1692 lines
48 KiB
C
1692 lines
48 KiB
C
/* Copyright (c) 2012-2015, 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-v2.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 <linux/msm_audio_ion.h>
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#include <sound/timer.h>
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#include "msm-compr-q6-v2.h"
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#include "msm-pcm-routing-v2.h"
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#include <sound/tlv.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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/* Changing period size to 4032. 4032 will make sure COMPRE_CAPTURE_PERIOD_SIZE
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* is 4096 with meta data size of 64 and MAX_NUM_FRAMES_PER_BUFFER 1
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*/
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#define COMPRE_CAPTURE_MAX_FRAME_SIZE (4032)
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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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#define COMPRESSED_LR_VOL_MAX_STEPS 0x20002000
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#define MAX_AC3_PARAM_SIZE (18*2*sizeof(int))
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#define AMR_WB_BAND_MODE 8
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#define AMR_WB_DTX_MODE 0
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const DECLARE_TLV_DB_LINEAR(compr_rx_vol_gain, 0,
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COMPRESSED_LR_VOL_MAX_STEPS);
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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 | SNDRV_PCM_FMTBIT_S24_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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/* Add supported codecs for compress capture path */
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static uint32_t supported_compr_capture_codecs[] = {
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SND_AUDIOCODEC_AMRWB
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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 bool msm_compr_capture_codecs(uint32_t req_codec)
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{
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int i;
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pr_debug("%s req_codec:%d\n", __func__, req_codec);
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if (req_codec == 0)
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return false;
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for (i = 0; i < ARRAY_SIZE(supported_compr_capture_codecs); i++) {
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if (req_codec == supported_compr_capture_codecs[i])
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return true;
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}
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return false;
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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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phys_addr_t temp;
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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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int stop_playback = 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_V2: {
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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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/*
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* check for underrun
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*/
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snd_pcm_stream_lock_irq(substream);
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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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stop_playback = 1;
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}
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snd_pcm_stream_unlock_irq(substream);
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if (stop_playback) {
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pr_err("underrun! render stopped\n");
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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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temp = buf[0].phys + (prtd->out_head * prtd->pcm_count);
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pr_debug("%s:writing buffer[%d] from 0x%pa\n",
