1398 lines
43 KiB
C
1398 lines
43 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/q6asm.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/timer.h>
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#include <mach/qdsp6v2/q6core.h>
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#include <sound/pcm.h>
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#include "msm-compr-q6.h"
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#include "msm-pcm-routing.h"
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#define COMPRE_CAPTURE_NUM_PERIODS 16
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/* Allocate the worst case frame size for compressed audio */
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#define COMPRE_CAPTURE_HEADER_SIZE (sizeof(struct snd_compr_audio_info))
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#define COMPRE_CAPTURE_MAX_FRAME_SIZE (6144)
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#define COMPRE_CAPTURE_PERIOD_SIZE ((COMPRE_CAPTURE_MAX_FRAME_SIZE + \
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COMPRE_CAPTURE_HEADER_SIZE) * \
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MAX_NUM_FRAMES_PER_BUFFER)
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#define COMPRE_OUTPUT_METADATA_SIZE (sizeof(struct output_meta_data_st))
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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 compressed_audio = {NULL, 0x2000} ;
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static struct audio_locks the_locks;
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static struct snd_pcm_hardware msm_compr_hardware_capture = {
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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,
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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 = 8,
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.buffer_bytes_max =
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COMPRE_CAPTURE_PERIOD_SIZE * COMPRE_CAPTURE_NUM_PERIODS ,
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.period_bytes_min = COMPRE_CAPTURE_PERIOD_SIZE,
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.period_bytes_max = COMPRE_CAPTURE_PERIOD_SIZE,
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.periods_min = COMPRE_CAPTURE_NUM_PERIODS,
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.periods_max = COMPRE_CAPTURE_NUM_PERIODS,
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.fifo_size = 0,
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};
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static struct snd_pcm_hardware msm_compr_hardware_playback = {
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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 = 8,
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.buffer_bytes_max = 1024 * 1024,
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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 compr_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 compr_audio *compr = priv;
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struct msm_audio *prtd = &compr->prtd;
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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_aio_read_param read_param;
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struct audio_buffer *buf = NULL;
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struct output_meta_data_st output_meta_data;
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uint32_t *ptrmem = (uint32_t *)payload;
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int i = 0;
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int time_stamp_flag = 0;
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int buffer_length = 0;
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pr_debug("%s opcode =%08x\n", __func__, opcode);
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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 (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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if (runtime->status->hw_ptr >= runtime->control->appl_ptr) {
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runtime->render_flag |= SNDRV_RENDER_STOPPED;
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atomic_set(&prtd->pending_buffer, 1);
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pr_debug("%s:compr driver underrun hw_ptr = %ld appl_ptr = %ld\n",
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__func__, runtime->status->hw_ptr,
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runtime->control->appl_ptr);
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break;
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}
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buf = prtd->audio_client->port[IN].buf;
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pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
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__func__, prtd->pcm_count, prtd->out_head);
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pr_debug("%s:writing buffer[%d] from 0x%08x\n",
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__func__, prtd->out_head,
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((unsigned int)buf[0].phys
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+ (prtd->out_head * prtd->pcm_count)));
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if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
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time_stamp_flag = SET_TIMESTAMP;
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else
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time_stamp_flag = NO_TIMESTAMP;
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memcpy(&output_meta_data, (char *)(buf->data +
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prtd->out_head * prtd->pcm_count),
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COMPRE_OUTPUT_METADATA_SIZE);
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buffer_length = output_meta_data.frame_size;
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pr_debug("meta_data_length: %d, frame_length: %d\n",
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output_meta_data.meta_data_length,
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output_meta_data.frame_size);
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pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
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output_meta_data.timestamp_msw,
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output_meta_data.timestamp_lsw);
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if (buffer_length == 0) {
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pr_debug("Recieved a zero length buffer-break out");
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break;
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}
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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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+ output_meta_data.meta_data_length;
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param.len = buffer_length;
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param.msw_ts = output_meta_data.timestamp_msw;
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param.lsw_ts = output_meta_data.timestamp_lsw;
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param.flags = time_stamp_flag;
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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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+ output_meta_data.meta_data_length);
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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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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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if (atomic_read(&prtd->eos)) {
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pr_debug("ASM_DATA_CMDRSP_EOS wake up\n");
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prtd->cmd_ack = 1;
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wake_up(&the_locks.eos_wait);
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atomic_set(&prtd->eos, 0);
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}
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atomic_set(&prtd->pending_buffer, 1);
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break;
