[PATCH 000/113] ALSA: add snd_pcm_is_playback/capture() macro
Hi ALSA-ML Cc Staging-ML, USB-ML
Many drivers are using below code to know the Sound direction.
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
This patch-set add snd_pcm_is_playback/capture() macro to handle it.
Thank you for your help !!
Best regards --- Kuninori Morimoto
Many drivers are using below code to know the direction.
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
Add snd_pcm_is_playback/capture() macro to handle it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/pcm.h | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index ac8f3aef92052..69e535aeb8e82 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -502,6 +502,35 @@ struct snd_pcm_substream {
#define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0)
+static inline int snd_pcm_direction_is_playback(const int stream) +{ + return stream == SNDRV_PCM_STREAM_PLAYBACK; +} + +static inline int snd_pcm_direction_is_capture(const int stream) +{ + return stream == SNDRV_PCM_STREAM_CAPTURE; +} + +static inline int snd_pcm_substream_is_playback(const struct snd_pcm_substream *substream) +{ + return snd_pcm_direction_is_playback(substream->stream); +} + +static inline int snd_pcm_substream_is_capture(const struct snd_pcm_substream *substream) +{ + return snd_pcm_direction_is_capture(substream->stream); +} + +#define snd_pcm_is_playback(x) _Generic((x), \ + struct snd_pcm_substream *: snd_pcm_substream_is_playback, \ + const struct snd_pcm_substream *: snd_pcm_substream_is_playback, \ + default : snd_pcm_direction_is_playback)(x) + +#define snd_pcm_is_capture(x) _Generic((x), \ + struct snd_pcm_substream *: snd_pcm_substream_is_capture, \ + const struct snd_pcm_substream *: snd_pcm_substream_is_capture, \ + default : snd_pcm_direction_is_capture)(x)
struct snd_pcm_str { int stream; /* stream (direction) */
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/slimbus/stream.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 863ab3075d7eb..e65a6bda2f5ba 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -6,6 +6,7 @@ #include <linux/slab.h> #include <linux/list.h> #include <linux/slimbus.h> +#include <sound/pcm.h> #include <uapi/sound/asound.h> #include "slimbus.h"
@@ -235,7 +236,7 @@ int slim_stream_prepare(struct slim_stream_runtime *rt, * data rate not exactly multiple of super frame, * use PUSH/PULL protocol */ - if (cfg->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(cfg->direction)) rt->prot = SLIM_PROTO_PUSH; else rt->prot = SLIM_PROTO_PULL; @@ -256,7 +257,7 @@ int slim_stream_prepare(struct slim_stream_runtime *rt, port->ch.aux_fmt = SLIM_CH_AUX_FMT_NOT_APPLICABLE; port->ch.state = SLIM_CH_STATE_ALLOCATED;
- if (cfg->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(cfg->direction)) port->direction = SLIM_PORT_SINK; else port->direction = SLIM_PORT_SOURCE;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/staging/greybus/audio_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/drivers/staging/greybus/audio_codec.c b/drivers/staging/greybus/audio_codec.c index 2f05e761fb9ad..3042a486c0c51 100644 --- a/drivers/staging/greybus/audio_codec.c +++ b/drivers/staging/greybus/audio_codec.c @@ -478,7 +478,7 @@ static int gbcodec_hw_params(struct snd_pcm_substream *substream,
gb_pm_runtime_put_noidle(bundle);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) sig_bits = dai->driver->playback.sig_bits; else sig_bits = dai->driver->capture.sig_bits;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/soundwire/amd_manager.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 0d01849c35861..6f9e075b49979 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -612,7 +612,7 @@ static int amd_sdw_hw_params(struct snd_pcm_substream *substream, return -EIO;
ch = params_channels(params); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) dir = SDW_DATA_DIR_RX; else dir = SDW_DATA_DIR_TX;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e6..0b86ee6a4d77c 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1168,7 +1168,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, int maxport, pn, nports = 0, ret = 0; unsigned int m_port;
- if (direction == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(direction)) sconfig.direction = SDW_DATA_DIR_TX; else sconfig.direction = SDW_DATA_DIR_RX;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/soundwire/intel.c | 4 ++-- drivers/soundwire/intel_ace2x.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 421da0f86fad6..17bad341f2336 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -734,7 +734,7 @@ static int intel_hw_params(struct snd_pcm_substream *substream, return -EIO;
ch = params_channels(params); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) dir = SDW_DATA_DIR_RX; else dir = SDW_DATA_DIR_TX; @@ -819,7 +819,7 @@ static int intel_prepare(struct snd_pcm_substream *substream,
/* configure stream */ ch = params_channels(hw_params); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) dir = SDW_DATA_DIR_RX; else dir = SDW_DATA_DIR_TX; diff --git a/drivers/soundwire/intel_ace2x.c b/drivers/soundwire/intel_ace2x.c index 781fe0aefa68f..b1cc9041ff917 100644 --- a/drivers/soundwire/intel_ace2x.c +++ b/drivers/soundwire/intel_ace2x.c @@ -319,7 +319,7 @@ static int intel_hw_params(struct snd_pcm_substream *substream, return -EIO;
ch = params_channels(params); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) dir = SDW_DATA_DIR_RX; else dir = SDW_DATA_DIR_TX; @@ -407,7 +407,7 @@ static int intel_prepare(struct snd_pcm_substream *substream,
/* configure stream */ ch = params_channels(hw_params); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) dir = SDW_DATA_DIR_RX; else dir = SDW_DATA_DIR_TX;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/soundwire/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 7aa4900dcf317..d471506327579 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1781,7 +1781,7 @@ int sdw_startup_stream(void *sdw_substream) char *name; int ret;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) name = kasprintf(GFP_KERNEL, "%s-Playback", substream->name); else name = kasprintf(GFP_KERNEL, "%s-Capture", substream->name);
We have for_each_pcm_streams() macro, let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/usb/gadget/function/u_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/drivers/usb/gadget/function/u_audio.c b/drivers/usb/gadget/function/u_audio.c index 89af0feb75120..40093fa1093d3 100644 --- a/drivers/usb/gadget/function/u_audio.c +++ b/drivers/usb/gadget/function/u_audio.c @@ -1290,7 +1290,7 @@ int g_audio_setup(struct g_audio *g_audio, const char *pcm_name, goto snd_fail; }
- for (i = 0; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_streams(i) { struct uac_rtd_params *prm; struct uac_fu_params *fu; char ctrl_name[24];
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- drivers/usb/gadget/function/u_audio.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-)
diff --git a/drivers/usb/gadget/function/u_audio.c b/drivers/usb/gadget/function/u_audio.c index 40093fa1093d3..09fbae35aea25 100644 --- a/drivers/usb/gadget/function/u_audio.c +++ b/drivers/usb/gadget/function/u_audio.c @@ -190,7 +190,7 @@ static void u_audio_iso_complete(struct usb_ep *ep, struct usb_request *req) goto exit; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* * For each IN packet, take the quotient of the current data * rate and the endpoint's interval as the base packet size. @@ -244,7 +244,7 @@ static void u_audio_iso_complete(struct usb_ep *ep, struct usb_request *req) /* Pack USB load in ALSA ring buffer */ pending = runtime->dma_bytes - hw_ptr;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (unlikely(pending < req->actual)) { memcpy(req->buf, runtime->dma_area + hw_ptr, pending); memcpy(req->buf + pending, runtime->dma_area, @@ -322,7 +322,7 @@ static int uac_pcm_trigger(struct snd_pcm_substream *substream, int cmd) audio_dev = uac->audio_dev; params = &audio_dev->params;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) prm = &uac->p_prm; else prm = &uac->c_prm; @@ -344,7 +344,7 @@ static int uac_pcm_trigger(struct snd_pcm_substream *substream, int cmd) }
/* Clear buffer after Play stops */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !prm->ss) + if (snd_pcm_is_playback(substream) && !prm->ss) memset(prm->rbuf, 0, prm->max_psize * params->req_number);
return err; @@ -355,7 +355,7 @@ static snd_pcm_uframes_t uac_pcm_pointer(struct snd_pcm_substream *substream) struct snd_uac_chip *uac = snd_pcm_substream_chip(substream); struct uac_rtd_params *prm;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) prm = &uac->p_prm; else prm = &uac->c_prm; @@ -402,7 +402,7 @@ static int uac_pcm_open(struct snd_pcm_substream *substream)
runtime->hw = uac_pcm_hardware;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { runtime->hw.formats = uac_ssize_to_fmt(p_ssize); runtime->hw.channels_min = num_channels(p_chmask); prm = &uac->p_prm; @@ -1299,7 +1299,7 @@ int g_audio_setup(struct g_audio *g_audio, const char *pcm_name, if (!pcm->streams[i].substream_count) continue;
- if (i == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(i)) { prm = &uac->p_prm; fu = ¶ms->p_fu; direction = "Playback";
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/isa/sb/sb16_main.c | 4 ++-- sound/isa/sb/sb8_main.c | 6 +++--- sound/isa/wss/wss_lib.c | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index a9b87e159b2d1..7145dcb4417a5 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -216,7 +216,7 @@ static void snd_sb16_setup_rate(struct snd_sb *chip, unsigned long flags;
spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->mode & (channel == SNDRV_PCM_STREAM_PLAYBACK ? SB_MODE_PLAYBACK_16 : SB_MODE_CAPTURE_16)) + if (chip->mode & (snd_pcm_is_playback(channel) ? SB_MODE_PLAYBACK_16 : SB_MODE_CAPTURE_16)) snd_sb_ack_16bit(chip); else snd_sb_ack_8bit(chip); @@ -860,7 +860,7 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device)
const struct snd_pcm_ops *snd_sb16dsp_get_pcm_ops(int direction) { - return direction == SNDRV_PCM_STREAM_PLAYBACK ? + return snd_pcm_is_playback(direction) ? &snd_sb16_playback_ops : &snd_sb16_capture_ops; }
diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 2ed176a5a5743..dbb08e9e0f367 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -473,7 +473,7 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } chip->open |= SB_OPEN_PCM; spin_unlock_irqrestore(&chip->open_lock, flags); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { chip->playback_substream = substream; runtime->hw = snd_sb8_playback; } else { @@ -501,7 +501,7 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, -1); break; case SB_HW_201: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { runtime->hw.rate_max = 44100; } else { runtime->hw.rate_max = 15000; @@ -532,7 +532,7 @@ static int snd_sb8_close(struct snd_pcm_substream *substream) chip->capture_substream = NULL; spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~SB_OPEN_PCM; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) chip->mode &= ~SB_MODE_PLAYBACK; else chip->mode &= ~SB_MODE_CAPTURE; diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 026061b55ee94..d3f1a80c3a574 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -2196,7 +2196,7 @@ EXPORT_SYMBOL(snd_wss_mixer);
const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction) { - return direction == SNDRV_PCM_STREAM_PLAYBACK ? + return snd_pcm_is_playback(direction) ? &snd_wss_playback_ops : &snd_wss_capture_ops; } EXPORT_SYMBOL(snd_wss_get_pcm_ops);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/arm/aaci.c | 4 ++-- sound/arm/pxa2xx-ac97.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index c3340b8ff3daf..5542c18cf2191 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -403,7 +403,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) struct aaci_runtime *aacirun; int ret = 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { aacirun = &aaci->playback; } else { aacirun = &aaci->capture; @@ -415,7 +415,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates = aacirun->pcm->rates; snd_pcm_limit_hw_rates(runtime);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { runtime->hw.channels_max = 6;
/* Add rule describing channel dependency. */ diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4c367e73b2c9b..bce0c7ca4b707 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -71,7 +71,7 @@ static int pxa2xx_ac97_pcm_open(struct snd_pcm_substream *substream) runtime->hw.channels_min = 2; runtime->hw.channels_max = 2;
- i = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + i = (snd_pcm_is_playback(substream)) ? AC97_RATES_FRONT_DAC : AC97_RATES_ADC; runtime->hw.rates = pxa2xx_ac97_ac97->rates[i]; snd_pcm_limit_hw_rates(runtime); @@ -100,7 +100,7 @@ static int pxa2xx_ac97_pcm_close(struct snd_pcm_substream *substream) static int pxa2xx_ac97_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + int reg = snd_pcm_is_playback(substream) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; int ret;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/xen/xen_snd_front_alsa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index b229eb6f70571..0f4f849bd20b3 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -240,7 +240,7 @@ stream_get(struct snd_pcm_substream *substream) snd_pcm_substream_chip(substream); struct xen_snd_front_pcm_stream_info *stream;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) stream = &pcm_instance->streams_pb[substream->number]; else stream = &pcm_instance->streams_cap[substream->number];
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/usb/6fire/pcm.c | 8 ++++---- sound/usb/caiaq/audio.c | 14 +++++++------- sound/usb/hiface/pcm.c | 4 ++-- sound/usb/line6/pcm.c | 16 ++++++++-------- sound/usb/media.c | 2 +- sound/usb/pcm.c | 20 ++++++++++---------- sound/usb/quirks.c | 6 +++--- sound/usb/stream.c | 6 +++--- sound/usb/usx2y/usbusx2yaudio.c | 2 +- sound/usb/usx2y/usx2yhwdeppcm.c | 6 +++--- 10 files changed, 42 insertions(+), 42 deletions(-)
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 32c39d8bd2e55..fa3dfceab11cc 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -119,9 +119,9 @@ static struct pcm_substream *usb6fire_pcm_get_substream( { struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub);
- if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(alsa_sub)) return &rt->playback; - else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(alsa_sub)) return &rt->capture; dev_err(&rt->chip->dev->dev, "error getting pcm substream slot.\n"); return NULL; @@ -395,12 +395,12 @@ static int usb6fire_pcm_open(struct snd_pcm_substream *alsa_sub) mutex_lock(&rt->stream_mutex); alsa_rt->hw = pcm_hw;
- if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(alsa_sub)) { if (rt->rate < ARRAY_SIZE(rates)) alsa_rt->hw.rates = rates_alsaid[rt->rate]; alsa_rt->hw.channels_max = OUT_N_CHANNELS; sub = &rt->playback; - } else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) { + } else if (snd_pcm_is_capture(alsa_sub)) { if (rt->rate < ARRAY_SIZE(rates)) alsa_rt->hw.rates = rates_alsaid[rt->rate]; alsa_rt->hw.channels_max = IN_N_CHANNELS; diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4981753652a7f..6996c5e07095f 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -53,7 +53,7 @@ activate_substream(struct snd_usb_caiaqdev *cdev, { spin_lock(&cdev->spinlock);
- if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(sub)) cdev->sub_playback[sub->number] = sub; else cdev->sub_capture[sub->number] = sub; @@ -68,7 +68,7 @@ deactivate_substream(struct snd_usb_caiaqdev *cdev, unsigned long flags; spin_lock_irqsave(&cdev->spinlock, flags);
- if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(sub)) cdev->sub_playback[sub->number] = NULL; else cdev->sub_capture[sub->number] = NULL; @@ -192,7 +192,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
dev_dbg(dev, "%s(%p)\n", __func__, substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { int out_pos;
switch (cdev->spec.data_alignment) { @@ -305,7 +305,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) goto unlock; }
- if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(sub)) ptr = bytes_to_frames(sub->runtime, cdev->audio_out_buf_pos[index]); else @@ -339,7 +339,7 @@ static void check_for_elapsed_periods(struct snd_usb_caiaqdev *cdev, continue;
pb = snd_pcm_lib_period_bytes(sub); - cnt = (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + cnt = snd_pcm_is_playback(sub) ? &cdev->period_out_count[stream] : &cdev->period_in_count[stream];
@@ -701,7 +701,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) struct usb_device *usb_dev = cdev->chip.dev; unsigned int pipe;
- pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = snd_pcm_is_playback(dir) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE);
@@ -741,7 +741,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *cdev, int dir, int *ret) urbs[i]->context = &cdev->data_cb_info[i]; urbs[i]->interval = 1; urbs[i]->number_of_packets = FRAMES_PER_URB; - urbs[i]->complete = (dir == SNDRV_PCM_STREAM_CAPTURE) ? + urbs[i]->complete = snd_pcm_is_capture(dir) ? read_completed : write_completed; }
diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index cf650fab54d7e..178f70fd64d51 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -172,7 +172,7 @@ static struct pcm_substream *hiface_pcm_get_substream(struct snd_pcm_substream struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); struct device *device = &rt->chip->dev->dev;
- if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(alsa_sub)) return &rt->playback;
dev_err(device, "Error getting pcm substream slot.\n"); @@ -359,7 +359,7 @@ static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) mutex_lock(&rt->stream_mutex); alsa_rt->hw = pcm_hw;
- if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(alsa_sub)) sub = &rt->playback;
if (!sub) { diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 6a4af725aedd2..db1d93de327fd 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -137,7 +137,7 @@ static void line6_wait_clear_audio_urbs(struct snd_line6_pcm *line6pcm, static inline struct line6_pcm_stream * get_stream(struct snd_line6_pcm *line6pcm, int direction) { - return (direction == SNDRV_PCM_STREAM_PLAYBACK) ? + return (snd_pcm_is_playback(direction)) ? &line6pcm->out : &line6pcm->in; }
@@ -148,7 +148,7 @@ static int line6_buffer_acquire(struct snd_line6_pcm *line6pcm, struct line6_pcm_stream *pstr, int direction, int type) { const int pkt_size = - (direction == SNDRV_PCM_STREAM_PLAYBACK) ? + snd_pcm_is_playback(direction) ? line6pcm->max_packet_size_out : line6pcm->max_packet_size_in;
@@ -191,7 +191,7 @@ static int line6_stream_start(struct snd_line6_pcm *line6pcm, int direction, !(pstr->active_urbs || pstr->unlink_urbs)) { pstr->count = 0; /* Submit all currently available URBs */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(direction)) ret = line6_submit_audio_out_all_urbs(line6pcm); else ret = line6_submit_audio_in_all_urbs(line6pcm); @@ -216,7 +216,7 @@ static void line6_stream_stop(struct snd_line6_pcm *line6pcm, int direction, spin_unlock_irqrestore(&pstr->lock, flags); line6_unlink_audio_urbs(line6pcm, pstr); spin_lock_irqsave(&pstr->lock, flags); - if (direction == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(direction)) { line6pcm->prev_fbuf = NULL; line6pcm->prev_fsize = 0; } @@ -240,7 +240,7 @@ int snd_line6_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - if (s->stream == SNDRV_PCM_STREAM_CAPTURE && + if (snd_pcm_is_capture(s) && (line6pcm->line6->properties->capabilities & LINE6_CAP_IN_NEEDS_OUT)) { err = line6_stream_start(line6pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -256,7 +256,7 @@ int snd_line6_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - if (s->stream == SNDRV_PCM_STREAM_CAPTURE && + if (snd_pcm_is_capture(s) && (line6pcm->line6->properties->capabilities & LINE6_CAP_IN_NEEDS_OUT)) { line6_stream_stop(line6pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -267,13 +267,13 @@ int snd_line6_trigger(struct snd_pcm_substream *substream, int cmd) break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (s->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(s)) return -EINVAL; set_bit(LINE6_FLAG_PAUSE_PLAYBACK, &line6pcm->flags); break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (s->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(s)) return -EINVAL; clear_bit(LINE6_FLAG_PAUSE_PLAYBACK, &line6pcm->flags); break; diff --git a/sound/usb/media.c b/sound/usb/media.c index d48db6f3ae659..3870df7214e53 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -54,7 +54,7 @@ int snd_media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, return -ENOMEM;
mctl->media_dev = mdev; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 08bf535ed1632..18a0b55e4159f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -37,7 +37,7 @@ static snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, int est_delay; int queued;
- if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(subs->direction)) { queued = bytes_to_frames(runtime, subs->inflight_bytes); if (!queued) return 0; @@ -57,7 +57,7 @@ static snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, some truncation for 44.1 but the estimate is good enough */ est_delay = frame_diff * runtime->rate / 1000;
- if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(subs->direction)) { est_delay = queued - est_delay; if (est_delay < 0) est_delay = 0; @@ -126,14 +126,14 @@ find_format(struct list_head *fmt_list_head, snd_pcm_format_t format, */ if (subs && attr != cur_attr) { if ((attr == USB_ENDPOINT_SYNC_ASYNC && - subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || + snd_pcm_is_playback(subs->direction)) || (attr == USB_ENDPOINT_SYNC_ADAPTIVE && - subs->direction == SNDRV_PCM_STREAM_CAPTURE)) + snd_pcm_is_capture(subs->direction))) continue; if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC && - subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || + snd_pcm_is_playback(subs->direction)) || (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE && - subs->direction == SNDRV_PCM_STREAM_CAPTURE)) { + snd_pcm_is_capture(subs->direction))) { found = fp; cur_attr = attr; continue; @@ -616,7 +616,7 @@ static int lowlatency_playback_available(struct snd_pcm_runtime *runtime, { struct snd_usb_audio *chip = subs->stream->chip;
- if (subs->direction == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(subs->direction)) return false; /* disabled via module option? */ if (!chip->lowlatency) @@ -678,7 +678,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) runtime->delay = 0;
subs->lowlatency_playback = lowlatency_playback_available(runtime, subs); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && !subs->lowlatency_playback) { ret = start_endpoints(subs); /* if XRUN happens at starting streams (possibly with implicit @@ -1212,7 +1212,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream)
runtime->hw = snd_usb_hardware; /* need an explicit sync to catch applptr update in low-latency mode */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(direction) && as->chip->lowlatency) runtime->hw.info |= SNDRV_PCM_INFO_SYNC_APPLPTR; runtime->private_data = subs; @@ -1770,7 +1770,7 @@ void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream) { const struct snd_pcm_ops *ops;
- ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + ops = snd_pcm_is_playback(stream) ? &snd_usb_playback_ops : &snd_usb_capture_ops; snd_pcm_set_ops(pcm, stream, ops); } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ea063a14cdd8f..b26d1aa12a698 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1685,7 +1685,7 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, * sample rate shouldn't be changed * by playback substream */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(subs->direction)) { if (subs->stream->substream[SNDRV_PCM_STREAM_CAPTURE].cur_audiofmt) return; } @@ -2017,7 +2017,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, * although it's really not... */ fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; else fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; @@ -2036,7 +2036,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, break; case USB_ID(0x3511, 0x2b1e): /* Opencomm2 UC USB Bluetooth dongle */ /* mic works only when ep pitch control is not set */ - if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) fp->attributes &= ~UAC_EP_CS_ATTR_PITCH_CONTROL; break; } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index e14c725acebf2..b4096a2b53210 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -524,8 +524,8 @@ static int __snd_usb_add_audio_stream(struct snd_usb_audio *chip, as->chip = chip; as->fmt_type = fp->fmt_type; err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1, + snd_pcm_is_playback(stream), + !snd_pcm_is_playback(stream), &pcm); if (err < 0) { kfree(as); @@ -1058,7 +1058,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, audioformat_free(fp); return NULL; } - pd->pd_id = (stream == SNDRV_PCM_STREAM_PLAYBACK) ? + pd->pd_id = snd_pcm_is_playback(stream) ? UAC3_BADD_PD_ID10 : UAC3_BADD_PD_ID11; pd->pd_d1d0_rec = UAC3_BADD_PD_RECOVER_D1D0; pd->pd_d2d0_rec = UAC3_BADD_PD_RECOVER_D2D0; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index ca7888495a9f4..7026e6a8f023b 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -816,7 +816,7 @@ static int snd_usx2y_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&subs->usx2y->pcm_mutex); snd_printdd("snd_usx2y_hw_free(%p)\n", substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { cap_subs = subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE]; atomic_set(&subs->state, STATE_STOPPED); usx2y_urbs_release(subs); diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 36f2e31168fb0..56111336d71f7 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -368,7 +368,7 @@ static int snd_usx2y_usbpcm_hw_free(struct snd_pcm_substream *substream) snd_printdd("%s(%p)\n", __func__, substream);
