diff options
Diffstat (limited to 'drivers/staging/dream/qdsp5/audio_mp3.c')
| -rw-r--r-- | drivers/staging/dream/qdsp5/audio_mp3.c | 971 |
1 files changed, 971 insertions, 0 deletions
diff --git a/drivers/staging/dream/qdsp5/audio_mp3.c b/drivers/staging/dream/qdsp5/audio_mp3.c new file mode 100644 index 000000000000..b95574f699ff --- /dev/null +++ b/drivers/staging/dream/qdsp5/audio_mp3.c @@ -0,0 +1,971 @@ +/* arch/arm/mach-msm/qdsp5/audio_mp3.c + * + * mp3 audio output device + * + * Copyright (C) 2008 Google, Inc. + * Copyright (C) 2008 HTC Corporation + * + * 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 <asm/atomic.h> +#include <asm/ioctls.h> +#include <mach/msm_adsp.h> + +#include <linux/msm_audio.h> + +#include "audmgr.h" + +#include <mach/qdsp5/qdsp5audppcmdi.h> +#include <mach/qdsp5/qdsp5audppmsg.h> +#include <mach/qdsp5/qdsp5audplaycmdi.h> +#include <mach/qdsp5/qdsp5audplaymsg.h> + +/* for queue ids - should be relative to module number*/ +#include "adsp.h" + +#ifdef DEBUG +#define dprintk(format, arg...) \ +printk(KERN_DEBUG format, ## arg) +#else +#define dprintk(format, arg...) do {} while (0) +#endif + +/* Size must be power of 2 */ +#define BUFSZ_MAX 32768 +#define BUFSZ_MIN 4096 +#define DMASZ_MAX (BUFSZ_MAX * 2) +#define DMASZ_MIN (BUFSZ_MIN * 2) + +#define AUDPLAY_INVALID_READ_PTR_OFFSET 0xFFFF +#define AUDDEC_DEC_MP3 2 + +#define PCM_BUFSZ_MIN 4800 /* Hold one stereo MP3 frame */ +#define PCM_BUF_MAX_COUNT 5 /* DSP only accepts 5 buffers at most + but support 2 buffers currently */ +#define ROUTING_MODE_FTRT 1 +#define ROUTING_MODE_RT 2 +/* Decoder status received from AUDPPTASK */ +#define AUDPP_DEC_STATUS_SLEEP 0 +#define AUDPP_DEC_STATUS_INIT 1 +#define AUDPP_DEC_STATUS_CFG 2 +#define AUDPP_DEC_STATUS_PLAY 3 + +struct buffer { + void *data; + unsigned size; + unsigned used; /* Input usage actual DSP produced PCM size */ + unsigned addr; +}; + +struct audio { + struct buffer out[2]; + + spinlock_t dsp_lock; + + uint8_t out_head; + uint8_t out_tail; + uint8_t out_needed; /* number of buffers the dsp is waiting for */ + unsigned out_dma_sz; + + atomic_t out_bytes; + + struct mutex lock; + struct mutex write_lock; + wait_queue_head_t write_wait; + + /* Host PCM section */ + struct buffer in[PCM_BUF_MAX_COUNT]; + struct mutex read_lock; + wait_queue_head_t read_wait; /* Wait queue for read */ + char *read_data; /* pointer to reader buffer */ + dma_addr_t read_phys; /* physical address of reader buffer */ + uint8_t read_next; /* index to input buffers to be read next */ + uint8_t fill_next; /* index to buffer that DSP should be filling */ + uint8_t pcm_buf_count; /* number of pcm buffer allocated */ + /* ---- End of Host PCM section */ + + struct msm_adsp_module *audplay; + + /* configuration to use on next enable */ + uint32_t out_sample_rate; + uint32_t out_channel_mode; + + struct audmgr audmgr; + + /* data allocated for various buffers */ + char *data; + dma_addr_t phys; + + int rflush; /* Read flush */ + int wflush; /* Write flush */ + int opened; + int enabled; + int running; + int stopped; /* set when stopped, cleared on flush */ + int pcm_feedback; + int buf_refresh; + + int reserved; /* A byte is being reserved */ + char rsv_byte; /* Handle odd length user data */ + + unsigned volume; + + uint16_t dec_id; + uint32_t read_ptr_offset; +}; + +static int auddec_dsp_config(struct audio *audio, int enable); +static void audpp_cmd_cfg_adec_params(struct audio *audio); +static void audpp_cmd_cfg_routing_mode(struct audio *audio); +static void audplay_send_data(struct audio *audio, unsigned needed); +static void audplay_config_hostpcm(struct audio *audio); +static void audplay_buffer_refresh(struct audio *audio); +static void audio_dsp_event(void *private, unsigned id, uint16_t *msg); + +/* must be called with audio->lock held */ +static int audio_enable(struct audio *audio) +{ + struct audmgr_config cfg; + int rc; + + pr_info("audio_enable()\n"); + + if (audio->enabled) + return 0; + + audio->out_tail = 0; + audio->out_needed = 0; + + cfg.tx_rate = RPC_AUD_DEF_SAMPLE_RATE_NONE; + cfg.rx_rate = RPC_AUD_DEF_SAMPLE_RATE_48000; + cfg.def_method = RPC_AUD_DEF_METHOD_PLAYBACK; + cfg.codec = RPC_AUD_DEF_CODEC_MP3; + cfg.snd_method = RPC_SND_METHOD_MIDI; + + rc = audmgr_enable(&audio->audmgr, &cfg); + if (rc < 0) + return rc; + + if (msm_adsp_enable(audio->audplay)) { + pr_err("audio: msm_adsp_enable(audplay) failed\n"); + audmgr_disable(&audio->audmgr); + return -ENODEV; + } + + if (audpp_enable(audio->dec_id, audio_dsp_event, audio)) { + pr_err("audio: audpp_enable() failed\n"); + msm_adsp_disable(audio->audplay); + audmgr_disable(&audio->audmgr); + return -ENODEV; + } + + audio->enabled = 1; + return 0; +} + +/* must be called with audio->lock held */ +static int audio_disable(struct audio *audio) +{ + pr_info("audio_disable()\n"); + if (audio->enabled) { + audio->enabled = 0; + auddec_dsp_config(audio, 0); + wake_up(&audio->write_wait); + wake_up(&audio->read_wait); + msm_adsp_disable(audio->audplay); + audpp_disable(audio->dec_id, audio); + audmgr_disable(&audio->audmgr); + audio->out_needed = 0; + } + return 0; +} + +/* ------------------- dsp --------------------- */ +static void audio_update_pcm_buf_entry(struct audio *audio, uint32_t *payload) +{ + uint8_t index; + unsigned long flags; + + if (audio->rflush) { + audio->buf_refresh = 1; + return; + } + spin_lock_irqsave(&audio->dsp_lock, flags); + for (index = 0; index < payload[1]; index++) { + if (audio->in[audio->fill_next].addr == + payload[2 + index * 2]) { + pr_info("audio_update_pcm_buf_entry: in[%d] ready\n", + audio->fill_next); + audio->in[audio->fill_next].used = + payload[3 + index * 2]; + if ((++audio->fill_next) == audio->pcm_buf_count) + audio->fill_next = 0; + + } else { + pr_err + ("audio_update_pcm_buf_entry: expected=%x ret=%x\n" + , audio->in[audio->fill_next].addr, + payload[1 + index * 2]); + break; + } + } + if (audio->in[audio->fill_next].used == 0) { + audplay_buffer_refresh(audio); + } else { + pr_info("audio_update_pcm_buf_entry: read cannot keep up\n"); + audio->buf_refresh = 1; + } + wake_up(&audio->read_wait); + spin_unlock_irqrestore(&audio->dsp_lock, flags); + +} + +static void audplay_dsp_event(void *data, unsigned id, size_t len, + void (*getevent) (void *ptr, size_t len)) +{ + struct audio *audio = data; + uint32_t msg[28]; + getevent(msg, sizeof(msg)); + + dprintk("audplay_dsp_event: msg_id=%x\n", id); + + switch (id) { + case AUDPLAY_MSG_DEC_NEEDS_DATA: + audplay_send_data(audio, 1); + break; + + case AUDPLAY_MSG_BUFFER_UPDATE: + audio_update_pcm_buf_entry(audio, msg); + break; + + default: + pr_err("unexpected message from decoder \n"); + break; + } +} + +static void audio_dsp_event(void *private, unsigned id, uint16_t *msg) +{ + struct audio *audio = private; + + switch (id) { + case AUDPP_MSG_STATUS_MSG:{ + unsigned status = msg[1]; + + switch (status) { + case