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__func__, prtd->out_head, &temp);
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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 = temp + 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 = prtd->session_id;
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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_EVENT_RENDERED_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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break;
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case ASM_DATA_EVENT_READ_DONE_V2: {
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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 + READDONE_IDX_SIZE),
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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",
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ptrmem[READDONE_IDX_SIZE],
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ptrmem[READDONE_IDX_BUFADD_LSW]);
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if (ptrmem[READDONE_IDX_SIZE] > 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%pa\n",
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prtd->pcm_irq_pos, &buf[0].phys);
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read_param.len = prtd->pcm_count - COMPRE_CAPTURE_HEADER_SIZE;
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read_param.paddr = 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 APR_BASIC_RSP_RESULT: {
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switch (payload[0]) {
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case ASM_SESSION_CMD_RUN_V2: {
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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 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%pa head %d count %d\n",
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__func__, prtd->out_head, &buf[0].phys,
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prtd->pcm_count, prtd->out_head);
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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 = 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 = prtd->session_id;
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param.metadata_len = COMPRE_OUTPUT_METADATA_SIZE;
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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_send_ddp_cfg(struct audio_client *ac,
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struct snd_dec_ddp *ddp)
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{
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int i, rc;
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pr_debug("%s\n", __func__);
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if (ddp->params_length / 2 > SND_DEC_DDP_MAX_PARAMS) {
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pr_err("%s: Invalid number of params %u, max allowed %u\n",
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__func__, ddp->params_length / 2,
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SND_DEC_DDP_MAX_PARAMS);
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return -EINVAL;
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}
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for (i = 0; i < ddp->params_length/2; i++) {
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rc = q6asm_ds1_set_endp_params(ac, ddp->params_id[i],
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ddp->params_value[i]);
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if (rc) {
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pr_err("sending params_id: %d failed\n",
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ddp->params_id[i]);
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return rc;
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}
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}
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return 0;
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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 snd_pcm_hw_params *params;
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struct asm_aac_cfg aac_cfg;
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uint16_t bits_per_sample = 16;
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int ret;
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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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pr_debug("%s\n", __func__);