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case ASM_DATA_EVENT_READ_DONE: {
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pr_debug("ASM_DATA_EVENT_READ_DONE\n");
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pr_debug("buf = %p, data = 0x%X, *data = %p,\n"
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"prtd->pcm_irq_pos = %d\n",
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prtd->audio_client->port[OUT].buf,
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*(uint32_t *)prtd->audio_client->port[OUT].buf->data,
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prtd->audio_client->port[OUT].buf->data,
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prtd->pcm_irq_pos);
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memcpy(prtd->audio_client->port[OUT].buf->data +
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prtd->pcm_irq_pos, (ptrmem + 2),
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COMPRE_CAPTURE_HEADER_SIZE);
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pr_debug("buf = %p, updated data = 0x%X, *data = %p\n",
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prtd->audio_client->port[OUT].buf,
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*(uint32_t *)(prtd->audio_client->port[OUT].buf->data +
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prtd->pcm_irq_pos),
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prtd->audio_client->port[OUT].buf->data);
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if (!atomic_read(&prtd->start))
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break;
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pr_debug("frame size=%d, buffer = 0x%X\n", ptrmem[2],
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ptrmem[1]);
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if (ptrmem[2] > COMPRE_CAPTURE_MAX_FRAME_SIZE) {
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pr_err("Frame length exceeded the max length");
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break;
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}
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buf = prtd->audio_client->port[OUT].buf;
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pr_debug("pcm_irq_pos=%d, buf[0].phys = 0x%X\n",
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prtd->pcm_irq_pos, (uint32_t)buf[0].phys);
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read_param.len = prtd->pcm_count - COMPRE_CAPTURE_HEADER_SIZE;
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read_param.paddr = (unsigned long)(buf[0].phys) +
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prtd->pcm_irq_pos + COMPRE_CAPTURE_HEADER_SIZE;
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prtd->pcm_irq_pos += prtd->pcm_count;
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if (atomic_read(&prtd->start))
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snd_pcm_period_elapsed(substream);
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q6asm_async_read(prtd->audio_client, &read_param);
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break;
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}
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case ASM_DATA_EVENT_READ_COMPRESSED_DONE: {
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pr_debug("ASM_DATA_EVENT_READ_COMPRESSED_DONE\n");
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pr_debug("buf = %p, data = 0x%X, *data = %p,\n"
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"prtd->pcm_irq_pos = %d\n",
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prtd->audio_client->port[OUT].buf,
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*(uint32_t *)prtd->audio_client->port[OUT].buf->data,
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prtd->audio_client->port[OUT].buf->data,
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prtd->pcm_irq_pos);
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if (!atomic_read(&prtd->start))
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break;
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buf = prtd->audio_client->port[OUT].buf;
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pr_debug("pcm_irq_pos=%d, buf[0].phys = 0x%X\n",
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prtd->pcm_irq_pos, (uint32_t)buf[0].phys);
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read_param.len = prtd->pcm_count;
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read_param.paddr = (unsigned long)(buf[0].phys) +
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prtd->pcm_irq_pos;
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prtd->pcm_irq_pos += prtd->pcm_count;
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if (atomic_read(&prtd->start))
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snd_pcm_period_elapsed(substream);
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q6asm_async_read_compressed(prtd->audio_client, &read_param);
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break;
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}
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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 (substream->stream
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!= SNDRV_PCM_STREAM_PLAYBACK) {
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atomic_set(&prtd->start, 1);
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break;
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}
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if (!atomic_read(&prtd->pending_buffer))
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break;
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pr_debug("%s:writing %d bytes"
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" of buffer[%d] to dsp\n",
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__func__, prtd->pcm_count, prtd->out_head);
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buf = prtd->audio_client->port[IN].buf;
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pr_debug("%s:writing buffer[%d] from 0x%08x\n",
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__func__, prtd->out_head,
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((unsigned int)buf[0].phys
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+ (prtd->out_head * prtd->pcm_count)));
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if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
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time_stamp_flag = SET_TIMESTAMP;
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else
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time_stamp_flag = NO_TIMESTAMP;
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memcpy(&output_meta_data, (char *)(buf->data +
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prtd->out_head * prtd->pcm_count),
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COMPRE_OUTPUT_METADATA_SIZE);
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buffer_length = output_meta_data.frame_size;
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pr_debug("meta_data_length: %d, frame_length: %d\n",
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output_meta_data.meta_data_length,
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output_meta_data.frame_size);
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pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
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output_meta_data.timestamp_msw,
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output_meta_data.timestamp_lsw);
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param.paddr = (unsigned long)buf[prtd->out_head].phys
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+ output_meta_data.meta_data_length;
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param.len = buffer_length;
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param.msw_ts = output_meta_data.timestamp_msw;
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param.lsw_ts = output_meta_data.timestamp_lsw;
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param.flags = time_stamp_flag;
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param.uid = (unsigned long)buf[prtd->out_head].phys
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+ output_meta_data.meta_data_length;
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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.flush_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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}