AUDPP_DEC_STATUS_SLEEP: + pr_info("decoder status: sleep \n"); + break; + + case AUDPP_DEC_STATUS_INIT: + pr_info("decoder status: init \n"); + audpp_cmd_cfg_routing_mode(audio); + break; + + case AUDPP_DEC_STATUS_CFG: + pr_info("decoder status: cfg \n"); + break; + case AUDPP_DEC_STATUS_PLAY: + pr_info("decoder status: play \n"); + if (audio->pcm_feedback) { + audplay_config_hostpcm(audio); + audplay_buffer_refresh(audio); + } + break; + default: + pr_err("unknown decoder status \n"); + break; + } + break; + } + case AUDPP_MSG_CFG_MSG: + if (msg[0] == AUDPP_MSG_ENA_ENA) { + pr_info("audio_dsp_event: CFG_MSG ENABLE\n"); + auddec_dsp_config(audio, 1); + audio->out_needed = 0; + audio->running = 1; + audpp_set_volume_and_pan(audio->dec_id, audio->volume, + 0); + audpp_avsync(audio->dec_id, 22050); + } else if (msg[0] == AUDPP_MSG_ENA_DIS) { + pr_info("audio_dsp_event: CFG_MSG DISABLE\n"); + audpp_avsync(audio->dec_id, 0); + audio->running = 0; + } else { + pr_err("audio_dsp_event: CFG_MSG %d?\n", msg[0]); + } + break; + case AUDPP_MSG_ROUTING_ACK: + pr_info("audio_dsp_event: ROUTING_ACK mode=%d\n", msg[1]); + audpp_cmd_cfg_adec_params(audio); + break; + + case AUDPP_MSG_FLUSH_ACK: + dprintk("%s: FLUSH_ACK\n", __func__); + audio->wflush = 0; + audio->rflush = 0; + if (audio->pcm_feedback) + audplay_buffer_refresh(audio); + break; + + default: + pr_err("audio_dsp_event: UNKNOWN (%d)\n", id); + } + +} + + +struct msm_adsp_ops audplay_adsp_ops = { + .event = audplay_dsp_event, +}; + + +#define audplay_send_queue0(audio, cmd, len) \ + msm_adsp_write(audio->audplay, QDSP_uPAudPlay0BitStreamCtrlQueue, \ + cmd, len) + +static int auddec_dsp_config(struct audio *audio, int enable) +{ + audpp_cmd_cfg_dec_type cmd; + + memset(&cmd, 0, sizeof(cmd)); + cmd.cmd_id = AUDPP_CMD_CFG_DEC_TYPE; + if (enable) + cmd.dec0_cfg = AUDPP_CMD_UPDATDE_CFG_DEC | + AUDPP_CMD_ENA_DEC_V | + AUDDEC_DEC_MP3; + else + cmd.dec0_cfg = AUDPP_CMD_UPDATDE_CFG_DEC | + AUDPP_CMD_DIS_DEC_V; + + return audpp_send_queue1(&cmd, sizeof(cmd)); +} + +static void audpp_cmd_cfg_adec_params(struct audio *audio) +{ + audpp_cmd_cfg_adec_params_mp3 cmd; + + memset(&cmd, 0, sizeof(cmd)); + cmd.common.cmd_id = AUDPP_CMD_CFG_ADEC_PARAMS; + cmd.common.length = AUDPP_CMD_CFG_ADEC_PARAMS_MP3_LEN; + cmd.common.dec_id = audio->dec_id; + cmd.common.input_sampling_frequency = audio->out_sample_rate; + + audpp_send_queue2(&cmd, sizeof(cmd)); +} + +static void audpp_cmd_cfg_routing_mode(struct audio *audio) +{ + struct audpp_cmd_routing_mode cmd; + pr_info("audpp_cmd_cfg_routing_mode()\n"); + memset(&cmd, 0, sizeof(cmd)); + cmd.cmd_id = AUDPP_CMD_ROUTING_MODE; + cmd.object_number = audio->dec_id; + if (audio->pcm_feedback) + cmd.routing_mode = ROUTING_MODE_FTRT; + else + cmd.routing_mode = ROUTING_MODE_RT; + + audpp_send_queue1(&cmd, sizeof(cmd)); +} + +static int audplay_dsp_send_data_avail(struct audio *audio, + unsigned idx, unsigned len) +{ + audplay_cmd_bitstream_data_avail cmd; + + cmd.cmd_id = AUDPLAY_CMD_BITSTREAM_DATA_AVAIL; + cmd.decoder_id = audio->dec_id; + cmd.buf_ptr = audio->out[idx].addr; + cmd.buf_size = len/2; + cmd.partition_number = 0; + return audplay_send_queue0(audio, &cmd, sizeof(cmd)); +} + +static void audplay_buffer_refresh(struct audio *audio) +{ + struct audplay_cmd_buffer_refresh refresh_cmd; + + refresh_cmd.cmd_id = AUDPLAY_CMD_BUFFER_REFRESH; + refresh_cmd.num_buffers = 1; + refresh_cmd.buf0_address = audio->in[audio->fill_next].addr; + refresh_cmd.buf0_length = audio->in[audio->fill_next].size - + (audio->in[audio->fill_next].size % 576); /* Mp3 frame size */ + refresh_cmd.buf_read_count = 0; + pr_info("audplay_buffer_fresh: buf0_addr=%x buf0_len=%d\n", + refresh_cmd.buf0_address, refresh_cmd.buf0_length); + (void)audplay_send_queue0(audio, &refresh_cmd, sizeof(refresh_cmd)); +} + +static void audplay_config_hostpcm(struct audio *audio) +{ + struct audplay_cmd_hpcm_buf_cfg cfg_cmd; + + pr_info("audplay_config_hostpcm()\n"); + cfg_cmd.cmd_id = AUDPLAY_CMD_HPCM_BUF_CFG; + cfg_cmd.max_buffers = 1; + cfg_cmd.byte_swap = 0; + cfg_cmd.hostpcm_config = (0x8000) | (0x4000); + cfg_cmd.feedback_frequency = 1; + cfg_cmd.partition_number = 0; + (void)audplay_send_queue0(audio, &cfg_cmd, sizeof(cfg_cmd)); + +} + +static void audplay_send_data(struct audio *audio, unsigned needed) +{ + struct buffer *frame; + unsigned long flags; + + spin_lock_irqsave(&audio->dsp_lock, flags); + if (!audio->running) + goto done; + + if (audio->wflush) { + audio->out_needed = 1; + goto done; + } + + if (needed && !audio->wflush) { + /* We were called from the callback because the DSP + * requested more data. Note that the DSP does want + * more data, and if a buffer was in-flight, mark it + * as available (since the DSP must now be done with + * it). + */ + audio->out_needed = 1; + frame = audio->out + audio->out_tail; + if (frame->used == 0xffffffff) { + dprintk("frame %d free\n", audio->out_tail); + frame->used = 0; + audio->out_tail ^= 1; + wake_up(&audio->write_wait); + } + } + + if (audio->out_needed) { + /* If the DSP currently wants data and we have a + * buffer available, we will send it and reset + * the needed flag. We'll mark the buffer as in-flight + * so that it won't be recycled until the next buffer + * is requested + */ + + frame = audio->out + audio->out_tail; + if (frame->used) { + BUG_ON(frame->used == 0xffffffff); + dprintk("frame %d busy\n", audio->out_tail); + audplay_dsp_send_data_avail(audio, audio->out_tail, + frame->used); + frame->used = 0xffffffff; + audio->out_needed = 0; + } + } +done: + spin_unlock_irqrestore(&audio->dsp_lock, flags); +} + +/* ------------------- device --------------------- */ + +static void audio_flush(struct audio *audio) +{ + audio->out[0].used = 0; + audio->out[1].used = 0; + audio->out_head = 0; + audio->out_tail = 0; + audio->reserved = 0; + atomic_set(&audio->out_bytes, 0); +} + +static void audio_flush_pcm_buf(struct audio *audio) +{ + uint8_t index; + + for (index = 0; index < PCM_BUF_MAX_COUNT; index++) + audio->in[index].used = 0; + + audio->read_next = 0; + audio->fill_next = 0; +} + +static void audio_ioport_reset(struct audio *audio) +{ + /* Make sure read/write thread are free from + * sleep and knowing that system is not able + * to process io request at the moment + */ + wake_up(&audio->write_wait); + mutex_lock(&audio->write_lock); + audio_flush(audio); + mutex_unlock(&audio->write_lock); + wake_up(&audio->read_wait); + mutex_lock(&audio->read_lock); + audio_flush_pcm_buf(audio); + mutex_unlock(&audio->read_lock); +} + +static long audio_ioctl(struct file *file, unsigned int cmd, unsigned long arg) +{ + struct audio *audio = file->private_data; + int rc = 0; + + pr_info("audio_ioctl() cmd = %d\n", cmd); + + if (cmd == AUDIO_GET_STATS) { + struct msm_audio_stats stats; + stats.byte_count = audpp_avsync_byte_count(audio->dec_id); + stats.sample_count = audpp_avsync_sample_count(audio->dec_id); + if (copy_to_user((void *) arg, &stats, sizeof(stats))) + return -EFAULT; + return 0; + } + if (cmd == AUDIO_SET_VOLUME) { + unsigned long flags; + spin_lock_irqsave(&audio->dsp_lock, flags); + audio->volume = arg; + if (audio->running) + audpp_set_volume_and_pan(audio->dec_id, arg, 0); + spin_unlock_irqrestore(&audio->dsp_lock, flags); + return 0; + } + mutex_lock(&audio->lock); + switch (cmd) { + case AUDIO_START: + rc = audio_enable(audio); + break; + case AUDIO_STOP: + rc = audio_disable(audio); + audio->stopped = 1; + audio_ioport_reset(audio); + audio->stopped = 0; + break; + case AUDIO_FLUSH: + dprintk("%s: AUDIO_FLUSH\n", __func__); + audio->rflush = 1; + audio->wflush = 1; + audio_ioport_reset(audio); + audio->rflush = 0; + audio->wflush = 0; + + if (audio->buf_refresh) { + audio->buf_refresh = 0; + audplay_buffer_refresh(audio); + } + break; + + case AUDIO_SET_CONFIG: { + struct msm_audio_config config; + if (copy_from_user(&config, (void *) arg, sizeof(config))) { + rc = -EFAULT; + break; + } + if (config.channel_count == 1) { + config.channel_count = AUDPP_CMD_PCM_INTF_MONO_V; + } else if (config.channel_count == 2) { + config.channel_count = AUDPP_CMD_PCM_INTF_STEREO_V; + } else { + rc = -EINVAL; + break; + } + audio->out_sample_rate = config.sample_rate; + audio->out_channel_mode = config.channel_count; + rc = 0; + break; + } + case AUDIO_GET_CONFIG: { + struct msm_audio_config config; + config.buffer_size = (audio->out_dma_sz >> 1); + config.buffer_count = 2; + config.sample_rate = audio->out_sample_rate; + if (audio->out_channel_mode == AUDPP_CMD_PCM_INTF_MONO_V) { + config.channel_count = 1; + } else { + config.channel_count = 2; + } + config.unused[0] = 0; + config.unused[1] = 0; + config.unused[2] = 0; + config.unused[3] = 0; + if (copy_to_user((void *) arg, &config, sizeof(config))) { + rc = -EFAULT; + } else { + rc = 0; + } + break; + } + case AUDIO_GET_PCM_CONFIG:{ + struct msm_audio_pcm_config config; + config.pcm_feedback = 0; + config.buffer_count = PCM_BUF_MAX_COUNT; + config.buffer_size = PCM_BUFSZ_MIN; + if (copy_to_user((void *)arg, &config, + sizeof(config))) + rc = -EFAULT; + else + rc = 0; + break; + } + case AUDIO_SET_PCM_CONFIG:{ + struct msm_audio_pcm_config config; + if (copy_from_user + (&config, (void *)arg, sizeof(config))) { + rc = -EFAULT; + break; + } + if ((config.buffer_count > PCM_BUF_MAX_COUNT) || + (config.buffer_count == 1)) + config.buffer_count = PCM_BUF_MAX_COUNT; + + if (config.buffer_size < PCM_BUFSZ_MIN) + config.buffer_size = PCM_BUFSZ_MIN; + + /* Check if pcm feedback is required */ + if ((config.pcm_feedback) && (!audio->read_data)) { + pr_info("ioctl: allocate PCM buffer %d\n", + config.buffer_count * + config.buffer_size); + audio->read_data = + dma_alloc_coherent(NULL, + config.buffer_size * + config.buffer_count, + &audio->read_phys, + GFP_KERNEL); + if (!audio->read_data) { + pr_err("audio_mp3: malloc pcm \ + buf failed\n"); + rc = -1; + } else { + uint8_t index; + uint32_t offset = 0; + audio->pcm_feedback = 1; + audio->buf_refresh = 0; + audio->pcm_buf_count = + config.buffer_count; + audio->read_next = 0; + audio->fill_next = 0; + + for (index = 0; + index < config.buffer_count; + index++) { + audio->in[index].data = + audio->read_data + offset; + audio->in[index].addr = + audio->read_phys + offset; + audio->in[index].size = + config.buffer_size; + audio->in[index].used = 0; + offset += config.buffer_size; + } + rc = 0; + } + } else { + rc = 0; + } + break; + } + case AUDIO_PAUSE: + dprintk("%s: AUDIO_PAUSE %ld\n", __func__, arg); + rc = audpp_pause(audio->dec_id, (int) arg); + break; + default: + rc = -EINVAL; + } + mutex_unlock(&audio->lock); + return rc; +} + +static ssize_t audio_read(struct file *file, char __user *buf, size_t count, + loff_t *pos) +{ + struct audio *audio = file->private_data; + const char __user *start = buf; + int rc = 0; + + if (!audio->pcm_feedback) + return 0; /* PCM feedback disabled. Nothing to read */ + + mutex_lock(&audio->read_lock); + pr_info("audio_read() %d \n", count); + while (count > 0) { + rc = wait_event_interruptible(audio->read_wait, + (audio->in[audio->read_next]. + used > 0) || (audio->stopped) + || (audio->rflush)); + + if (rc < 0) + break; + + if (audio->stopped || audio->rflush) { + rc = -EBUSY; + break; + } + + if (count < audio->in[audio->read_next].used) { + /* Read must happen in frame boundary. Since + * driver does not know frame size, read count + * must be greater or equal + * to size of PCM samples + */ + pr_info("audio_read: no partial frame done reading\n"); + break; + } else { + pr_info("audio_read: read from in[%d]\n", + audio->read_next); + if (copy_to_user + (buf, audio->in[audio->read_next].data, + audio->in[audio->read_next].used)) { + pr_err("audio_read: invalid addr %x \n", + (unsigned int)buf); + rc = -EFAULT; + break; + } + count -= audio->in[audio->read_next].used; + buf += audio->in[audio->read_next].used; + audio->in[audio->read_next].used = 0; + if ((++audio->read_next) == audio->pcm_buf_count) + audio->read_next = 0; + if (audio->in[audio->read_next].used == 0) + break; /* No data ready at this moment + * Exit while loop to prevent + * output thread sleep too long + */ + } + } + + /* don't feed output buffer to HW decoder during flushing + * buffer refresh command will be sent once flush completes + * send buf refresh command here can confuse HW decoder + */ + if (audio->buf_refresh && !audio->rflush) { + audio->buf_refresh = 0; + pr_info("audio_read: kick start pcm feedback again\n"); + audplay_buffer_refresh(audio); + } + + mutex_unlock(&audio->read_lock); + + if (buf > start) + rc = buf - start; + + pr_info("audio_read: read %d bytes\n", rc); + return rc; +} + +static ssize_t audio_write(struct file *file, const char __user *buf, + size_t count, loff_t *pos) +{ + struct audio *audio = file->private_data; + const char __user *start = buf; + struct buffer *frame; + size_t xfer; + char *cpy_ptr; + int rc = 0; + unsigned dsize; + + mutex_lock(&audio->write_lock); + while (count > 0) { + frame = audio->out + audio->out_head; + cpy_ptr = frame->data; + dsize = 0; + rc = wait_event_interruptible(audio->write_wait, + (frame->used == 0) + || (audio->stopped) + || (audio->wflush)); + if (rc < 0) + break; + if (audio->stopped || audio->wflush) { + rc = -EBUSY; + break; + } + + if (audio->reserved) { + dprintk("%s: append reserved byte %x\n", + __func__, audio->rsv_byte); + *cpy_ptr = audio->rsv_byte; + xfer = (count > (frame->size - 1)) ? + frame->size - 1 : count; + cpy_ptr++; + dsize = 1; + audio->reserved = 0; + } else + xfer = (count > frame->size) ? frame->size : count; + + if (copy_from_user(cpy_ptr, buf, xfer)) { + rc = -EFAULT; + break; + } + + dsize += xfer; + if (dsize & 1) { + audio->rsv_byte = ((char *) frame->data)[dsize - 1]; + dprintk("%s: odd length buf reserve last byte %x\n", + __func__, audio->rsv_byte); + audio->reserved = 1; + dsize--; + } + count -= xfer; + buf += xfer; + + if (dsize > 0) { + audio->out_head ^= 1; + frame->used = dsize; + audplay_send_data(audio, 0); + } + } + mutex_unlock(&audio->write_lock); + if (buf > start) + return buf - start; + return rc; +} + +static int audio_release(struct inode *inode, struct file *file) +{ + struct audio *audio = file->private_data; + + dprintk("audio_release()\n"); + + mutex_lock(&audio->lock); + audio_disable(audio); + audio_flush(audio); + audio_flush_pcm_buf(audio); + msm_adsp_put(audio->audplay); + audio->audplay = NULL; + audio->opened = 0; + audio->reserved = 0; + dma_free_coherent(NULL, audio->out_dma_sz, audio->data, audio->phys); + audio->data = NULL; + if (audio->read_data != NULL) { + dma_free_coherent(NULL, + audio->in[0].size * audio->pcm_buf_count, + audio->read_data, audio->read_phys); + audio->read_data = NULL; + } + audio->pcm_feedback = 0; + mutex_unlock(&audio->lock); + return 0; +} + +static struct audio the_mp3_audio; + +static int audio_open(struct inode *inode, struct file *file) +{ + struct audio *audio = &the_mp3_audio; + int rc; + unsigned pmem_sz; + + mutex_lock(&audio->lock); + + if (audio->opened) { + pr_err("audio: busy\n"); + rc = -EBUSY; + goto done; + } + + pmem_sz = DMASZ_MAX; + + while (pmem_sz >= DMASZ_MIN) { + audio->data = dma_alloc_coherent(NULL, pmem_sz, + &audio->phys, GFP_KERNEL); + if (audio->data) + break; + else if (pmem_sz == DMASZ_MIN) { + pr_err("audio: could not allocate DMA buffers\n"); + rc = -ENOMEM; + goto done; + } else + pmem_sz >>= 1; + } + + dprintk("%s: allocated %d bytes DMA buffer\n", __func__, pmem_sz); + + rc = audmgr_open(&audio->audmgr); + if (rc) { + dma_free_coherent(NULL, pmem_sz, + audio->data, audio->phys); + goto done; + } + + rc = msm_adsp_get("AUDPLAY0TASK", &audio->audplay, &audplay_adsp_ops, + audio); + if (rc) { + pr_err("audio: failed to get audplay0 dsp module\n"); + dma_free_coherent(NULL, pmem_sz, + audio->data, audio->phys); + audmgr_close(&audio->audmgr); + goto done; + } + + audio->out_dma_sz = pmem_sz; + pmem_sz >>= 1; /* Shift by 1 to get size of ping pong buffer */ + + audio->out_sample_rate = 44100; + audio->out_channel_mode = AUDPP_CMD_PCM_INTF_STEREO_V; + audio->dec_id = 0; + + audio->out[0].data = audio->data + 0; + audio->out[0].addr = audio->phys + 0; + audio->out[0].size = pmem_sz; + + audio->out[1].data = audio->data + pmem_sz; + audio->out[1].addr = audio->phys + pmem_sz; + audio->out[1].size = pmem_sz; + + audio->volume = 0x2000; /* equal to Q13 number 1.0 Unit Gain */ + + audio_flush(audio); + + file->private_data = audio; + audio->opened = 1; + rc = 0; +done: + mutex_unlock(&audio->lock); + return rc; +} + +static struct file_operations audio_mp3_fops = { + .owner = THIS_MODULE, + .open = audio_open, + .release = audio_release, + .read = audio_read, + .write = audio_write, + .unlocked_ioctl = audio_ioctl, +}; + +struct miscdevice audio_mp3_misc = { + .minor = MISC_DYNAMIC_MINOR, + .name = "msm_mp3", + .fops = &audio_mp3_fops, +}; + +static int __init audio_init(void) +{ + mutex_init(&the_mp3_audio.lock); + mutex_init(&the_mp3_audio.write_lock); + mutex_init(&the_mp3_audio.read_lock); + spin_lock_init(&the_mp3_audio.dsp_lock); + init_waitqueue_head(&the_mp3_audio.write_wait); + init_waitqueue_head(&the_mp3_audio.read_wait); + the_mp3_audio.read_data = NULL; + return misc_register(&audio_mp3_misc); +} + +device_initcall(audio_init); |