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params = &soc_prtd->dpcm[substream->stream].hw_params;
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if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
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bits_per_sample = 24;
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ret = q6asm_open_write_v2(prtd->audio_client,
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compr->codec, bits_per_sample);
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if (ret < 0) {
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pr_err("%s: Session out open failed\n",
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__func__);
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return -ENOMEM;
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}
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msm_pcm_routing_reg_phy_stream(
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soc_prtd->dai_link->be_id,
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prtd->audio_client->perf_mode,
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prtd->session_id,
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substream->stream);
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/*
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* the number of channels are required to call volume api
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* accoridngly. So, get channels from hw params
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*/
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if ((params_channels(params) > 0) &&
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(params_periods(params) <= runtime->hw.channels_max))
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prtd->channel_mode = params_channels(params);
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ret = q6asm_set_softpause(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(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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ret = q6asm_set_io_mode(prtd->audio_client,
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(COMPRESSED_IO | 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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return -ENOMEM;
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}
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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 */
|
|
prtd->samp_rate = runtime->rate;
|
|
prtd->channel_mode = runtime->channels;
|
|
prtd->out_head = 0;
|
|
atomic_set(&prtd->out_count, runtime->periods);
|
|
|
|
if (prtd->enabled)
|
|
return 0;
|
|
|
|
switch (compr->info.codec_param.codec.id) {
|
|
case SND_AUDIOCODEC_MP3:
|
|
/* No media format block for mp3 */
|
|
break;
|
|
case SND_AUDIOCODEC_AAC:
|
|
pr_debug("%s: SND_AUDIOCODEC_AAC\n", __func__);
|
|
memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
|
|
aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
|
|
aac_cfg.format = 0x03;
|
|
aac_cfg.ch_cfg = runtime->channels;
|
|
aac_cfg.sample_rate = runtime->rate;
|
|
ret = q6asm_media_format_block_aac(prtd->audio_client,
|
|
&aac_cfg);
|
|
if (ret < 0)
|
|
pr_err("%s: CMD Format block failed\n", __func__);
|
|
break;
|
|
case SND_AUDIOCODEC_AC3: {
|
|
struct snd_dec_ddp *ddp =
|
|
&compr->info.codec_param.codec.options.ddp;
|
|
pr_debug("%s: SND_AUDIOCODEC_AC3\n", __func__);
|
|
ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
|
|
if (ret < 0)
|
|
pr_err("%s: DDP CMD CFG failed\n", __func__);
|
|
break;
|
|
}
|
|
case SND_AUDIOCODEC_EAC3: {
|
|
struct snd_dec_ddp *ddp =
|
|
&compr->info.codec_param.codec.options.ddp;
|
|
pr_debug("%s: SND_AUDIOCODEC_EAC3\n", __func__);
|
|
ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
|
|
if (ret < 0)
|
|
pr_err("%s: DDP CMD CFG failed\n", __func__);
|
|
break;
|
|
}
|
|
default:
|
|
return -EINVAL;
|
|
}
|
|
|
|
prtd->enabled = 1;
|
|
prtd->cmd_ack = 0;
|
|
prtd->cmd_interrupt = 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 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 snd_soc_pcm_runtime *soc_prtd = substream->private_data;
|
|
struct audio_aio_read_param read_param;
|
|
uint16_t bits_per_sample = 16;
|
|
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;
|
|
|
|
if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
|
|
bits_per_sample = 24;
|
|
|
|
if (!msm_compr_capture_codecs(
|
|
compr->info.codec_param.codec.id)) {
|
|
/*
|
|
* request codec invalid or not supported,
|
|
* use default compress format
|
|
*/
|
|
