M7350/kernel/sound/soc/msm/qdsp6v2/msm-pcm-lpa-v2.c
2024-09-09 08:52:07 +00:00

665 lines
18 KiB
C

/* Copyright (c) 2012-2013, The Linux Foundation. All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 and
* only 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.
*/
#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 <linux/slab.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 <sound/pcm_params.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <linux/of_device.h>
#include <linux/msm_audio_ion.h>
#include <sound/compress_params.h>
#include <sound/compress_offload.h>
#include <sound/compress_driver.h>
#include <sound/timer.h>
#include <sound/pcm_params.h>
#include "msm-pcm-q6-v2.h"
#include "msm-pcm-routing-v2.h"
#include "audio_ocmem.h"
static struct audio_locks the_locks;
struct snd_msm {
struct msm_audio *prtd;
unsigned volume;
atomic_t audio_ocmem_req;
};
static struct snd_msm lpa_audio;
static struct snd_pcm_hardware msm_pcm_hardware = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
.rates = SNDRV_PCM_RATE_8000_192000 |
SNDRV_PCM_RATE_KNOT,
.rate_min = 8000,
.rate_max = 192000,
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 1024 * 1024,
.period_bytes_min = 128 * 1024,
.period_bytes_max = 256 * 1024,
.periods_min = 4,
.periods_max = 8,
.fifo_size = 0,
};
/* Conventional and unconventional sample rate supported */
static unsigned int supported_sample_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
96000, 192000
};
static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
.count = ARRAY_SIZE(supported_sample_rates),
.list = supported_sample_rates,
.mask = 0,
};
static void event_handler(uint32_t opcode,
uint32_t token, uint32_t *payload, void *priv)
{
struct msm_audio *prtd = priv;
struct snd_pcm_substream *substream = prtd->substream;
struct snd_pcm_runtime *runtime = substream->runtime;
struct audio_aio_write_param param;
struct audio_buffer *buf = NULL;
unsigned long flag = 0;
int i = 0;
spin_lock_irqsave(&the_locks.event_lock, flag);
switch (opcode) {
case ASM_DATA_EVENT_WRITE_DONE_V2: {
uint32_t *ptrmem = (uint32_t *)&param;
pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2\n");
pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
prtd->pcm_irq_pos += prtd->pcm_count;
if (prtd->pcm_irq_pos >= prtd->pcm_size)
prtd->pcm_irq_pos = 0;
if (atomic_read(&prtd->start))
snd_pcm_period_elapsed(substream);
else
if (substream->timer_running)
snd_timer_interrupt(substream->timer, 1);
atomic_inc(&prtd->out_count);
wake_up(&the_locks.write_wait);
if (!atomic_read(&prtd->start)) {
atomic_set(&prtd->pending_buffer, 1);
break;
} else
atomic_set(&prtd->pending_buffer, 0);
buf = prtd->audio_client->port[IN].buf;
if (runtime->status->hw_ptr >= runtime->control->appl_ptr) {
memset((void *)buf[0].data +
(prtd->out_head * prtd->pcm_count),
0, prtd->pcm_count);
}
pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
__func__, prtd->pcm_count);
param.paddr = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count);
param.len = prtd->pcm_count;
param.msw_ts = 0;
param.lsw_ts = 0;
param.flags = NO_TIMESTAMP;
param.uid = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count);
for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
i++, ++ptrmem)
pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
if (q6asm_async_write(prtd->audio_client,
&param) < 0)
pr_err("%s:q6asm_async_write failed\n",
__func__);
else
prtd->out_head =
(prtd->out_head + 1) & (runtime->periods - 1);
atomic_set(&prtd->pending_buffer, 0);
break;
}
case ASM_DATA_EVENT_RENDERED_EOS:
pr_debug("ASM_DATA_CMDRSP_EOS\n");
prtd->cmd_ack = 1;
wake_up(&the_locks.eos_wait);
break;
case APR_BASIC_RSP_RESULT: {
switch (payload[0]) {
case ASM_SESSION_CMD_RUN_V2: {
if (!atomic_read(&prtd->pending_buffer))
break;
if (runtime->status->hw_ptr >=
runtime->control->appl_ptr)
break;
pr_debug("%s:writing %d bytes of buffer to dsp\n",
__func__, prtd->pcm_count);
buf = prtd->audio_client->port[IN].buf;
param.paddr = (unsigned long)buf[prtd->out_head].phys;