cap_subs2 = subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { cap_subs = subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE]; atomic_set(&subs->state, STATE_STOPPED); usx2y_usbpcm_urbs_release(subs); @@ -414,7 +414,7 @@ static int usx2y_usbpcm_urbs_start(struct snd_usx2y_substream *subs) struct urb *urb; unsigned long pack;
- if (stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(stream)) { usx2y->hwdep_pcm_shm->captured_iso_head = -1; usx2y->hwdep_pcm_shm->captured_iso_frames = 0; } @@ -592,7 +592,7 @@ static int snd_usx2y_usbpcm_open(struct snd_pcm_substream *substream) if (!(subs->usx2y->chip_status & USX2Y_STAT_CHIP_MMAP_PCM_URBS)) return -EBUSY;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) runtime->hw = snd_usx2y_2c; else runtime->hw = (subs->usx2y->subs[3] ? snd_usx2y_4c : snd_usx2y_2c);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/hda/hda_controller.c | 6 +++--- sound/pci/hda/hda_intel.c | 6 +++--- sound/pci/hda/patch_si3054.c | 4 ++-- 3 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 5d86e5a9c814a..9efa43afd0c16 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -75,7 +75,7 @@ static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream, codec_nsecs = div_u64(codec_frames * 1000000000LL, substream->runtime->rate);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) return nsec + codec_nsecs;
return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; @@ -385,7 +385,7 @@ static int azx_get_sync_time(ktime_t *device,
runtime = substream->runtime;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) direction = 1; else direction = 0; @@ -659,7 +659,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
/* disable LINK_ATIME timestamps for capture streams until we figure out how to handle digital inputs */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */ runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b79020adce63b..c9552b71c0e9d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -583,7 +583,7 @@ static int azx_get_delay_from_lpib(struct azx *chip, struct azx_dev *azx_dev, unsigned int lpib_pos = azx_get_pos_lpib(chip, azx_dev); int delay;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) delay = pos - lpib_pos; else delay = lpib_pos - pos; @@ -800,7 +800,7 @@ static unsigned int azx_via_get_position(struct azx *chip, unsigned int fifo_size;
link_pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev)); - if (azx_dev->core.substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(azx_dev->core.substream)) { /* Playback, no problem using link position */ return link_pos; } @@ -869,7 +869,7 @@ static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev) }
/* correct the DMA position for capture stream */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { if (pos < delay) pos += azx_dev->core.bufsize; pos -= delay; diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 763eae80a148e..9f13732a89064 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -142,8 +142,8 @@ static int si3054_pcm_prepare(struct hda_pcm_stream *hinfo,
SET_REG(codec, SI3054_LINE_RATE, substream->runtime->rate); val = GET_REG(codec, SI3054_LINE_LEVEL); - val &= 0xff << (8 * (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)); - val |= ((stream_tag & 0xf) << 4) << (8 * (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)); + val &= 0xff << (8 * (!snd_pcm_is_playback(substream))); + val |= ((stream_tag & 0xf) << 4) << (8 * (snd_pcm_is_playback(substream))); SET_REG(codec, SI3054_LINE_LEVEL, val);
snd_hda_codec_setup_stream(codec, hinfo->nid,
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/ac97/ac97_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index 5fee8e89790fb..0d29676a8ecfa 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -150,7 +150,7 @@ static unsigned char get_slot_reg(struct ac97_pcm *pcm, unsigned short cidx, return 0xff; if (pcm->spdif) return AC97_SPDIF; /* pseudo register */ - if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(pcm->stream)) return rate_reg_tables[dbl][pcm->r[dbl].rate_table[cidx]][slot - 3]; else return rate_cregs[slot - 3]; @@ -512,7 +512,7 @@ int snd_ac97_pcm_assign(struct snd_ac97_bus *bus, rpcm->rates &= rates; } /* for double rate, we check the first codec only */ - if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(pcm->stream) && bus->codec[0] && (bus->codec[0]->flags & AC97_DOUBLE_RATE) && rate_table[pcm->stream][0] == 0) { tmp = (1<<AC97_SLOT_PCM_LEFT) | (1<<AC97_SLOT_PCM_RIGHT) |
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/nm256/nm256.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 11ba7d4eac2a4..4ad3734796ed6 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -332,7 +332,7 @@ snd_nm256_load_one_coefficient(struct nm256 *chip, int stream, u32 port, int whi snd_nm256_write_buffer(chip, coefficients + offset, coeff_buf, size); snd_nm256_writel(chip, port, coeff_buf); /* ??? Record seems to behave differently than playback. */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) size--; snd_nm256_writel(chip, port + 4, coeff_buf + size); } @@ -341,11 +341,11 @@ static void snd_nm256_load_coefficient(struct nm256 *chip, int stream, int number) { /* The enable register for the specified engine. */ - u32 poffset = (stream == SNDRV_PCM_STREAM_CAPTURE ? + u32 poffset = (snd_pcm_is_capture(stream) ? NM_RECORD_ENABLE_REG : NM_PLAYBACK_ENABLE_REG); u32 addr = NM_COEFF_START_OFFSET;
- addr += (stream == SNDRV_PCM_STREAM_CAPTURE ? + addr += (snd_pcm_is_capture(stream) ? NM_RECORD_REG_OFFSET : NM_PLAYBACK_REG_OFFSET);
if (snd_nm256_readb(chip, poffset) & 1) { @@ -356,7 +356,7 @@ snd_nm256_load_coefficient(struct nm256 *chip, int stream, int number)
/* The recording engine uses coefficient values 8-15. */ number &= 7; - if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) number += 8;
if (! chip->use_cache) { @@ -372,7 +372,7 @@ snd_nm256_load_coefficient(struct nm256 *chip, int stream, int number) u32 offset = snd_nm256_get_start_offset(number); u32 end_offset = offset + coefficient_sizes[number]; snd_nm256_writel(chip, addr, base + offset); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) end_offset--; snd_nm256_writel(chip, addr + 4, base + end_offset); }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/pcxhr/pcxhr.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 242bd7e04b3e1..2de8ccb33d17e 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -626,7 +626,7 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) struct snd_pcm_substream *subs = stream->substream; struct snd_pcxhr *chip = snd_pcm_substream_chip(subs);
- is_capture = (subs->stream == SNDRV_PCM_STREAM_CAPTURE); + is_capture = snd_pcm_is_capture(subs); stream_num = is_capture ? 0 : subs->number;
dev_dbg(chip->card->dev, @@ -995,7 +995,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) /* copy the struct snd_pcm_hardware struct */ runtime->hw = pcxhr_caps;
- if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) { + if(snd_pcm_is_playback(subs)) { dev_dbg(chip->card->dev, "%s playback chip%d subs%d\n", __func__, chip->chip_idx, subs->number); stream = &chip->playback_stream[subs->number];
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/asihpi/asihpi.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 001786e2aba13..69eb696012933 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -447,7 +447,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, params_channels(params), format, params_rate(params), 0, 0));
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { if (hpi_instream_reset(dpcm->h_stream) != 0) return -EINVAL;
@@ -582,7 +582,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, continue;
ds->drained_count = 0; - if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(s)) { /* How do I know how much valid data is present * in buffer? Must be at least one period! * Guessing 2 periods, but if @@ -615,7 +615,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } /* start the master stream */ card->pcm_start(substream); - if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) || + if (snd_pcm_is_capture(substream) || !card->can_dma) hpi_handle_error(hpi_stream_start(dpcm->h_stream)); break; @@ -643,7 +643,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
/* _prepare and _hwparams reset the stream */ hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) hpi_handle_error( hpi_outstream_reset(dpcm->h_stream));
@@ -755,7 +755,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) if (!card->can_dma) on_card_bytes = bytes_avail;
- if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(s)) { pcm_buf_dma_ofs = ds->pcm_buf_host_rw_ofs - bytes_avail; if (state == HPI_STATE_STOPPED) { if (bytes_avail == 0) { @@ -837,7 +837,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) if (xfercount && /* Limit use of on card fifo for playback */ ((on_card_bytes <= ds->period_bytes) || - (s->stream == SNDRV_PCM_STREAM_CAPTURE))) + (snd_pcm_is_capture(s))))
{
@@ -853,7 +853,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) xfer2 = xfercount - xfer1; }
- if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(s)) { snd_printddd("write1, P=%d, xfer=%d, buf_ofs=%d\n", s->number, xfer1, buf_ofs); hpi_handle_error(
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/au88x0/au88x0_core.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index e5d8676373368..192b5fe0d54d7 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2132,7 +2132,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->type = type;
/* PLAYBACK ROUTES. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { int src[4], mix[4], ch_top; #ifndef CHIP_AU8820 int a3d = 0; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 546f712206040..53000e486986c 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -167,7 +167,7 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) || VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_I2S) runtime->hw = snd_vortex_playback_hw_adb; #ifdef CHIP_AU8830 - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && VORTEX_IS_QUAD(vortex) && VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { runtime->hw.channels_max = 4; @@ -308,7 +308,7 @@ static int snd_vortex_pcm_prepare(struct snd_pcm_substream *substream) int dma = stream->dma, fmt, dir;
// set up the hardware with the current configuration. - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dir = 1; else dir = 0;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/ca0106/ca0106_main.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index cf1bac7a435f1..abee20e6dab39 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -943,7 +943,7 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, } snd_pcm_group_for_each_entry(s, substream) { if (snd_pcm_substream_chip(s) != emu || - s->stream != SNDRV_PCM_STREAM_PLAYBACK) + !snd_pcm_is_playback(s)) continue; runtime = s->runtime; epcm = runtime->private_data;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/mixart/mixart.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7ceaf6a7a77ea..ed4959c517d2d 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -391,7 +391,7 @@ static int mixart_set_stream_state(struct mixart_stream *stream, int start) stream_state_req.stream_info.stream_desc.uid_pipe = stream->pipe->group_uid; stream_state_req.stream_info.stream_desc.stream_idx = stream->substream->number;
- if (stream->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream->substream)) request.message_id = start ? MSG_STREAM_START_INPUT_STAGE_PACKET : MSG_STREAM_STOP_INPUT_STAGE_PACKET; else request.message_id = start ? MSG_STREAM_START_OUTPUT_STAGE_PACKET : MSG_STREAM_STOP_OUTPUT_STAGE_PACKET; @@ -608,7 +608,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, /* update the stream levels */ if( stream->pcm_number <= MIXART_PCM_DIGITAL ) { int is_aes = stream->pcm_number > MIXART_PCM_ANALOG; - if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) + if(snd_pcm_is_playback(subs)) mixart_update_playback_stream_level(chip, is_aes, subs->number); else mixart_update_capture_stream_level( chip, is_aes); @@ -626,7 +626,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, if (subs->runtime->buffer_changed) { struct mixart_bufferinfo *bufferinfo; int i = (chip->chip_idx * MIXART_MAX_STREAM_PER_CARD) + (stream->pcm_number * (MIXART_PLAYBACK_STREAMS+MIXART_CAPTURE_STREAMS)) + subs->number; - if( subs->stream == SNDRV_PCM_STREAM_CAPTURE ) { + if(snd_pcm_is_capture(subs)) { i += MIXART_PLAYBACK_STREAMS; /* in array capture is behind playback */ }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/emu10k1/p16v.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index a9a75891f1da4..b174b392b6aee 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -422,7 +422,7 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, } snd_pcm_group_for_each_entry(s, substream) { if (snd_pcm_substream_chip(s) != emu || - s->stream != SNDRV_PCM_STREAM_PLAYBACK) + !snd_pcm_is_playback(s)) continue; runtime = s->runtime; channel = substream->pcm->device-emu->p16v_device_offset;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/juli.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 3b0c3e70987b9..12f0cd9b8fbb1 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -907,7 +907,7 @@ static int snd_ice1712_pro_trigger(struct snd_pcm_substream *substream, { unsigned int what; unsigned int old; - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -EINVAL; what = ICE1712_PLAYBACK_PAUSE; snd_pcm_trigger_done(substream, substream); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index d679842ae1bd7..b52426540c529 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -170,7 +170,7 @@ static void juli_spdif_in_open(struct snd_ice1712 *ice, struct snd_pcm_runtime *runtime = substream->runtime; int rate;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(substream) || !ice->is_spdif_master(ice)) return; rate = snd_ak4114_external_rate(spec->ak4114);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/rme9652/hdsp.c | 14 +++++++------- sound/pci/rme9652/hdspm.c | 23 +++++++++++------------ sound/pci/rme9652/rme9652.c | 14 +++++++------- 3 files changed, 25 insertions(+), 26 deletions(-)
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index e7d1b43471a29..b99f32d24f7df 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3953,7 +3953,7 @@ static signed char *hdsp_channel_buffer_location(struct hdsp *hdsp, if (mapped_channel < 0) return NULL;
- if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); else return hdsp->playback_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); @@ -4014,7 +4014,7 @@ static int snd_hdsp_reset(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct hdsp *hdsp = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) other = hdsp->capture_substream; else other = hdsp->playback_substream; @@ -4051,7 +4051,7 @@ static int snd_hdsp_hw_params(struct snd_pcm_substream *substream,
spin_lock_irq(&hdsp->lock);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream->pstr->stream)) { hdsp->control_register &= ~(HDSP_SPDIFProfessional | HDSP_SPDIFNonAudio | HDSP_SPDIFEmphasis); hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register |= hdsp->creg_spdif_stream); this_pid = hdsp->playback_pid; @@ -4172,7 +4172,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) spin_unlock(&hdsp->lock); return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) other = hdsp->capture_substream; else other = hdsp->playback_substream; @@ -4191,15 +4191,15 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) } if (cmd == SNDRV_PCM_TRIGGER_START) { if (!(running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) && - substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_is_capture(substream)) hdsp_silence_playback(hdsp); } else { if (running && - substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_is_playback(substream)) hdsp_silence_playback(hdsp); } } else { - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) hdsp_silence_playback(hdsp); } _ok: diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 56d335f0e1960..a85e09535636a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5465,7 +5465,7 @@ static int snd_hdspm_reset(struct snd_pcm_substream *substream) struct hdspm *hdspm = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) other = hdspm->capture_substream; else other = hdspm->playback_substream; @@ -5499,7 +5499,7 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
spin_lock_irq(&hdspm->lock);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream->pstr->stream)) { this_pid = hdspm->playback_pid; other_pid = hdspm->capture_pid; } else { @@ -5570,7 +5570,7 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream, return err; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) {
for (i = 0; i < params_channels(params); ++i) { int c = hdspm->channel_map_out[i]; @@ -5656,7 +5656,7 @@ static int snd_hdspm_hw_free(struct snd_pcm_substream *substream) int i; struct hdspm *hdspm = snd_pcm_substream_chip(substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* Just disable all channels. The saving when disabling a */ /* smaller set is not worth the trouble. */ for (i = 0; i < HDSPM_MAX_CHANNELS; ++i) @@ -5682,7 +5682,7 @@ static int snd_hdspm_channel_info(struct snd_pcm_substream *substream, struct hdspm *hdspm = snd_pcm_substream_chip(substream); unsigned int channel = info->channel;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (snd_BUG_ON(channel >= hdspm->max_channels_out)) { dev_info(hdspm->card->dev, "snd_hdspm_channel_info: output channel out of range (%d)\n", @@ -5765,7 +5765,7 @@ static int snd_hdspm_trigger(struct snd_pcm_substream *substream, int cmd) spin_unlock(&hdspm->lock); return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) other = hdspm->capture_substream; else other = hdspm->playback_substream; @@ -5784,16 +5784,15 @@ static int snd_hdspm_trigger(struct snd_pcm_substream *substream, int cmd) } if (cmd == SNDRV_PCM_TRIGGER_START) { if (!(running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) - && substream->stream == - SNDRV_PCM_STREAM_CAPTURE) + && snd_pcm_is_capture(substream)) hdspm_silence_playback(hdspm); } else { if (running && - substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_is_playback(substream)) hdspm_silence_playback(hdspm); } } else { - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) hdspm_silence_playback(hdspm); } _ok: @@ -6046,7 +6045,7 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool playback = snd_pcm_is_playback(substream);
spin_lock_irq(&hdspm->lock); snd_pcm_set_sync(substream); @@ -6121,7 +6120,7 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream) static int snd_hdspm_release(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); - bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool playback = snd_pcm_is_playback(substream);
spin_lock_irq(&hdspm->lock);
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index d066c70ae1600..84e18197f26f7 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1833,7 +1833,7 @@ static signed char *rme9652_channel_buffer_location(struct snd_rme9652 *rme9652, if (mapped_channel < 0) return NULL; - if (stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(stream)) { return rme9652->capture_buffer + (mapped_channel * RME9652_CHANNEL_BUFFER_BYTES); } else { @@ -1903,7 +1903,7 @@ static int snd_rme9652_reset(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_rme9652 *rme9652 = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) other = rme9652->capture_substream; else other = rme9652->playback_substream; @@ -1934,7 +1934,7 @@ static int snd_rme9652_hw_params(struct snd_pcm_substream *substream,
spin_lock_irq(&rme9652->lock);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream->pstr->stream)) { rme9652->control_register &= ~(RME9652_PRO | RME9652_Dolby | RME9652_EMP); rme9652_write(rme9652, RME9652_control_register, rme9652->control_register |= rme9652->creg_spdif_stream); this_pid = rme9652->playback_pid; @@ -2056,7 +2056,7 @@ static int snd_rme9652_trigger(struct snd_pcm_substream *substream, spin_unlock(&rme9652->lock); return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) other = rme9652->capture_substream; else other = rme9652->playback_substream; @@ -2075,15 +2075,15 @@ static int snd_rme9652_trigger(struct snd_pcm_substream *substream, } if (cmd == SNDRV_PCM_TRIGGER_START) { if (!(running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) && - substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_is_capture(substream)) rme9652_silence_playback(rme9652); } else { if (running && - substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_is_playback(substream)) rme9652_silence_playback(rme9652); } } else { - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) rme9652_silence_playback(rme9652); } _ok:
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/korg1212/korg1212.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 5c2cac201a281..f6c7edd1f045b 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -1494,7 +1494,7 @@ static int snd_korg1212_hw_params(struct snd_pcm_substream *substream,
spin_lock_irqsave(&korg1212->lock, flags);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream->pstr->stream)) { this_pid = korg1212->playback_pid; other_pid = korg1212->capture_pid; } else {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/lx6464es/lx6464es.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index bd9b6148dd6fb..721b4ea3103ac 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -98,7 +98,7 @@ static int lx_hardware_open(struct lx6464es *chip, int err = 0; struct snd_pcm_runtime *runtime = substream->runtime; int channels = runtime->channels; - int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int is_capture = snd_pcm_is_capture(substream);
snd_pcm_uframes_t period_size = runtime->period_size;
@@ -124,7 +124,7 @@ static int lx_hardware_start(struct lx6464es *chip, { int err = 0; struct snd_pcm_runtime *runtime = substream->runtime; - int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int is_capture = snd_pcm_is_capture(substream);
dev_dbg(chip->card->dev, "setting stream format\n"); err = lx_stream_set_format(chip, runtime, 0, is_capture); @@ -155,7 +155,7 @@ static int lx_hardware_stop(struct lx6464es *chip, struct snd_pcm_substream *substream) { int err = 0; - int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int is_capture = snd_pcm_is_capture(substream);
dev_dbg(chip->card->dev, "pausing pipe\n"); err = lx_pipe_pause(chip, 0, is_capture); @@ -186,7 +186,7 @@ static int lx_hardware_close(struct lx6464es *chip, struct snd_pcm_substream *substream) { int err = 0; - int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int is_capture = snd_pcm_is_capture(substream);
dev_dbg(chip->card->dev, "releasing pipe\n"); err = lx_pipe_release(chip, 0, is_capture); @@ -268,7 +268,7 @@ static snd_pcm_uframes_t lx_pcm_stream_pointer(struct snd_pcm_substream { struct lx6464es *chip = snd_pcm_substream_chip(substream); snd_pcm_uframes_t pos; - int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int is_capture = snd_pcm_is_capture(substream);
struct lx_stream *lx_stream = is_capture ? &chip->capture_stream : &chip->playback_stream; @@ -287,7 +287,7 @@ static int lx_pcm_prepare(struct snd_pcm_substream *substream) { struct lx6464es *chip = snd_pcm_substream_chip(substream); int err = 0; - const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + const int is_capture = snd_pcm_is_capture(substream);
dev_dbg(chip->card->dev, "->lx_pcm_prepare\n");
@@ -370,7 +370,7 @@ static int lx_pcm_hw_free(struct snd_pcm_substream *substream) { struct lx6464es *chip = snd_pcm_substream_chip(substream); int err = 0; - int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int is_capture = snd_pcm_is_capture(substream);