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static int msm_compr_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 compr_audio *compr = runtime->private_data;
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struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
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struct msm_audio *prtd = &compr->prtd;
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struct asm_aac_cfg aac_cfg;
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struct asm_wma_cfg wma_cfg;
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struct asm_wmapro_cfg wma_pro_cfg;
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struct asm_amrwbplus_cfg amrwb_cfg;
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int ret;
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pr_debug("compressed stream prepare\n");
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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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atomic_set(&prtd->out_count, runtime->periods);
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if (prtd->enabled)
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return 0;
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switch (compr->info.codec_param.codec.id) {
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case SND_AUDIOCODEC_MP3:
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pr_debug("%s: SND_AUDIOCODEC_MP3\n", __func__);
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ret = q6asm_media_format_block(prtd->audio_client,
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compr->codec);
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if (ret < 0)
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pr_info("%s: CMD Format block failed\n", __func__);
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break;
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case SND_AUDIOCODEC_AAC:
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pr_debug("%s: SND_AUDIOCODEC_AAC\n", __func__);
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memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
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aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
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aac_cfg.format = 0x03;
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aac_cfg.ch_cfg = runtime->channels;
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aac_cfg.sample_rate = runtime->rate;
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ret = q6asm_media_format_block_aac(prtd->audio_client,
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&aac_cfg);
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if (ret < 0)
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pr_err("%s: CMD Format block failed\n", __func__);
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break;
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case SND_AUDIOCODEC_AC3_PASS_THROUGH:
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case SND_AUDIOCODEC_DTS_PASS_THROUGH:
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case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
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pr_debug("compressd playback, no need to send decoder params");
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pr_debug("decoder id: %d\n",
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compr->info.codec_param.codec.id);
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msm_pcm_routing_reg_psthr_stream(
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soc_prtd->dai_link->be_id,
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prtd->session_id, substream->stream,
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1);
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break;
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case SND_AUDIOCODEC_WMA:
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pr_debug("SND_AUDIOCODEC_WMA\n");
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memset(&wma_cfg, 0x0, sizeof(struct asm_wma_cfg));
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wma_cfg.format_tag = compr->info.codec_param.codec.format;
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wma_cfg.ch_cfg = compr->info.codec_param.codec.ch_in;
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wma_cfg.sample_rate = compr->info.codec_param.codec.sample_rate;
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wma_cfg.avg_bytes_per_sec =
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compr->info.codec_param.codec.bit_rate/8;
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wma_cfg.block_align = compr->info.codec_param.codec.align;
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wma_cfg.valid_bits_per_sample =
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compr->info.codec_param.codec.options.wma.bits_per_sample;
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wma_cfg.ch_mask =
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compr->info.codec_param.codec.options.wma.channelmask;
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wma_cfg.encode_opt =
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compr->info.codec_param.codec.options.wma.encodeopt;
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ret = q6asm_media_format_block_wma(prtd->audio_client,
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&wma_cfg);
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if (ret < 0)
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pr_err("%s: CMD Format block failed\n", __func__);
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break;
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|
case SND_AUDIOCODEC_WMA_PRO:
|
|
pr_debug("SND_AUDIOCODEC_WMA_PRO\n");
|
|
memset(&wma_pro_cfg, 0x0, sizeof(struct asm_wmapro_cfg));
|
|
wma_pro_cfg.format_tag = compr->info.codec_param.codec.format;
|
|
wma_pro_cfg.ch_cfg = compr->info.codec_param.codec.ch_in;
|
|
wma_pro_cfg.sample_rate =
|
|
compr->info.codec_param.codec.sample_rate;
|
|
wma_pro_cfg.avg_bytes_per_sec =
|
|
compr->info.codec_param.codec.bit_rate/8;
|
|
wma_pro_cfg.block_align = compr->info.codec_param.codec.align;
|
|
wma_pro_cfg.valid_bits_per_sample =
|
|
compr->info.codec_param.codec\
|
|
.options.wma.bits_per_sample;
|
|
wma_pro_cfg.ch_mask =
|
|
compr->info.codec_param.codec.options.wma.channelmask;
|
|
wma_pro_cfg.encode_opt =
|
|
compr->info.codec_param.codec.options.wma.encodeopt;
|
|
wma_pro_cfg.adv_encode_opt =
|
|
compr->info.codec_param.codec.options.wma.encodeopt1;
|
|
wma_pro_cfg.adv_encode_opt2 =
|
|
compr->info.codec_param.codec.options.wma.encodeopt2;
|
|
ret = q6asm_media_format_block_wmapro(prtd->audio_client,
|
|
&wma_pro_cfg);
|
|
if (ret < 0)
|
|
pr_err("%s: CMD Format block failed\n", __func__);
|
|
break;
|
|
case SND_AUDIOCODEC_DTS:
|
|
case SND_AUDIOCODEC_DTS_LBR:
|
|
pr_debug("SND_AUDIOCODEC_DTS\n");
|
|
ret = q6asm_media_format_block(prtd->audio_client,
|
|
compr->codec);
|
|
if (ret < 0) {
|
|
pr_err("%s: CMD Format block failed\n", __func__);
|
|
return ret;
|
|
}
|
|
break;
|
|
case SND_AUDIOCODEC_AMRWB:
|
|
pr_debug("SND_AUDIOCODEC_AMRWB\n");
|
|
ret = q6asm_media_format_block(prtd->audio_client,
|
|
compr->codec);
|
|
if (ret < 0) {
|
|
pr_err("%s: CMD Format block failed\n", __func__);
|
|
return ret;
|
|
}
|
|
break;
|
|
case SND_AUDIOCODEC_AMRWBPLUS:
|
|
pr_debug("SND_AUDIOCODEC_AMRWBPLUS\n");
|
|
memset(&amrwb_cfg, 0x0, sizeof(struct asm_amrwbplus_cfg));
|
|
amrwb_cfg.size_bytes = sizeof(struct asm_amrwbplus_cfg);
|
|
pr_debug("calling q6asm_media_format_block_amrwbplus");
|
|
ret = q6asm_media_format_block_amrwbplus(prtd->audio_client,
|
|
&amrwb_cfg);
|
|
if (ret < 0) {
|
|
pr_err("%s: CMD Format block failed\n", __func__);
|
|
return ret;
|
|
}
|
|
break;
|
|
case SND_AUDIOCODEC_MP2:
|
|
pr_debug("%s: SND_AUDIOCODEC_MP2\n", __func__);
|
|
break;
|
|
default:
|
|
return -EINVAL;
|
|
}
|
|
if (compr->info.codec_param.codec.transcode_dts) {
|
|
msm_pcm_routing_reg_pseudo_stream(
|
|
MSM_FRONTEND_DAI_PSEUDO,
|
|
prtd->enc_audio_client->perf_mode,
|
|
prtd->enc_audio_client->session,
|
|
SNDRV_PCM_STREAM_CAPTURE,
|
|
48000, runtime->channels > 6 ?
|
|
6 : runtime->channels);
|
|
pr_debug("%s: cmd: DTS ENCDEC CFG BLK\n", __func__);
|
|
ret = q6asm_enc_cfg_blk_dts(prtd->enc_audio_client,
|
|
DTS_ENC_SAMPLE_RATE48k,
|
|
runtime->channels > 6 ?
