619 lines
17 KiB
C
619 lines
17 KiB
C
/* Copyright (c) 2011-2013, The Linux Foundation. All rights reserved.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2 and
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* only version 2 as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*/
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#include <linux/init.h>
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#include <linux/err.h>
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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/time.h>
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#include <linux/wait.h>
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#include <linux/platform_device.h>
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#include <linux/slab.h>
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#include <sound/core.h>
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#include <sound/soc.h>
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#include <sound/soc-dapm.h>
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#include <sound/pcm.h>
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#include <sound/initval.h>
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#include <sound/control.h>
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#include <sound/pcm_params.h>
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#include <asm/dma.h>
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#include <linux/dma-mapping.h>
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#include <sound/compress_params.h>
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#include <sound/compress_offload.h>
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#include <sound/compress_driver.h>
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#include <sound/timer.h>
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#include "msm-pcm-q6.h"
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#include "msm-pcm-routing.h"
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static struct audio_locks the_locks;
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struct snd_msm {
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struct msm_audio *prtd;
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unsigned volume;
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};
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static struct snd_msm lpa_audio;
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static struct snd_pcm_hardware msm_pcm_hardware = {
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.info = (SNDRV_PCM_INFO_MMAP |
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SNDRV_PCM_INFO_BLOCK_TRANSFER |
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SNDRV_PCM_INFO_MMAP_VALID |
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SNDRV_PCM_INFO_INTERLEAVED |
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SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
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.formats = SNDRV_PCM_FMTBIT_S16_LE,
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.rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
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.rate_min = 8000,
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.rate_max = 48000,
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.channels_min = 1,
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.channels_max = 2,
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.buffer_bytes_max = 1024 * 1024,
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/* TODO: Check on the lowest period size we can support */
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.period_bytes_min = 128 * 1024,
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.period_bytes_max = 256 * 1024,
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.periods_min = 4,
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.periods_max = 8,
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.fifo_size = 0,
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};
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/* Conventional and unconventional sample rate supported */
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static unsigned int supported_sample_rates[] = {
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8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
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};
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static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
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.count = ARRAY_SIZE(supported_sample_rates),
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.list = supported_sample_rates,
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.mask = 0,
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};
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static void event_handler(uint32_t opcode,
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uint32_t token, uint32_t *payload, void *priv)
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{
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struct msm_audio *prtd = priv;
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struct snd_pcm_substream *substream = prtd->substream;
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct audio_aio_write_param param;
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struct audio_buffer *buf = NULL;
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unsigned long flag = 0;
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int i = 0;
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pr_debug("%s\n", __func__);
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spin_lock_irqsave(&the_locks.event_lock, flag);
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switch (opcode) {
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case ASM_DATA_EVENT_WRITE_DONE: {
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uint32_t *ptrmem = (uint32_t *)¶m;
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pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
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pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
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prtd->pcm_irq_pos += prtd->pcm_count;
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if (prtd->pcm_irq_pos >= prtd->pcm_size)
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prtd->pcm_irq_pos = 0;
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if (atomic_read(&prtd->start))
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snd_pcm_period_elapsed(substream);
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else
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if (substream->timer_running)
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snd_timer_interrupt(substream->timer, 1);
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atomic_inc(&prtd->out_count);
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wake_up(&the_locks.write_wait);
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if (!atomic_read(&prtd->start)) {
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atomic_set(&prtd->pending_buffer, 1);
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break;
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} else
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atomic_set(&prtd->pending_buffer, 0);
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buf = prtd->audio_client->port[IN].buf;
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if (runtime->status->hw_ptr >= runtime->control->appl_ptr) {
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memset((void *)buf[0].data +
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(prtd->out_head * prtd->pcm_count),
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0, prtd->pcm_count);
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}
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pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
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__func__, prtd->pcm_count);
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param.paddr = (unsigned long)buf[0].phys
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+ (prtd->out_head * prtd->pcm_count);
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param.len = prtd->pcm_count;
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param.msw_ts = 0;
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param.lsw_ts = 0;
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param.flags = NO_TIMESTAMP;
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param.uid = (unsigned long)buf[0].phys
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+ (prtd->out_head * prtd->pcm_count);
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for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
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i++, ++ptrmem)
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pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
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if (q6asm_async_write(prtd->audio_client,
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¶m) < 0)
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pr_err("%s:q6asm_async_write failed\n",
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__func__);
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else
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prtd->out_head =
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(prtd->out_head + 1) & (runtime->periods - 1);
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atomic_set(&prtd->pending_buffer, 0);
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break;
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}
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case ASM_DATA_CMDRSP_EOS:
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pr_debug("ASM_DATA_CMDRSP_EOS\n");
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prtd->cmd_ack = 1;
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wake_up(&the_locks.eos_wait);
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break;
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case APR_BASIC_RSP_RESULT: {
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switch (payload[0]) {
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case ASM_SESSION_CMD_RUN: {
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if (!atomic_read(&prtd->pending_buffer))
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break;
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if (runtime->status->hw_ptr >=
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runtime->control->appl_ptr)
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break;