compr->info.codec_param.codec.id =
|
|
SND_AUDIOCODEC_AMRWB;
|
|
}
|
|
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;
|
|
default:
|
|
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);
|
|
*/
|
|
ret = -EINVAL;
|
|
break;
|
|
}
|
|
|
|
if (ret < 0) {
|
|
pr_err("%s: compressed Session out open failed\n",
|
|
__func__);
|
|
return -ENOMEM;
|
|
}
|
|
|
|
ret = q6asm_set_io_mode(prtd->audio_client,
|
|
(COMPRESSED_IO | ASYNC_IO_MODE));
|
|
if (ret < 0) {
|
|
pr_err("%s: Set IO mode failed\n", __func__);
|
|
return -ENOMEM;
|
|
}
|
|
|
|
if (!msm_compr_capture_codecs(codec->id)) {
|
|
/*
|
|
* request codec invalid or not supported,
|
|
* use default compress format
|
|
*/
|
|
codec->id = SND_AUDIOCODEC_AMRWB;
|
|
}
|
|
/* 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,
|
|
/*
|
|
* use fixed band mode and dtx mode
|
|
* band mode - 23.85 kbps
|
|
*/
|
|
AMR_WB_BAND_MODE,
|
|
/* dtx mode - disable */
|
|
AMR_WB_DTX_MODE);
|
|
if (ret < 0)
|
|
pr_err("%s: CMD Format block failed: %d\n",
|
|
__func__, ret);
|
|
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:
|
|
read_param.len = prtd->pcm_count
|
|
- COMPRE_CAPTURE_HEADER_SIZE;
|
|
read_param.paddr = buf[i].phys
|
|
+ COMPRE_CAPTURE_HEADER_SIZE;
|
|
pr_debug("Push buffer [%d] to DSP, paddr: %pa, vaddr: %p\n",
|
|
i, &read_param.paddr,
|
|
buf[i].data);
|
|
q6asm_async_read(prtd->audio_client, &read_param);
|
|
break;
|
|
default:
|
|
read_param.paddr = buf[i].phys;
|
|
/*q6asm_async_read_compressed(prtd->audio_client,
|
|
&read_param);*/
|
|
pr_debug("%s: To add support for read compressed\n",
|
|
__func__);
|
|
ret = -EINVAL;
|
|
break;
|
|
}
|
|
}
|
|
prtd->periods = runtime->periods;
|
|
|
|
prtd->enabled = 1;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
|
|
{
|
|
int ret = 0;
|
|
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;
|
|
|
|
pr_debug("%s\n", __func__);
|
|
switch (cmd) {
|
|
case SNDRV_PCM_TRIGGER_START:
|
|
prtd->pcm_irq_pos = 0;
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
|
|
if (!msm_compr_capture_codecs(
|
|
compr->info.codec_param.codec.id)) {
|
|
/*
|
|
* request codec invalid or not supported,
|
|
* use default compress format
|
|
*/
|
|
compr->info.codec_param.codec.id =
|
|
SND_AUDIOCODEC_AMRWB;
|
|
}
|
|
switch (compr->info.codec_param.codec.id) {
|
|
case SND_AUDIOCODEC_AMRWB:
|
|
break;
|
|
default:
|
|
msm_pcm_routing_reg_psthr_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->session_id, substream->stream);
|
|
break;
|
|
}
|
|
}
|
|
atomic_set(&prtd->pending_buffer, 1);
|
|
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);
|
|
atomic_set(&prtd->start, 1);
|
|
break;
|
|
case SNDRV_PCM_TRIGGER_STOP:
|
|
pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
|
|
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
|
|
switch (compr->info.codec_param.codec.id) {
|
|
case SND_AUDIOCODEC_AMRWB:
|
|
break;
|
|
default:
|
|
msm_pcm_routing_reg_psthr_stream(
|
|
soc_prtd->dai_link->be_id,
|
|
prtd->session_id, substream->stream);
|
|
break;
|
|
}
|
|
}
|
|
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);
|
|
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 = 5;
|
|
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;
|
|
compr->info.compr_cap.codecs[3] = SND_AUDIOCODEC_EAC3;
|
|
compr->info.compr_cap.codecs[4] = SND_AUDIOCODEC_AMRWB;
|
|
/* Add new codecs here */
|
|
}
|
|
|
|
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;
|
|
|
|
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);
|
|
return 0;
|
|
}
|
|
|
|
static int compressed_set_volume(struct msm_audio *prtd, uint32_t volume)
|
|
{
|
|
int rc = 0;
|
|
int avg_vol = 0;
|
|
int lgain = (volume >> 16) & 0xFFFF;
|
|
int rgain = volume & 0xFFFF;
|
|
if (prtd && prtd->audio_client) {
|
|
pr_debug("%s: channels %d volume 0x%x\n", __func__,
|
|
prtd->channel_mode, volume);
|
|
if ((prtd->channel_mode == 2) &&
|
|
(lgain != rgain)) {
|
|
pr_debug("%s: call q6asm_set_lrgain\n", __func__);
|
|
rc = q6asm_set_lrgain(prtd->audio_client, lgain, rgain);
|
|
} else {
|
|
avg_vol = (lgain + rgain)/2;
|
|
pr_debug("%s: call q6asm_set_volume\n", __func__);
|
|
rc = q6asm_set_volume(prtd->audio_client, avg_vol);
|
|
}
|
|
if (rc < 0) {
|
|
pr_err("%s: Send Volume command failed rc=%d\n",
|
|
__func__, rc);
|
|
}
|
|
}
|
|
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);
|
|
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);
|
|
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);
|
|
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_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)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct msm_audio *prtd = runtime->private_data;