param.len = prtd->pcm_count;
param.msw_ts = 0;
param.lsw_ts = 0;
param.flags = NO_TIMESTAMP;
param.uid = (unsigned long)buf[prtd->out_head].phys;
if (q6asm_async_write(prtd->audio_client,
&param) < 0)
pr_err("%s:q6asm_async_write failed\n",
__func__);
else
prtd->out_head =
(prtd->out_head + 1)
& (runtime->periods - 1);
atomic_set(&prtd->pending_buffer, 0);
}
break;
case ASM_STREAM_CMD_FLUSH:
pr_debug("ASM_STREAM_CMD_FLUSH\n");
prtd->cmd_ack = 1;
wake_up(&the_locks.eos_wait);
break;
default:
break;
}
break;
}
default:
pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
break;
}
spin_unlock_irqrestore(&the_locks.event_lock, flag);
}
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;
int ret;
uint16_t bits_per_sample = 16;
pr_debug("%s\n", __func__);
prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
prtd->pcm_irq_pos = 0;
/* rate and channels are sent to audio driver */
prtd->samp_rate = runtime->rate;
prtd->channel_mode = runtime->channels;
prtd->out_head = 0;
if (prtd->enabled)
return 0;
switch (runtime->format) {
case SNDRV_PCM_FORMAT_S16_LE:
bits_per_sample = 16;
break;
case SNDRV_PCM_FORMAT_S24_LE:
bits_per_sample = 24;
break;
}
ret = q6asm_media_format_block_pcm_format_support(
prtd->audio_client, runtime->rate,
runtime->channels, bits_per_sample);
if (ret < 0)
pr_debug("%s: CMD Format block failed\n", __func__);
atomic_set(&prtd->out_count, runtime->periods);
prtd->enabled = 1;
prtd->cmd_ack = 0;
prtd->cmd_interrupt = 0;
return 0;
}
static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct msm_audio *prtd = runtime->private_data;
pr_debug("%s\n", __func__);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
prtd->pcm_irq_pos = 0;
if (!atomic_cmpxchg(&lpa_audio.audio_ocmem_req, 0, 1))
audio_ocmem_process_req(AUDIO, true);
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
pr_debug("SNDRV_PCM_TRIGGER_START\n");
q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
atomic_set(&prtd->start, 1);
atomic_set(&prtd->stop, 0);
break;
case SNDRV_PCM_TRIGGER_STOP:
pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
atomic_set(&prtd->start, 0);
atomic_set(&prtd->stop, 1);
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
break;
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
atomic_set(&prtd->start, 0);
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static int msm_pcm_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct msm_audio *prtd;
int ret = 0;
pr_debug("%s\n", __func__);
prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
if (prtd == NULL) {
pr_err("Failed to allocate memory for msm_audio\n");
return -ENOMEM;
}
runtime->hw = msm_pcm_hardware;
prtd->substream = substream;
prtd->audio_client = q6asm_audio_client_alloc(
(app_cb)event_handler, prtd);
if (!prtd->audio_client) {
pr_debug("%s: Could not allocate memory\n", __func__);
kfree(prtd);
return -ENOMEM;
}
/* Capture path */
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
return -EPERM;
ret = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_sample_rates);
if (ret < 0)
pr_debug("snd_pcm_hw_constraint_list failed\n");
/* Ensure that buffer size is a multiple of period size */
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
pr_debug("snd_pcm_hw_constraint_integer failed\n");
prtd->dsp_cnt = 0;
atomic_set(&prtd->pending_buffer, 1);
atomic_set(&prtd->stop, 1);
atomic_set(&lpa_audio.audio_ocmem_req, 0);
runtime->private_data = prtd;
lpa_audio.prtd = prtd;
return 0;
}
int lpa_set_volume(unsigned volume)
{
int rc = 0;
if (lpa_audio.prtd && lpa_audio.prtd->audio_client) {
rc = q6asm_set_lrgain(lpa_audio.prtd->audio_client,
(volume >> 16) & 0xFFFF,
volume & 0xFFFF);
if (rc < 0) {
pr_err("%s: Send Volume command failed rc=%d\n",
__func__, rc);
}
}
lpa_audio.volume = volume;
return rc;
}
static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct msm_audio *prtd = runtime->private_data;
int dir = 0;
int rc = 0;
/*
If routing is still enabled, we need to issue EOS to
the DSP
To issue EOS to dsp, we need to be run state otherwise
EOS is not honored.