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pr_debug("%s:writing %d bytes"
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" of buffer to dsp\n",
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__func__, prtd->pcm_count);
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buf = prtd->audio_client->port[IN].buf;
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param.paddr = (unsigned long)buf[prtd->out_head].phys;
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param.len = prtd->pcm_count;
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param.msw_ts = 0;
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param.lsw_ts = 0;
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param.flags = NO_TIMESTAMP;
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param.uid = (unsigned long)buf[prtd->out_head].phys;
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if (q6asm_async_write(prtd->audio_client,
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¶m) < 0)
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pr_err("%s:q6asm_async_write failed\n",
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__func__);
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else
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prtd->out_head =
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(prtd->out_head + 1)
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& (runtime->periods - 1);
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atomic_set(&prtd->pending_buffer, 0);
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}
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break;
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case ASM_STREAM_CMD_FLUSH:
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pr_debug("ASM_STREAM_CMD_FLUSH\n");
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prtd->cmd_ack = 1;
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wake_up(&the_locks.eos_wait);
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break;
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default:
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break;
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}
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break;
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}
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default:
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pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
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break;
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}
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spin_unlock_irqrestore(&the_locks.event_lock, flag);
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}
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static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct msm_audio *prtd = runtime->private_data;
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int ret;
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pr_debug("%s\n", __func__);
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prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
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prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
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prtd->pcm_irq_pos = 0;
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/* rate and channels are sent to audio driver */
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prtd->samp_rate = runtime->rate;
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prtd->channel_mode = runtime->channels;
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prtd->out_head = 0;
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if (prtd->enabled)
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return 0;
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ret = q6asm_media_format_block_pcm(prtd->audio_client, runtime->rate,
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runtime->channels);
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if (ret < 0)
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pr_debug("%s: CMD Format block failed\n", __func__);
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atomic_set(&prtd->out_count, runtime->periods);
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prtd->enabled = 1;
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prtd->cmd_ack = 0;
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return 0;
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}
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static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
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{
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int ret = 0;
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct msm_audio *prtd = runtime->private_data;
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pr_debug("%s\n", __func__);
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switch (cmd) {
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case SNDRV_PCM_TRIGGER_START:
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prtd->pcm_irq_pos = 0;
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case SNDRV_PCM_TRIGGER_RESUME:
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case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
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pr_debug("SNDRV_PCM_TRIGGER_START\n");
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q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
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atomic_set(&prtd->start, 1);
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atomic_set(&prtd->stop, 0);
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break;
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case SNDRV_PCM_TRIGGER_STOP:
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pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
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atomic_set(&prtd->start, 0);
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atomic_set(&prtd->stop, 1);
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if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
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break;
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break;
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case SNDRV_PCM_TRIGGER_SUSPEND:
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case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
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pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
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q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
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atomic_set(&prtd->start, 0);
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break;
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default:
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ret = -EINVAL;
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break;
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}
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return ret;
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}
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static int msm_pcm_open(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
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struct msm_audio *prtd;
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struct asm_softpause_params softpause = {
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.enable = SOFT_PAUSE_ENABLE,
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.period = SOFT_PAUSE_PERIOD,
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.step = SOFT_PAUSE_STEP,
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.rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
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};
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struct asm_softvolume_params softvol = {
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.period = SOFT_VOLUME_PERIOD,
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.step = SOFT_VOLUME_STEP,
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.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
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};
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int ret = 0;
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pr_debug("%s\n", __func__);
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prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
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if (prtd == NULL) {
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pr_err("Failed to allocate memory for msm_audio\n");
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return -ENOMEM;
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}
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runtime->hw = msm_pcm_hardware;
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prtd->substream = substream;
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prtd->audio_client = q6asm_audio_client_alloc(
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(app_cb)event_handler, prtd);
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if (!prtd->audio_client) {
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pr_debug("%s: Could not allocate memory\n", __func__);
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kfree(prtd);
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return -ENOMEM;
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}
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prtd->audio_client->perf_mode = false;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM);
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if (ret < 0) {
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pr_err("%s: pcm out open failed\n", __func__);
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q6asm_audio_client_free(prtd->audio_client);
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kfree(prtd);
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return -ENOMEM;
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}
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ret = q6asm_set_io_mode(prtd->audio_client, ASYNC_IO_MODE);
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if (ret < 0) {
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pr_err("%s: Set IO mode failed\n", __func__);
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q6asm_audio_client_free(prtd->audio_client);
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kfree(prtd);
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return -ENOMEM;
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}
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}