|
|
struct audio_client *ac = prtd->audio_client;
|
|
struct audio_port_data *apd = ac->port;
|
|
struct audio_buffer *ab;
|
|
int dir = -1;
|
|
|
|
prtd->mmap_flag = 1;
|
|
runtime->render_flag = SNDRV_NON_DMA_MODE;
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
dir = IN;
|
|
else
|
|
dir = OUT;
|
|
ab = &(apd[dir].buf[0]);
|
|
|
|
return msm_audio_ion_mmap(ab, vma);
|
|
}
|
|
|
|
static int msm_compr_hw_params(struct snd_pcm_substream *substream,
|
|
struct snd_pcm_hw_params *params)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
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;
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
dir = IN;
|
|
else
|
|
dir = OUT;
|
|
/* 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;
|
|
pr_debug("allocate %zd buffers each of size %d\n",
|
|
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[%pa]\n"
|
|
"dma_buf->bytes[%zd]\n", __func__,
|
|
(void *)buf, (void *)dma_buf->area,
|
|
&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_shared(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");
|
|
tstamp = arg;
|
|
memset(tstamp, 0x0, sizeof(*tstamp));
|
|
rc = q6asm_get_session_time(prtd->audio_client, ×tamp);
|
|
if (rc < 0) {
|
|
pr_err("%s: Get Session Time return value =%lld\n",
|
|
__func__, timestamp);
|
|
return -EAGAIN;
|
|
}
|
|
temp = (timestamp * 2 * runtime->channels);
|
|
temp = temp * (runtime->rate/1000);
|
|
temp = div_u64(temp, 1000);
|
|
tstamp->sampling_rate = runtime->rate;
|
|
tstamp->timestamp = timestamp;
|
|
pr_debug("%s: bytes_consumed:,timestamp = %lld,\n",
|
|
__func__,
|
|
tstamp->timestamp);
|
|
return 0;
|
|
}
|
|
case SNDRV_COMPRESS_GET_CAPS: {
|
|
struct snd_compr_caps *caps;
|
|
caps = arg;
|
|
memset(caps, 0, sizeof(*caps));
|
|
pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
|
|
memcpy(caps, &compr->info.compr_cap, sizeof(*caps));
|
|
return 0;
|
|
}
|
|
case SNDRV_COMPRESS_SET_PARAMS:
|
|
pr_debug("SNDRV_COMPRESS_SET_PARAMS:\n");
|
|
memcpy(&compr->info.codec_param, (void *) arg,
|
|
sizeof(struct snd_compr_params));
|
|
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: {
|
|
char params_value[MAX_AC3_PARAM_SIZE];
|
|
int *params_value_data = (int *)params_value;
|
|
/* 36 is the max param length for ddp */
|
|
int i;
|
|
struct snd_dec_ddp *ddp =
|
|
&compr->info.codec_param.codec.options.ddp;
|
|
uint32_t params_length = 0;
|
|
/* check integer overflow */
|
|
if (ddp->params_length > UINT_MAX/sizeof(int)) {
|
|
pr_err("%s: Integer overflow ddp->params_length %d\n",
|
|
__func__, ddp->params_length);
|
|
return -EINVAL;
|
|
}
|
|
params_length = ddp->params_length*sizeof(int);
|
|
if (params_length > MAX_AC3_PARAM_SIZE) {
|
|
/*MAX is 36*sizeof(int) this should not happen*/
|
|
pr_err("%s: params_length(%d) is greater than %zd\n",
|
|
__func__, params_length, MAX_AC3_PARAM_SIZE);
|
|
return -EINVAL;
|
|
}
|
|
pr_debug("SND_AUDIOCODEC_AC3\n");
|
|
compr->codec = FORMAT_AC3;
|
|
pr_debug("params_length: %d\n", ddp->params_length);
|
|
for (i = 0; i < params_length/sizeof(int); i++)
|
|
pr_debug("params_value[%d]: %x\n", i,
|
|
params_value_data[i]);
|
|
for (i = 0; i < ddp->params_length/2; i++) {
|
|
ddp->params_id[i] = params_value_data[2*i];
|
|
ddp->params_value[i] = params_value_data[2*i+1];
|
|
}
|
|
if (atomic_read(&prtd->start)) {
|
|
rc = msm_compr_send_ddp_cfg(prtd->audio_client,
|
|
ddp);
|
|
if (rc < 0)
|
|
pr_err("%s: DDP CMD CFG failed\n",
|
|
__func__);
|
|
}
|
|
break;
|
|
}
|
|
case SND_AUDIOCODEC_EAC3: {
|
|
char params_value[MAX_AC3_PARAM_SIZE];
|
|
int *params_value_data = (int *)params_value;
|
|
/* 36 is the max param length for ddp */
|
|
int i;
|
|
struct snd_dec_ddp *ddp =
|
|
&compr->info.codec_param.codec.options.ddp;
|
|
uint32_t params_length = 0;
|
|
/* check integer overflow */
|
|
if (ddp->params_length > UINT_MAX/sizeof(int)) {
|
|
pr_err("%s: Integer overflow ddp->params_length %d\n",
|
|
__func__, ddp->params_length);
|
|
return -EINVAL;
|
|
}
|
|
if (params_length > MAX_AC3_PARAM_SIZE) {
|
|
/*MAX is 36*sizeof(int) this should not happen*/
|
|
pr_err("%s: params_length(%d) is greater than %zd\n",
|
|
__func__, params_length, MAX_AC3_PARAM_SIZE);
|
|
return -EINVAL;
|
|
}
|
|
pr_debug("SND_AUDIOCODEC_EAC3\n");
|
|
compr->codec = FORMAT_EAC3;
|
|
pr_debug("params_length: %d\n", ddp->params_length);
|
|
for (i = 0; i < ddp->params_length; i++)
|
|
pr_debug("params_value[%d]: %x\n", i,
|
|
params_value_data[i]);
|
|
for (i = 0; i < ddp->params_length/2; i++) {
|
|