|
|
6 : runtime->channels);
|
|
if (ret < 0)
|
|
pr_err("%s: CMD: DTS ENCDEC CFG BLK failed\n",
|
|
__func__);
|
|
}
|
|
prtd->enabled = 1;
|
|
prtd->cmd_ack = 0;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int msm_compr_capture_prepare(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct compr_audio *compr = runtime->private_data;
|
|
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
struct audio_buffer *buf = prtd->audio_client->port[OUT].buf;
|
|
struct snd_codec *codec = &compr->info.codec_param.codec;
|
|
struct audio_aio_read_param read_param;
|
|
int ret = 0;
|
|
int i;
|
|
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 ret;
|
|
read_param.len = prtd->pcm_count;
|
|
|
|
switch (codec->id) {
|
|
case SND_AUDIOCODEC_AMRWB:
|
|
pr_debug("SND_AUDIOCODEC_AMRWB\n");
|
|
ret = q6asm_enc_cfg_blk_amrwb(prtd->audio_client,
|
|
MAX_NUM_FRAMES_PER_BUFFER,
|
|
codec->options.generic.reserved[0] /*bitrate 0-8*/,
|
|
codec->options.generic.reserved[1] /*dtx mode 0/1*/);
|
|
if (ret < 0)
|
|
pr_err("%s: CMD Format block" \
|
|
"failed: %d\n", __func__, ret);
|
|
break;
|
|
case SND_AUDIOCODEC_PCM:
|
|
pr_debug("SND_AUDIOCODEC_PCM\n");
|
|
ret = q6asm_enc_cfg_blk_multi_ch_pcm(prtd->audio_client,
|
|
prtd->samp_rate, prtd->channel_mode);
|
|
if (ret < 0)
|
|
pr_info("%s: CMD Format block failed\n", __func__);
|
|
break;
|
|
default:
|
|
pr_debug("No config for codec %d\n", codec->id);
|
|
}
|
|
pr_debug("%s: Samp_rate = %d, Channel = %d, pcm_size = %d,\n"
|
|
"pcm_count = %d, periods = %d\n",
|
|
__func__, prtd->samp_rate, prtd->channel_mode,
|
|
prtd->pcm_size, prtd->pcm_count, runtime->periods);
|
|
|
|
for (i = 0; i < runtime->periods; i++) {
|
|
read_param.uid = i;
|
|
switch (codec->id) {
|
|
case SND_AUDIOCODEC_AMRWB:
|
|
case SND_AUDIOCODEC_PCM:
|
|
read_param.len = prtd->pcm_count
|
|
- COMPRE_CAPTURE_HEADER_SIZE;
|
|
read_param.paddr = (unsigned long)(buf[i].phys)
|
|
+ COMPRE_CAPTURE_HEADER_SIZE;
|
|
pr_debug("Push buffer [%d] to DSP, "\
|
|
"paddr: %p, vaddr: %p\n",
|
|
i, (void *) read_param.paddr,
|
|
buf[i].data);
|
|
q6asm_async_read(prtd->audio_client, &read_param);
|
|
break;
|
|
case SND_AUDIOCODEC_PASS_THROUGH:
|
|
read_param.paddr = (unsigned long)(buf[i].phys);
|
|
q6asm_async_read_compressed(prtd->audio_client,
|
|
&read_param);
|
|
break;
|
|
default:
|
|
pr_err("Invalid format");
|
|
ret = -EINVAL;
|
|
break;
|
|
}
|
|
}
|
|
prtd->periods = runtime->periods;
|
|
|
|
prtd->enabled = 1;
|
|
|
|
if (compr->info.codec_param.codec.id ==
|
|
SND_AUDIOCODEC_PASS_THROUGH)
|
|
msm_pcm_routing_reg_psthr_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->session_id, substream->stream,
|
|
1);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int msm_compr_restart(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
struct audio_aio_write_param param;
|
|
struct audio_buffer *buf = NULL;
|
|
struct output_meta_data_st output_meta_data;
|
|
int time_stamp_flag = 0;
|
|
int buffer_length = 0;
|
|
|
|
pr_err("msm_compr_restart\n");
|
|
if (runtime->render_flag & SNDRV_RENDER_STOPPED) {
|
|
buf = prtd->audio_client->port[IN].buf;
|
|
pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
|
|
__func__, prtd->pcm_count, prtd->out_head);
|
|
pr_debug("%s:writing buffer[%d] from 0x%08x\n",
|
|
__func__, prtd->out_head,
|
|
((unsigned int)buf[0].phys
|
|
+ (prtd->out_head * prtd->pcm_count)));
|
|
|
|
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
|
|
time_stamp_flag = SET_TIMESTAMP;
|
|
else
|
|
time_stamp_flag = NO_TIMESTAMP;
|
|
memcpy(&output_meta_data, (char *)(buf->data +
|
|
prtd->out_head * prtd->pcm_count),
|
|
COMPRE_OUTPUT_METADATA_SIZE);
|
|
|
|
buffer_length = output_meta_data.frame_size;
|
|
pr_debug("meta_data_length: %d, frame_length: %d\n",
|
|
output_meta_data.meta_data_length,
|
|
output_meta_data.frame_size);
|
|
pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
|
|
output_meta_data.timestamp_msw,
|
|
output_meta_data.timestamp_lsw);
|
|
if (buffer_length == 0) {
|
|
pr_debug("Recieved a zero length buffer-break out");
|
|
return -EINVAL;
|
|
}
|
|
param.paddr = (unsigned long)buf[0].phys
|
|
+ (prtd->out_head * prtd->pcm_count)
|
|
+ output_meta_data.meta_data_length;
|
|
param.len = buffer_length;
|
|
param.msw_ts = output_meta_data.timestamp_msw;
|
|
param.lsw_ts = output_meta_data.timestamp_lsw;
|
|
param.flags = time_stamp_flag;
|
|
param.uid = (unsigned long)buf[0].phys
|
|
+ (prtd->out_head * prtd->pcm_count
|
|
+ output_meta_data.meta_data_length);
|
|
if (q6asm_async_write(prtd->audio_client,
|
|
¶m) < 0)
|
|
pr_err("%s:q6asm_async_write failed\n",
|
|
__func__);
|
|
else
|
|
prtd->out_head =
|
|
(prtd->out_head + 1) & (runtime->periods - 1);
|
|
|
|
runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
|
|
return 0;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
|
|
{
|
|
int ret = 0;
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
|
|
pr_debug("%s\n", __func__);
|
|
switch (cmd) {
