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ANDROID: ASoC: core - add hostless DAI support

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Allow DAI's to be hostless so that no PCM data is sent between DAI
and CPU. This allows for power savings as there is no DMA or CPU
interaction required.

BUG: 144369166
Change-Id: I8947f1ad2c4a7013e92e21078b35e3cad332cf6f
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Meng Wang <mwang@codeaurora.org>
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
Signed-off-by: Neema Shetty <nshetty@codeaurora.org>
Signed-off-by: Vidyakumar Athota <vathota@codeaurora.org>
Signed-off-by: Anish Kumar <kanish@codeaurora.org>
Signed-off-by: Shiv Maliyappanahalli <smaliyap@codeaurora.org>
Signed-off-by: Sudheer Papothi <spapothi@codeaurora.org>
This commit is contained in:
Liam Girdwood 2017-01-10 17:10:17 -08:00 committed by Todd Kjos
parent 934b595640
commit 7fe5afb291
5 changed files with 86 additions and 0 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
include/sound/pcm.h
View File

@ -475,6 +475,7 @@ struct snd_pcm_substream {
#endif /* CONFIG_SND_VERBOSE_PROCFS */
/* misc flags */
unsigned int hw_opened: 1;
unsigned int hw_no_buffer: 1; /* substream may not have a buffer */
};
#define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0)

11
include/sound/soc.h
View File

@ -370,6 +370,10 @@
#define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
/* DAI Link Host Mode Support */
#define SND_SOC_DAI_LINK_NO_HOST 0x1
#define SND_SOC_DAI_LINK_OPT_HOST 0x2
/*
* Bias levels
*
@ -723,6 +727,7 @@ struct snd_soc_pcm_stream {
unsigned int channels_min; /* min channels */
unsigned int channels_max; /* max channels */
unsigned int sig_bits; /* number of bits of content */
const char *aif_name; /* DAPM AIF widget name */
};
/* SoC audio ops */
@ -837,6 +842,12 @@ struct snd_soc_dai_link {
/* This DAI link can route to other DAI links at runtime (Frontend)*/
unsigned int dynamic:1;
/*
* This DAI can support no host IO (no pcm data is
* copied to from host)
*/
unsigned int no_host_mode:2;
/* DPCM capture and Playback support */
unsigned int dpcm_capture:1;
unsigned int dpcm_playback:1;

2
sound/core/pcm_lib.c
View File

@ -2050,6 +2050,8 @@ static int pcm_sanity_check(struct snd_pcm_substream *substream)
struct snd_pcm_runtime *runtime;
if (PCM_RUNTIME_CHECK(substream))
return -ENXIO;
if (substream->hw_no_buffer)
snd_printd("%s: warning this PCM is host less\n", __func__);
runtime = substream->runtime;
if (snd_BUG_ON(!substream->ops->copy_user && !runtime->dma_area))
return -EINVAL;

1
sound/core/pcm_native.c
View File

@ -1199,6 +1199,7 @@ static int snd_pcm_pre_start(struct snd_pcm_substream *substream, int state)
if (runtime->status->state != SNDRV_PCM_STATE_PREPARED)
return -EBADFD;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
!substream->hw_no_buffer &&
!snd_pcm_playback_data(substream))
return -EPIPE;
runtime->trigger_tstamp_latched = false;

71
sound/soc/soc-pcm.c
View File

@ -19,6 +19,7 @@
#include <linux/workqueue.h>
#include <linux/export.h>
#include <linux/debugfs.h>
#include <linux/dma-mapping.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
@ -28,6 +29,27 @@
#define DPCM_MAX_BE_USERS 8
/* ASoC no host IO hardware */
static const struct snd_pcm_hardware no_host_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S32_LE,
.period_bytes_min = PAGE_SIZE >> 2,
.period_bytes_max = PAGE_SIZE >> 1,
.periods_min = 2,
.periods_max = 4,
/*
* Increase the max buffer bytes as PAGE_SIZE bytes is
* not enough to encompass all the scenarios sent by
* userspapce.
*/
.buffer_bytes_max = PAGE_SIZE * 4,
};
/**
* snd_soc_runtime_activate() - Increment active count for PCM runtime components
* @rtd: ASoC PCM runtime that is activated
@ -138,6 +160,8 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
const struct snd_pcm_hardware *hw)
{
struct snd_pcm_runtime *runtime = substream->runtime;
if (!runtime)
return 0;
runtime->hw.info = hw->info;
runtime->hw.formats = hw->formats;
runtime->hw.period_bytes_min = hw->period_bytes_min;
@ -508,6 +532,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
if (rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST)
snd_soc_set_runtime_hwparams(substream, &no_host_hardware);
/* startup the audio subsystem */
ret = snd_soc_dai_startup(cpu_dai, substream);
if (ret < 0) {
@ -877,6 +904,16 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
int i, ret = 0;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
/* perform any hw_params fixups */
if ((rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST) &&
rtd->dai_link->be_hw_params_fixup) {
ret = rtd->dai_link->be_hw_params_fixup(rtd,
params);
if (ret < 0)
dev_err(rtd->card->dev, "ASoC: fixup failed for %s\n",
rtd->dai_link->name);
}
if (rtd->dai_link->ops->hw_params) {
ret = rtd->dai_link->ops->hw_params(substream, params);
if (ret < 0) {
@ -961,6 +998,21 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
ret = soc_pcm_params_symmetry(substream, params);
if (ret)
goto component_err;
/* malloc a page for hostless IO */
if (rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST) {
substream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV;
substream->dma_buffer.dev.dev = rtd->dev;
substream->dma_buffer.dev.dev->coherent_dma_mask =
DMA_BIT_MASK(sizeof(dma_addr_t) * 8);
substream->dma_buffer.private_data = NULL;
arch_setup_dma_ops(substream->dma_buffer.dev.dev,
0, 0, NULL, 0);
ret = snd_pcm_lib_malloc_pages(substream, PAGE_SIZE);
if (ret < 0)
goto component_err;
}
out:
mutex_unlock(&rtd->card->pcm_mutex);
return ret;
@ -1043,6 +1095,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
snd_soc_dai_hw_free(cpu_dai, substream);
if (rtd->dai_link->no_host_mode == SND_SOC_DAI_LINK_NO_HOST)
snd_pcm_lib_free_pages(substream);
mutex_unlock(&rtd->card->pcm_mutex);
return 0;
}
@ -2825,6 +2879,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_rtdcom_list *rtdcom;
struct snd_pcm *pcm;
struct snd_pcm_str *stream;
char new_name[64];
int ret = 0, playback = 0, capture = 0;
int i;
@ -2914,6 +2969,22 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
goto out;
}
/* setup any hostless PCMs - i.e. no host IO is performed */
if (rtd->dai_link->no_host_mode) {
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
stream = &pcm->streams[SNDRV_PCM_STREAM_PLAYBACK];
stream->substream->hw_no_buffer = 1;
snd_soc_set_runtime_hwparams(stream->substream,
&no_host_hardware);
}
if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
stream = &pcm->streams[SNDRV_PCM_STREAM_CAPTURE];
stream->substream->hw_no_buffer = 1;
snd_soc_set_runtime_hwparams(stream->substream,
&no_host_hardware);
}
}
/* ASoC PCM operations */
if (rtd->dai_link->dynamic) {
rtd->ops.open = dpcm_fe_dai_open;