Donate to e Foundation | Murena handsets with /e/OS | Own a part of Murena! Learn more

Commit 57321c27 authored by Anish Kumar's avatar Anish Kumar
Browse files

ASoC: msm: qdsp6v2: removing multi-pcm driver 



Remove this driver as it is not used in B-family.

Change-Id: Ic2a06caa51a2cfbaeaf4076302f37c2549aad33d
Signed-off-by: default avatarAnish Kumar <kanish@codeaurora.org>
parent 7138c71e

sound/soc/msm/qdsp6v2/Makefile

+1 −1
Original line number Diff line number Diff line 
snd-soc-qdsp6v2-objs += msm-dai-q6-v2.o msm-pcm-q6-v2.o msm-pcm-routing-v2.o \

			msm-compress-q6-v2.o msm-compr-q6-v2.o \

			msm-multi-ch-pcm-q6-v2.o msm-pcm-lpa-v2.o \

			msm-pcm-lpa-v2.o \

			msm-pcm-afe-v2.o msm-pcm-voip-v2.o \

			msm-pcm-voice-v2.o msm-dai-q6-hdmi-v2.o \

			msm-lsm-client.o msm-pcm-host-voice-v2.o \

sound/soc/msm/qdsp6v2/msm-multi-ch-pcm-q6-v2.c

deleted100644 → 0
+0 −794
Original line number Diff line number Diff line 
/* Copyright (c) 2012-2014, 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 <linux/dma-mapping.h>



#include <asm/dma.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 "msm-pcm-q6-v2.h"

#include "msm-pcm-routing-v2.h"



static struct audio_locks the_locks;



struct snd_msm {

	struct snd_card *card;

	struct snd_pcm *pcm;

};



struct snd_msm_volume {

	struct msm_audio *prtd;

	unsigned volume;

};

static struct snd_msm_volume multi_ch_pcm_audio = {NULL, 0x2000};



#define PLAYBACK_NUM_PERIODS	8

#define PLAYBACK_PERIOD_SIZE	4032

#define CAPTURE_NUM_PERIODS	16

#define CAPTURE_PERIOD_SIZE	320



static struct snd_pcm_hardware msm_pcm_hardware_capture = {

	.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,

	.rates =                SNDRV_PCM_RATE_8000_48000,

	.rate_min =             8000,

	.rate_max =             48000,

	.channels_min =         1,

	.channels_max =         2,

	.buffer_bytes_max =     CAPTURE_NUM_PERIODS * CAPTURE_PERIOD_SIZE,

	.period_bytes_min =	CAPTURE_PERIOD_SIZE,

	.period_bytes_max =     CAPTURE_PERIOD_SIZE,

	.periods_min =          CAPTURE_NUM_PERIODS,

	.periods_max =          CAPTURE_NUM_PERIODS,

	.fifo_size =            0,

};



static struct snd_pcm_hardware msm_pcm_hardware_playback = {

	.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_96000,

	.rate_min =             8000,

	.rate_max =             96000,

	.channels_min =         1,

	.channels_max =         6,

	.buffer_bytes_max =     PLAYBACK_NUM_PERIODS * PLAYBACK_PERIOD_SIZE,

	.period_bytes_min =	PLAYBACK_PERIOD_SIZE,

	.period_bytes_max =     PLAYBACK_PERIOD_SIZE,

	.periods_min =          PLAYBACK_NUM_PERIODS,

	.periods_max =          PLAYBACK_NUM_PERIODS,

	.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

};



static uint32_t in_frame_info[CAPTURE_NUM_PERIODS][2];



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;

	uint32_t *ptrmem = (uint32_t *)payload;

	int i = 0;

	uint32_t idx = 0;

	uint32_t size = 0;



	pr_debug("%s\n", __func__);

	switch (opcode) {

	case ASM_DATA_EVENT_WRITE_DONE_V2: {

		pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");

		pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);

		prtd->pcm_irq_pos += prtd->pcm_count;

		if (atomic_read(&prtd->start))

			snd_pcm_period_elapsed(substream);

		atomic_inc(&prtd->out_count);

		wake_up(&the_locks.write_wait);

		if (!atomic_read(&prtd->start))

			break;

		if (!prtd->mmap_flag)

			break;

		if (q6asm_is_cpu_buf_avail_nolock(IN,

				prtd->audio_client,

				&size, &idx)) {

			pr_debug("%s:writing %d bytes of buffer to dsp 2\n",

					__func__, prtd->pcm_count);

			q6asm_write_nolock(prtd->audio_client,

				prtd->pcm_count, 0, 0, NO_TIMESTAMP);