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/* Capture path */
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if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
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return -EPERM;
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pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
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prtd->session_id = prtd->audio_client->session;
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msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
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prtd->audio_client->perf_mode,
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prtd->session_id, substream->stream);
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ret = snd_pcm_hw_constraint_list(runtime, 0,
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SNDRV_PCM_HW_PARAM_RATE,
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&constraints_sample_rates);
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if (ret < 0)
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pr_debug("snd_pcm_hw_constraint_list failed\n");
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/* Ensure that buffer size is a multiple of period size */
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ret = snd_pcm_hw_constraint_integer(runtime,
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SNDRV_PCM_HW_PARAM_PERIODS);
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if (ret < 0)
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pr_debug("snd_pcm_hw_constraint_integer failed\n");
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prtd->dsp_cnt = 0;
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atomic_set(&prtd->pending_buffer, 1);
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atomic_set(&prtd->stop, 1);
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runtime->private_data = prtd;
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lpa_audio.prtd = prtd;
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lpa_set_volume(0);
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ret = q6asm_set_softpause(lpa_audio.prtd->audio_client, &softpause);
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if (ret < 0)
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pr_err("%s: Send SoftPause Param failed ret=%d\n",
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__func__, ret);
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ret = q6asm_set_softvolume(lpa_audio.prtd->audio_client, &softvol);
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if (ret < 0)
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pr_err("%s: Send SoftVolume Param failed ret=%d\n",
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__func__, ret);
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return 0;
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}
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int lpa_set_volume(unsigned volume)
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{
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int rc = 0;
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if (lpa_audio.prtd && lpa_audio.prtd->audio_client) {
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rc = q6asm_set_volume(lpa_audio.prtd->audio_client, volume);
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if (rc < 0) {
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pr_err("%s: Send Volume command failed"
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" rc=%d\n", __func__, rc);
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}
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}
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lpa_audio.volume = volume;
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return rc;
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}
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static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
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struct msm_audio *prtd = runtime->private_data;
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int dir = 0;
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int rc = 0;
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/*
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If routing is still enabled, we need to issue EOS to
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the DSP
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To issue EOS to dsp, we need to be run state otherwise
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EOS is not honored.
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*/
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if (msm_routing_check_backend_enabled(soc_prtd->dai_link->be_id) &&
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(!atomic_read(&prtd->stop))) {
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rc = q6asm_run(prtd->audio_client, 0, 0, 0);
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atomic_set(&prtd->pending_buffer, 0);
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prtd->cmd_ack = 0;
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q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
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pr_debug("%s\n", __func__);
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rc = wait_event_timeout(the_locks.eos_wait,
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prtd->cmd_ack, 5 * HZ);
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if (!rc)
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pr_err("EOS cmd timeout\n");
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prtd->pcm_irq_pos = 0;
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}
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dir = IN;
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atomic_set(&prtd->pending_buffer, 0);
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lpa_audio.prtd = NULL;
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q6asm_cmd(prtd->audio_client, CMD_CLOSE);
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q6asm_audio_client_buf_free_contiguous(dir,
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prtd->audio_client);
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atomic_set(&prtd->stop, 1);
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pr_debug("%s\n", __func__);
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msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
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SNDRV_PCM_STREAM_PLAYBACK);
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pr_debug("%s\n", __func__);
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q6asm_audio_client_free(prtd->audio_client);
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kfree(prtd);
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return 0;
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}
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static int msm_pcm_close(struct snd_pcm_substream *substream)
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{
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int ret = 0;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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ret = msm_pcm_playback_close(substream);
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return ret;
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}
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static int msm_pcm_prepare(struct snd_pcm_substream *substream)
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{
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int ret = 0;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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ret = msm_pcm_playback_prepare(substream);
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return ret;
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}
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static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct msm_audio *prtd = runtime->private_data;
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pr_debug("%s: pcm_irq_pos = %d\n", __func__, prtd->pcm_irq_pos);
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return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
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}
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static int msm_pcm_mmap(struct snd_pcm_substream *substream,
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struct vm_area_struct *vma)
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{
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int result = 0;
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct msm_audio *prtd = runtime->private_data;
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pr_debug("%s\n", __func__);
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prtd->mmap_flag = 1;
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if (runtime->dma_addr && runtime->dma_bytes) {
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vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
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result = remap_pfn_range(vma, vma->vm_start,
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runtime->dma_addr >> PAGE_SHIFT,
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runtime->dma_bytes,
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vma->vm_page_prot);
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} else {
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pr_err("Physical address or size of buf is NULL");
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return -EINVAL;
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}
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return result;
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}
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static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct msm_audio *prtd = runtime->private_data;
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struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
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struct audio_buffer *buf;
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int dir, ret;
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|
|
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, ×tamp);
|
|
if (rc < 0) {
|
|
pr_err("%s: fail to get session tstamp\n", __func__);
|
|
return rc;
|
|
}
|
|
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)
|
|
{
|
|
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 struct platform_driver msm_pcm_driver = {
|
|
.driver = {
|
|
.name = "msm-pcm-lpa",
|
|
.owner = THIS_MODULE,
|
|
},
|
|
.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");
|