dev_dbg(chip->card->dev, "->lx_pcm_hw_free\n"); mutex_lock(&chip->setup_mutex); @@ -515,7 +515,7 @@ static int lx_pcm_trigger_dispatch(struct lx6464es *chip, static int lx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct lx6464es *chip = snd_pcm_substream_chip(substream); - const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + const int is_capture = snd_pcm_is_capture(substream); struct lx_stream *stream = is_capture ? &chip->capture_stream : &chip->playback_stream;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/pci/intel8x0.c | 2 +- sound/pci/maestro3.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index dae3e15ba534d..d9a6a9477bccc 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -838,7 +838,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* clear FIFO for synchronization of channels */ fifo = igetdword(chip, fiforeg[ichdev->ali_slot / 4]); fifo &= ~(0xff << (ichdev->ali_slot % 4)); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index f4d211970d7ec..28634b2d8e5bd 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -1130,7 +1130,7 @@ snd_m3_pcm_setup1(struct snd_m3 *chip, struct m3_dma *s, struct snd_pcm_substrea int dsp_in_size, dsp_out_size, dsp_in_buffer, dsp_out_buffer; struct snd_pcm_runtime *runtime = subs->runtime;
- if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(subs)) { dsp_in_size = MINISRC_IN_BUFFER_SIZE - (0x20 * 2); dsp_out_size = MINISRC_OUT_BUFFER_SIZE - (0x20 * 2); } else { @@ -1416,7 +1416,7 @@ snd_m3_pcm_prepare(struct snd_pcm_substream *subs)
snd_m3_pcm_setup1(chip, s, subs);
- if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(subs)) snd_m3_playback_setup(chip, s, subs); else snd_m3_capture_setup(chip, s, subs); @@ -1724,7 +1724,7 @@ snd_m3_substream_open(struct snd_m3 *chip, struct snd_pcm_substream *subs) s->substream = subs;
/* set list owners */ - if (subs->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(subs)) { s->index_list[0] = &chip->mixer_list; } else s->index_list[0] = &chip->adc1_list;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/ppc/pmac.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 84058bbf9d127..168c1e2535faf 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -103,7 +103,7 @@ unsigned int snd_pmac_rate_index(struct snd_pmac *chip, struct pmac_stream *rec, */ static inline int another_stream(int stream) { - return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? + return snd_pcm_is_playback(stream) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; }
@@ -251,7 +251,7 @@ static int snd_pmac_pcm_trigger(struct snd_pmac *chip, struct pmac_stream *rec, case SNDRV_PCM_TRIGGER_RESUME: if (rec->running) return -EBUSY; - command = (subs->stream == SNDRV_PCM_STREAM_PLAYBACK ? + command = (snd_pcm_is_playback(subs) ? OUTPUT_MORE : INPUT_MORE) + INTR_ALWAYS; spin_lock(&chip->reg_lock); snd_pmac_beep_stop(chip);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/mips/sgio2audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index a8551ccdd1bf8..a16f63f21dd18 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -447,7 +447,7 @@ static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream) udelay(10); writeq(0, &mace->perif.audio.chan[ch].control);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* push a full buffer */ snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32); }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/core/pcm.c | 8 ++++---- sound/core/pcm_compat.c | 4 ++-- sound/core/pcm_dmaengine.c | 6 +++--- sound/core/pcm_lib.c | 14 +++++++------- sound/core/pcm_local.h | 4 ++-- sound/core/pcm_memory.c | 2 +- sound/core/pcm_native.c | 30 +++++++++++++++--------------- 7 files changed, 34 insertions(+), 34 deletions(-)
diff --git a/sound/core/pcm.c b/sound/core/pcm.c index dc37f3508dc7a..fdbfb13e4d18f 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -497,7 +497,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) char name[16];
sprintf(name, "pcm%i%c", pcm->device, - pstr->stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c'); + snd_pcm_is_playback(pstr->stream) ? 'p' : 'c'); entry = snd_info_create_card_entry(pcm->card, name, pcm->card->proc_root); if (!entry) @@ -642,7 +642,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) if (err < 0) return err; dev_set_name(pstr->dev, "pcmC%iD%i%c", pcm->card->number, pcm->device, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c'); + snd_pcm_is_playback(stream) ? 'p' : 'c'); pstr->dev->groups = pcm_dev_attr_groups; pstr->dev->type = &pcm_dev_type; dev_set_drvdata(pstr->dev, pstr); @@ -884,8 +884,8 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
if (snd_BUG_ON(!pcm || !rsubstream)) return -ENXIO; - if (snd_BUG_ON(stream != SNDRV_PCM_STREAM_PLAYBACK && - stream != SNDRV_PCM_STREAM_CAPTURE)) + if (snd_BUG_ON(!snd_pcm_is_playback(stream) && + !snd_pcm_is_capture(stream))) return -EINVAL; *rsubstream = NULL; pstr = &pcm->streams[stream]; diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index a42ec7f5a1daf..47fbbbdb5a8c7 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -300,7 +300,7 @@ static int snd_pcm_ioctl_xferi_compat(struct snd_pcm_substream *substream, get_user(frames, &data32->frames)) return -EFAULT;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) err = snd_pcm_lib_write(substream, compat_ptr(buf), frames); else err = snd_pcm_lib_read(substream, compat_ptr(buf), frames); @@ -359,7 +359,7 @@ static int snd_pcm_ioctl_xfern_compat(struct snd_pcm_substream *substream, bufs[i] = compat_ptr(ptr); bufptr++; } - if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) err = snd_pcm_lib_writev(substream, bufs, frames); else err = snd_pcm_lib_readv(substream, bufs, frames); diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index b134a51b3fd58..30db37652038f 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -72,7 +72,7 @@ int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, else buswidth = DMA_SLAVE_BUSWIDTH_8_BYTES;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { slave_config->direction = DMA_MEM_TO_DEV; slave_config->dst_addr_width = buswidth; } else { @@ -108,7 +108,7 @@ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_dmaengine_dai_dma_data *dma_data, struct dma_slave_config *slave_config) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { slave_config->dst_addr = dma_data->addr; slave_config->dst_maxburst = dma_data->maxburst; if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) @@ -444,7 +444,7 @@ int snd_dmaengine_pcm_refine_runtime_hwparams( if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) hw->info |= SNDRV_PCM_INFO_BATCH;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) addr_widths = dma_caps.dst_addr_widths; else addr_widths = dma_caps.src_addr_widths; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e7905749c4a3..34f45deaf493b 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -253,7 +253,7 @@ static void update_audio_tstamp(struct snd_pcm_substream *substream, audio_frames = runtime->hw_ptr_wrap + runtime->status->hw_ptr;
if (runtime->audio_tstamp_config.report_delay) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) audio_frames -= runtime->delay; else audio_frames += runtime->delay; @@ -464,7 +464,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return 0; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr);
@@ -1947,7 +1947,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t *availp) { struct snd_pcm_runtime *runtime = substream->runtime; - int is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int is_playback = snd_pcm_is_playback(substream); wait_queue_entry_t wait; int err = 0; snd_pcm_uframes_t avail = 0; @@ -2069,7 +2069,7 @@ static int fill_silence(struct snd_pcm_substream *substream, int channel, { struct snd_pcm_runtime *runtime = substream->runtime;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return 0; if (substream->ops->fill_silence) return substream->ops->fill_silence(substream, channel, @@ -2100,7 +2100,7 @@ static int do_transfer(struct snd_pcm_substream *substream, int c, struct iov_iter iter; int err, type;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) type = ITER_SOURCE; else type = ITER_DEST; @@ -2283,7 +2283,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (err < 0) return err;
- is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + is_playback = snd_pcm_is_playback(substream); if (interleaved) { if (runtime->access != SNDRV_PCM_ACCESS_RW_INTERLEAVED && runtime->channels > 1) @@ -2605,7 +2605,7 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, info->stream = stream; info->chmap = chmap; info->max_channels = max_channels; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) knew.name = "Playback Channel Map"; else knew.name = "Capture Channel Map"; diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h index ecb21697ae3a4..f08030e56aab6 100644 --- a/sound/core/pcm_local.h +++ b/sound/core/pcm_local.h @@ -35,7 +35,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, static inline snd_pcm_uframes_t snd_pcm_avail(struct snd_pcm_substream *substream) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return snd_pcm_playback_avail(substream->runtime); else return snd_pcm_capture_avail(substream->runtime); @@ -44,7 +44,7 @@ snd_pcm_avail(struct snd_pcm_substream *substream) static inline snd_pcm_uframes_t snd_pcm_hw_avail(struct snd_pcm_substream *substream) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return snd_pcm_playback_hw_avail(substream->runtime); else return snd_pcm_capture_hw_avail(substream->runtime); diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 506386959f084..2d55d3bf72d5b 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -63,7 +63,7 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, __update_allocated_size(card, size); }
- if (str == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(str)) dir = DMA_TO_DEVICE; else dir = DMA_FROM_DEVICE; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4057f9f10aeec..5e57ab50f4c0f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -972,7 +972,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, runtime->silence_size = params->silence_size; params->boundary = runtime->boundary; if (snd_pcm_running(substream)) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); err = snd_pcm_update_state(substream, runtime); @@ -999,7 +999,7 @@ snd_pcm_calc_delay(struct snd_pcm_substream *substream) { snd_pcm_uframes_t delay;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) delay = snd_pcm_playback_hw_avail(substream->runtime); else delay = snd_pcm_capture_avail(substream->runtime); @@ -1419,7 +1419,7 @@ static int snd_pcm_pre_start(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; if (runtime->state != SNDRV_PCM_STATE_PREPARED) return -EBADFD; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && !snd_pcm_playback_data(substream)) return -EPIPE; runtime->trigger_tstamp_latched = false; @@ -1459,7 +1459,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / runtime->rate; __snd_pcm_set_state(runtime, state); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); @@ -1798,7 +1798,7 @@ static int snd_pcm_do_resume(struct snd_pcm_substream *substream, /* DMA not running previously? */ if (runtime->suspended_state != SNDRV_PCM_STATE_RUNNING && (runtime->suspended_state != SNDRV_PCM_STATE_DRAINING || - substream->stream != SNDRV_PCM_STREAM_PLAYBACK)) + !snd_pcm_is_playback(substream))) return 0; return substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_RESUME); } @@ -1904,7 +1904,7 @@ static void snd_pcm_post_reset(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; guard(pcm_stream_lock_irq)(substream); runtime->control->appl_ptr = runtime->status->hw_ptr; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); } @@ -2021,7 +2021,7 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, snd_pcm_state_t state) { struct snd_pcm_runtime *runtime = substream->runtime; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (runtime->state) { case SNDRV_PCM_STATE_PREPARED: /* start playback stream if possible */ @@ -2130,7 +2130,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, to_check = NULL; group = snd_pcm_stream_group_ref(substream); snd_pcm_group_for_each_entry(s, substream) { - if (s->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(s)) continue; runtime = s->runtime; if (runtime->state == SNDRV_PCM_STATE_DRAINING) { @@ -2916,7 +2916,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream) { switch (substream->runtime->state) { case SNDRV_PCM_STATE_DRAINING: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) return -EBADFD; fallthrough; case SNDRV_PCM_STATE_RUNNING: @@ -3215,7 +3215,7 @@ static int snd_pcm_xferi_frames_ioctl(struct snd_pcm_substream *substream, return -EFAULT; if (copy_from_user(&xferi, _xferi, sizeof(xferi))) return -EFAULT; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) result = snd_pcm_lib_write(substream, xferi.buf, xferi.frames); else result = snd_pcm_lib_read(substream, xferi.buf, xferi.frames); @@ -3244,7 +3244,7 @@ static int snd_pcm_xfern_frames_ioctl(struct snd_pcm_substream *substream, bufs = memdup_user(xfern.bufs, sizeof(void *) * runtime->channels); if (IS_ERR(bufs)) return PTR_ERR(no_free_ptr(bufs)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) result = snd_pcm_lib_writev(substream, bufs, xfern.frames); else result = snd_pcm_lib_readv(substream, bufs, xfern.frames); @@ -3433,7 +3433,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, case SNDRV_PCM_IOCTL_FORWARD: { /* provided only for OSS; capture-only and no value returned */ - if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream)) return -EINVAL; result = snd_pcm_forward(substream, *frames); return result < 0 ? result : 0; @@ -3596,7 +3596,7 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) pcm_file = file->private_data;
substream = pcm_file->substream; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ok = EPOLLOUT | EPOLLWRNORM; else ok = EPOLLIN | EPOLLRDNORM; @@ -3620,7 +3620,7 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) mask = ok; break; case SNDRV_PCM_STATE_DRAINING: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { mask = ok; if (!avail) mask |= EPOLLERR; @@ -3876,7 +3876,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, size_t dma_bytes; int err;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (!(area->vm_flags & (VM_WRITE|VM_READ))) return -EINVAL; } else {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/core/oss/io.c | 2 +- sound/core/oss/pcm_oss.c | 20 ++++++++++---------- sound/core/oss/pcm_plugin.c | 10 +++++----- 3 files changed, 16 insertions(+), 16 deletions(-)
diff --git a/sound/core/oss/io.c b/sound/core/oss/io.c index d870b2d93135d..86d25f2e01ea6 100644 --- a/sound/core/oss/io.c +++ b/sound/core/oss/io.c @@ -128,7 +128,7 @@ int snd_pcm_plugin_build_io(struct snd_pcm_substream *plug, if (err < 0) return err; plugin->access = params_access(params); - if (snd_pcm_plug_stream(plug) == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(snd_pcm_plug_stream(plug))) { plugin->transfer = io_playback_transfer; if (plugin->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED) plugin->client_channels = io_src_channels; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 7386982cf40ed..aeaf9c69e791f 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -986,7 +986,7 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) "snd_pcm_plugin_build_io failed: %i\n", err); goto failure; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { err = snd_pcm_plugin_append(plugin); } else { err = snd_pcm_plugin_insert(plugin); @@ -1003,13 +1003,13 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) sw_params->start_threshold = runtime->boundary; } if (atomic_read(&substream->mmap_count) || - substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_is_capture(substream)) sw_params->stop_threshold = runtime->boundary; else sw_params->stop_threshold = runtime->buffer_size; sw_params->tstamp_mode = SNDRV_PCM_TSTAMP_NONE; sw_params->period_step = 1; - sw_params->avail_min = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + sw_params->avail_min = snd_pcm_is_playback(substream) ? 1 : runtime->period_size; if (atomic_read(&substream->mmap_count) || substream->oss.setup.nosilence) { @@ -2017,7 +2017,7 @@ static int snd_pcm_oss_get_caps1(struct snd_pcm_substream *substream, int res) return res; } #ifdef DSP_CAP_MULTI - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) if (substream->pstr->substream_count > 1) res |= DSP_CAP_MULTI; #endif @@ -2201,7 +2201,7 @@ static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream return -EFAULT; return 0; } - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DELAY, &delay); if (err == -EPIPE || err == -ESTRPIPE || (! err && delay < 0)) { err = 0; @@ -2225,12 +2225,12 @@ static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream n += runtime->boundary; info.blocks = n / runtime->period_size; runtime->oss.prev_hw_ptr_period = delay; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_pcm_oss_simulate_fill(substream, delay); info.bytes = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr) & INT_MAX; } else { delay = snd_pcm_oss_bytes(substream, delay); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { if (substream->oss.setup.buggyptr) info.blocks = (runtime->oss.buffer_bytes - delay - fixup) / runtime->oss.period_bytes; else @@ -2272,7 +2272,7 @@ static int snd_pcm_oss_get_space(struct snd_pcm_oss_file *pcm_oss_file, int stre info.fragsize = runtime->oss.period_bytes; info.fragstotal = runtime->periods; if (runtime->oss.prepare) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { info.bytes = runtime->oss.period_bytes * runtime->oss.periods; info.fragments = runtime->oss.periods; } else { @@ -2280,7 +2280,7 @@ static int snd_pcm_oss_get_space(struct snd_pcm_oss_file *pcm_oss_file, int stre info.fragments = 0; } } else { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DELAY, &avail); if (err == -EPIPE || err == -ESTRPIPE || (! err && avail < 0)) { avail = runtime->buffer_size; @@ -2429,7 +2429,7 @@ static int snd_pcm_oss_open_file(struct file *file, continue; if (! pcm->streams[idx].substream_count) continue; /* no matching substream */ - if (idx == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(idx)) { if (! (f_mode & FMODE_WRITE)) continue; } else { diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 82e180c776ae1..47168e175c966 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -54,7 +54,7 @@ static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t unsigned int channel; struct snd_pcm_plugin_channel *c;
- if (plugin->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(plugin->stream)) { format = &plugin->src_format; } else { format = &plugin->dst_format; @@ -110,7 +110,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) int err; if (snd_BUG_ON(!snd_pcm_plug_first(plug))) return -ENXIO; - if (snd_pcm_plug_stream(plug) == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(snd_pcm_plug_stream(plug))) { struct snd_pcm_plugin *plugin = snd_pcm_plug_first(plug); while (plugin->next) { if (plugin->dst_frames) @@ -174,7 +174,7 @@ int snd_pcm_plugin_build(struct snd_pcm_substream *plug, plugin->dst_format = *dst_format; plugin->dst_width = snd_pcm_format_physical_width(dst_format->format); snd_BUG_ON(plugin->dst_width <= 0); - if (plugin->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(plugin->stream)) channels = src_format->channels; else channels = dst_format->channels; @@ -567,7 +567,7 @@ snd_pcm_sframes_t snd_pcm_plug_client_channels_buf(struct snd_pcm_substream *plu
if (snd_BUG_ON(!buf)) return -ENXIO; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { plugin = snd_pcm_plug_first(plug); format = &plugin->src_format; } else { @@ -586,7 +586,7 @@ snd_pcm_sframes_t snd_pcm_plug_client_channels_buf(struct snd_pcm_substream *plu for (channel = 0; channel < nchannels; channel++, v++) { v->frames = count; v->enabled = 1; - v->wanted = (stream == SNDRV_PCM_STREAM_CAPTURE); + v->wanted = snd_pcm_is_capture(stream); v->area.addr = buf; v->area.first = channel * width; v->area.step = nchannels * width;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/virtio/virtio_card.h | 2 +- sound/virtio/virtio_pcm_msg.c | 4 ++-- sound/virtio/virtio_pcm_ops.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/virtio/virtio_card.h b/sound/virtio/virtio_card.h index 3ceee4e416fc7..db6d164fada99 100644 --- a/sound/virtio/virtio_card.h +++ b/sound/virtio/virtio_card.h @@ -107,7 +107,7 @@ virtsnd_rx_queue(struct virtio_snd *snd) static inline struct virtio_snd_queue * virtsnd_pcm_queue(struct virtio_pcm_substream *vss) { - if (vss->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(vss->direction)) return virtsnd_tx_queue(vss->snd); else return virtsnd_rx_queue(vss->snd); diff --git a/sound/virtio/virtio_pcm_msg.c b/sound/virtio/virtio_pcm_msg.c index 8c32efaf4c529..40e113a2de73f 100644 --- a/sound/virtio/virtio_pcm_msg.c +++ b/sound/virtio/virtio_pcm_msg.c @@ -230,7 +230,7 @@ int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss, unsigned long offset, msg->xfer.stream_id = cpu_to_le32(vss->sid); memset(&msg->status, 0, sizeof(msg->status));
- if (vss->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(vss->direction)) rc = virtqueue_add_sgs(vqueue, psgs, 2, 1, msg, GFP_ATOMIC); else @@ -313,7 +313,7 @@ static void virtsnd_pcm_msg_complete(struct virtio_pcm_msg *msg, * If the capture substream returned an incorrect status, then just * increase the hw_ptr by the message size. */ - if (vss->direction == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(vss->direction) || written_bytes <= sizeof(msg->status)) vss->hw_ptr += msg->length; else diff --git a/sound/virtio/virtio_pcm_ops.c b/sound/virtio/virtio_pcm_ops.c index ad12aae52fc32..5d93d50f24023 100644 --- a/sound/virtio/virtio_pcm_ops.c +++ b/sound/virtio/virtio_pcm_ops.c @@ -337,7 +337,7 @@ static int virtsnd_pcm_trigger(struct snd_pcm_substream *substream, int command)
spin_lock_irqsave(&queue->lock, flags); spin_lock(&vss->lock); - if (vss->direction == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(vss->direction)) rc = virtsnd_pcm_msg_send(vss, 0, vss->buffer_bytes); if (!rc) vss->xfer_enabled = true;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/dmaengine_pcm.h | 2 +- include/sound/pcm.h | 4 ++-- include/sound/sdw.h | 2 +- include/sound/soc-dai.h | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-)
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f6baa9a018681..3b2e31922f0f1 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -21,7 +21,7 @@ static inline enum dma_transfer_direction snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return DMA_MEM_TO_DEV; else return DMA_DEV_TO_MEM; diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 69e535aeb8e82..00603c0e568e1 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -734,7 +734,7 @@ static inline int snd_pcm_running(struct snd_pcm_substream *substream) { return (substream->runtime->state == SNDRV_PCM_STATE_RUNNING || (substream->runtime->state == SNDRV_PCM_STATE_DRAINING && - substream->stream == SNDRV_PCM_STREAM_PLAYBACK)); + snd_pcm_is_playback(substream))); }
/** @@ -1521,7 +1521,7 @@ const char *snd_pcm_format_name(snd_pcm_format_t format); */ static inline const char *snd_pcm_direction_name(int direction) { - if (direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(direction)) return "Playback"; else return "Capture"; diff --git a/include/sound/sdw.h b/include/sound/sdw.h index 6dcdb3228dba6..ab752dadea3bc 100644 --- a/include/sound/sdw.h +++ b/include/sound/sdw.h @@ -38,7 +38,7 @@ static inline void snd_sdw_params_to_config(struct snd_pcm_substream *substream, stream_config->ch_count = params_channels(params); stream_config->bps = snd_pcm_format_width(params_format(params));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) stream_config->direction = SDW_DATA_DIR_RX; else stream_config->direction = SDW_DATA_DIR_TX; diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..577bbaede90af 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -476,7 +476,7 @@ struct snd_soc_dai { static inline const struct snd_soc_pcm_stream * snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) { - return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? + return (snd_pcm_is_playback(stream)) ? &dai->driver->playback : &dai->driver->capture; }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/drivers/aloop.c | 18 +++++++++--------- sound/drivers/dummy.c | 2 +- sound/drivers/pcmtest.c | 4 ++-- 3 files changed, 12 insertions(+), 12 deletions(-)
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 439d12ad87879..9d33aef59e95b 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -193,7 +193,7 @@ static inline struct loopback_setup *get_setup(struct loopback_pcm *dpcm) { int device = dpcm->substream->pstr->pcm->device; - if (dpcm->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dpcm->substream)) device ^= 1; return &dpcm->loopback->setup[dpcm->substream->number][device]; } @@ -341,7 +341,7 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) int check;