		}

		break;

	}

	case ASM_DATA_EVENT_RENDERED_EOS:

		pr_debug("ASM_DATA_CMDRSP_EOS\n");

		clear_bit(CMD_EOS, &prtd->cmd_pending);

		wake_up(&the_locks.eos_wait);

		break;

	case ASM_DATA_EVENT_READ_DONE_V2: {

		pr_debug("ASM_DATA_EVENT_READ_DONE\n");

		pr_debug("token = 0x%08x\n", token);

		for (i = 0; i < 8; i++, ++ptrmem)

			pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);

		in_frame_info[token][0] = payload[2];

		in_frame_info[token][1] = payload[3];

		prtd->pcm_irq_pos += in_frame_info[token][0];

		pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);

		if (atomic_read(&prtd->start))

			snd_pcm_period_elapsed(substream);

		if (atomic_read(&prtd->in_count) <= prtd->periods)

			atomic_inc(&prtd->in_count);

		wake_up(&the_locks.read_wait);

		if (prtd->mmap_flag

			&& q6asm_is_cpu_buf_avail_nolock(OUT,

				prtd->audio_client,

				&size, &idx))

			q6asm_read_nolock(prtd->audio_client);

		break;

	}

	case APR_BASIC_RSP_RESULT: {

		switch (payload[0]) {

		case ASM_SESSION_CMD_RUN_V2:

			if (substream->stream

				!= SNDRV_PCM_STREAM_PLAYBACK) {

				atomic_set(&prtd->start, 1);

				break;

			}

			if (prtd->mmap_flag) {

				pr_debug("%s:writing %d bytes of buffer to dsp\n",

					__func__,

					prtd->pcm_count);

				q6asm_write_nolock(prtd->audio_client,

					prtd->pcm_count,

					0, 0, NO_TIMESTAMP);

			} else {

				while (atomic_read(&prtd->out_needed)) {

					pr_debug("%s:writing %d bytes of buffer to dsp\n",

						__func__,

						prtd->pcm_count);

					q6asm_write_nolock(prtd->audio_client,

						prtd->pcm_count,

						0, 0, NO_TIMESTAMP);

					atomic_dec(&prtd->out_needed);

					wake_up(&the_locks.write_wait);

				};

			}

			atomic_set(&prtd->start, 1);

			break;

		default:

			break;

		}

	}

	break;

	default:

		pr_debug("Not Supported Event opcode[0x%x]\n", opcode);

		break;

	}

}



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;

	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_info("%s: CMD Format block failed\n", __func__);



	atomic_set(&prtd->out_count, runtime->periods);



	prtd->enabled = 1;

	prtd->cmd_pending = 0;

	prtd->cmd_interrupt = 0;



	return 0;

}



static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)

{

	struct snd_pcm_runtime *runtime = substream->runtime;

	struct msm_audio *prtd = runtime->private_data;

	int ret = 0;

	int i = 0;

	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;



	if (prtd->enabled)

		return 0;



	pr_debug("Samp_rate = %d\n", prtd->samp_rate);

	pr_debug("Channel = %d\n", prtd->channel_mode);

	ret = q6asm_enc_cfg_blk_pcm(prtd->audio_client, prtd->samp_rate,

					prtd->channel_mode);

	if (ret < 0)

		pr_debug("%s: cmd cfg pcm was block failed", __func__);



	for (i = 0; i < runtime->periods; i++)

		q6asm_read(prtd->audio_client);

	prtd->periods = runtime->periods;



	prtd->enabled = 1;



	return ret;

}



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;



	switch (cmd) {

	case SNDRV_PCM_TRIGGER_START:

	case SNDRV_PCM_TRIGGER_RESUME:

	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:

		pr_debug("%s: Trigger start\n", __func__);

		q6asm_run_nowait(prtd->audio_client, 0, 0, 0);

		break;

	case SNDRV_PCM_TRIGGER_STOP:

		pr_debug("SNDRV_PCM_TRIGGER_STOP\n");

		atomic_set(&prtd->start, 0);

		if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)

			break;

		/* pending CMD_EOS isn't expected */

		WARN_ON_ONCE(test_bit(CMD_EOS, &prtd->cmd_pending));

		set_bit(CMD_EOS, &prtd->cmd_pending);

		q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);

		if (ret)

			clear_bit(CMD_EOS, &prtd->cmd_pending);

		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 snd_soc_pcm_runtime *soc_prtd = substream->private_data;

	struct msm_audio *prtd;

	int ret = 0;

	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,

	};



	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;

	}

	prtd->substream = substream;

	prtd->audio_client = q6asm_audio_client_alloc(

				(app_cb)event_handler, prtd);

	if (!prtd->audio_client) {

		pr_err("%s: Could not allocate memory\n", __func__);

		kfree(prtd);

		return -ENOMEM;

	}

	prtd->audio_client->perf_mode = false;

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {

		runtime->hw = msm_pcm_hardware_playback;

		ret = q6asm_open_write(prtd->audio_client,

				FORMAT_MULTI_CHANNEL_LINEAR_PCM);

		if (ret < 0) {

			pr_err("%s: pcm out open failed\n", __func__);

			q6asm_audio_client_free(prtd->audio_client);

			kfree(prtd);

			return -ENOMEM;

		}

	}

	/* Capture path */

	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {

		runtime->hw = msm_pcm_hardware_capture;

		ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM);

		if (ret < 0) {

			pr_err("%s: pcm in open failed\n", __func__);

			q6asm_audio_client_free(prtd->audio_client);

			kfree(prtd);

			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);



	ret = snd_pcm_hw_constraint_list(runtime, 0,

				SNDRV_PCM_HW_PARAM_RATE,

				&constraints_sample_rates);

	if (ret < 0)

		pr_err("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_err("snd_pcm_hw_constraint_integer failed\n");



	prtd->dsp_cnt = 0;

	runtime->private_data = prtd;

	pr_debug("substream->pcm->device = %d\n", substream->pcm->device);

	pr_debug("soc_prtd->dai_link->be_id = %d\n", soc_prtd->dai_link->be_id);

	multi_ch_pcm_audio.prtd = prtd;

	ret = multi_ch_pcm_set_volume(multi_ch_pcm_audio.volume);

	if (ret < 0)

		pr_err("%s : Set Volume failed : %d", __func__, ret);



	ret = q6asm_set_softpause(multi_ch_pcm_audio.prtd->audio_client,

								 &softpause);

	if (ret < 0)

		pr_err("%s: Send SoftPause Param failed ret=%d\n",

			__func__, ret);

	ret = q6asm_set_softvolume(multi_ch_pcm_audio.prtd->audio_client,

								 &softvol);

	if (ret < 0)

		pr_err("%s: Send SoftVolume Param failed ret=%d\n",

			__func__, ret);



	return 0;

}



int multi_ch_pcm_set_volume(unsigned volume)

{

	int rc = 0;

	pr_err("multi_ch_pcm_set_volume\n");



	if (multi_ch_pcm_audio.prtd && multi_ch_pcm_audio.prtd->audio_client) {

		pr_err("%s q6asm_set_volume\n", __func__);

		rc = q6asm_set_volume(multi_ch_pcm_audio.prtd->audio_client,

								volume);

		if (rc < 0) {

			pr_err("%s: Send Volume command failed rc=%d\n",

							__func__, rc);

		}

	}

	multi_ch_pcm_audio.volume = volume;

	return rc;

}





static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,

	snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)

{

	int ret = 0;

	int fbytes = 0;

	int xfer = 0;

	char *bufptr = NULL;

	void *data = NULL;

	uint32_t idx = 0;

	uint32_t size = 0;



	struct snd_pcm_runtime *runtime = substream->runtime;

	struct msm_audio *prtd = runtime->private_data;



	fbytes = frames_to_bytes(runtime, frames);

	pr_debug("%s: prtd->out_count = %d\n",

				__func__, atomic_read(&prtd->out_count));

	ret = wait_event_timeout(the_locks.write_wait,

			(atomic_read(&prtd->out_count)), 5 * HZ);

	if (!ret) {

		pr_err("%s: wait_event_timeout failed\n", __func__);

		goto fail;