|
|
case SNDRV_PCM_TRIGGER_START:
|
|
prtd->pcm_irq_pos = 0;
|
|
/* intentional fall-through */
|
|
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);
|
|
if (prtd->enc_audio_client)
|
|
q6asm_run_nowait(prtd->enc_audio_client, 0, 0, 0);
|
|
atomic_set(&prtd->start, 1);
|
|
break;
|
|
case SNDRV_PCM_TRIGGER_STOP:
|
|
atomic_set(&prtd->start, 0);
|
|
runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
|
|
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);
|
|
if (prtd->enc_audio_client)
|
|
q6asm_cmd_nowait(prtd->enc_audio_client, CMD_PAUSE);
|
|
atomic_set(&prtd->start, 0);
|
|
runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
|
|
break;
|
|
default:
|
|
ret = -EINVAL;
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void populate_codec_list(struct compr_audio *compr,
|
|
struct snd_pcm_runtime *runtime)
|
|
{
|
|
pr_debug("%s\n", __func__);
|
|
/* MP3 Block */
|
|
compr->info.compr_cap.num_codecs = 14;
|
|
compr->info.compr_cap.min_fragment_size = runtime->hw.period_bytes_min;
|
|
compr->info.compr_cap.max_fragment_size = runtime->hw.period_bytes_max;
|
|
compr->info.compr_cap.min_fragments = runtime->hw.periods_min;
|
|
compr->info.compr_cap.max_fragments = runtime->hw.periods_max;
|
|
compr->info.compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
|
|
compr->info.compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
|
|
compr->info.compr_cap.codecs[2] = SND_AUDIOCODEC_AC3_PASS_THROUGH;
|
|
compr->info.compr_cap.codecs[3] = SND_AUDIOCODEC_WMA;
|
|
compr->info.compr_cap.codecs[4] = SND_AUDIOCODEC_WMA_PRO;
|
|
compr->info.compr_cap.codecs[5] = SND_AUDIOCODEC_DTS;
|
|
compr->info.compr_cap.codecs[6] = SND_AUDIOCODEC_DTS_LBR;
|
|
compr->info.compr_cap.codecs[7] = SND_AUDIOCODEC_DTS_PASS_THROUGH;
|
|
compr->info.compr_cap.codecs[8] = SND_AUDIOCODEC_AMRWB;
|
|
compr->info.compr_cap.codecs[9] = SND_AUDIOCODEC_AMRWBPLUS;
|
|
compr->info.compr_cap.codecs[10] = SND_AUDIOCODEC_PASS_THROUGH;
|
|
compr->info.compr_cap.codecs[11] = SND_AUDIOCODEC_PCM;
|
|
compr->info.compr_cap.codecs[12] = SND_AUDIOCODEC_MP2;
|
|
compr->info.compr_cap.codecs[13] = SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH;
|
|
/* Add new codecs here and update num_codecs*/
|
|
}
|
|
|
|
static int msm_compr_open(struct snd_pcm_substream *substream)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct compr_audio *compr;
|
|
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__);
|
|
compr = kzalloc(sizeof(struct compr_audio), GFP_KERNEL);
|
|
if (compr == NULL) {
|
|
pr_err("Failed to allocate memory for msm_audio\n");
|
|
return -ENOMEM;
|
|
}
|
|
prtd = &compr->prtd;
|
|
prtd->substream = substream;
|
|
runtime->render_flag = SNDRV_DMA_MODE;
|
|
prtd->audio_client = q6asm_audio_client_alloc(
|
|
(app_cb)compr_event_handler, compr);
|
|
if (!prtd->audio_client) {
|
|
pr_info("%s: Could not allocate memory\n", __func__);
|
|
kfree(prtd);
|
|
return -ENOMEM;
|
|
}
|
|
prtd->audio_client->perf_mode = false;
|
|
pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
|
|
|
|
prtd->session_id = prtd->audio_client->session;
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
runtime->hw = msm_compr_hardware_playback;
|
|
prtd->cmd_ack = 1;
|
|
} else {
|
|
runtime->hw = msm_compr_hardware_capture;
|
|
}
|
|
|
|
|
|
ret = snd_pcm_hw_constraint_list(runtime, 0,
|
|
SNDRV_PCM_HW_PARAM_RATE,
|
|
&constraints_sample_rates);
|
|
if (ret < 0)
|
|
pr_info("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_info("snd_pcm_hw_constraint_integer failed\n");
|
|
|
|
prtd->dsp_cnt = 0;
|
|
atomic_set(&prtd->pending_buffer, 1);
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
compr->codec = FORMAT_MP3;
|
|
populate_codec_list(compr, runtime);
|
|
runtime->private_data = compr;
|
|
atomic_set(&prtd->eos, 0);
|
|
compressed_audio.prtd = &compr->prtd;
|
|
ret = compressed_set_volume(0);
|
|
if (ret < 0)
|
|
pr_err("%s : Set Volume failed : %d", __func__, ret);
|
|
|
|
ret = q6asm_set_softpause(compressed_audio.prtd->audio_client,
|
|
&softpause);
|
|
if (ret < 0)
|
|
pr_err("%s: Send SoftPause Param failed ret=%d\n",
|
|
__func__, ret);
|
|
ret = q6asm_set_softvolume(compressed_audio.prtd->audio_client,
|
|
&softvol);
|
|
if (ret < 0)
|
|
pr_err("%s: Send SoftVolume Param failed ret=%d\n",
|
|
__func__, ret);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int compressed_set_volume(unsigned volume)
|
|
{
|
|
int rc = 0;
|
|
if (compressed_audio.prtd && compressed_audio.prtd->audio_client) {
|
|
rc = q6asm_set_volume(compressed_audio.prtd->audio_client,
|
|
volume);
|
|
if (rc < 0) {
|
|
pr_err("%s: Send Volume command failed"
|
|
" rc=%d\n", __func__, rc);
|
|
}
|
|
}
|
|
compressed_audio.volume = volume;
|
|
return rc;
|
|
}
|
|
|
|
static int msm_compr_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 compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
int dir = 0;
|
|
|
|
pr_debug("%s\n", __func__);
|
|
|
|
dir = IN;
|
|