ddp->params_id[i] = params_value_data[2*i];
|
|
ddp->params_value[i] = params_value_data[2*i+1];
|
|
}
|
|
if (atomic_read(&prtd->start)) {
|
|
rc = msm_compr_send_ddp_cfg(prtd->audio_client,
|
|
ddp);
|
|
if (rc < 0)
|
|
pr_err("%s: DDP CMD CFG failed\n",
|
|
__func__);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
pr_debug("FORMAT_LINEAR_PCM\n");
|
|
compr->codec = FORMAT_LINEAR_PCM;
|
|
break;
|
|
}
|
|
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_interrupt = 1;
|
|
wake_up(&the_locks.eos_wait);
|
|
atomic_set(&prtd->eos, 0);
|
|
}
|
|
|
|
/* 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__);
|
|
if (atomic_read(&prtd->pending_buffer)) {
|
|
pr_debug("%s: no pending writes, drain would block\n",
|
|
__func__);
|
|
return -EWOULDBLOCK;
|
|
}
|
|
|
|
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 || prtd->cmd_interrupt));
|
|
|
|
if (rc < 0)
|
|
pr_err("EOS cmd interrupted\n");
|
|
pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait\n", __func__);
|
|
|
|
if (prtd->cmd_interrupt)
|
|
rc = -EINTR;
|
|
|
|
prtd->cmd_interrupt = 0;
|
|
return rc;
|
|
default:
|
|
break;
|
|
}
|
|
return snd_pcm_lib_ioctl(substream, cmd, arg);
|
|
}
|
|
#ifdef CONFIG_COMPAT
|
|
struct snd_enc_wma32 {
|
|
u32 super_block_align; /* WMA Type-specific data */
|
|
u32 encodeopt1;
|
|
u32 encodeopt2;
|
|
};
|
|
|
|
struct snd_enc_vorbis32 {
|
|
s32 quality;
|
|
u32 managed;
|
|
u32 max_bit_rate;
|
|
u32 min_bit_rate;
|
|
u32 downmix;
|
|
};
|
|
|
|
struct snd_enc_real32 {
|
|
u32 quant_bits;
|
|
u32 start_region;
|
|
u32 num_regions;
|
|
};
|
|
|
|
struct snd_enc_flac32 {
|
|
u32 num;
|
|
u32 gain;
|
|
};
|
|
|
|
struct snd_enc_generic32 {
|
|
u32 bw; /* encoder bandwidth */
|
|
s32 reserved[15];
|
|
};
|
|
struct snd_dec_ddp32 {
|
|
u32 params_length;
|
|
u32 params_id[18];
|
|
u32 params_value[18];
|
|
};
|
|
|
|
union snd_codec_options32 {
|
|
struct snd_enc_wma32 wma;
|
|
struct snd_enc_vorbis32 vorbis;
|
|
struct snd_enc_real32 real;
|
|
struct snd_enc_flac32 flac;
|
|
struct snd_enc_generic32 generic;
|
|
struct snd_dec_ddp32 ddp;
|
|
};
|
|
|
|
struct snd_codec32 {
|
|
u32 id;
|
|
u32 ch_in;
|
|
u32 ch_out;
|
|
u32 sample_rate;
|
|
u32 bit_rate;
|
|
u32 rate_control;
|
|
u32 profile;
|
|
u32 level;
|
|
u32 ch_mode;
|
|
u32 format;
|
|
u32 align;
|
|
union snd_codec_options32 options;
|
|
u32 reserved[3];
|
|
};
|
|
|
|
struct snd_compressed_buffer32 {
|
|
u32 fragment_size;
|
|
u32 fragments;
|
|
};
|
|
|
|
struct snd_compr_params32 {
|
|
struct snd_compressed_buffer32 buffer;
|
|
struct snd_codec32 codec;
|
|
u8 no_wake_mode;
|
|
};
|
|
|
|
struct snd_compr_caps32 {
|
|
u32 num_codecs;
|
|
u32 direction;
|
|
u32 min_fragment_size;
|
|
u32 max_fragment_size;
|
|
u32 min_fragments;
|
|
u32 max_fragments;
|
|
u32 codecs[MAX_NUM_CODECS];
|
|
u32 reserved[11];
|
|
};
|
|
struct snd_compr_tstamp32 {
|
|
u32 byte_offset;
|
|
u32 copied_total;
|
|
compat_ulong_t pcm_frames;
|
|
compat_ulong_t pcm_io_frames;
|
|
u32 sampling_rate;
|
|
compat_u64 timestamp;
|
|
};
|
|
enum {
|
|
SNDRV_COMPRESS_TSTAMP32 = _IOR('C', 0x20, struct snd_compr_tstamp32),
|
|
SNDRV_COMPRESS_GET_CAPS32 = _IOWR('C', 0x10, struct snd_compr_caps32),
|
|
SNDRV_COMPRESS_SET_PARAMS32 =
|
|
_IOW('C', 0x12, struct snd_compr_params32),
|
|
};
|
|
static int msm_compr_compat_ioctl(struct snd_pcm_substream *substream,
|
|
unsigned int cmd, void *arg)
|
|
{
|
|
int err = 0;
|
|
switch (cmd) {
|
|
case SNDRV_COMPRESS_TSTAMP32: {
|
|
struct snd_compr_tstamp tstamp;
|
|
struct snd_compr_tstamp32 tstamp32;
|
|
memset(&tstamp, 0, sizeof(tstamp));
|
|
memset(&tstamp32, 0, sizeof(tstamp32));
|
|
cmd = SNDRV_COMPRESS_TSTAMP;
|
|
err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
|
|
if (err) {
|
|
pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
|
|
__func__, err);
|
|
goto bail_out;
|
|
}
|
|
tstamp32.byte_offset = tstamp.byte_offset;
|
|
tstamp32.copied_total = tstamp.copied_total;
|
|
tstamp32.pcm_frames = tstamp.pcm_frames;
|
|
tstamp32.pcm_io_frames = tstamp.pcm_io_frames;
|
|
tstamp32.sampling_rate = tstamp.sampling_rate;
|
|
tstamp32.timestamp = tstamp.timestamp;
|
|
if (copy_to_user(arg, &tstamp32, sizeof(tstamp32))) {
|
|
pr_err("%s: copytouser failed COMPRESS_TSTAMP32\n",
|
|
__func__);
|
|
err = -EFAULT;
|
|
}
|
|
break;
|
|
}
|
|
case SNDRV_COMPRESS_GET_CAPS32: {
|
|
struct snd_compr_caps caps;
|
|
struct snd_compr_caps32 caps32;
|
|
u32 i;
|
|
memset(&caps, 0, sizeof(caps));
|
|
memset(&caps32, 0, sizeof(caps32));
|
|
cmd = SNDRV_COMPRESS_GET_CAPS;
|
|
err = msm_compr_ioctl_shared(substream, cmd, &caps);
|
|
if (err) {
|
|