	}



	if (!atomic_read(&prtd->out_count)) {

		pr_err("%s: pcm stopped out_count 0\n", __func__);

		return 0;

	}



	data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size, &idx);

	bufptr = data;

	if (bufptr) {

		pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",

					__func__, fbytes, xfer, size);

		xfer = fbytes;

		if (copy_from_user(bufptr, buf, xfer)) {

			ret = -EFAULT;

			goto fail;

		}

		buf += xfer;

		fbytes -= xfer;

		pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes, xfer);

		if (atomic_read(&prtd->start)) {

			pr_debug("%s:writing %d bytes of buffer to dsp\n",

					__func__, xfer);

			ret = q6asm_write(prtd->audio_client, xfer,

						0, 0, NO_TIMESTAMP);

			if (ret < 0) {

				ret = -EFAULT;

				goto fail;

			}

		} else

			atomic_inc(&prtd->out_needed);

		atomic_dec(&prtd->out_count);

	}

fail:

	return  ret;

}



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 ret = 0;



	pr_debug("%s: cmd_pending 0x%lx\n", __func__, prtd->cmd_pending);



	dir = IN;

	ret = wait_event_timeout(the_locks.eos_wait,

				 !test_bit(CMD_EOS, &prtd->cmd_pending),

				 5 * HZ);

	if (!ret)

		pr_err("%s: CMD_EOS failed, cmd_pending 0x%lx\n",

		       __func__, prtd->cmd_pending);

	q6asm_cmd(prtd->audio_client, CMD_CLOSE);

	q6asm_audio_client_buf_free_contiguous(dir,

				prtd->audio_client);



	msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,

	SNDRV_PCM_STREAM_PLAYBACK);

	multi_ch_pcm_audio.prtd = NULL;

	q6asm_audio_client_free(prtd->audio_client);

	kfree(prtd);

	return 0;

}



static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,

		 int channel, snd_pcm_uframes_t hwoff, void __user *buf,

						 snd_pcm_uframes_t frames)

{

	int ret = 0;

	int fbytes = 0;

	int xfer;

	char *bufptr;

	void *data = NULL;

	static uint32_t idx;

	static uint32_t size;

	uint32_t offset = 0;

	struct snd_pcm_runtime *runtime = substream->runtime;

	struct msm_audio *prtd = substream->runtime->private_data;





	pr_debug("%s\n", __func__);

	fbytes = frames_to_bytes(runtime, frames);



	pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);

	pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);

	pr_debug("avail_min %d\n", (int)runtime->control->avail_min);



	ret = wait_event_timeout(the_locks.read_wait,

			(atomic_read(&prtd->in_count)), 5 * HZ);

	if (!ret) {

		pr_debug("%s: wait_event_timeout failed\n", __func__);

		goto fail;

	}

	if (!atomic_read(&prtd->in_count)) {

		pr_debug("%s: pcm stopped in_count 0\n", __func__);

		return 0;

	}

	pr_debug("Checking if valid buffer is available...%p\n", data);

	data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);

	bufptr = data;

	pr_debug("Size = %d\n", size);

	pr_debug("fbytes = %d\n", fbytes);

	pr_debug("idx = %d\n", idx);

	if (bufptr) {

		xfer = fbytes;

		if (xfer > size)

			xfer = size;

		offset = in_frame_info[idx][1];

		pr_debug("Offset value = %d\n", offset);

		if (copy_to_user(buf, bufptr+offset, xfer)) {

			pr_err("Failed to copy buf to user\n");

			ret = -EFAULT;

			goto fail;

		}

		fbytes -= xfer;

		size -= xfer;

		in_frame_info[idx][1] += xfer;

		pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",

					__func__, fbytes, size, xfer);

		pr_debug(" Sending next buffer to dsp\n");

		memset(&in_frame_info[idx], 0,

			sizeof(uint32_t) * 2);

		atomic_dec(&prtd->in_count);

		ret = q6asm_read(prtd->audio_client);

		if (ret < 0) {

			pr_err("q6asm read failed\n");

			ret = -EFAULT;

			goto fail;