atomic_set(&prtd->pending_buffer, 0);
|
|
prtd->pcm_irq_pos = 0;
|
|
q6asm_cmd(prtd->audio_client, CMD_CLOSE);
|
|
if (prtd->enc_audio_client)
|
|
q6asm_cmd(prtd->enc_audio_client, CMD_CLOSE);
|
|
compressed_audio.prtd = NULL;
|
|
q6asm_audio_client_buf_free_contiguous(dir,
|
|
prtd->audio_client);
|
|
switch (compr->info.codec_param.codec.id) {
|
|
case SND_AUDIOCODEC_AC3_PASS_THROUGH:
|
|
case SND_AUDIOCODEC_DTS_PASS_THROUGH:
|
|
case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
|
|
msm_pcm_routing_reg_psthr_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->session_id, substream->stream,
|
|
0);
|
|
default:
|
|
msm_pcm_routing_dereg_phy_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
SNDRV_PCM_STREAM_PLAYBACK);
|
|
}
|
|
if (compr->info.codec_param.codec.transcode_dts) {
|
|
msm_pcm_routing_dereg_pseudo_stream(MSM_FRONTEND_DAI_PSEUDO,
|
|
prtd->enc_audio_client->session);
|
|
}
|
|
if (prtd->enc_audio_client)
|
|
q6asm_audio_client_free(prtd->enc_audio_client);
|
|
q6asm_audio_client_free(prtd->audio_client);
|
|
kfree(prtd);
|
|
return 0;
|
|
}
|
|
|
|
static int msm_compr_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 compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
int dir = OUT;
|
|
|
|
pr_debug("%s\n", __func__);
|
|
atomic_set(&prtd->pending_buffer, 0);
|
|
q6asm_cmd(prtd->audio_client, CMD_CLOSE);
|
|
compressed_audio.prtd = NULL;
|
|
q6asm_audio_client_buf_free_contiguous(dir,
|
|
prtd->audio_client);
|
|
if (compr->info.codec_param.codec.id ==
|
|
SND_AUDIOCODEC_PASS_THROUGH)
|
|
msm_pcm_routing_reg_psthr_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->session_id, substream->stream,
|
|
0);
|
|
else
|
|
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_compr_close(struct snd_pcm_substream *substream)
|
|
{
|
|
int ret = 0;
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
ret = msm_compr_playback_close(substream);
|
|
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
|
|
ret = msm_compr_capture_close(substream);
|
|
return ret;
|
|
}
|
|
static int msm_compr_prepare(struct snd_pcm_substream *substream)
|
|
{
|
|
int ret = 0;
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
ret = msm_compr_playback_prepare(substream);
|
|
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
|
|
ret = msm_compr_capture_prepare(substream);
|
|
return ret;
|
|
}
|
|
|
|
static snd_pcm_uframes_t msm_compr_pointer(struct snd_pcm_substream *substream)
|
|
{
|
|
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
|
|
if (prtd->pcm_irq_pos >= prtd->pcm_size)
|
|
prtd->pcm_irq_pos = 0;
|
|
|
|
pr_debug("%s: pcm_irq_pos = %d, pcm_size = %d, sample_bits = %d,\n"
|
|
"frame_bits = %d\n", __func__, prtd->pcm_irq_pos,
|
|
prtd->pcm_size, runtime->sample_bits,
|
|
runtime->frame_bits);
|
|
return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
|
|
}
|
|
|
|
static int msm_compr_mmap(struct snd_pcm_substream *substream,
|
|
struct vm_area_struct *vma)
|
|
{
|
|
int result = 0;
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
|
|
pr_debug("%s\n", __func__);
|
|
prtd->mmap_flag = 1;
|
|
runtime->render_flag = SNDRV_NON_DMA_MODE;
|
|
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_compr_hw_params(struct snd_pcm_substream *substream,
|
|
struct snd_pcm_hw_params *params)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
|
|
struct compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
|
|
struct audio_buffer *buf;
|
|
int dir, ret;
|
|
|
|
pr_debug("%s\n", __func__);
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
dir = IN;
|
|
else
|
|
dir = OUT;
|
|
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
|
|
switch (compr->info.codec_param.codec.id) {
|
|
case SND_AUDIOCODEC_AC3_PASS_THROUGH:
|
|
case SND_AUDIOCODEC_DTS_PASS_THROUGH:
|
|
case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
|
|
ret = q6asm_open_write_compressed(prtd->audio_client,
|
|
compr->codec);
|
|
|
|
if (ret < 0) {
|
|
pr_err("%s: Session out open failed\n",
|
|
__func__);
|
|
return -ENOMEM;
|
|
}
|
|
break;
|
|
default:
|
|
ret = q6asm_open_write(prtd->audio_client,
|
|
compr->codec);
|
|
if (ret < 0) {
|
|
pr_err("%s: Session out open failed\n",
|
|
__func__);
|
|
return -ENOMEM;
|
|
}
|
|
msm_pcm_routing_reg_phy_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->audio_client->perf_mode,
|
|
prtd->session_id,
|
|
substream->stream);
|
|
|
|
if (compr->info.codec_param.codec.transcode_dts) {
|
|
prtd->enc_audio_client =
|
|
q6asm_audio_client_alloc(
|
|
(app_cb)compr_event_handler, compr);
|
|
if (!prtd->enc_audio_client) {
|
|
pr_err("%s: Could not allocate " \
|
|
"memory\n", __func__);
|
|
return -ENOMEM;
|
|
}
|
|
prtd->enc_audio_client->perf_mode = false;
|
|
pr_debug("%s Setting up loopback path\n",
|
|
__func__);
|
|
ret = q6asm_open_transcode_loopback(
|
|
prtd->enc_audio_client,
|
|
params_channels(params));
|
|
if (ret < 0) {
|
|
pr_err("%s: Session transcode " \
|
|
"loopback open failed\n",
|
|
__func__);
|
|