pr_err("%s: GET_CAPS failed rc %d\n",
|
|
__func__, err);
|
|
goto bail_out;
|
|
}
|
|
pr_debug("SNDRV_COMPRESS_GET_CAPS_32\n");
|
|
if (!err && caps.num_codecs >= MAX_NUM_CODECS) {
|
|
pr_err("%s: Invalid number of codecs\n", __func__);
|
|
err = -EINVAL;
|
|
goto bail_out;
|
|
}
|
|
caps32.direction = caps.direction;
|
|
caps32.max_fragment_size = caps.max_fragment_size;
|
|
caps32.max_fragments = caps.max_fragments;
|
|
caps32.min_fragment_size = caps.min_fragment_size;
|
|
caps32.num_codecs = caps.num_codecs;
|
|
for (i = 0; i < caps.num_codecs; i++)
|
|
caps32.codecs[i] = caps.codecs[i];
|
|
if (copy_to_user(arg, &caps32, sizeof(caps32))) {
|
|
pr_err("%s: copytouser failed COMPRESS_GETCAPS32\n",
|
|
__func__);
|
|
err = -EFAULT;
|
|
}
|
|
break;
|
|
}
|
|
case SNDRV_COMPRESS_SET_PARAMS32: {
|
|
struct snd_compr_params32 params32;
|
|
struct snd_compr_params params;
|
|
memset(¶ms32, 0 , sizeof(params32));
|
|
memset(¶ms, 0 , sizeof(params));
|
|
cmd = SNDRV_COMPRESS_SET_PARAMS;
|
|
if (copy_from_user(¶ms32, arg, sizeof(params32))) {
|
|
pr_err("%s: copyfromuser failed SET_PARAMS32\n",
|
|
__func__);
|
|
err = -EFAULT;
|
|
goto bail_out;
|
|
}
|
|
params.no_wake_mode = params32.no_wake_mode;
|
|
params.codec.id = params32.codec.id;
|
|
params.codec.ch_in = params32.codec.ch_in;
|
|
params.codec.ch_out = params32.codec.ch_out;
|
|
params.codec.sample_rate = params32.codec.sample_rate;
|
|
params.codec.bit_rate = params32.codec.bit_rate;
|
|
params.codec.rate_control = params32.codec.rate_control;
|
|
params.codec.profile = params32.codec.profile;
|
|
params.codec.level = params32.codec.level;
|
|
params.codec.ch_mode = params32.codec.ch_mode;
|
|
params.codec.format = params32.codec.format;
|
|
params.codec.align = params32.codec.align;
|
|
|
|
switch (params.codec.id) {
|
|
case SND_AUDIOCODEC_WMA:
|
|
case SND_AUDIOCODEC_WMA_PRO:
|
|
params.codec.options.wma.encodeopt1 =
|
|
params32.codec.options.wma.encodeopt1;
|
|
params.codec.options.wma.encodeopt2 =
|
|
params32.codec.options.wma.encodeopt2;
|
|
params.codec.options.wma.super_block_align =
|
|
params32.codec.options.wma.super_block_align;
|
|
break;
|
|
case SND_AUDIOCODEC_VORBIS:
|
|
params.codec.options.vorbis.downmix =
|
|
params32.codec.options.vorbis.downmix;
|
|
params.codec.options.vorbis.managed =
|
|
params32.codec.options.vorbis.managed;
|
|
params.codec.options.vorbis.max_bit_rate =
|
|
params32.codec.options.vorbis.max_bit_rate;
|
|
params.codec.options.vorbis.min_bit_rate =
|
|
params32.codec.options.vorbis.min_bit_rate;
|
|
params.codec.options.vorbis.quality =
|
|
params32.codec.options.vorbis.quality;
|
|
break;
|
|
case SND_AUDIOCODEC_REAL:
|
|
params.codec.options.real.num_regions =
|
|
params32.codec.options.real.num_regions;
|
|
params.codec.options.real.quant_bits =
|
|
params32.codec.options.real.quant_bits;
|
|
params.codec.options.real.start_region =
|
|
params32.codec.options.real.start_region;
|
|
break;
|
|
case SND_AUDIOCODEC_FLAC:
|
|
params.codec.options.flac.gain =
|
|
params32.codec.options.flac.gain;
|
|
params.codec.options.flac.num =
|
|
params32.codec.options.flac.num;
|
|
break;
|
|
case SND_AUDIOCODEC_DTS:
|
|
case SND_AUDIOCODEC_DTS_PASS_THROUGH:
|
|
case SND_AUDIOCODEC_DTS_LBR:
|
|
case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
|
|
case SND_AUDIOCODEC_DTS_TRANSCODE_LOOPBACK:
|
|
break;
|
|
case SND_AUDIOCODEC_AC3:
|
|
case SND_AUDIOCODEC_EAC3:
|
|
params.codec.options.ddp.params_length =
|
|
params32.codec.options.ddp.params_length;
|
|
memcpy(params.codec.options.ddp.params_value,
|
|
params32.codec.options.ddp.params_value,
|
|
sizeof(params32.codec.options.ddp.params_value));
|
|
memcpy(params.codec.options.ddp.params_id,
|
|
params32.codec.options.ddp.params_id,
|
|
sizeof(params32.codec.options.ddp.params_id));
|
|
break;
|
|
default:
|
|
params.codec.options.generic.bw =
|
|
params32.codec.options.generic.bw;
|
|
break;
|
|
}
|
|
if (!err)
|
|
err = msm_compr_ioctl_shared(substream, cmd, ¶ms);
|
|
break;
|
|
}
|
|
default:
|
|
err = msm_compr_ioctl_shared(substream, cmd, arg);
|
|
}
|
|
bail_out:
|
|
return err;
|
|
|
|
}
|
|
#endif
|
|
static int msm_compr_ioctl(struct snd_pcm_substream *substream,
|
|
unsigned int cmd, void *arg)
|
|
{
|
|
int err = 0;
|
|
if (!substream) {
|
|
pr_err("%s: Invalid params\n", __func__);
|
|
return -EINVAL;
|
|
}
|
|
pr_debug("%s called with cmd = %d\n", __func__, cmd);
|
|
switch (cmd) {
|
|
case SNDRV_COMPRESS_TSTAMP: {
|
|
struct snd_compr_tstamp tstamp;
|
|
if (!arg) {
|
|
pr_err("%s: Invalid params Tstamp\n", __func__);
|
|
return -EINVAL;
|
|
}
|
|