*/
if (msm_routing_check_backend_enabled(soc_prtd->dai_link->be_id) &&
(!atomic_read(&prtd->stop))) {
rc = q6asm_run(prtd->audio_client, 0, 0, 0);
atomic_set(&prtd->pending_buffer, 0);
prtd->cmd_ack = 0;
q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
pr_debug("%s\n", __func__);
rc = wait_event_timeout(the_locks.eos_wait,
prtd->cmd_ack, 5 * HZ);
if (!rc)
pr_err("EOS cmd timeout\n");
prtd->pcm_irq_pos = 0;
}
if (prtd->audio_client) {
dir = IN;
atomic_set(&prtd->pending_buffer, 0);
if (atomic_cmpxchg(&lpa_audio.audio_ocmem_req, 1, 0))
audio_ocmem_process_req(AUDIO, false);
lpa_audio.prtd = NULL;
q6asm_cmd(prtd->audio_client, CMD_CLOSE);
q6asm_audio_client_buf_free_contiguous(dir,
prtd->audio_client);
atomic_set(&prtd->stop, 1);
q6asm_audio_client_free(prtd->audio_client);
pr_debug("%s\n", __func__);
}
msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
SNDRV_PCM_STREAM_PLAYBACK);
pr_debug("%s\n", __func__);
kfree(prtd);
return 0;
}
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);
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);
return ret;
}
static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct msm_audio *prtd = runtime->private_data;
pr_debug("%s: pcm_irq_pos = %d\n", __func__, prtd->pcm_irq_pos);
return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
}
static int msm_pcm_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct msm_audio *prtd = runtime->private_data;
struct audio_client *ac = prtd->audio_client;
struct audio_port_data *apd = ac->port;
struct audio_buffer *ab;
int dir = -1;
prtd->mmap_flag = 1;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dir = IN;
else
dir = OUT;
ab = &(apd[dir].buf[0]);
return msm_audio_ion_mmap(ab, vma);
}
static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct msm_audio *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
struct audio_buffer *buf;
uint16_t bits_per_sample = 16;
int dir, ret;
struct asm_softpause_params softpause = {
.enable = SOFT_PAUSE_ENABLE,
.period = SOFT_PAUSE_PERIOD,
.step = SOFT_PAUSE_STEP,
.rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
};
struct asm_softvolume_params softvol = {
.period = SOFT_VOLUME_PERIOD,
.step = SOFT_VOLUME_STEP,
.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
};
prtd->audio_client->perf_mode = false;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (params_format(params) == SNDRV_PCM_FORMAT_S24_LE)
bits_per_sample = 24;
ret = q6asm_open_write_v2(prtd->audio_client,
FORMAT_LINEAR_PCM, bits_per_sample);
if (ret < 0) {
pr_err("%s: pcm out open failed\n", __func__);
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
return -ENOMEM;
}
ret = q6asm_set_io_mode(prtd->audio_client, ASYNC_IO_MODE);
if (ret < 0) {
pr_err("%s: Set IO mode failed\n", __func__);
q6asm_audio_client_free(prtd->audio_client);
prtd->audio_client = NULL;
return -ENOMEM;
}
}
pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
prtd->session_id = prtd->audio_client->session;
msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
prtd->audio_client->perf_mode,
prtd->session_id, substream->stream);
lpa_set_volume(0);
ret = q6asm_set_softpause(lpa_audio.prtd->audio_client, &softpause);
if (ret < 0)
pr_err("%s: Send SoftPause Param failed ret=%d\n",
__func__, ret);
ret = q6asm_set_softvolume(lpa_audio.prtd->audio_client, &softvol);
if (ret < 0)
pr_err("%s: Send SoftVolume Param failed ret=%d\n",
__func__, ret);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dir = IN;
else
return -EPERM;
ret = q6asm_audio_client_buf_alloc_contiguous(dir,
prtd->audio_client,
params_period_bytes(params),
params_periods(params));
if (ret < 0) {
pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