return -ENODEV;
|
|
}
|
|
}
|
|
|
|
break;
|
|
}
|
|
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
|
|
switch (compr->info.codec_param.codec.id) {
|
|
case SND_AUDIOCODEC_AMRWB:
|
|
pr_debug("q6asm_open_read(FORMAT_AMRWB)\n");
|
|
ret = q6asm_open_read(prtd->audio_client,
|
|
FORMAT_AMRWB);
|
|
if (ret < 0) {
|
|
pr_err("%s: compressed Session out open failed\n",
|
|
__func__);
|
|
return -ENOMEM;
|
|
}
|
|
pr_debug("msm_pcm_routing_reg_phy_stream\n");
|
|
msm_pcm_routing_reg_phy_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->audio_client->perf_mode,
|
|
prtd->session_id, substream->stream);
|
|
break;
|
|
case SND_AUDIOCODEC_PCM:
|
|
pr_debug("q6asm_open_read(FORMAT_PCM)\n");
|
|
ret = q6asm_open_read(prtd->audio_client,
|
|
FORMAT_MULTI_CHANNEL_LINEAR_PCM);
|
|
if (ret < 0) {
|
|
pr_err("%s: compressed Session open failed\n",
|
|
__func__);
|
|
return -ENOMEM;
|
|
}
|
|
pr_debug("msm_pcm_routing_reg_phy_stream\n");
|
|
msm_pcm_routing_reg_phy_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->audio_client->perf_mode,
|
|
prtd->session_id, substream->stream);
|
|
break;
|
|
case SND_AUDIOCODEC_PASS_THROUGH:
|
|
pr_debug("q6asm_open_read_compressed(COMPRESSED_META_DATA_MODE)\n");
|
|
ret = q6asm_open_read_compressed(prtd->audio_client,
|
|
MAX_NUM_FRAMES_PER_BUFFER,
|
|
COMPRESSED_META_DATA_MODE);
|
|
break;
|
|
default:
|
|
pr_err("Invalid codec for compressed session open\n");
|
|
return -EFAULT;
|
|
}
|
|
|
|
if (ret < 0) {
|
|
pr_err("%s: compressed Session out open failed\n",
|
|
__func__);
|
|
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__);
|
|
return -ENOMEM;
|
|
}
|
|
/* Modifying kernel hardware params based on userspace config */
|
|
if (params_periods(params) > 0 &&
|
|
(params_periods(params) != runtime->hw.periods_max)) {
|
|
runtime->hw.periods_max = params_periods(params);
|
|
}
|
|
if (params_period_bytes(params) > 0 &&
|
|
(params_period_bytes(params) != runtime->hw.period_bytes_min)) {
|
|
runtime->hw.period_bytes_min = params_period_bytes(params);
|
|
}
|
|
runtime->hw.buffer_bytes_max =
|
|
runtime->hw.period_bytes_min * runtime->hw.periods_max;
|
|
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;
|
|
|
|
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;
|
|
|
|
pr_debug("%s: buf[%p]dma_buf->area[%p]dma_buf->addr[%p]\n"
|
|
"dma_buf->bytes[%d]\n", __func__,
|
|
(void *)buf, (void *)dma_buf->area,
|
|
(void *)dma_buf->addr, dma_buf->bytes);
|
|
if (!dma_buf->area)
|
|
return -ENOMEM;
|
|
|
|
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
|
|
return 0;
|
|
}
|
|
|
|
static int msm_compr_ioctl(struct snd_pcm_substream *substream,
|
|
unsigned int cmd, void *arg)
|
|
{
|
|
int rc = 0;
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct compr_audio *compr = runtime->private_data;
|
|
struct msm_audio *prtd = &compr->prtd;
|
|
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_COMPRESS_GET_CAPS:
|
|
pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
|
|
if (copy_to_user((void *) arg, &compr->info.compr_cap,
|
|
sizeof(struct snd_compr_caps))) {
|
|
rc = -EFAULT;
|
|
pr_err("%s: ERROR: copy to user\n", __func__);
|
|
return rc;
|
|
}
|
|
return 0;
|
|
case SNDRV_COMPRESS_SET_PARAMS:
|
|
pr_debug("SNDRV_COMPRESS_SET_PARAMS: ");
|
|
if (copy_from_user(&compr->info.codec_param, (void *) arg,
|
|
sizeof(struct snd_compr_params))) {
|
|
rc = -EFAULT;
|
|
pr_err("%s: ERROR: copy from user\n", __func__);
|
|
return rc;
|
|
}
|
|
/*
|
|
* DTS Security needed for the transcode path
|
|
*/
|
|
if (compr->info.codec_param.codec.transcode_dts) {
|
|
char modelId[128];
|
|
struct snd_dec_dts opt_dts =
|
|
compr->info.codec_param.codec.dts;
|
|
int modelIdLength = opt_dts.modelIdLength;
|
|
if (copy_from_user(modelId, (void *)opt_dts.modelId,
|
|
modelIdLength))
|
|
pr_err("%s: ERROR: copy modelId\n", __func__);
|
|
modelId[modelIdLength] = '\0';
|
|
pr_debug("%s: Received modelId =%s,length=%d\n",
|
|
__func__, modelId, modelIdLength);
|
|
core_set_dts_model_id(modelIdLength, modelId);
|
|
}
|
|
switch (compr->info.codec_param.codec.id) {
|
|
case SND_AUDIOCODEC_MP3:
|
|
/* For MP3 we dont need any other parameter */
|
|
pr_debug("SND_AUDIOCODEC_MP3\n");
|
|
compr->codec = FORMAT_MP3;
|
|
break;
|
|
case SND_AUDIOCODEC_AAC:
|
|
pr_debug("SND_AUDIOCODEC_AAC\n");
|
|
compr->codec = FORMAT_MPEG4_AAC;
|
|
break;
|
|
case SND_AUDIOCODEC_AC3_PASS_THROUGH:
|
|
pr_debug("SND_AUDIOCODEC_AC3_PASS_THROUGH\n");
|
|
compr->codec = FORMAT_AC3;
|
|
break;
|
|
case SND_AUDIOCODEC_WMA:
|
|
pr_debug("SND_AUDIOCODEC_WMA\n");
|
|
compr->codec = FORMAT_WMA_V9;
|
|
break;
|
|
case SND_AUDIOCODEC_WMA_PRO:
|
|
pr_debug("SND_AUDIOCODEC_WMA_PRO\n");
|
|
compr->codec = FORMAT_WMA_V10PRO;
|
|
break;
|
|
case SND_AUDIOCODEC_DTS_PASS_THROUGH:
|
|
pr_debug("SND_AUDIOCODEC_DTS_PASS_THROUGH\n");
|
|
compr->codec = FORMAT_DTS;
|
|
break;
|
|
case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
|
|