		}

	} else

		pr_err("No valid buffer\n");



	pr_debug("Returning from capture_copy... %d\n", ret);

fail:

	return ret;

}



static int msm_pcm_capture_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 = OUT;



	pr_debug("%s\n", __func__);

	q6asm_cmd(prtd->audio_client, CMD_CLOSE);

	q6asm_audio_client_buf_free_contiguous(dir,

				prtd->audio_client);

	msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,

	SNDRV_PCM_STREAM_CAPTURE);

	q6asm_audio_client_free(prtd->audio_client);

	kfree(prtd);



	return 0;

}



static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,

	 snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)

{

	int ret = 0;



	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

		ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);

	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)

		ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);

	return ret;

}



static int msm_pcm_close(struct snd_pcm_substream *substream)

{

	int ret = 0;



	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

		ret = msm_pcm_playback_close(substream);

	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)

		ret = msm_pcm_capture_close(substream);

	return ret;

}

static int msm_pcm_prepare(struct snd_pcm_substream *substream)

{

	int ret = 0;



	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

		ret = msm_pcm_playback_prepare(substream);

	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)

		ret = msm_pcm_capture_prepare(substream);

	return ret;

}



static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)

{



	struct snd_pcm_runtime *runtime = substream->runtime;

	struct msm_audio *prtd = runtime->private_data;



	if (prtd->pcm_irq_pos >= prtd->pcm_size)

		prtd->pcm_irq_pos = 0;



	pr_debug("pcm_irq_pos = %d\n", 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)

{

	int result = 0;

	struct snd_pcm_runtime *runtime = substream->runtime;

	struct msm_audio *prtd = runtime->private_data;



	pr_debug("%s\n", __func__);

	prtd->mmap_flag = 1;



	if (runtime->dma_addr && runtime->dma_bytes) {

		vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);

		result = remap_pfn_range(vma, vma->vm_start,

				runtime->dma_addr >> PAGE_SHIFT,

				runtime->dma_bytes,

				vma->vm_page_prot);

	} else {

		pr_err("Physical address or size of buf is NULL");

		return -EINVAL;

	}



	return result;

}



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_dma_buffer *dma_buf = &substream->dma_buffer;

	struct audio_buffer *buf;

	int dir, ret;



	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

		dir = IN;

	else

		dir = OUT;



	ret = q6asm_audio_client_buf_alloc_contiguous(dir,

			prtd->audio_client,

			runtime->hw.period_bytes_min,

			runtime->hw.periods_max);

	if (ret < 0) {

		pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);

		return -ENOMEM;

	}

	buf = prtd->audio_client->port[dir].buf;



	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 = runtime->hw.buffer_bytes_max;

	if (!dma_buf->area)

		return -ENOMEM;



	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);

	return 0;

}



static struct snd_pcm_ops msm_pcm_ops = {

	.open           = msm_pcm_open,

	.copy		= msm_pcm_copy,

	.hw_params	= msm_pcm_hw_params,

	.close          = msm_pcm_close,

	.ioctl          = snd_pcm_lib_ioctl,

	.prepare        = msm_pcm_prepare,

	.trigger        = msm_pcm_trigger,

	.pointer        = msm_pcm_pointer,

	.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 int msm_pcm_probe(struct platform_device *pdev)

{

	pr_info("%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-multi-ch-pcm-dsp",

		.owner = THIS_MODULE,

	},

	.probe = msm_pcm_probe,

	.remove = msm_pcm_remove,

};



static int __init msm_soc_platform_init(void)

{

	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("Multi channel PCM module platform driver");

MODULE_LICENSE("GPL v2");

sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.h

+0 −2
Original line number Diff line number Diff line
@@ -219,8 +219,6 @@ void msm_pcm_routing_dereg_phy_stream(int fedai_id, int stream_type); 


int msm_routing_check_backend_enabled(int fedai_id);



int multi_ch_pcm_set_volume(unsigned volume);



uint32_t get_adm_rx_topology(void);



uint32_t get_adm_tx_topology(void);