if (cable->valid != CABLE_VALID_BOTH) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) goto __notify; return 0; } @@ -356,7 +356,7 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) is_access_interleaved(cruntime->access); if (!check) return 0; - if (stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(stream)) { return -EIO; } else { snd_pcm_stop(cable->streams[SNDRV_PCM_STREAM_CAPTURE]-> @@ -418,7 +418,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause &= ~stream; err = cable->ops->start(dpcm); spin_unlock(&cable->lock); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_STOP: @@ -427,7 +427,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause &= ~stream; err = cable->ops->stop(dpcm); spin_unlock(&cable->lock); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -436,7 +436,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause |= stream; err = cable->ops->stop(dpcm); spin_unlock(&cable->lock); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -446,7 +446,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause &= ~stream; err = cable->ops->start(dpcm); spin_unlock(&cable->lock); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) loopback_active_notify(dpcm); break; default: @@ -497,7 +497,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) dpcm->buf_pos = 0; dpcm->pcm_buffer_size = frames_to_bytes(runtime, runtime->buffer_size); dpcm->channel_buf_n = dpcm->pcm_buffer_size / runtime->channels; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { /* clear capture buffer */ dpcm->silent_size = dpcm->pcm_buffer_size; snd_pcm_format_set_silence(runtime->format, runtime->dma_area, @@ -513,7 +513,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) mutex_lock(&dpcm->loopback->cable_lock); if (!(cable->valid & ~(1 << substream->stream)) || (get_setup(dpcm)->notify && - substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + snd_pcm_is_playback(substream))) params_change(substream); cable->valid |= 1 << substream->stream; mutex_unlock(&dpcm->loopback->cable_lock); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 52ff6ac3f7435..5e440f952449e 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -570,7 +570,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) if (model == NULL) return 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (model->playback_constraints) err = model->playback_constraints(substream->runtime); } else { diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 21cefaf5419aa..ae308ede80929 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -351,9 +351,9 @@ static void timer_timeout(struct timer_list *data) if (v_iter->suspend) return;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !v_iter->is_buf_corrupted) + if (snd_pcm_is_playback(substream) && !v_iter->is_buf_corrupted) check_buf_block(v_iter, substream->runtime); - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) fill_block(v_iter, substream->runtime); else inc_buf_pos(v_iter, v_iter->b_rw, substream->runtime->dma_bytes);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/motu/motu-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index d410c2efbde57..411d2b3dccbb8 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -101,7 +101,7 @@ static int init_hw_info(struct snd_motu *motu, struct snd_motu_packet_format *formats; int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { hw->formats = SNDRV_PCM_FMTBIT_S32; stream = &motu->tx_stream; formats = &motu->tx_packet_formats;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/oxfw/oxfw-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 5f43a0b826d2e..c0e67a0d10f1b 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -114,7 +114,7 @@ static int init_hw_params(struct snd_oxfw *oxfw, struct amdtp_stream *stream; int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; stream = &oxfw->tx_stream; formats = oxfw->tx_stream_formats; @@ -150,7 +150,7 @@ static int limit_to_current_params(struct snd_pcm_substream *substream) enum avc_general_plug_dir dir; int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) dir = AVC_GENERAL_PLUG_DIR_OUT; else dir = AVC_GENERAL_PLUG_DIR_IN;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/dice/dice-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index d64366217d572..063e16d153fa5 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -26,7 +26,7 @@ static int dice_rate_constraint(struct snd_pcm_hw_params *params, enum snd_dice_rate_mode mode; unsigned int i, rate;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) pcm_channels = dice->tx_pcm_chs[index]; else pcm_channels = dice->rx_pcm_chs[index]; @@ -64,7 +64,7 @@ static int dice_channels_constraint(struct snd_pcm_hw_params *params, enum snd_dice_rate_mode mode; unsigned int i, rate;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) pcm_channels = dice->tx_pcm_chs[index]; else pcm_channels = dice->rx_pcm_chs[index]; @@ -132,7 +132,7 @@ static int init_hw_info(struct snd_dice *dice, struct amdtp_stream *stream; int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { hw->formats = AM824_IN_PCM_FORMAT_BITS; dir = AMDTP_IN_STREAM; stream = &dice->tx_stream[index];
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/bebob/bebob_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index ce49eef0fcbaa..882aab28dd96d 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -100,7 +100,7 @@ pcm_init_hw_params(struct snd_bebob *bebob, struct snd_bebob_stream_formation *formations; int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &bebob->tx_stream; formations = bebob->tx_stream_formations;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/tascam/tascam-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index f6da571707ac2..29a8e7db7c30f 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -15,7 +15,7 @@ static int pcm_init_hw_params(struct snd_tscm *tscm, struct amdtp_stream *stream; unsigned int pcm_channels;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; stream = &tscm->tx_stream; pcm_channels = tscm->spec->pcm_capture_analog_channels;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/digi00x/digi00x-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 3bd1575c9d9c1..42ffed7b19e36 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -63,7 +63,7 @@ static int pcm_init_hw_params(struct snd_dg00x *dg00x, int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; s = &dg00x->tx_stream; } else {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/fireface/ff-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index ec915671a79b3..d17abff5bb1e0 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -109,7 +109,7 @@ static int pcm_init_hw_params(struct snd_ff *ff, const unsigned int *pcm_channels; int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { runtime->hw.formats = SNDRV_PCM_FMTBIT_S32; s = &ff->tx_stream; pcm_channels = ff->spec->pcm_capture_channels;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/firewire/fireworks/fireworks_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index c3c21860b245b..7ab42a6903e40 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -137,7 +137,7 @@ pcm_init_hw_params(struct snd_efw *efw, unsigned int *pcm_channels; int err;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS; s = &efw->tx_stream; pcm_channels = efw->pcm_capture_channels;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/ti/davinci-i2s.c | 12 ++++++------ sound/soc/ti/davinci-mcasp.c | 18 +++++++++--------- sound/soc/ti/omap-mcbsp.c | 18 +++++++++--------- sound/soc/ti/omap-mcpdm.c | 10 +++++----- 4 files changed, 29 insertions(+), 29 deletions(-)
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 0f15a743c7982..f509aaafa411f 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -190,7 +190,7 @@ static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback) static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev, struct snd_pcm_substream *substream) { - int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int playback = snd_pcm_is_playback(substream); u32 spcr; u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST;
@@ -485,7 +485,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, }
/* general line settings */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { spcr |= DAVINCI_MCBSP_SPCR_RINTM(3); davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } else { @@ -641,7 +641,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); @@ -656,7 +656,7 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int playback = snd_pcm_is_playback(substream); u32 spcr; u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST;
@@ -700,7 +700,7 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); int ret = 0; - int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int playback = snd_pcm_is_playback(substream);
switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -723,7 +723,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int playback = snd_pcm_is_playback(substream); davinci_mcbsp_stop(dev, playback); }
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 2b1ed91a736c9..e7eabbd972e79 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -289,7 +289,7 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) { mcasp->streams++;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) mcasp_start_tx(mcasp); else mcasp_start_rx(mcasp); @@ -354,7 +354,7 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) { mcasp->streams--;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) mcasp_stop_tx(mcasp); else mcasp_stop_rx(mcasp); @@ -873,7 +873,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); max_tx_serializers = max_active_serializers; @@ -913,7 +913,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, } }
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { active_serializers = tx_ser; numevt = mcasp->txnumevt; reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET; @@ -1026,12 +1026,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, if (!mcasp->dat_port) busel = TXSEL;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXMOD(total_slots), FSXMOD(0x1FF)); - } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { + } else if (snd_pcm_is_capture(stream)) { mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, @@ -1190,7 +1190,7 @@ static snd_pcm_sframes_t davinci_mcasp_delay( struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); u32 fifo_use;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) fifo_use = davinci_mcasp_tx_delay(mcasp); else fifo_use = davinci_mcasp_rx_delay(mcasp); @@ -1509,7 +1509,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, * Limit the maximum allowed channels for the first stream: * number of serializers for the direction * tdm slots per serializer */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dir = TX_MODE; else dir = RX_MODE; @@ -1591,7 +1591,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; }
- numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + numevt = snd_pcm_is_playback(substream) ? &mcasp->txnumevt : &mcasp->rxnumevt; snd_pcm_hw_rule_add(substream->runtime, 0, diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 2110ffe5281ce..bb6a01b41ac86 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -217,7 +217,7 @@ static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, { int data_reg;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { if (mcbsp->pdata->reg_size == 2) data_reg = OMAP_MCBSP_REG_DXR1; else @@ -413,7 +413,7 @@ static void omap_mcbsp_free(struct omap_mcbsp *mcbsp) */ static void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream) { - int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int tx = snd_pcm_is_playback(stream); int rx = !tx; int enable_srg = 0; u16 w; @@ -472,7 +472,7 @@ static void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream)
static void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream) { - int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + int tx = snd_pcm_is_playback(stream); int rx = !tx; int idle; u16 w; @@ -742,7 +742,7 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream, words = 1;
/* Configure McBSP internal buffer usage */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) omap_mcbsp_set_tx_threshold(mcbsp, words); else omap_mcbsp_set_rx_threshold(mcbsp, words); @@ -797,7 +797,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * smaller buffer than the FIFO size to avoid underruns. * This applies only for the playback stream. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, omap_mcbsp_hwrule_min_buffersize, @@ -816,7 +816,7 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int tx = snd_pcm_is_playback(substream); int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
@@ -839,7 +839,7 @@ static int omap_mcbsp_dai_prepare(struct snd_pcm_substream *substream, { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req; - int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int tx = snd_pcm_is_playback(substream); int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; int latency = mcbsp->latency[stream2]; @@ -896,7 +896,7 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( if (mcbsp->pdata->buffer_size == 0) return 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) fifo_use = omap_mcbsp_get_tx_delay(mcbsp); else fifo_use = omap_mcbsp_get_rx_delay(mcbsp); @@ -944,7 +944,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int divider = 0;
period_words = params_period_bytes(params) / (wlen / 8); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) max_thrsh = mcbsp->max_tx_thres; else max_thrsh = mcbsp->max_rx_thres; diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index 1a5d19937c642..43637ce12b665 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -265,7 +265,7 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int tx = snd_pcm_is_playback(substream); int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
@@ -305,13 +305,13 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, channels = params_channels(params); switch (channels) { case 5: - if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 4; fallthrough; case 4: - if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; @@ -334,7 +334,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
threshold = mcpdm->config[stream].threshold; /* Configure McPDM channels, and DMA packet size */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { link_mask <<= 3;
/* If capture is not running assume a stereo stream to come */ @@ -377,7 +377,7 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req; - int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int tx = snd_pcm_is_playback(substream); int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; int latency = mcpdm->latency[stream2];
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sh/dma-sh7760.c | 12 ++++++------ sound/soc/sh/fsi.c | 7 +------ sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rz-ssi.c | 14 ++++---------- sound/soc/sh/siu_dai.c | 4 ++-- sound/soc/sh/siu_pcm.c | 14 +++++++------- sound/soc/sh/ssi.c | 2 +- 8 files changed, 24 insertions(+), 35 deletions(-)
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index c53539482c208..32b30bbfaa88f 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -120,7 +120,7 @@ static int camelot_pcm_open(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct camelot_pcm *cam = &cam_pcm_data[snd_soc_rtd_to_cpu(rtd, 0)->id]; - int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int recv = snd_pcm_is_capture(substream); int ret, dmairq;
snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); @@ -154,7 +154,7 @@ static int camelot_pcm_close(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct camelot_pcm *cam = &cam_pcm_data[snd_soc_rtd_to_cpu(rtd, 0)->id]; - int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int recv = snd_pcm_is_capture(substream); int dmairq;
dmairq = (recv) ? cam->txid + 2 : cam->txid; @@ -176,7 +176,7 @@ static int camelot_hw_params(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct camelot_pcm *cam = &cam_pcm_data[snd_soc_rtd_to_cpu(rtd, 0)->id]; - int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int recv = snd_pcm_is_capture(substream);
if (recv) { cam->rx_period_size = params_period_bytes(hw_params); @@ -198,7 +198,7 @@ static int camelot_prepare(struct snd_soc_component *component, pr_debug("PCM data: addr %pad len %zu\n", &runtime->dma_addr, runtime->dma_bytes);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { BRGREG(BRGATXSAR) = (unsigned long)runtime->dma_area; BRGREG(BRGATXTCR) = runtime->dma_bytes; } else { @@ -242,7 +242,7 @@ static int camelot_trigger(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct camelot_pcm *cam = &cam_pcm_data[snd_soc_rtd_to_cpu(rtd, 0)->id]; - int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int recv = snd_pcm_is_capture(substream);
switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -270,7 +270,7 @@ static snd_pcm_uframes_t camelot_pos(struct snd_soc_component *component, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct camelot_pcm *cam = &cam_pcm_data[snd_soc_rtd_to_cpu(rtd, 0)->id]; - int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int recv = snd_pcm_is_capture(substream); unsigned long pos;
/* cannot use the DMABRG pointer register: under load, by the diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 087e379aa3bc4..59198f615ed6a 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -398,11 +398,6 @@ static int fsi_is_enable_stream(struct fsi_priv *fsi) return fsi->enable_stream; }
-static int fsi_is_play(struct snd_pcm_substream *substream) -{ - return substream->stream == SNDRV_PCM_STREAM_PLAYBACK; -} - static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); @@ -492,7 +487,7 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi) static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi, struct snd_pcm_substream *substream) { - return fsi_is_play(substream) ? &fsi->playback : &fsi->capture; + return snd_pcm_is_playback(substream) ? &fsi->playback : &fsi->capture; }
static int fsi_stream_is_working(struct fsi_priv *fsi, diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index cc200f45826c3..dc724042d336e 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -237,7 +237,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hac_priv *hac = &hac_cpu_data[dai->id]; - int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + int d = snd_pcm_is_capture(substream);
switch (params->msbits) { case 16: diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 15cb5e7008f9f..9e719a01769ba 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -683,7 +683,7 @@ static struct rsnd_dai_stream *rsnd_rdai_to_io(struct rsnd_dai *rdai, struct snd_pcm_substream *substream) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return &rdai->playback; else return &rdai->capture; @@ -1004,7 +1004,7 @@ static int rsnd_soc_dai_startup(struct snd_pcm_substream *substream, * It depends on Clock Master Mode */ if (rsnd_rdai_is_clk_master(rdai)) { - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int is_play = snd_pcm_is_playback(substream);
snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, rsnd_soc_hw_rule_rate, diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index d0bf0487bf1bd..d0afc9ced950b 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -171,18 +171,12 @@ rz_ssi_get_dai(struct snd_pcm_substream *substream) return snd_soc_rtd_to_cpu(rtd, 0); }
-static inline bool rz_ssi_stream_is_play(struct rz_ssi_priv *ssi, - struct snd_pcm_substream *substream) -{ - return substream->stream == SNDRV_PCM_STREAM_PLAYBACK; -} - static inline struct rz_ssi_stream * rz_ssi_stream_get(struct rz_ssi_priv *ssi, struct snd_pcm_substream *substream) { struct rz_ssi_stream *stream = &ssi->playback;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) stream = &ssi->capture;
return stream; @@ -349,7 +343,7 @@ static void rz_ssi_set_idle(struct rz_ssi_priv *ssi)
static int rz_ssi_start(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) { - bool is_play = rz_ssi_stream_is_play(ssi, strm->substream); + bool is_play = snd_pcm_is_playback(strm->substream); bool is_full_duplex; u32 ssicr, ssifcr;
@@ -682,7 +676,7 @@ static int rz_ssi_dma_transfer(struct rz_ssi_priv *ssi, */ return 0;
- dir = rz_ssi_stream_is_play(ssi, substream) ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + dir = snd_pcm_is_playback(substream) ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
/* Always transfer 1 period */ amount = runtime->period_size; @@ -808,7 +802,7 @@ static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd, if (ssi->dma_rt) { bool is_playback;
- is_playback = rz_ssi_stream_is_play(ssi, substream); + is_playback = snd_pcm_is_playback(substream); ret = rz_ssi_dma_slave_config(ssi, ssi->playback.dma_ch, is_playback); /* Fallback to pio */ diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index d0b5c543fd2f8..e747d34b51580 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -521,7 +521,7 @@ static void siu_dai_shutdown(struct snd_pcm_substream *substream, dev_dbg(substream->pcm->card->dev, "%s: port=%d@%p\n", __func__, info->port_id, port_info);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) port_info->play_cap &= ~PLAYBACK_ENABLED; else port_info->play_cap &= ~CAPTURE_ENABLED; @@ -550,7 +550,7 @@ static int siu_dai_prepare(struct snd_pcm_substream *substream, "%s: port %d, active streams %lx, %d channels\n", __func__, info->port_id, port_info->play_cap, rt->channels);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { self = PLAYBACK_ENABLED; siu_stream = &port_info->playback; } else { diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index f15ff36e79345..27ee6fd6d35c2 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -214,7 +214,7 @@ static void siu_io_work(struct work_struct *work) return; }
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { dma_addr_t buff; size_t count;
@@ -306,7 +306,7 @@ static int siu_pcm_open(struct snd_soc_component *component,
dev_dbg(dev, "%s, port=%d@%p\n", __func__, port, port_info);
- if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(ss)) { siu_stream = &port_info->playback; param = &siu_stream->param; param->shdma_slave.slave_id = port ? pdata->dma_slave_tx_b : @@ -340,7 +340,7 @@ static int siu_pcm_close(struct snd_soc_component *component,
dev_dbg(dev, "%s: port=%d\n", __func__, info->port_id);
- if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(ss)) siu_stream = &port_info->playback; else siu_stream = &port_info->capture; @@ -363,7 +363,7 @@ static int siu_pcm_prepare(struct snd_soc_component *component, struct siu_stream *siu_stream; snd_pcm_sframes_t xfer_cnt;
- if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(ss)) siu_stream = &port_info->playback; else siu_stream = &port_info->capture; @@ -413,7 +413,7 @@ static int siu_pcm_trigger(struct snd_soc_component *component,
switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(ss)) ret = siu_pcm_stmwrite_start(port_info); else ret = siu_pcm_stmread_start(port_info); @@ -424,7 +424,7 @@ static int siu_pcm_trigger(struct snd_soc_component *component,
break; case SNDRV_PCM_TRIGGER_STOP: - if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(ss)) siu_pcm_stmwrite_stop(port_info); else siu_pcm_stmread_stop(port_info); @@ -455,7 +455,7 @@ siu_pcm_pointer_dma(struct snd_soc_component *component, size_t ptr; struct siu_stream *siu_stream;
- if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(ss)) siu_stream = &port_info->playback; else siu_stream = &port_info->capture; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 96cf523c22734..f77b4d9a4a205 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -135,7 +135,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream,
channels = params_channels(params); bits = params->msbits; - recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + recv = snd_pcm_is_capture(substream);
pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); pr_debug("bits: %u channels: %u\n", bits, channels);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/adi/axi-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7b25630757436..7c950a7b71eec 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -60,7 +60,7 @@ static int axi_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct axi_i2s *i2s = snd_soc_dai_get_drvdata(dai); unsigned int mask, val;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) mask = AXI_I2S_CTRL_RX_EN; else mask = AXI_I2S_CTRL_TX_EN; @@ -110,7 +110,7 @@ static int axi_i2s_startup(struct snd_pcm_substream *substream, uint32_t mask; int ret;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) mask = AXI_I2S_RESET_RX_FIFO; else mask = AXI_I2S_RESET_TX_FIFO;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/amd/acp-pcm-dma.c | 24 ++++++++++++------------ sound/soc/amd/acp/acp-i2s.c | 24 ++++++++++++------------ sound/soc/amd/acp/acp-legacy-common.c | 8 ++++---- sound/soc/amd/acp/acp-platform.c | 2 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 4 ++-- sound/soc/amd/acp/amd.h | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 4 ++-- sound/soc/amd/ps/ps-sdw-dma.c | 4 ++-- sound/soc/amd/raven/acp3x-i2s.c | 8 ++++---- sound/soc/amd/raven/acp3x-pcm-dma.c | 10 +++++----- sound/soc/amd/raven/acp3x.h | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 4 ++-- sound/soc/amd/vangogh/acp5x-i2s.c | 8 ++++---- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 10 +++++----- sound/soc/amd/vangogh/acp5x.h | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 4 ++-- 16 files changed, 60 insertions(+), 60 deletions(-)
diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b857e2676fe8c..39d2b1538b27b 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -210,7 +210,7 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio,
for (i = 0; i < NUM_DSCRS_PER_CHANNEL; i++) { dmadscr[i].xfer_val = 0; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { dma_dscr_idx = dma_dscr_idx + i; dmadscr[i].dest = sram_bank + (i * (size / 2)); dmadscr[i].src = ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS @@ -268,7 +268,7 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size,
for (i = 0; i < NUM_DSCRS_PER_CHANNEL; i++) { dmadscr[i].xfer_val = 0; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { dma_dscr_idx = dma_dscr_idx + i; dmadscr[i].src = sram_bank + (i * (size / 2)); /* dmadscr[i].dest is unused by hardware. */ @@ -336,7 +336,7 @@ static void config_acp_dma(void __iomem *acp_mmio, acp_pte_config(acp_mmio, rtd->dma_addr, rtd->num_of_pages, rtd->pte_offset);
- if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(rtd->direction)) { ch_acp_sysmem = rtd->ch1; ch_acp_i2s = rtd->ch2; } else { @@ -779,7 +779,7 @@ static int acp_dma_open(struct snd_soc_component *component, if (!adata) return -ENOMEM;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (intr_data->asic_type) { case CHIP_STONEY: runtime->hw = acp_st_pcm_hardware_playback; @@ -819,7 +819,7 @@ static int acp_dma_open(struct snd_soc_component *component, !intr_data->play_i2s_micsp_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* * For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. @@ -861,7 +861,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, return -EINVAL;
if (pinfo) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { rtd->i2s_instance = pinfo->play_i2s_instance; } else { rtd->i2s_instance = pinfo->cap_i2s_instance; @@ -871,7 +871,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: val |= ACP_I2S_BT_16BIT_RESOLUTION_EN; @@ -898,7 +898,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, mmACP_I2S_16BIT_RESOLUTION_EN); }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: rtd->pte_offset = ACP_ST_BT_PLAYBACK_PTE_OFFSET; @@ -1043,7 +1043,7 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_soc_component *component, if (!rtd) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { period_bytes = frames_to_bytes(runtime, runtime->period_size); bytescount = acp_get_byte_count(rtd); if (bytescount >= rtd->bytescount) @@ -1092,7 +1092,7 @@ static int acp_dma_prepare(struct snd_soc_component *component, if (!rtd) return -EINVAL;
- if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(rtd->direction)) { ch_acp_sysmem = rtd->ch1; ch_acp_i2s = rtd->ch2; } else { @@ -1125,7 +1125,7 @@ static int acp_dma_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: rtd->bytescount = acp_get_byte_count(rtd); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { if (rtd->capture_channel == CAP_CHANNEL0) { acp_dma_cap_channel_disable(rtd->acp_mmio, CAP_CHANNEL1); @@ -1190,7 +1190,7 @@ static int acp_dma_close(struct snd_soc_component *component, struct audio_substream_data *rtd = runtime->private_data; struct audio_drv_data *adata = dev_get_drvdata(component->dev);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: adata->play_i2sbt_stream = NULL; diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 97258b4cf89b0..eafa6262e6feb 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -132,10 +132,10 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas
spin_lock_irq(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { - if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + if (tx_mask && snd_pcm_is_playback(stream->dir)) adata->tdm_tx_fmt[stream->dai_id - 1] = FRM_LEN | (slots << 15) | (slot_len << 18); - else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) + else if (rx_mask && snd_pcm_is_capture(stream->dir)) adata->tdm_rx_fmt[stream->dai_id - 1] = FRM_LEN | (slots << 15) | (slot_len << 18); } @@ -176,7 +176,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_ITER; @@ -224,7 +224,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ if (adata->tdm_mode) { val = readl(adata->acp_base + reg_val); writel(val | BIT(1), adata->acp_base + reg_val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tdm_fmt = adata->tdm_tx_fmt[dai->driver->id - 1]; else tdm_fmt = adata->tdm_rx_fmt[dai->driver->id - 1]; @@ -318,7 +318,7 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: stream->bytescount = acp_get_byte_count(adata, stream->dai_id, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (dai->driver->id) { case I2S_BT_INSTANCE: water_val = ACP_BT_TX_INTR_WATERMARK_SIZE; @@ -379,7 +379,7 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_ITER; @@ -444,7 +444,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d
switch (dai->driver->id) { case I2S_SP_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { reg_dma_size = ACP_I2S_TX_DMA_SIZE; acp_fifo_addr = rsrc->sram_pte_offset + SP_PB_FIFO_ADDR_OFFSET; @@ -464,7 +464,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d } break; case I2S_BT_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { reg_dma_size = ACP_BT_TX_DMA_SIZE; acp_fifo_addr = rsrc->sram_pte_offset + BT_PB_FIFO_ADDR_OFFSET; @@ -485,7 +485,7 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d } break; case I2S_HS_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { reg_dma_size = ACP_HS_TX_DMA_SIZE; acp_fifo_addr = rsrc->sram_pte_offset + HS_PB_FIFO_ADDR_OFFSET; @@ -538,7 +538,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d
switch (dai->driver->id) { case I2S_SP_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { irq_bit = BIT(I2S_TX_THRESHOLD(rsrc->offset)); stream->pte_offset = ACP_SRAM_SP_PB_PTE_OFFSET; stream->fifo_offset = SP_PB_FIFO_ADDR_OFFSET; @@ -549,7 +549,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d } break; case I2S_BT_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { irq_bit = BIT(BT_TX_THRESHOLD(rsrc->offset)); stream->pte_offset = ACP_SRAM_BT_PB_PTE_OFFSET; stream->fifo_offset = BT_PB_FIFO_ADDR_OFFSET; @@ -560,7 +560,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d } break; case I2S_HS_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { irq_bit = BIT(HS_TX_THRESHOLD(rsrc->offset)); stream->pte_offset = ACP_SRAM_HS_PB_PTE_OFFSET; stream->fifo_offset = HS_PB_FIFO_ADDR_OFFSET; diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 4422cec81e3c4..35cd4b2b86cf2 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -112,7 +112,7 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream,
switch (dai->driver->id) { case I2S_SP_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { reg_dma_size = ACP_I2S_TX_DMA_SIZE; acp_fifo_addr = rsrc->sram_pte_offset + SP_PB_FIFO_ADDR_OFFSET; @@ -131,7 +131,7 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, } break; case I2S_BT_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { reg_dma_size = ACP_BT_TX_DMA_SIZE; acp_fifo_addr = rsrc->sram_pte_offset + BT_PB_FIFO_ADDR_OFFSET; @@ -150,7 +150,7 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, } break; case I2S_HS_INSTANCE: - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { reg_dma_size = ACP_HS_TX_DMA_SIZE; acp_fifo_addr = rsrc->sram_pte_offset + HS_PB_FIFO_ADDR_OFFSET; @@ -199,7 +199,7 @@ int restore_acp_i2s_params(struct snd_pcm_substream *substream,
soc_runtime = snd_soc_substream_to_rtd(substream); dai = snd_soc_rtd_to_cpu(soc_runtime, 0); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { tdm_fmt = adata->tdm_tx_fmt[stream->dai_id - 1]; switch (stream->dai_id) { case I2S_BT_INSTANCE: diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 4f409cd09c11c..d4b7355a1d989 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -192,7 +192,7 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs
stream->substream = substream;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) runtime->hw = acp_pcm_hardware_playback; else runtime->hw = acp_pcm_hardware_capture; diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 3419675e45a98..85f9fcbfe822b 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -450,8 +450,8 @@ static int create_sdw_dailink(struct snd_soc_card *card, cpus[k].dai_name = cpu_name; }
- playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); - capture = (stream == SNDRV_PCM_STREAM_CAPTURE); + playback = snd_pcm_is_playback(stream); + capture = snd_pcm_is_capture(stream); asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, cpus, cpu_dai_num, diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 87a4813783f91..90b0716e5f2f5 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -253,7 +253,7 @@ static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int { u64 byte_count = 0, low = 0, high = 0;
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { switch (dai_id) { case I2S_BT_INSTANCE: high = readl(adata->acp_base + ACP_BT_TX_LINEARPOSITIONCNTR_HIGH); diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index 7bbacbab10950..256271a270286 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -193,7 +193,7 @@ static int acp63_pdm_dma_open(struct snd_soc_component *component, if (!pdm_data) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) runtime->hw = acp63_pdm_hardware_capture;
ret = snd_pcm_hw_constraint_integer(runtime, @@ -206,7 +206,7 @@ static int acp63_pdm_dma_open(struct snd_soc_component *component,
acp63_enable_pdm_interrupts(adata);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) adata->capture_stream = substream;
pdm_data->acp63_base = adata->acp63_base; diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 2f630753278dc..0f253912f032e 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -228,7 +228,7 @@ static int acp63_sdw_dma_open(struct snd_soc_component *component, if (!stream) return -ENOMEM;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) runtime->hw = acp63_sdw_hardware_playback; else runtime->hw = acp63_sdw_hardware_capture; @@ -270,7 +270,7 @@ static int acp63_sdw_dma_hw_params(struct snd_soc_component *component, sdw_data->sdw0_dma_stream[stream_id] = substream; water_mark_size_reg = sdw0_dma_ring_buf_reg[stream_id].water_mark_size_reg; acp_ext_intr_cntl_reg = ACP_EXTERNAL_INTR_CNTL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) irq_mask = BIT(SDW0_DMA_TX_IRQ_MASK(stream_id)); else irq_mask = BIT(SDW0_DMA_RX_IRQ_MASK(stream_id)); diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index e7f2a05e802cf..8debcd2487fe6 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -86,7 +86,7 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, adata = snd_soc_dai_get_drvdata(dai); pinfo = snd_soc_card_get_drvdata(card); if (pinfo) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) rtd->i2s_instance = pinfo->play_i2s_instance; else rtd->i2s_instance = pinfo->cap_i2s_instance; @@ -110,7 +110,7 @@ static int acp3x_i2s_hwparams(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_ITER; @@ -163,7 +163,7 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: rtd->bytescount = acp_get_byte_count(rtd, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: water_val = @@ -209,7 +209,7 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_val = mmACP_BTTDM_ITER; diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 3a50558f67516..0a6d63db2e52f 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -113,7 +113,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
addr = rtd->dma_addr;
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: val = ACP_SRAM_BT_PB_PTE_OFFSET; @@ -152,7 +152,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction) addr += PAGE_SIZE; }
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: reg_dma_size = mmACP_BT_TX_DMA_SIZE; @@ -222,7 +222,7 @@ static int acp3x_dma_open(struct snd_soc_component *component, if (!i2s_data) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) runtime->hw = acp3x_pcm_hardware_playback; else runtime->hw = acp3x_pcm_hardware_capture; @@ -261,7 +261,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, return -EINVAL;
if (pinfo) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { rtd->i2s_instance = pinfo->play_i2s_instance; switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: @@ -334,7 +334,7 @@ static int acp3x_dma_close(struct snd_soc_component *component, if (!ins) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (ins->i2s_instance) { case I2S_BT_INSTANCE: adata->play_stream = NULL; diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 7702f628ecd68..2200c4f5ebfdd 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -126,7 +126,7 @@ static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, { u64 byte_count;
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { switch (rtd->i2s_instance) { case I2S_BT_INSTANCE: byte_count = rv_readl(rtd->acp3x_base + diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index c3b47e9bd2392..0bcd20ec70a6e 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -215,7 +215,7 @@ static int acp_pdm_dma_open(struct snd_soc_component *component, if (!pdm_data) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) runtime->hw = acp_pdm_hardware_capture;
ret = snd_pcm_hw_constraint_integer(runtime, @@ -228,7 +228,7 @@ static int acp_pdm_dma_open(struct snd_soc_component *component,
enable_pdm_interrupts(adata->acp_base);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) adata->capture_stream = substream;
pdm_data->acp_base = adata->acp_base; diff --git a/sound/soc/amd/vangogh/acp5x-i2s.c b/sound/soc/amd/vangogh/acp5x-i2s.c index 7dbe33f4b8678..9e05816d4f5c1 100644 --- a/sound/soc/amd/vangogh/acp5x-i2s.c +++ b/sound/soc/amd/vangogh/acp5x-i2s.c @@ -101,7 +101,7 @@ static int acp5x_i2s_hwparams(struct snd_pcm_substream *substream, adata = snd_soc_dai_get_drvdata(dai); pinfo = snd_soc_card_get_drvdata(card); if (pinfo) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) rtd->i2s_instance = pinfo->play_i2s_instance; else rtd->i2s_instance = pinfo->cap_i2s_instance; @@ -125,7 +125,7 @@ static int acp5x_i2s_hwparams(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_ITER; @@ -249,7 +249,7 @@ static int acp5x_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: rtd->bytescount = acp_get_byte_count(rtd, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_HS_INSTANCE: water_val = @@ -297,7 +297,7 @@ static int acp5x_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (rtd->i2s_instance) { case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_ITER; diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 491b16e52a72a..73b4d879bbfcc 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -108,7 +108,7 @@ static void config_acp5x_dma(struct i2s_stream_instance *rtd, int direction) dma_addr_t addr;
addr = rtd->dma_addr; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { switch (rtd->i2s_instance) { case I2S_HS_INSTANCE: val = ACP_SRAM_HS_PB_PTE_OFFSET; @@ -146,7 +146,7 @@ static void config_acp5x_dma(struct i2s_stream_instance *rtd, int direction) addr += PAGE_SIZE; }
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { switch (rtd->i2s_instance) { case I2S_HS_INSTANCE: reg_dma_size = ACP_HS_TX_DMA_SIZE; @@ -217,7 +217,7 @@ static int acp5x_dma_open(struct snd_soc_component *component, if (!i2s_data) return -ENOMEM;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) runtime->hw = acp5x_pcm_hardware_playback; else runtime->hw = acp5x_pcm_hardware_capture; @@ -255,7 +255,7 @@ static int acp5x_dma_hw_params(struct snd_soc_component *component, return -EINVAL;
if (pinfo) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { rtd->i2s_instance = pinfo->play_i2s_instance; switch (rtd->i2s_instance) { case I2S_HS_INSTANCE: @@ -328,7 +328,7 @@ static int acp5x_dma_close(struct snd_soc_component *component, ins = substream->runtime->private_data; if (!ins) return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (ins->i2s_instance) { case I2S_HS_INSTANCE: adata->play_stream = NULL; diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index ac1936a8c43ff..0d08e734f0db6 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -154,7 +154,7 @@ static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, { union acp_dma_count byte_count;
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { switch (rtd->i2s_instance) { case I2S_HS_INSTANCE: byte_count.bcount.high = diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index 72c4591e451bd..74ecea36e3dd5 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -191,7 +191,7 @@ static int acp6x_pdm_dma_open(struct snd_soc_component *component, if (!pdm_data) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) runtime->hw = acp6x_pdm_hardware_capture;
ret = snd_pcm_hw_constraint_integer(runtime, @@ -204,7 +204,7 @@ static int acp6x_pdm_dma_open(struct snd_soc_component *component,
acp6x_enable_pdm_interrupts(adata->acp6x_base);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) adata->capture_stream = substream;
pdm_data->acp6x_base = adata->acp6x_base;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/bcm/bcm2835-i2s.c | 8 ++++---- sound/soc/bcm/bcm63xx-i2s-whistler.c | 4 ++-- sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 ++-- sound/soc/bcm/cygnus-pcm.c | 18 +++++++++--------- sound/soc/bcm/cygnus-ssp.c | 12 ++++++------ 5 files changed, 23 insertions(+), 23 deletions(-)
diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 9bda6499e66e1..54fcf85a1158c 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -628,10 +628,10 @@ static int bcm2835_i2s_prepare(struct snd_pcm_substream *substream, */ regmap_read(dev->i2s_regmap, BCM2835_I2S_CS_A_REG, &cs_reg);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK + if (snd_pcm_is_playback(substream) && !(cs_reg & BCM2835_I2S_TXE)) bcm2835_i2s_clear_fifos(dev, true, false); - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + else if (snd_pcm_is_capture(substream) && (cs_reg & BCM2835_I2S_RXD)) bcm2835_i2s_clear_fifos(dev, false, true);
@@ -644,7 +644,7 @@ static void bcm2835_i2s_stop(struct bcm2835_i2s_dev *dev, { uint32_t mask;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) mask = BCM2835_I2S_RXON; else mask = BCM2835_I2S_TXON; @@ -669,7 +669,7 @@ static int bcm2835_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: bcm2835_i2s_start_clock(dev);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) mask = BCM2835_I2S_RXON; else mask = BCM2835_I2S_TXON; diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c index c64609718738b..0980e35d12830 100644 --- a/sound/soc/bcm/bcm63xx-i2s-whistler.c +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -93,7 +93,7 @@ static int bcm63xx_i2s_startup(struct snd_pcm_substream *substream, struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); struct regmap *regmap_i2s = i2s_priv->regmap_i2s;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(regmap_i2s, I2S_TX_CFG, I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, @@ -146,7 +146,7 @@ static void bcm63xx_i2s_shutdown(struct snd_pcm_substream *substream, struct bcm_i2s_priv *i2s_priv = snd_soc_dai_get_drvdata(dai); struct regmap *regmap_i2s = i2s_priv->regmap_i2s;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(regmap_i2s, I2S_TX_CFG, I2S_TX_OUT_R | I2S_TX_DATA_ALIGNMENT | I2S_TX_DATA_ENABLE | I2S_TX_CLOCK_ENABLE, 0); diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index 018f2372e892c..e8542b2009f63 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -81,7 +81,7 @@ static int bcm63xx_pcm_trigger(struct snd_soc_component *component, i2s_priv = dev_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0)->dev); regmap_i2s = i2s_priv->regmap_i2s;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (cmd) { case SNDRV_PCM_TRIGGER_START: regmap_update_bits(regmap_i2s, @@ -153,7 +153,7 @@ static int bcm63xx_pcm_prepare(struct snd_soc_component *component, dma_desc->dma_addr = runtime->dma_addr; dma_desc->dma_area = runtime->dma_area;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regaddr_desclen = I2S_TX_DESC_IFF_LEN; regaddr_descaddr = I2S_TX_DESC_IFF_ADDR; } else { diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c index 2d1e241d83673..3dc204d0a34c6 100644 --- a/sound/soc/bcm/cygnus-pcm.c +++ b/sound/soc/bcm/cygnus-pcm.c @@ -252,7 +252,7 @@ static int configure_ringbuf_regs(struct snd_pcm_substream *substream) aio = cygnus_dai_get_dma_data(substream);
/* Map the ssp portnum to a set of ring buffers. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { p_rbuf = &aio->play_rb_regs;
switch (aio->portnum) { @@ -299,7 +299,7 @@ static struct ringbuf_regs *get_ringbuf(struct snd_pcm_substream *substream)
aio = cygnus_dai_get_dma_data(substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) p_rbuf = &aio->play_rb_regs; else p_rbuf = &aio->capture_rb_regs; @@ -317,7 +317,7 @@ static void enable_intr(struct snd_pcm_substream *substream) /* The port number maps to the bit position to be cleared */ clear_mask = BIT(aio->portnum);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* Clear interrupt status before enabling them */ writel(clear_mask, aio->cygaud->audio + ESR0_STATUS_CLR_OFFSET); writel(clear_mask, aio->cygaud->audio + ESR1_STATUS_CLR_OFFSET); @@ -354,7 +354,7 @@ static void disable_intr(struct snd_pcm_substream *substream) /* The port number maps to the bit position to be set */ set_mask = BIT(aio->portnum);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* Mask the interrupts of the given port*/ writel(set_mask, aio->cygaud->audio + ESR0_MASK_SET_OFFSET); writel(set_mask, aio->cygaud->audio + ESR1_MASK_SET_OFFSET); @@ -404,7 +404,7 @@ static void cygnus_pcm_period_elapsed(struct snd_pcm_substream *substream) */ snd_pcm_period_elapsed(substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* Set the ring buffer to full */ regval = readl(aio->cygaud->audio + p_rbuf->rdaddr); regval = regval ^ BIT(31); @@ -597,7 +597,7 @@ static int cygnus_pcm_open(struct snd_soc_component *component, * Keep track of which substream belongs to which port. * This info is needed by snd_pcm_period_elapsed() in irq_handler */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) aio->play_stream = substream; else aio->capture_stream = substream; @@ -615,7 +615,7 @@ static int cygnus_pcm_close(struct snd_soc_component *component,
dev_dbg(snd_soc_rtd_to_cpu(rtd, 0)->dev, "%s port %d\n", __func__, aio->portnum);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) aio->play_stream = NULL; else aio->capture_stream = NULL; @@ -652,7 +652,7 @@ static int cygnus_pcm_prepare(struct snd_soc_component *component,
start = runtime->dma_addr;
- is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0; + is_play = snd_pcm_is_playback(substream);
ringbuf_set_initial(aio->cygaud->audio, p_rbuf, is_play, start, periodsize, bufsize); @@ -674,7 +674,7 @@ static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_soc_component *component, * index (for capture). Report this value back to the asoc framework. */ p_rbuf = get_ringbuf(substream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) cur = readl(aio->cygaud->audio + p_rbuf->rdaddr); else cur = readl(aio->cygaud->audio + p_rbuf->wraddr); diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 90088516fed01..73c231f5c1ed8 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -642,7 +642,7 @@ static int cygnus_ssp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { value = readl(aio->cygaud->audio + aio->regs.bf_sourcech_cfg); value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE); value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE); @@ -736,7 +736,7 @@ static int cygnus_ssp_startup(struct snd_pcm_substream *substream, struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
snd_soc_dai_set_dma_data(dai, substream, aio); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) aio->clk_trace.play_en = true; else aio->clk_trace.cap_en = true; @@ -754,7 +754,7 @@ static void cygnus_ssp_shutdown(struct snd_pcm_substream *substream, { struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) aio->clk_trace.play_en = false; else aio->clk_trace.cap_en = false; @@ -770,7 +770,7 @@ static void cygnus_ssp_shutdown(struct snd_pcm_substream *substream, return; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (aio->clk_trace.play_clk_en) { clk_disable_unprepare(aio->cygaud-> audio_clk[val]); @@ -932,7 +932,7 @@ static int cygnus_ssp_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) audio_ssp_out_enable(aio); else audio_ssp_in_enable(aio); @@ -943,7 +943,7 @@ static int cygnus_ssp_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) audio_ssp_out_disable(aio); else audio_ssp_in_disable(aio);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/dwc/dwc-i2s.c | 20 ++++++++++---------- sound/soc/dwc/dwc-pcm.c | 6 +++--- 2 files changed, 13 insertions(+), 13 deletions(-)
diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index c04466f5492e9..874d5bf2985e6 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -42,7 +42,7 @@ static inline void i2s_disable_channels(struct dw_i2s_dev *dev, u32 stream) { u32 i = 0;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { for (i = 0; i < 4; i++) i2s_write_reg(dev->i2s_base, TER(i), 0); } else { @@ -55,7 +55,7 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) { u32 i = 0;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { for (i = 0; i < 4; i++) i2s_read_reg(dev->i2s_base, TOR(i)); } else { @@ -69,7 +69,7 @@ static inline void i2s_disable_irqs(struct dw_i2s_dev *dev, u32 stream, { u32 i, irq;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { for (i = 0; i < (chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30); @@ -87,7 +87,7 @@ static inline void i2s_enable_irqs(struct dw_i2s_dev *dev, u32 stream, { u32 i, irq;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { for (i = 0; i < (chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); @@ -156,7 +156,7 @@ static void i2s_enable_dma(struct dw_i2s_dev *dev, u32 stream) u32 dma_reg = i2s_read_reg(dev->i2s_base, I2S_DMACR);
/* Enable DMA handshake for stream */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) dma_reg |= I2S_DMAEN_TXBLOCK; else dma_reg |= I2S_DMAEN_RXBLOCK; @@ -169,7 +169,7 @@ static void i2s_disable_dma(struct dw_i2s_dev *dev, u32 stream) u32 dma_reg = i2s_read_reg(dev->i2s_base, I2S_DMACR);
/* Disable DMA handshake for stream */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { dma_reg &= ~I2S_DMAEN_TXBLOCK; i2s_write_reg(dev->i2s_base, I2S_RTXDMA, 1); } else { @@ -194,7 +194,7 @@ static void i2s_start(struct dw_i2s_dev *dev,
i2s_write_reg(dev->i2s_base, IER, reg);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) i2s_write_reg(dev->i2s_base, ITER, 1); else i2s_write_reg(dev->i2s_base, IRER, 1); @@ -213,7 +213,7 @@ static void i2s_stop(struct dw_i2s_dev *dev, {
i2s_clear_irqs(dev, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) i2s_write_reg(dev->i2s_base, ITER, 0); else i2s_write_reg(dev->i2s_base, IRER, 0); @@ -253,7 +253,7 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) i2s_disable_channels(dev, stream);
for (ch_reg = 0; ch_reg < (config->chan_nr / 2); ch_reg++) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), dev->xfer_resolution); i2s_write_reg(dev->i2s_base, TFCR(ch_reg), @@ -352,7 +352,7 @@ static int dw_i2s_prepare(struct snd_pcm_substream *substream, { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) i2s_write_reg(dev->i2s_base, TXFFR, 1); else i2s_write_reg(dev->i2s_base, RXFFR, 1); diff --git a/sound/soc/dwc/dwc-pcm.c b/sound/soc/dwc/dwc-pcm.c index a418265c030a5..673218e010607 100644 --- a/sound/soc/dwc/dwc-pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -200,7 +200,7 @@ static int dw_pcm_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { WRITE_ONCE(dev->tx_ptr, 0); rcu_assign_pointer(dev->tx_substream, substream); } else { @@ -211,7 +211,7 @@ static int dw_pcm_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) rcu_assign_pointer(dev->tx_substream, NULL); else rcu_assign_pointer(dev->rx_substream, NULL); @@ -231,7 +231,7 @@ static snd_pcm_uframes_t dw_pcm_pointer(struct snd_soc_component *component, struct dw_i2s_dev *dev = runtime->private_data; snd_pcm_uframes_t pos;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) pos = READ_ONCE(dev->tx_ptr); else pos = READ_ONCE(dev->rx_ptr);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 8 ++++---- sound/soc/fsl/fsl_audmix.c | 2 +- sound/soc/fsl/fsl_dma.c | 10 +++++----- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 4 ++-- sound/soc/fsl/fsl_qmc_audio.c | 10 +++++----- sound/soc/fsl/fsl_sai.c | 8 ++++---- sound/soc/fsl/fsl_spdif.c | 8 ++++---- sound/soc/fsl/fsl_ssi.c | 6 +++--- sound/soc/fsl/fsl_xcvr.c | 8 ++++---- sound/soc/fsl/imx-audmix.c | 4 ++-- sound/soc/fsl/imx-hdmi.c | 2 +- sound/soc/fsl/imx-pcm-fiq.c | 8 ++++---- sound/soc/fsl/imx-pcm-rpmsg.c | 24 ++++++++++++------------ sound/soc/fsl/lpc3xxx-i2s.c | 10 +++++----- sound/soc/fsl/mpc5200_dma.c | 10 +++++----- sound/soc/fsl/mpc5200_dma.h | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- 20 files changed, 66 insertions(+), 66 deletions(-)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index f6c3aeff0d8ea..29f32bf65c19f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -171,7 +171,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); struct codec_priv *codec_priv; struct snd_soc_dai *codec_dai; struct cpu_priv *cpu_priv = &priv->cpu_priv; diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index b793263291dc8..42d08db984e17 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -719,7 +719,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { config.input_format = params_format(params); config.output_format = asrc->asrc_format; config.input_sample_rate = rate; diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index f501f47242fb0..abde5badf8383 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -55,7 +55,7 @@ static void fsl_asrc_dma_complete(void *arg) static int fsl_asrc_dma_prepare_and_submit(struct snd_pcm_substream *substream, struct snd_soc_component *component) { - u8 dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? OUT : IN; + u8 dir = snd_pcm_is_playback(substream) ? OUT : IN; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct device *dev = component->dev; @@ -131,7 +131,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; enum sdma_peripheral_type be_peripheral_type = IMX_DMATYPE_SSI; struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); struct snd_dmaengine_dai_dma_data *dma_params_fe = NULL; struct snd_dmaengine_dai_dma_data *dma_params_be = NULL; struct snd_pcm_runtime *runtime = substream->runtime; @@ -308,7 +308,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, static int fsl_asrc_dma_hw_free(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; u8 dir = tx ? OUT : IN; @@ -329,7 +329,7 @@ static int fsl_asrc_dma_hw_free(struct snd_soc_component *component, static int fsl_asrc_dma_startup(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_dmaengine_dai_dma_data *dma_data; diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 1671a3037c604..1ee3f8f919695 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -283,7 +283,7 @@ static int fsl_audmix_dai_trigger(struct snd_pcm_substream *substream, int cmd, unsigned long lock_flags;
/* Capture stream shall not be handled */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) return 0;
switch (cmd) { diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index c4bc9395dff7d..6754ea7372da0 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -166,7 +166,7 @@ static void fsl_dma_update_pointers(struct fsl_dma_private *dma_private) * system, we also need to update the ESAD bits. We also set (keep) the * snoop bits. See the comments in fsl_dma_hw_params() about snooping. */ - if (dma_private->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dma_private->substream)) { link->source_addr = cpu_to_be32(dma_private->dma_buf_next); #ifdef CONFIG_PHYS_64BIT link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP | @@ -395,7 +395,7 @@ static int fsl_dma_open(struct snd_soc_component *component, dev_err(dev, "can't allocate dma private data\n"); return -ENOMEM; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dma_private->ssi_sxx_phys = dma->ssi_stx_phys; else dma_private->ssi_sxx_phys = dma->ssi_srx_phys; @@ -473,7 +473,7 @@ static int fsl_dma_open(struct snd_soc_component *component,
/* For playback, we want the destination address to be held. For capture, set the source address to be held. */ - mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + mr |= (snd_pcm_is_playback(substream)) ? CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE;
out_be32(&dma_channel->mr, mr); @@ -633,7 +633,7 @@ static int fsl_dma_hw_params(struct snd_soc_component *component, * get more performance by not snooping, and you'll still be * okay. You'll need to update fsl_dma_update_pointers() also. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { link->source_addr = cpu_to_be32(temp_addr); link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP | upper_32_bits(temp_addr)); @@ -683,7 +683,7 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_soc_component *component, * only have 32-bit DMA addresses. This function is typically called * in interrupt context, so we need to optimize it. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { position = in_be32(&dma_channel->sar); #ifdef CONFIG_PHYS_64BIT position |= (u64)(in_be32(&dma_channel->satr) & diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f309120918..ba577bd9ab477 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1461,7 +1461,7 @@ static int fsl_easrc_hw_params(struct snd_pcm_substream *substream, * Set the input and output ratio so we can compute * the resampling ratio in RS_LOW/HIGH */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ctx_priv->in_params.sample_rate = rate; ctx_priv->in_params.sample_format = format; ctx_priv->out_params.sample_rate = easrc->asrc_rate; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0d8a01da9bdd..000abee37d0da 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -537,7 +537,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); u32 width = params_width(params); u32 channels = params_channels(params); u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); @@ -758,7 +758,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); unsigned long lock_flags;
esai_priv->channels[tx] = substream->runtime->channels; diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c index 8668abd352080..bc7eef100c8ad 100644 --- a/sound/soc/fsl/fsl_qmc_audio.c +++ b/sound/soc/fsl/fsl_qmc_audio.c @@ -250,7 +250,7 @@ static int qmc_audio_pcm_trigger(struct snd_soc_component *component, switch (cmd) { case SNDRV_PCM_TRIGGER_START: bitmap_zero(prtd->chans_pending, 64); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { for (i = 0; i < prtd->channels; i++) prtd->qmc_dai->chans[i].prtd_tx = prtd;
@@ -513,7 +513,7 @@ static int qmc_dai_constraints_interleaved(struct snd_pcm_substream *substream, u64 access; int ret;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { hw_rule_channels_by_format = qmc_dai_hw_rule_capture_channels_by_format; hw_rule_format_by_channels = qmc_dai_hw_rule_capture_format_by_channels; frame_bits = qmc_dai->nb_rx_ts * 8; @@ -566,7 +566,7 @@ static int qmc_dai_constraints_noninterleaved(struct snd_pcm_substream *substrea u64 access; int ret;
- frame_bits = (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ? + frame_bits = snd_pcm_is_capture(substream) ? qmc_dai->nb_rx_ts * 8 : qmc_dai->nb_tx_ts * 8; ret = snd_pcm_hw_constraint_single(substream->runtime, SNDRV_PCM_HW_PARAM_FRAME_BITS, @@ -637,7 +637,7 @@ static int qmc_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { chan_param.mode = QMC_TRANSPARENT; chan_param.transp.max_rx_buf_size = params_period_bytes(params) / nb_chans_used; for (i = 0; i < nb_chans_used; i++) { @@ -672,7 +672,7 @@ static int qmc_dai_trigger(struct snd_pcm_substream *substream, int cmd, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { direction = QMC_CHAN_WRITE; nb_chans_used = qmc_dai->nb_chans_used_tx; } else { diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d03b0172b8ad2..c0bc992e22b5f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -525,7 +525,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); unsigned int ofs = sai->soc_data->reg_offset; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); unsigned int channels = params_channels(params); struct snd_dmaengine_dai_dma_data *dma_params; struct fsl_sai_dl_cfg *dl_cfg = sai->dl_cfg; @@ -721,7 +721,7 @@ static int fsl_sai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); unsigned int ofs = sai->soc_data->reg_offset;
/* Clear xMR to avoid channel swap with mclk_with_tere enabled case */ @@ -783,7 +783,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); unsigned int ofs = sai->soc_data->reg_offset;
- bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); int adir = tx ? RX : TX; int dir = tx ? TX : RX; u32 xcsr; @@ -868,7 +868,7 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); int ret;
/* diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index a63121c888e02..d860e54b1f5e6 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -622,7 +622,7 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0); }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { scr = SCR_TXFIFO_AUTOSYNC | SCR_TXFIFO_CTRL_NORMAL | SCR_TXSEL_NORMAL | SCR_USRC_SEL_CHIP | SCR_TXFIFO_FSEL_IF8; @@ -650,7 +650,7 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, struct regmap *regmap = spdif_priv->regmap; u32 scr, mask;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { scr = 0; mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | @@ -706,7 +706,7 @@ static int fsl_spdif_hw_params(struct snd_pcm_substream *substream, u32 sample_rate = params_rate(params); int ret = 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = spdif_reparent_rootclk(spdif_priv, sample_rate); if (ret) { dev_err(&pdev->dev, "%s: reparent root clk failed: %d\n", @@ -737,7 +737,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0)); struct regmap *regmap = spdif_priv->regmap; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); u32 intr = SIE_INTR_FOR(tx); u32 dmaen = SCR_DMA_xX_EN(tx);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4ca3a16f7ac0d..ad02a9e0154ac 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -680,7 +680,7 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct snd_soc_dai *dai, struct snd_pcm_hw_params *hw_params) { - bool tx2, tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx2, tx = snd_pcm_is_playback(substream); struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(dai); struct regmap *regs = ssi->regs; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; @@ -805,7 +805,7 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) { - bool tx2, tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx2, tx = snd_pcm_is_playback(substream); struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(dai); struct fsl_ssi_regvals *vals = ssi->regvals; struct regmap *regs = ssi->regs; @@ -1109,7 +1109,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0)); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream);
switch (cmd) { case SNDRV_PCM_TRIGGER_START: diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index bf9a4e90978ef..f89f778a0d89a 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -437,7 +437,7 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_xcvr *xcvr = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); u32 m_ctl = 0, v_ctl = 0; u32 r = substream->runtime->rate, ch = substream->runtime->channels; u32 fout = 32 * r * ch * 10; @@ -562,7 +562,7 @@ static int fsl_xcvr_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_xcvr *xcvr = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); int ret = 0;
if (xcvr->streams & BIT(substream->stream)) { @@ -614,7 +614,7 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_xcvr *xcvr = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); u32 mask = 0, val = 0; int ret;
@@ -662,7 +662,7 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct fsl_xcvr *xcvr = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); int ret;
switch (cmd) { diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 6fbcf33fd0dea..8e03eadc33888 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -74,7 +74,7 @@ static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct device *dev = rtd->card->dev; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; u32 channels = params_channels(params); int ret, dir; @@ -113,7 +113,7 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct device *dev = rtd->card->dev; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; int ret;
diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index fe47b439a8183..b6ce395f91bbf 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -34,7 +34,7 @@ static int imx_hdmi_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct imx_hdmi_data *data = snd_soc_card_get_drvdata(rtd->card); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct snd_soc_card *card = rtd->card; struct device *dev = card->dev; diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 3391430e42532..9005150012c15 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -53,7 +53,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
get_fiq_regs(®s);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) iprtd->offset = regs.ARM_r8 & 0xffff; else iprtd->offset = regs.ARM_r9 & 0xffff; @@ -93,7 +93,7 @@ static int snd_imx_pcm_prepare(struct snd_soc_component *component, struct pt_regs regs;
get_fiq_regs(®s); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16; else regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16; @@ -115,7 +115,7 @@ static int snd_imx_pcm_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) atomic_set(&iprtd->playing, 1); else atomic_set(&iprtd->capturing, 1); @@ -127,7 +127,7 @@ static int snd_imx_pcm_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) atomic_set(&iprtd->playing, 0); else atomic_set(&iprtd->capturing, 0); diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index b0944a07ab470..3653eed2d5bc9 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -142,7 +142,7 @@ static int imx_rpmsg_pcm_hw_params(struct snd_soc_component *component, struct rpmsg_info *info = dev_get_drvdata(component->dev); struct rpmsg_msg *msg;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_HW_PARAM]; msg->s_msg.header.cmd = TX_HW_PARAM; } else { @@ -195,7 +195,7 @@ static snd_pcm_uframes_t imx_rpmsg_pcm_pointer(struct snd_soc_component *compone unsigned int pos = 0; int buffer_tail = 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) msg = &info->msg[TX_PERIOD_DONE + MSG_TYPE_A_NUM]; else msg = &info->msg[RX_PERIOD_DONE + MSG_TYPE_A_NUM]; @@ -214,7 +214,7 @@ static void imx_rpmsg_timer_callback(struct timer_list *t) struct rpmsg_info *info = stream_timer->info; struct rpmsg_msg *msg;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_PERIOD_DONE + MSG_TYPE_A_NUM]; msg->s_msg.header.cmd = TX_PERIOD_DONE; } else { @@ -237,7 +237,7 @@ static int imx_rpmsg_pcm_open(struct snd_soc_component *component, int ret = 0; int cmd;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_OPEN]; msg->s_msg.header.cmd = TX_OPEN;
@@ -291,7 +291,7 @@ static int imx_rpmsg_pcm_close(struct snd_soc_component *component, /* Flush work in workqueue to make TX_CLOSE is the last message */ flush_workqueue(info->rpmsg_wq);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_CLOSE]; msg->s_msg.header.cmd = TX_CLOSE; } else { @@ -353,7 +353,7 @@ static int imx_rpmsg_prepare_and_submit(struct snd_soc_component *component, struct rpmsg_info *info = dev_get_drvdata(component->dev); struct rpmsg_msg *msg;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_BUFFER]; msg->s_msg.header.cmd = TX_BUFFER; } else { @@ -382,7 +382,7 @@ static int imx_rpmsg_async_issue_pending(struct snd_soc_component *component, struct rpmsg_info *info = dev_get_drvdata(component->dev); struct rpmsg_msg *msg;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_START]; msg->s_msg.header.cmd = TX_START; } else { @@ -399,7 +399,7 @@ static int imx_rpmsg_restart(struct snd_soc_component *component, struct rpmsg_info *info = dev_get_drvdata(component->dev); struct rpmsg_msg *msg;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_RESTART]; msg->s_msg.header.cmd = TX_RESTART; } else { @@ -416,7 +416,7 @@ static int imx_rpmsg_pause(struct snd_soc_component *component, struct rpmsg_info *info = dev_get_drvdata(component->dev); struct rpmsg_msg *msg;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_PAUSE]; msg->s_msg.header.cmd = TX_PAUSE; } else { @@ -434,7 +434,7 @@ static int imx_rpmsg_terminate_all(struct snd_soc_component *component, struct rpmsg_msg *msg; int cmd;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_TERMINATE]; msg->s_msg.header.cmd = TX_TERMINATE; /* Clear buffer count*/ @@ -530,7 +530,7 @@ static int imx_rpmsg_pcm_ack(struct snd_soc_component *component, if (!rpmsg->force_lpa) return 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { msg = &info->msg[TX_PERIOD_DONE + MSG_TYPE_A_NUM]; msg->s_msg.header.cmd = TX_PERIOD_DONE; } else { @@ -559,7 +559,7 @@ static int imx_rpmsg_pcm_ack(struct snd_soc_component *component, info->notify_updated[substream->stream] = true; spin_unlock_irqrestore(&info->lock[substream->stream], flags);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) avail = snd_pcm_playback_hw_avail(runtime); else avail = snd_pcm_capture_hw_avail(runtime); diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index c65c17dfa1747..536ace6de0088 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -75,7 +75,7 @@ static int lpc3xxx_i2s_startup(struct snd_pcm_substream *substream, struct snd_s
guard(mutex)(&i2s_info_p->lock);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) flag = I2S_PLAYBACK_FLAG; else flag = I2S_CAPTURE_FLAG; @@ -107,7 +107,7 @@ static void lpc3xxx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd
guard(mutex)(&i2s_info_p->lock);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { flag = I2S_PLAYBACK_FLAG; regmap_write(regs, LPC3XXX_REG_I2S_TX_RATE, 0); regmap_update_bits(regs, LPC3XXX_REG_I2S_DAO, stop_bits, stop_bits); @@ -197,7 +197,7 @@ static int lpc3xxx_i2s_hw_params(struct snd_pcm_substream *substream, dev_dbg(dev, "Channels : %d\n", params_channels(params)); dev_dbg(dev, "Data format : %s\n", "I2S");
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_write(regs, LPC3XXX_REG_I2S_DMA1, LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4)); regmap_write(regs, LPC3XXX_REG_I2S_TX_RATE, (clkx << 8) | clky); @@ -223,7 +223,7 @@ static int lpc3xxx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(regs, LPC3XXX_REG_I2S_DAO, LPC3XXX_I2S_STOP, LPC3XXX_I2S_STOP); else @@ -234,7 +234,7 @@ static int lpc3xxx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(regs, LPC3XXX_REG_I2S_DAO, (LPC3XXX_I2S_RESET | LPC3XXX_I2S_STOP), 0); else diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 345f338251ace..07ddc3cd31890 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -137,7 +137,7 @@ static int psc_dma_trigger(struct snd_soc_component *component, */ spin_lock_irqsave(&psc_dma->lock, flags);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) bcom_gen_bd_rx_reset(s->bcom_task); else bcom_gen_bd_tx_reset(s->bcom_task); @@ -160,7 +160,7 @@ static int psc_dma_trigger(struct snd_soc_component *component,
spin_lock_irqsave(&psc_dma->lock, flags); bcom_disable(s->bcom_task); - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) bcom_gen_bd_rx_reset(s->bcom_task); else bcom_gen_bd_tx_reset(s->bcom_task); @@ -219,7 +219,7 @@ static int psc_dma_open(struct snd_soc_component *component,
dev_dbg(psc_dma->dev, "psc_dma_open(substream=%p)\n", substream);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) s = &psc_dma->capture; else s = &psc_dma->playback; @@ -246,7 +246,7 @@ static int psc_dma_close(struct snd_soc_component *component,
dev_dbg(psc_dma->dev, "psc_dma_close(substream=%p)\n", substream);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) s = &psc_dma->capture; else s = &psc_dma->playback; @@ -271,7 +271,7 @@ psc_dma_pointer(struct snd_soc_component *component, struct psc_dma_stream *s; dma_addr_t count;
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) s = &psc_dma->capture; else s = &psc_dma->playback; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index d7ee33b5b9a8d..42460f2b3906d 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -77,7 +77,7 @@ struct psc_dma { static inline struct psc_dma_stream * to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) { - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) return &psc_dma->capture; return &psc_dma->playback; } diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 0423cf43c7a02..cb96b0ff74396 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -151,7 +151,7 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
/* Determine the set of enable bits to turn on */ s->ac97_slot_bits = (params_channels(params) == 1) ? 0x100 : 0x300; - if (substream->pstr->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream->pstr->stream)) s->ac97_slot_bits <<= 16; return 0; }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/mxs/mxs-saif.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 3e3a62df3d7e3..2a3d8038d4362 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -489,7 +489,7 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, }
/* Tx/Rx config */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* enable TX mode */ scr &= ~BM_SAIF_CTRL_READ_MODE; } else { @@ -560,7 +560,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, __raw_writel(BM_SAIF_CTRL_RUN, master_saif->base + SAIF_CTRL + MXS_SET_ADDR);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { /* * write data to saif data register to trigger * the transfer.