err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
|
|
if (err)
|
|
pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
|
|
__func__, err);
|
|
if (!err && copy_to_user(arg, &tstamp, sizeof(tstamp))) {
|
|
pr_err("%s: copytouser failed COMPRESS_TSTAMP\n",
|
|
__func__);
|
|
err = -EFAULT;
|
|
}
|
|
break;
|
|
}
|
|
case SNDRV_COMPRESS_GET_CAPS: {
|
|
struct snd_compr_caps cap;
|
|
if (!arg) {
|
|
pr_err("%s: Invalid params getcaps\n", __func__);
|
|
return -EINVAL;
|
|
}
|
|
pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
|
|
err = msm_compr_ioctl_shared(substream, cmd, &cap);
|
|
if (err)
|
|
pr_err("%s: GET_CAPS failed rc %d\n",
|
|
__func__, err);
|
|
if (!err && copy_to_user(arg, &cap, sizeof(cap))) {
|
|
pr_err("%s: copytouser failed GET_CAPS\n",
|
|
__func__);
|
|
err = -EFAULT;
|
|
}
|
|
break;
|
|
}
|
|
case SNDRV_COMPRESS_SET_PARAMS: {
|
|
struct snd_compr_params params;
|
|
if (!arg) {
|
|
pr_err("%s: Invalid params setparam\n", __func__);
|
|
return -EINVAL;
|
|
}
|
|
if (copy_from_user(¶ms, arg,
|
|
sizeof(struct snd_compr_params))) {
|
|
pr_err("%s: SET_PARAMS\n", __func__);
|
|
return -EFAULT;
|
|
}
|
|
err = msm_compr_ioctl_shared(substream, cmd, ¶ms);
|
|
if (err)
|
|
pr_err("%s: SET_PARAMS failed rc %d\n",
|
|
__func__, err);
|
|
break;
|
|
}
|
|
default:
|
|
err = msm_compr_ioctl_shared(substream, cmd, arg);
|
|
}
|
|
return err;
|
|
}
|
|
|
|
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_debug("%s, trigger restart\n", __func__);
|
|
|
|
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);
|
|
|
|
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 = prtd->session_id;
|
|
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_volume_ctl_put(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
int rc = 0;
|
|
struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
|
|
struct snd_pcm_substream *substream =
|
|
vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
|
|
struct msm_audio *prtd;
|
|
int volume = ucontrol->value.integer.value[0];
|
|
|
|
pr_debug("%s: volume : %x\n", __func__, volume);
|
|
if (!substream)
|
|
return -ENODEV;
|
|
if (!substream->runtime)
|
|
return 0;
|
|
prtd = substream->runtime->private_data;
|
|
if (prtd)
|
|
rc = compressed_set_volume(prtd, volume);
|
|
|
|
return rc;
|
|
}
|
|
|
|
static int msm_compr_volume_ctl_get(struct snd_kcontrol *kcontrol,
|
|
struct snd_ctl_elem_value *ucontrol)
|
|
{
|
|
struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
|
|
struct snd_pcm_substream *substream =
|
|
vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
|
|
struct msm_audio *prtd;
|
|
|
|
pr_debug("%s\n", __func__);
|
|
if (!substream)
|
|
return -ENODEV;
|
|
if (!substream->runtime)
|
|
return 0;
|
|
prtd = substream->runtime->private_data;
|
|
if (prtd)
|
|
ucontrol->value.integer.value[0] = prtd->volume;
|
|
return 0;
|
|
}
|
|
|
|
static int msm_compr_add_controls(struct snd_soc_pcm_runtime *rtd)
|
|
{
|
|
int ret = 0;
|
|
struct snd_pcm *pcm = rtd->pcm;
|
|
struct snd_pcm_volume *volume_info;
|
|
struct snd_kcontrol *kctl;
|
|
|
|
dev_dbg(rtd->dev, "%s, Volume cntrl add\n", __func__);
|
|
ret = snd_pcm_add_volume_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
|
|
NULL, 1, rtd->dai_link->be_id,
|
|
&volume_info);
|
|
if (ret < 0)
|
|
return ret;
|
|
kctl = volume_info->kctl;
|
|
kctl->put = msm_compr_volume_ctl_put;
|
|
kctl->get = msm_compr_volume_ctl_get;
|
|
kctl->tlv.p = compr_rx_vol_gain;
|
|
return 0;
|
|
}
|
|
|
|
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,
|
|
#ifdef CONFIG_COMPAT
|
|
.compat_ioctl = msm_compr_compat_ioctl,
|
|
#endif
|
|
};
|
|
|
|
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);
|
|
|
|
ret = msm_compr_add_controls(rtd);
|
|
if (ret)
|
|
pr_err("%s, kctl add failed\n", __func__);
|
|
return ret;
|
|
}
|
|
|
|
static struct snd_soc_platform_driver msm_soc_platform = {
|
|
.ops = &msm_compr_ops,
|
|
.pcm_new = msm_asoc_pcm_new,
|
|
};
|
|
|
|
static int msm_compr_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_compr_remove(struct platform_device *pdev)
|
|
{
|
|
snd_soc_unregister_platform(&pdev->dev);
|
|
return 0;
|
|
}
|
|
|
|
static const struct of_device_id msm_compr_dt_match[] = {
|
|
{.compatible = "qcom,msm-compr-dsp"},
|
|
{}
|
|
};
|
|
MODULE_DEVICE_TABLE(of, msm_compr_dt_match);
|
|
|
|
static struct platform_driver msm_compr_driver = {
|
|
.driver = {
|
|
.name = "msm-compr-dsp",
|
|
.owner = THIS_MODULE,
|
|
.of_match_table = msm_compr_dt_match,
|
|
},
|
|
.probe = msm_compr_probe,
|
|
.remove = 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");
|