pr_debug("SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH\n");
|
|
compr->codec = FORMAT_DTS_LBR;
|
|
break;
|
|
case SND_AUDIOCODEC_DTS: {
|
|
char modelId[128];
|
|
struct snd_dec_dts opt_dts =
|
|
compr->info.codec_param.codec.dts;
|
|
int modelIdLength = opt_dts.modelIdLength;
|
|
pr_debug("SND_AUDIOCODEC_DTS\n");
|
|
if (copy_from_user(modelId, (void *)opt_dts.modelId,
|
|
modelIdLength))
|
|
pr_err("%s: ERROR: copy modelId\n", __func__);
|
|
modelId[modelIdLength] = '\0';
|
|
pr_debug("%s: Received modelId =%s,length=%d\n",
|
|
__func__, modelId, modelIdLength);
|
|
core_set_dts_model_id(modelIdLength, modelId);
|
|
compr->codec = FORMAT_DTS;
|
|
}
|
|
break;
|
|
case SND_AUDIOCODEC_DTS_LBR:{
|
|
char modelId[128];
|
|
struct snd_dec_dts opt_dts =
|
|
compr->info.codec_param.codec.dts;
|
|
int modelIdLength = opt_dts.modelIdLength;
|
|
pr_debug("SND_AUDIOCODEC_DTS_LBR\n");
|
|
if (copy_from_user(modelId, (void *)opt_dts.modelId,
|
|
modelIdLength))
|
|
pr_err("%s: ERROR: copy modelId\n", __func__);
|
|
modelId[modelIdLength] = '\0';
|
|
pr_debug("%s: Received modelId =%s,length=%d\n",
|
|
__func__, modelId, modelIdLength);
|
|
core_set_dts_model_id(modelIdLength, modelId);
|
|
compr->codec = FORMAT_DTS_LBR;
|
|
}
|
|
break;
|
|
case SND_AUDIOCODEC_AMRWB:
|
|
pr_debug("msm_compr_ioctl SND_AUDIOCODEC_AMRWB\n");
|
|
compr->codec = FORMAT_AMRWB;
|
|
break;
|
|
case SND_AUDIOCODEC_AMRWBPLUS:
|
|
pr_debug("msm_compr_ioctl SND_AUDIOCODEC_AMRWBPLUS\n");
|
|
compr->codec = FORMAT_AMR_WB_PLUS;
|
|
break;
|
|
case SND_AUDIOCODEC_PASS_THROUGH:
|
|
/* format pass through is used for HDMI IN compressed
|
|
where the decoder format is indicated by LPASS */
|
|
pr_debug("msm_compr_ioctl SND_AUDIOCODEC_PASSTHROUGH\n");
|
|
compr->codec = FORMAT_PASS_THROUGH;
|
|
break;
|
|
case SND_AUDIOCODEC_PCM:
|
|
pr_debug("msm_compr_ioctl SND_AUDIOCODEC_PCM\n");
|
|
compr->codec = FORMAT_MULTI_CHANNEL_LINEAR_PCM;
|
|
break;
|
|
case SND_AUDIOCODEC_MP2:
|
|
pr_debug("SND_AUDIOCODEC_MP2\n");
|
|
compr->codec = FORMAT_MP2;
|
|
break;
|
|
default:
|
|
pr_err("msm_compr_ioctl failed..unknown codec\n");
|
|
return -EFAULT;
|
|
}
|
|
return 0;
|
|
case SNDRV_PCM_IOCTL1_RESET:
|
|
pr_debug("SNDRV_PCM_IOCTL1_RESET\n");
|
|
/* Flush only when session is started during CAPTURE,
|
|
while PLAYBACK has no such restriction. */
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
|
|
(substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
|
|
atomic_read(&prtd->start))) {
|
|
if (atomic_read(&prtd->eos)) {
|
|
prtd->cmd_ack = 1;
|
|
wake_up(&the_locks.eos_wait);
|
|
atomic_set(&prtd->eos, 0);
|
|
atomic_set(&prtd->pending_buffer, 1);
|
|
}
|
|
|
|
/* A unlikely race condition possible with FLUSH
|
|
DRAIN if ack is set by flush and reset by drain */
|
|
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);
|
|
return rc;
|
|
}
|
|
rc = wait_event_timeout(the_locks.flush_wait,
|
|
prtd->cmd_ack, 5 * HZ);
|
|
if (!rc)
|
|
pr_err("Flush cmd timeout\n");
|
|
prtd->pcm_irq_pos = 0;
|
|
}
|
|
break;
|
|
case SNDRV_COMPRESS_DRAIN:
|
|
pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__);
|
|
atomic_set(&prtd->eos, 1);
|
|
atomic_set(&prtd->pending_buffer, 0);
|
|
prtd->cmd_ack = 0;
|
|
q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
|
|
/* Wait indefinitely for DRAIN. Flush can also signal this*/
|
|
rc = wait_event_interruptible(the_locks.eos_wait,
|
|
prtd->cmd_ack);
|
|
if (rc < 0)
|
|
pr_err("EOS cmd interrupted\n");
|
|
pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait\n", __func__);
|
|
return 0;
|
|
default:
|
|
break;
|
|
}
|
|
return snd_pcm_lib_ioctl(substream, cmd, arg);
|
|
}
|
|
|
|
static struct snd_pcm_ops msm_compr_ops = {
|
|
.open = msm_compr_open,
|
|
.hw_params = msm_compr_hw_params,
|
|
.close = msm_compr_close,
|
|
.ioctl = msm_compr_ioctl,
|
|
.prepare = msm_compr_prepare,
|
|
.trigger = msm_compr_trigger,
|
|
.pointer = msm_compr_pointer,
|
|
.mmap = msm_compr_mmap,
|
|
.restart = msm_compr_restart,
|
|
};
|
|
|
|
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_compr_ops,
|
|
.pcm_new = msm_asoc_pcm_new,
|
|
};
|
|
|
|
static __devinit int msm_compr_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_compr_remove(struct platform_device *pdev)
|
|
{
|
|
snd_soc_unregister_platform(&pdev->dev);
|
|
return 0;
|
|
}
|
|
|
|
static struct platform_driver msm_compr_driver = {
|
|
.driver = {
|
|
.name = "msm-compr-dsp",
|
|
.owner = THIS_MODULE,
|
|
},
|
|
.probe = msm_compr_probe,
|
|
.remove = __devexit_p(msm_compr_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);
|
|
init_waitqueue_head(&the_locks.flush_wait);
|
|
|
|
return platform_driver_register(&msm_compr_driver);
|
|
}
|
|
module_init(msm_soc_platform_init);
|
|
|
|
static void __exit msm_soc_platform_exit(void)
|
|
{
|
|
platform_driver_unregister(&msm_compr_driver);
|
|
}
|
|
module_exit(msm_soc_platform_exit);
|
|
|
|
MODULE_DESCRIPTION("PCM module platform driver");
|
|
MODULE_LICENSE("GPL v2");
|