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/pxa/mmp-sspa.c | 6 +++--- sound/soc/pxa/pxa-ssp.c | 6 +++--- sound/soc/pxa/pxa2xx-ac97.c | 6 +++--- sound/soc/pxa/pxa2xx-i2s.c | 8 ++++---- 4 files changed, 13 insertions(+), 13 deletions(-)
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index abfaf3cdf5bb6..229b1ebe720d9 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -266,7 +266,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, params_channels(params) * bits); }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { __raw_writel(sspa_ctrl, sspa->tx_base + SSPA_CTL); __raw_writel(0x1, sspa->tx_base + SSPA_FIFO_UL); } else { @@ -296,7 +296,7 @@ static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd, if (!sspa->running_cnt) mmp_sspa_rx_enable(sspa);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) mmp_sspa_tx_enable(sspa);
sspa->running_cnt++; @@ -307,7 +307,7 @@ static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sspa->running_cnt--;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) mmp_sspa_tx_disable(sspa);
/* have no capture stream, disable rx port */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index b8a3cb8b75978..82f91f951619a 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -88,7 +88,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) return -ENOMEM; - dma->chan_name = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + dma->chan_name = snd_pcm_is_playback(substream) ? "tx" : "rx";
snd_soc_dai_set_dma_data(cpu_dai, substream, dma); @@ -551,7 +551,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, */ pxa_ssp_set_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), - substream->stream == SNDRV_PCM_STREAM_PLAYBACK, dma_data); + snd_pcm_is_playback(substream), dma_data);
/* we can only change the settings if the port is not in use */ if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) @@ -683,7 +683,7 @@ static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream, if (value && (sscr0 & SSCR0_SSE)) pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (value) sscr1 |= SSCR1_TSRE; else diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 80e0ea0ec9fb3..9a8e08b30ebfa 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -95,7 +95,7 @@ static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, { struct snd_dmaengine_dai_dma_data *dma_data;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dma_data = &pxa2xx_ac97_pcm_stereo_out; else dma_data = &pxa2xx_ac97_pcm_stereo_in; @@ -110,7 +110,7 @@ static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream, { struct snd_dmaengine_dai_dma_data *dma_data;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dma_data = &pxa2xx_ac97_pcm_aux_mono_out; else dma_data = &pxa2xx_ac97_pcm_aux_mono_in; @@ -123,7 +123,7 @@ static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream, static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return -ENODEV; snd_soc_dai_set_dma_data(cpu_dai, substream, &pxa2xx_ac97_pcm_mic_mono_in); diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 849fbf176a70f..664116396e8ff 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -163,7 +163,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, clk_ena = 1; pxa_i2s_wait();
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dma_data = &pxa2xx_i2s_pcm_stereo_out; else dma_data = &pxa2xx_i2s_pcm_stereo_in; @@ -179,7 +179,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, writel(readl(i2s_reg_base + SACR0) | (SACR0_RFTH(14) | SACR0_TFTH(1)), i2s_reg_base + SACR0); writel(readl(i2s_reg_base + SACR1) | (pxa_i2s.fmt), i2s_reg_base + SACR1); } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) writel(readl(i2s_reg_base + SAIMR) | (SAIMR_TFS), i2s_reg_base + SAIMR); else writel(readl(i2s_reg_base + SAIMR) | (SAIMR_RFS), i2s_reg_base + SAIMR); @@ -218,7 +218,7 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) writel(readl(i2s_reg_base + SACR1) & (~SACR1_DRPL), i2s_reg_base + SACR1); else writel(readl(i2s_reg_base + SACR1) & (~SACR1_DREC), i2s_reg_base + SACR1); @@ -240,7 +240,7 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { writel(readl(i2s_reg_base + SACR1) | (SACR1_DRPL), i2s_reg_base + SACR1); writel(readl(i2s_reg_base + SAIMR) & (~SAIMR_TFS), i2s_reg_base + SAIMR); } else {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sti/sti_uniperif.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39cf..75520b8827ca1 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -337,7 +337,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) struct uniperif *uni = priv->dai_data.uni; int ret;
- if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(priv->dai_data.stream)) { ret = uni_player_resume(uni); if (ret) return ret; @@ -358,7 +358,7 @@ static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) struct sti_uniperiph_dai *dai_data = &priv->dai_data;
/* DMA settings*/ - if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(priv->dai_data.stream)) snd_soc_dai_init_dma_data(dai, &dai_data->dma_data, NULL); else snd_soc_dai_init_dma_data(dai, NULL, &dai_data->dma_data); @@ -440,7 +440,7 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, dai_data->uni = uni; dai_data->stream = dev_data->stream;
- if (priv->dai_data.stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(priv->dai_data.stream)) { ret = uni_player_init(priv->pdev, uni); stream = &dai->playback; } else {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index a96aa308681a2..d6654fc9f1a7c 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -813,7 +813,7 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); - bool playback_flg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool playback_flg = snd_pcm_is_playback(substream); u32 cfg1_mask, ier; int ret;
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ad2492efb1cdc..ff8d98e574f2b 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -38,8 +38,8 @@
#define STM_SAI_DAI_NAME_SIZE 15
-#define STM_SAI_IS_PLAYBACK(ip) ((ip)->dir == SNDRV_PCM_STREAM_PLAYBACK) -#define STM_SAI_IS_CAPTURE(ip) ((ip)->dir == SNDRV_PCM_STREAM_CAPTURE) +#define STM_SAI_IS_PLAYBACK(ip) snd_pcm_is_playback((ip)->dir) +#define STM_SAI_IS_CAPTURE(ip) snd_pcm_is_capture((ip)->dir)
#define STM_SAI_A_ID 0x0 #define STM_SAI_B_ID 0x1 @@ -1406,7 +1406,7 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->spdif = false; if (of_property_present(np, "st,iec60958")) { if (!STM_SAI_HAS_SPDIF(sai) || - sai->dir == SNDRV_PCM_STREAM_CAPTURE) { + STM_SAI_IS_CAPTURE(sai)) { dev_err(&pdev->dev, "S/PDIF IEC60958 not supported\n"); return -EINVAL; }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sof/ipc4-pcm.c | 6 +++--- sound/soc/sof/ipc4-topology.c | 10 +++++----- sound/soc/sof/sof-audio.c | 8 ++++---- 3 files changed, 12 insertions(+), 12 deletions(-)
diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 4df2be3d39eba..52e6983acba64 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -345,7 +345,7 @@ static int sof_ipc4_chain_dma_trigger(struct snd_sof_dev *sdev, msg.extension |= pipeline->msg.extension; }
- if (direction == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(direction)) { /* * For ChainDMA the DMA ids are unique with the following mapping: * playback: 0 - (num_playback_streams - 1) @@ -681,7 +681,7 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, if (pipeline->use_chain_dma) return 0;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { if (sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->output_pin_fmts, available_fmt->num_output_formats)) { @@ -1044,7 +1044,7 @@ static int sof_ipc4_pcm_pointer(struct snd_soc_component *component, /* Wrap the dai counter at the boundary where the host counter wraps */ div64_u64_rem(dai_cnt, time_info->boundary, &dai_cnt);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { head_cnt = host_cnt; tail_cnt = dai_cnt; } else { diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 87be7f16e8c2b..ce14acb6770eb 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -511,7 +511,7 @@ static int sof_ipc4_widget_setup_pcm(struct snd_sof_widget *swidget) if (ret) goto free_available_fmt;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { struct snd_sof_pcm_stream *sps = &spcm->stream[dir];
sof_update_ipc_object(scomp, &sps->dsp_max_burst_size_in_ms, @@ -1668,7 +1668,7 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, * of the RATE, CHANNELS, bit depth is static among the formats then * narrow the params to only allow that specific parameter value. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { pin_fmts = available_fmt->output_pin_fmts; num_pin_fmts = available_fmt->num_output_formats; } else { @@ -1783,7 +1783,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, * Use the input_pin_fmts to match pcm params for playback and the output_pin_fmts * for capture. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) ref_params = *fe_params; else ref_params = *pipeline_params; @@ -1828,7 +1828,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, * For playback the pipeline_params needs to be used to find the * input configuration of the copier. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) ref_params = *pipeline_params;
break; @@ -2225,7 +2225,7 @@ static int sof_ipc4_prepare_src_module(struct snd_sof_widget *swidget, * For playback, the SRC sink rate will be configured based on the requested output * format, which is restricted to only deal with DAI's with a single format for now. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK && available_fmt->num_output_formats > 1) { + if (snd_pcm_is_playback(dir) && available_fmt->num_output_formats > 1) { dev_err(sdev->dev, "Invalid number of output formats: %d for SRC %s\n", available_fmt->num_output_formats, swidget->widget->name); return -EINVAL; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 9a52781bf8d8b..9ac03dc5a24d4 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -308,7 +308,7 @@ static int sof_setup_pipeline_connections(struct snd_sof_dev *sdev, * purpose of connecting a pipeline from a host to a DAI in order to receive the DAPM * events. But they are not handled by the firmware. So ignore them. */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(dir)) { for_each_dapm_widgets(list, i, widget) { if (!widget->dobj.private) continue; @@ -623,11 +623,11 @@ sof_walk_widgets_in_order(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, continue;
/* starting widget for playback is AIF type */ - if (dir == SNDRV_PCM_STREAM_PLAYBACK && widget->id != snd_soc_dapm_aif_in) + if (snd_pcm_is_playback(dir) && widget->id != snd_soc_dapm_aif_in) continue;
/* starting widget for capture is DAI type */ - if (dir == SNDRV_PCM_STREAM_CAPTURE && widget->id != snd_soc_dapm_dai_out) + if (snd_pcm_is_capture(dir) && widget->id != snd_soc_dapm_dai_out) continue;
switch (op) { @@ -950,7 +950,7 @@ snd_sof_find_swidget_sname(struct snd_soc_component *scomp, struct snd_sof_widget *swidget; enum snd_soc_dapm_type type;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) type = snd_soc_dapm_aif_in; else type = snd_soc_dapm_aif_out;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sof/intel/hda-dai-ops.c | 2 +- sound/soc/sof/intel/hda-dai.c | 4 ++-- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-loader.c | 2 +- sound/soc/sof/intel/hda-pcm.c | 4 ++-- sound/soc/sof/intel/hda-stream.c | 6 +++--- 6 files changed, 10 insertions(+), 10 deletions(-)
diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 484c761478853..c00fc981f8059 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -198,7 +198,7 @@ static unsigned int hda_calc_stream_format(struct snd_sof_dev *sdev, unsigned int format_val; unsigned int bits;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) link_bps = codec_dai->driver->playback.sig_bits; else link_bps = codec_dai->driver->capture.sig_bits; diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 1c823f9eea570..0b5d3c5693ab0 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -123,7 +123,7 @@ int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_st if (!hlink) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { stream_tag = hdac_stream(hext_stream)->stream_tag; snd_hdac_ext_bus_link_clear_stream_id(hlink, stream_tag); } @@ -174,7 +174,7 @@ static int hda_link_dma_hw_params(struct snd_pcm_substream *substream, hstream = &hext_stream->hstream; stream_tag = hstream->stream_tag;
- if (hext_stream->hstream.direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(hext_stream->hstream.direction)) snd_hdac_ext_bus_link_set_stream_id(hlink, stream_tag);
/* set the hdac_stream in the codec dai */ diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 4c88522d40484..f5be61a6f4ba5 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -542,7 +542,7 @@ static bool hda_dsp_d0i3_streaming_applicable(struct snd_sof_dev *sdev) if (!spcm->stream[dir].d0i3_compatible) return false;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) playback_active = true; } } diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 75f6240cf3e1d..ec46529974a5e 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -262,7 +262,7 @@ int hda_cl_cleanup(struct device *dev, struct snd_dma_buffer *dmab, int sd_offset = SOF_STREAM_SD_OFFSET(hstream); int ret = 0;
- if (hstream->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(hstream->direction)) ret = hda_dsp_stream_spib_config(sdev, hext_stream, HDA_DSP_SPIB_DISABLE, 0); else snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index f6e24edd7adbe..d5a630da5a218 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -237,11 +237,11 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, * All playback streams are DMI L1 capable, capture streams need * pause push/release to be disabled */ - if (hda_always_enable_dmi_l1 && direction == SNDRV_PCM_STREAM_CAPTURE) + if (hda_always_enable_dmi_l1 && snd_pcm_is_capture(direction)) runtime->hw.info &= ~SNDRV_PCM_INFO_PAUSE;
if (hda_always_enable_dmi_l1 || - direction == SNDRV_PCM_STREAM_PLAYBACK || + snd_pcm_is_playback(direction) || spcm->stream[substream->stream].d0i3_compatible) flags |= SOF_HDA_STREAM_DMI_L1_COMPATIBLE;
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 3ac63ce67ab1c..c83b260c35f92 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -33,7 +33,7 @@ EXPORT_SYMBOL_NS(sof_hda_position_quirk, SND_SOC_SOF_INTEL_HDA_COMMON);
static inline const char *hda_hstream_direction_str(struct hdac_stream *hstream) { - if (hstream->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(hstream->direction)) return "Playback"; else return "Capture"; @@ -667,7 +667,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, SOF_HDA_CL_DMA_SD_INT_MASK);
/* read FIFO size */ - if (hstream->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(hstream->direction)) { hstream->fifo_size = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, sd_offset + @@ -1030,7 +1030,7 @@ snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, * is not accurate enough, its update may be completed * earlier than the data written to DDR. */ - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL *
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/qcom/apq8096.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-cpu.c | 12 ++++++------ sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-platform.c | 8 ++++---- sound/soc/qcom/lpass-sc7180.c | 4 ++-- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/qcom/qdsp6/audioreach.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 10 +++++----- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 6 +++--- sound/soc/qcom/qdsp6/q6apm.c | 12 ++++++------ sound/soc/qcom/qdsp6/q6asm-dai.c | 16 ++++++++-------- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 4 ++-- 14 files changed, 42 insertions(+), 42 deletions(-)
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 4f6594cc723ce..a5305f33c32c5 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -46,7 +46,7 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return 0; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, rx_ch_cnt, rx_ch); else diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 9005c85f8c547..5dfcd547cfcd5 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -126,7 +126,7 @@ static int apq8016_lpass_alloc_dma_channel(struct lpass_data *drvdata, const struct lpass_variant *v = drvdata->variant; int chan = 0;
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { chan = find_first_zero_bit(&drvdata->dma_ch_bit_map, v->rdma_channels);
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6f..81036c49bce1b 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -113,7 +113,7 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, * Will not impact if disabled in lpass_cpu_daiops_trigger() * suspend. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); else regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); @@ -185,7 +185,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return ret; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) mode = drvdata->mi2s_playback_sd_mode[id]; else mode = drvdata->mi2s_capture_sd_mode[id]; @@ -249,7 +249,7 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = regmap_fields_write(i2sctl->spkmode, id, LPAIF_I2SCTL_SPKMODE(mode)); if (ret) { @@ -320,7 +320,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, * turn off the shared BCLK while other devices are using * it. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); } else { @@ -345,7 +345,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, * To ensure lpass BCLK/LRCLK is disabled during * device suspend. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); } else { @@ -378,7 +378,7 @@ static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, * the data flow. * (ex: to drop start up pop noise before capture starts). */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); else ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); diff --git a/sound/soc/qcom/lpass-ipq806x.c b/sound/soc/qcom/lpass-ipq806x.c index 5c874139f39d4..dbaaed1f3d8e3 100644 --- a/sound/soc/qcom/lpass-ipq806x.c +++ b/sound/soc/qcom/lpass-ipq806x.c @@ -97,7 +97,7 @@ static int ipq806x_lpass_exit(struct platform_device *pdev)
static int ipq806x_lpass_alloc_dma_channel(struct lpass_data *drvdata, int dir, unsigned int dai_id) { - if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) return IPQ806X_LPAIF_RDMA_CHAN_MI2S; else /* Capture currently not implemented */ return -EINVAL; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index addd2c4bdd3e8..f8e223e73fa02 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -329,7 +329,7 @@ static struct lpaif_dmactl *__lpass_get_dmactl_handle(const struct snd_pcm_subst
switch (cpu_dai->driver->id) { case MI2S_PRIMARY ... MI2S_QUINARY: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dmactl = drvdata->rd_dmactl; else dmactl = drvdata->wr_dmactl; @@ -364,7 +364,7 @@ static int __lpass_get_id(const struct snd_pcm_substream *substream,
switch (cpu_dai->driver->id) { case MI2S_PRIMARY ... MI2S_QUINARY: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) id = pcm_data->dma_ch; else id = pcm_data->dma_ch - v->wrdma_channel_start; @@ -1230,14 +1230,14 @@ static int lpass_platform_copy(struct snd_soc_component *component, void __iomem *dma_buf = (void __iomem *) (rt->dma_area + pos + channel * (rt->dma_bytes / rt->channels));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (is_cdc_dma_port(dai_id)) { ret = copy_from_iter_toio(dma_buf, buf, bytes); } else { if (copy_from_iter((void __force *)dma_buf, bytes, buf) != bytes) ret = -EFAULT; } - } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + } else if (snd_pcm_is_capture(substream)) { if (is_cdc_dma_port(dai_id)) { ret = copy_to_iter_fromio(buf, dma_buf, bytes); } else { diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index e6bcdf6ed7965..6898e9254a78d 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -80,7 +80,7 @@ static int sc7180_lpass_alloc_dma_channel(struct lpass_data *drvdata, int chan = 0;
if (dai_id == LPASS_DP_RX) { - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { chan = find_first_zero_bit(&drvdata->hdmi_dma_ch_bit_map, v->hdmi_rdma_channels);
@@ -89,7 +89,7 @@ static int sc7180_lpass_alloc_dma_channel(struct lpass_data *drvdata, } set_bit(chan, &drvdata->hdmi_dma_ch_bit_map); } else { - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { chan = find_first_zero_bit(&drvdata->dma_ch_bit_map, v->rdma_channels);
diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 47c622327a8d3..d5a1c27652e48 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -115,7 +115,7 @@ static int sc7280_lpass_alloc_dma_channel(struct lpass_data *drvdata,
switch (dai_id) { case MI2S_PRIMARY ... MI2S_QUINARY: - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { chan = find_first_zero_bit(&drvdata->dma_ch_bit_map, v->rdma_channels);
