].size = 0;
+ audio->in_tail = (audio->in_tail + 1) & (FRAME_NUM - 1);
+ audio->in_count--;
+ spin_unlock_irqrestore(&audio->dsp_lock, flags);
+ count -= size;
+ buf += size;
+ } else {
+ MM_ERR("short read\n");
+ break;
+ }
+ if (audio->type == AUDREC_CMD_TYPE_0_INDEX_AAC)
+ break; /* AAC only read one frame */
+ }
+ mutex_unlock(&audio->read_lock);
+
+ if (buf > start)
+ return buf - start;
+
+ return rc;
+}
+
+static ssize_t audio_in_write(struct file *file,
+ const char __user *buf,
+ size_t count, loff_t *pos)
+{
+ return -EINVAL;
+}
+
+static int audio_in_release(struct inode *inode, struct file *file)
+{
+ struct audio_in *audio = file->private_data;
+
+ mutex_lock(&audio->lock);
+ audio_in_disable(audio);
+ audio_flush(audio);
+ msm_adsp_put(audio->audrec);
+ msm_adsp_put(audio->audpre);
+ audio->audrec = NULL;
+ audio->audpre = NULL;
+ audio->opened = 0;
+ mutex_unlock(&audio->lock);
+ return 0;
+}
+
+struct audio_in the_audio_in;
+
+static int audio_in_open(struct inode *inode, struct file *file)
+{
+ struct audio_in *audio = &the_audio_in;
+ int rc;
+
+ mutex_lock(&audio->lock);
+ if (audio->opened) {
+ rc = -EBUSY;
+ goto done;
+ }
+
+ /* Settings will be re-config at AUDIO_SET_CONFIG,
+ * but at least we need to have initial config
+ */
+ audio->samp_rate = RPC_AUD_DEF_SAMPLE_RATE_11025;
+ audio->samp_rate_index = AUDREC_CMD_SAMP_RATE_INDX_11025;
+ audio->channel_mode = AUDREC_CMD_STEREO_MODE_MONO;
+ audio->buffer_size = MONO_DATA_SIZE;
+ audio->type = AUDREC_CMD_TYPE_0_INDEX_WAV;
+
+ /* For AAC, bit rate hard coded, default settings is
+ * sample rate (11025) x channel count (1) x recording quality (1.75)
+ * = 19293 bps */
+ audio->bit_rate = 19293;
+ audio->record_quality = 0x1c00;
+
+ rc = audmgr_open(&audio->audmgr);
+ if (rc)
+ goto done;
+ rc = msm_adsp_get("AUDPREPROCTASK", &audio->audpre,
+ &audpre_adsp_ops, audio);
+ if (rc)
+ goto done;
+ rc = msm_adsp_get("AUDRECTASK", &audio->audrec,
+ &audrec_adsp_ops, audio);
+ if (rc)
+ goto done;
+
+ audio->dsp_cnt = 0;
+ audio->stopped = 0;
+ audio->buffer_cfg_ioctl = 0; /* No valid ioctl set */
+
+ audio_flush(audio);
+
+ file->private_data = audio;
+ audio->opened = 1;
+ rc = 0;
+done:
+ mutex_unlock(&audio->lock);
+ return rc;
+}
+
+static long audpre_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
+{
+ struct audio_in *audio = file->private_data;
+ int rc = 0, enable;
+ uint16_t enable_mask;
+
+ mutex_lock(&audio->lock);
+ switch (cmd) {
+ case AUDIO_ENABLE_AUDPRE:
+ if (copy_from_user(&enable_mask, (void *) arg,
+ sizeof(enable_mask))) {
+ rc = -EFAULT;
+ break;
+ }
+
+ enable = (enable_mask & AGC_ENABLE) ? 1 : 0;
+ audio_enable_tx_agc(audio, enable);
+ enable = (enable_mask & NS_ENABLE) ? 1 : 0;
+ audio_enable_ns(audio, enable);
+ enable = (enable_mask & TX_IIR_ENABLE) ? 1 : 0;
+ audio_enable_iir(audio, enable);
+ break;
+
+ case AUDIO_SET_AGC:
+ if (copy_from_user(&audio->tx_agc_cfg, (void *) arg,
+ sizeof(audio->tx_agc_cfg)))
+ rc = -EFAULT;
+ break;
+
+ case AUDIO_SET_NS:
+ if (copy_from_user(&audio->ns_cfg, (void *) arg,
+ sizeof(audio->ns_cfg)))
+ rc = -EFAULT;
+ break;
+
+ case AUDIO_SET_TX_IIR:
+ if (copy_from_user(&audio->iir_cfg, (void *) arg,
+ sizeof(audio->iir_cfg)))
+ rc = -EFAULT;
+ break;
+
+ default:
+ rc = -EINVAL;
+ }
+
+ mutex_unlock(&audio->lock);
+ return rc;
+}
+
+static int audpre_open(struct inode *inode, struct file *file)
+{
+ struct audio_in *audio = &the_audio_in;
+
+ file->private_data = audio;
+
+ return 0;
+}
+
+static struct file_operations audio_fops = {
+ .owner = THIS_MODULE,
+ .open = audio_in_open,
+ .release = audio_in_release,
+ .read = audio_in_read,
+ .write = audio_in_write,
+ .unlocked_ioctl = audio_in_ioctl,
+};
+
+struct miscdevice audio_in_misc = {
+ .minor = MISC_DYNAMIC_MINOR,
+ .name = "msm_pcm_in",
+ .fops = &audio_fops,
+};
+
+static const struct file_operations audpre_fops = {
+ .owner = THIS_MODULE,
+ .open = audpre_open,
+ .unlocked_ioctl = audpre_ioctl,
+};
+
+struct miscdevice audpre_misc = {
+ .minor = MISC_DYNAMIC_MINOR,
+ .name = "msm_preproc_ctl",
+ .fops = &audpre_fops,
+};
+
+static int __init audio_in_init(void)
+{
+ the_audio_in.data = dma_alloc_coherent(NULL, DMASZ,
+ &the_audio_in.phys, GFP_KERNEL);
+ if (!the_audio_in.data) {
+ MM_ERR("Unable to allocate DMA buffer\n");
+ return -ENOMEM;
+ }
+
+ mutex_init(&the_audio_in.lock);
+ mutex_init(&the_audio_in.read_lock);
+ spin_lock_init(&the_audio_in.dsp_lock);
+ init_waitqueue_head(&the_audio_in.wait);
+ return misc_register(&audio_in_misc) || misc_register(&audpre_misc);
+}
+
+device_initcall(audio_in_init);