ret);
return -ENOMEM;
}
buf = prtd->audio_client->port[dir].buf;
if (buf == NULL || buf[0].data == NULL)
return -ENOMEM;
pr_debug("%s:buf = %p\n", __func__, buf);
dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
dma_buf->dev.dev = substream->pcm->card->dev;
dma_buf->private_data = NULL;
dma_buf->area = buf[0].data;
dma_buf->addr = buf[0].phys;
dma_buf->bytes = params_period_bytes(params) * params_periods(params);
if (!dma_buf->area)
return -ENOMEM;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
return 0;
}
static int msm_pcm_ioctl(struct snd_pcm_substream *substream,
unsigned int cmd, void *arg)
{
int rc = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct msm_audio *prtd = runtime->private_data;
uint64_t timestamp;
uint64_t temp;
switch (cmd) {
case SNDRV_COMPRESS_TSTAMP: {
struct snd_compr_tstamp tstamp;
pr_debug("SNDRV_COMPRESS_TSTAMP\n");
memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
rc = q6asm_get_session_time(prtd->audio_client, &timestamp);
if (rc < 0) {
pr_err("%s: Fail to get session time stamp, rc:%d\n",
__func__, rc);
return -EAGAIN;
}
temp = (timestamp * 2 * runtime->channels);
temp = temp * (runtime->rate/1000);
temp = div_u64(temp, 1000);
tstamp.sampling_rate = runtime->rate;
tstamp.timestamp = timestamp;
pr_debug("%s: bytes_consumed:timestamp = %lld,\n",
__func__,
tstamp.timestamp);
if (copy_to_user((void *) arg, &tstamp,
sizeof(struct snd_compr_tstamp)))
return -EFAULT;
return 0;
}
case SNDRV_PCM_IOCTL1_RESET:
prtd->cmd_ack = 0;
rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
if (rc < 0)
pr_err("%s: flush cmd failed rc=%d\n", __func__, rc);
rc = wait_event_timeout(the_locks.eos_wait,
prtd->cmd_ack, 5 * HZ);
if (!rc)
pr_err("Flush cmd timeout\n");
prtd->pcm_irq_pos = 0;
break;
default:
break;
}
return snd_pcm_lib_ioctl(substream, cmd, arg);
}
static struct snd_pcm_ops msm_pcm_ops = {
.open = msm_pcm_open,
.hw_params = msm_pcm_hw_params,
.close = msm_pcm_close,
.ioctl = msm_pcm_ioctl,
.prepare = msm_pcm_prepare,
.trigger = msm_pcm_trigger,
.pointer = msm_pcm_pointer,
.mmap = msm_pcm_mmap,
};
static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
int ret = 0;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
return ret;
}
static struct snd_soc_platform_driver msm_soc_platform = {
.ops = &msm_pcm_ops,
.pcm_new = msm_asoc_pcm_new,
};
static __devinit int msm_pcm_probe(struct platform_device *pdev)
{
if (pdev->dev.of_node)
dev_set_name(&pdev->dev, "%s", "msm-pcm-lpa");
dev_info(&pdev->dev, "%s: dev name %s\n",
__func__, dev_name(&pdev->dev));
return snd_soc_register_platform(&pdev->dev,
&msm_soc_platform);
}
static int msm_pcm_remove(struct platform_device *pdev)
{
snd_soc_unregister_platform(&pdev->dev);
return 0;
}
static const struct of_device_id msm_pcm_lpa_dt_match[] = {
{.compatible = "qcom,msm-pcm-lpa"},
{}
};
MODULE_DEVICE_TABLE(of, msm_pcm_lpa_dt_match);
static struct platform_driver msm_pcm_driver = {
.driver = {
.name = "msm-pcm-lpa",
.owner = THIS_MODULE,
.of_match_table = msm_pcm_lpa_dt_match,
},
.probe = msm_pcm_probe,
.remove = __devexit_p(msm_pcm_remove),
};
static int __init msm_soc_platform_init(void)
{
spin_lock_init(&the_locks.event_lock);
init_waitqueue_head(&the_locks.enable_wait);
init_waitqueue_head(&the_locks.eos_wait);
init_waitqueue_head(&the_locks.write_wait);
init_waitqueue_head(&the_locks.read_wait);
return platform_driver_register(&msm_pcm_driver);
}
module_init(msm_soc_platform_init);
static void __exit msm_soc_platform_exit(void)
{
platform_driver_unregister(&msm_pcm_driver);
}
module_exit(msm_soc_platform_exit);
MODULE_DESCRIPTION("PCM module platform driver");
MODULE_LICENSE("GPL v2");