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 4ebaaf736fb98..cd7d99f9b8b40 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -1309,7 +1309,7 @@ int audioreach_map_memory_regions(struct q6apm_graph *graph, unsigned int dir, s void *p; int rc, i;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) data = &graph->rx_data; else data = &graph->tx_data; diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index c9404b5934c7e..26c6051a53a0a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -280,7 +280,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, return ret; }
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { int i; /* Queue the buffers for Capture ONLY after graph is started */ for (i = 0; i < runtime->periods; i++) @@ -306,7 +306,7 @@ static int q6apm_dai_trigger(struct snd_soc_component *component, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* start writing buffers for playback only as we already queued capture buffers */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, 0); break; case SNDRV_PCM_TRIGGER_STOP: @@ -356,9 +356,9 @@ static int q6apm_dai_open(struct snd_soc_component *component, goto err; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) runtime->hw = q6apm_dai_hardware_playback; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) runtime->hw = q6apm_dai_hardware_capture;
/* Ensure that buffer size is a multiple of period size */ @@ -368,7 +368,7 @@ static int q6apm_dai_open(struct snd_soc_component *component, goto err; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, BUFFER_BYTES_MIN, BUFFER_BYTES_MAX); if (ret < 0) { diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c index 9c98a35ad0994..3189a10b2f28a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c +++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c @@ -171,7 +171,7 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s q6apm_graph_stop(dai_data->graph[dai->id]); dai_data->is_port_started[dai->id] = false;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { q6apm_graph_close(dai_data->graph[dai->id]); dai_data->graph[dai->id] = NULL; } @@ -181,7 +181,7 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s * It is recommend to load DSP with source graph first and then sink * graph, so sequence for playback and capture will be different */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { graph = q6apm_graph_open(dai->dev, NULL, dai->dev, graph_id); if (IS_ERR(graph)) { dev_err(dai->dev, "Failed to open graph (%d)\n", graph_id); @@ -224,7 +224,7 @@ static int q6apm_lpass_dai_startup(struct snd_pcm_substream *substream, struct s struct q6apm_graph *graph; int graph_id = dai->id;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { graph = q6apm_graph_open(dai->dev, NULL, dai->dev, graph_id); if (IS_ERR(graph)) { dev_err(dai->dev, "Failed to open graph (%d)\n", graph_id); diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 2a2a5bd98110b..38d8aaab876d2 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -195,7 +195,7 @@ int q6apm_graph_media_format_shmem(struct q6apm_graph *graph, { struct audioreach_module *module;
- if (cfg->direction == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(cfg->direction)) module = q6apm_find_module_by_mid(graph, MODULE_ID_RD_SHARED_MEM_EP); else module = q6apm_find_module_by_mid(graph, MODULE_ID_WR_SHARED_MEM_EP); @@ -218,7 +218,7 @@ int q6apm_map_memory_regions(struct q6apm_graph *graph, unsigned int dir, phys_a int cnt; int rc;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) data = &graph->rx_data; else data = &graph->tx_data; @@ -236,7 +236,7 @@ int q6apm_map_memory_regions(struct q6apm_graph *graph, unsigned int dir, phys_a return -ENOMEM; }
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) data = &graph->rx_data; else data = &graph->tx_data; @@ -273,7 +273,7 @@ int q6apm_unmap_memory_regions(struct q6apm_graph *graph, unsigned int dir) struct gpr_pkt *pkt; int rc;
- if (dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(dir)) data = &graph->rx_data; else data = &graph->tx_data; @@ -538,7 +538,7 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) graph->result.status = 0; rsp = data->payload;
- if (hdr->token == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(hdr->token)) graph->rx_data.mem_map_handle = rsp->mem_map_handle; else graph->tx_data.mem_map_handle = rsp->mem_map_handle; @@ -575,7 +575,7 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) case APM_CMD_SHARED_MEM_UNMAP_REGIONS: graph->result.opcode = result->opcode; graph->result.status = 0; - if (hdr->token == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(hdr->token)) graph->rx_data.mem_map_handle = 0; else graph->tx_data.mem_map_handle = 0; diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 3913706ccdc5f..3e3d2847f992b 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -187,7 +187,7 @@ static void event_handler(uint32_t opcode, uint32_t token,
switch (opcode) { case ASM_CLIENT_EVENT_CMD_RUN_DONE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) q6asm_write_async(prtd->audio_client, prtd->stream_id, prtd->pcm_count, 0, 0, 0); break; @@ -258,11 +258,11 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, return -ENOMEM; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, FORMAT_LINEAR_PCM, 0, prtd->bits_per_sample, false); - } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + } else if (snd_pcm_is_capture(substream)) { ret = q6asm_open_read(prtd->audio_client, prtd->stream_id, FORMAT_LINEAR_PCM, prtd->bits_per_sample); @@ -281,12 +281,12 @@ static int q6asm_dai_prepare(struct snd_soc_component *component, goto routing_err; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = q6asm_media_format_block_multi_ch_pcm( prtd->audio_client, prtd->stream_id, runtime->rate, runtime->channels, NULL, prtd->bits_per_sample); - } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + } else if (snd_pcm_is_capture(substream)) { ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client, prtd->stream_id, runtime->rate, @@ -385,9 +385,9 @@ static int q6asm_dai_open(struct snd_soc_component *component, /* DSP expects stream id from 1 */ prtd->stream_id = 1;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) runtime->hw = q6asm_dai_hardware_playback; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) runtime->hw = q6asm_dai_hardware_capture;
ret = snd_pcm_hw_constraint_list(runtime, 0, @@ -401,7 +401,7 @@ static int q6asm_dai_open(struct snd_soc_component *component, if (ret < 0) dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE, diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 81fde0681f952..7e7ad072700d2 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -1055,7 +1055,7 @@ static int routing_hw_params(struct snd_soc_component *component, struct session_data *session; int path_type;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) path_type = ADM_PATH_PLAYBACK; else path_type = ADM_PATH_LIVE_REC; diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 75701546b6ea8..daa38d07a50f2 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -84,7 +84,7 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, continue; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, rx_ch_cnt, rx_ch); else @@ -115,7 +115,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, }
channels = params_channels(params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0x3, 8, slot_width); if (ret < 0) {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sprd/sprd-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/sprd/sprd-pcm-dma.c b/sound/soc/sprd/sprd-pcm-dma.c index d6b96cc2f7087..b7149322b9858 100644 --- a/sound/soc/sprd/sprd-pcm-dma.c +++ b/sound/soc/sprd/sprd-pcm-dma.c @@ -195,7 +195,7 @@ static int sprd_pcm_hw_params(struct snd_soc_component *component, size_t totsize = params_buffer_bytes(params); size_t period = params_period_bytes(params); int channels = params_channels(params); - int is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int is_playback = snd_pcm_is_playback(substream); struct scatterlist *sg; unsigned long flags; int ret, i, j, sg_num;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/au1x/dbdma2.c | 4 ++-- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 10 +++++----- sound/soc/au1x/psc-i2s.c | 8 ++++---- 4 files changed, 12 insertions(+), 12 deletions(-)
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index ea01d6490cec0..307cfbc7f713f 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -158,7 +158,7 @@ static int au1x_pcm_dbdma_realloc(struct au1xpsc_audio_dmadata *pcd,
au1x_pcm_dbdma_free(pcd);
- if (stype == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stype)) pcd->ddma_chan = au1xxx_dbdma_chan_alloc(pcd->ddma_id, DSCR_CMD0_ALWAYS, au1x_pcm_dmarx_cb, (void *)pcd); @@ -235,7 +235,7 @@ static int au1xpsc_pcm_prepare(struct snd_soc_component *component,
au1xxx_dbdma_reset(pcd->ddma_chan);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { au1x_pcm_queue_rx(pcd); au1x_pcm_queue_rx(pcd); } else { diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index d2fdebd8881bb..7a8ff759ab4b1 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -200,7 +200,7 @@ static int alchemy_pcm_open(struct snd_soc_component *component, return -ENODEV; /* whoa, has ordering changed? */
/* DMA setup */ - name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx"; + name = snd_pcm_is_playback(s) ? "audio-tx" : "audio-rx"; ctx->stream[s].dma = request_au1000_dma(dmaids[s], name, au1000_dma_interrupt, 0, &ctx->stream[s]); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 1727eeb12b64e..62627ec731063 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -37,14 +37,14 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3BE)
#define AC97PCR_START(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) + (snd_pcm_is_playback(stype) ? PSC_AC97PCR_TS : PSC_AC97PCR_RS) #define AC97PCR_STOP(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) + (snd_pcm_is_playback(stype) ? PSC_AC97PCR_TP : PSC_AC97PCR_RP) #define AC97PCR_CLRFIFO(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) + (snd_pcm_is_playback(stype) ? PSC_AC97PCR_TC : PSC_AC97PCR_RC)
#define AC97STAT_BUSY(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + (snd_pcm_is_playback(stype) ? PSC_AC97STAT_TB : PSC_AC97STAT_RB)
/* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -230,7 +230,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_SET_LEN(params->msbits);
/* channels: enable slots for front L/R channel */ - if (stype == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stype)) { r &= ~PSC_AC97CFG_TXSLOT_MASK; r |= PSC_AC97CFG_TXSLOT_ENA(3); r |= PSC_AC97CFG_TXSLOT_ENA(4); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 52734dec82472..bd4a75fec9822 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -39,13 +39,13 @@ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
#define I2SSTAT_BUSY(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) + (snd_pcm_is_playback(stype) ? PSC_I2SSTAT_TB : PSC_I2SSTAT_RB) #define I2SPCR_START(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) + (snd_pcm_is_playback(stype) ? PSC_I2SPCR_TS : PSC_I2SPCR_RS) #define I2SPCR_STOP(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) + (snd_pcm_is_playback(stype) ? PSC_I2SPCR_TP : PSC_I2SPCR_RP) #define I2SPCR_CLRFIFO(stype) \ - ((stype) == SNDRV_PCM_STREAM_PLAYBACK ? PSC_I2SPCR_TC : PSC_I2SPCR_RC) + (snd_pcm_is_playback(stype) ? PSC_I2SPCR_TC : PSC_I2SPCR_RC)
static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/apple/mca.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index 3780aca710769..60a5ecbd5f756 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -193,7 +193,7 @@ static void mca_fe_early_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct mca_cluster *cl = mca_dai_to_cluster(dai); - bool is_tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool is_tx = snd_pcm_is_playback(substream); int serdes_unit = is_tx ? CLUSTER_TX_OFF : CLUSTER_RX_OFF; int serdes_conf = serdes_unit + (is_tx ? REG_TX_SERDES_CONF : REG_RX_SERDES_CONF); @@ -230,7 +230,7 @@ static int mca_fe_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct mca_cluster *cl = mca_dai_to_cluster(dai); - bool is_tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool is_tx = snd_pcm_is_playback(substream); int serdes_unit = is_tx ? CLUSTER_TX_OFF : CLUSTER_RX_OFF;
switch (cmd) { @@ -570,7 +570,7 @@ static int mca_fe_hw_params(struct snd_pcm_substream *substream, struct mca_data *mca = cl->host; struct device *dev = mca->dev; unsigned int samp_rate = params_rate(params); - bool is_tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool is_tx = snd_pcm_is_playback(substream); bool refine_tdm = false; unsigned long bclk_ratio; unsigned int tdm_slots, tdm_slot_width, tdm_mask; @@ -844,7 +844,7 @@ static int mca_hw_params(struct snd_soc_component *component, if (ret < 0) return ret;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) slave_config.dst_port_window_size = min_t(u32, params_channels(params), 4); else @@ -895,7 +895,7 @@ static snd_pcm_uframes_t mca_pointer(struct snd_soc_component *component,
static struct dma_chan *mca_request_dma_channel(struct mca_cluster *cl, unsigned int stream) { - bool is_tx = (stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_tx = snd_pcm_is_playback(stream); #ifndef USE_RXB_FOR_CAPTURE char *name = devm_kasprintf(cl->host->dev, GFP_KERNEL, is_tx ? "tx%da" : "rx%da", cl->no);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/atmel/atmel-i2s.c | 6 +++--- sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 10 +++++----- sound/soc/atmel/mchp-i2s-mcc.c | 6 +++--- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- 7 files changed, 15 insertions(+), 15 deletions(-)
diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6c20c643f3218..97bf80ba45531 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -272,7 +272,7 @@ static int atmel_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct atmel_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_playback = snd_pcm_is_playback(substream); unsigned int rhr, sr = 0;
if (is_playback) { @@ -324,7 +324,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct atmel_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_playback = snd_pcm_is_playback(substream); unsigned int mr = 0, mr_mask; int ret;
@@ -477,7 +477,7 @@ static int atmel_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct atmel_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_playback = snd_pcm_is_playback(substream); bool is_master, mck_enabled; unsigned int cr, mr; int err; diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 7306e04da513b..f65c30a5dd4f7 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -60,7 +60,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, if (ssc_sr & prtd->mask->ssc_error) { if (snd_pcm_running(substream)) pr_warn("atmel-pcm: buffer %s on %s (SSC_SR=%#x)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK + snd_pcm_is_playback(substream) ? "underrun" : "overrun", prtd->name, ssc_sr);
diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index 7db8df85c54f3..81ad08d436a34 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -96,7 +96,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
if (ssc_sr & params->mask->ssc_endbuf) { pr_warn("atmel-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK + snd_pcm_is_playback(substream) ? "underrun" : "overrun", params->name, ssc_sr, count);
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 3763454436c15..3a8dd39537db8 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -290,7 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, if (!ssc_p->initialized) ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { dir = 0; dir_mask = SSC_DIR_MASK_PLAYBACK; } else { @@ -337,7 +337,7 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, dir_mask;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dir = 0; else dir = 1; @@ -476,7 +476,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * each direction. If more are added, this code will * have to be changed to select the proper set. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dir = 0; else dir = 1; @@ -717,7 +717,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dir = 0; else dir = 1; @@ -741,7 +741,7 @@ static int atmel_ssc_trigger(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dir = 0; else dir = 1; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 193dd7acceb08..017f363ed389d 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -517,7 +517,7 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, unsigned int bclk_rate; int set_divs = 0; int ret; - bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_playback = snd_pcm_is_playback(substream);
dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", __func__, params_rate(params), params_format(params), @@ -733,7 +733,7 @@ static int mchp_i2s_mcc_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); - bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_playback = snd_pcm_is_playback(substream); long err;
if (is_playback) { @@ -789,7 +789,7 @@ static int mchp_i2s_mcc_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); - bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_playback = snd_pcm_is_playback(substream); u32 cr = 0; u32 iera = 0, ierb = 0; u32 sr; diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index 33ce5e54482be..653c575e9a1f6 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -436,7 +436,7 @@ static int mchp_spdifrx_hw_params(struct snd_pcm_substream *substream, __func__, params_rate(params), params_format(params), params_width(params), params_channels(params));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { dev_err(dev->dev, "Playback is not supported\n"); return -EINVAL; } diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index a201a96fa6906..1e73a720ff6bb 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -358,7 +358,7 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream, __func__, params_rate(params), params_format(params), params_width(params), params_channels(params));
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { dev_err(dev->dev, "Capture is not supported\n"); return -EINVAL; }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/pcm.c | 16 ++++++++-------- 2 files changed, 9 insertions(+), 9 deletions(-)
diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index f31d5e2caa7b0..dab52a90f4c31 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -171,7 +171,7 @@ static int avs_copier_create(struct avs_dev *adev, struct avs_path_module *mod)
if (t->cfg_ext->copier.blob_fmt) fmt = t->cfg_ext->copier.blob_fmt; - else if (direction == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(direction)) fmt = t->in_fmt; else fmt = t->cfg_ext->copier.out_fmt; diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index c76b86254a8b4..df71a0292d4a6 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -40,7 +40,7 @@ avs_dai_find_path_template(struct snd_soc_dai *dai, bool is_fe, int direction) struct snd_soc_dapm_path *dp; enum snd_soc_dapm_direction dir;
- if (direction == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(direction)) { dir = is_fe ? SND_SOC_DAPM_DIR_OUT : SND_SOC_DAPM_DIR_IN; } else { dir = is_fe ? SND_SOC_DAPM_DIR_IN : SND_SOC_DAPM_DIR_OUT; @@ -331,7 +331,7 @@ static int avs_dai_hda_be_hw_free(struct snd_pcm_substream *substream, struct sn if (!link) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_hdac_ext_bus_link_clear_stream_id(link, hdac_stream(link_stream)->stream_tag);
return 0; @@ -372,7 +372,7 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn if (!link) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_hdac_ext_bus_link_set_stream_id(link, hdac_stream(link_stream)->stream_tag);
ret = avs_dai_prepare(substream, dai); @@ -695,7 +695,7 @@ static void avs_hda_stream_start(struct hdac_bus *bus, struct hdac_ext_stream *h * disable L1SEN to avoid sound clipping. */ if (!first_running) { - if (hdac_stream(host_stream)->direction == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(hdac_stream(host_stream)->direction)) avs_hda_l1sen_enable(adev, false); snd_hdac_stream_start(hdac_stream(host_stream)); return; @@ -707,7 +707,7 @@ static void avs_hda_stream_start(struct hdac_bus *bus, struct hdac_ext_stream *h * re-enable L1SEN. */ if (list_entry_is_head(pos, &bus->stream_list, list) && - first_running->direction == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_is_capture(first_running->direction)) avs_hda_l1sen_enable(adev, true); }
@@ -733,7 +733,7 @@ static void avs_hda_stream_stop(struct hdac_bus *bus, struct hdac_ext_stream *ho */ if (!first_running) { snd_hdac_stream_stop(hdac_stream(host_stream)); - if (hdac_stream(host_stream)->direction == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(hdac_stream(host_stream)->direction)) avs_hda_l1sen_enable(adev, true); return; } @@ -743,7 +743,7 @@ static void avs_hda_stream_stop(struct hdac_bus *bus, struct hdac_ext_stream *ho * left, disable L1SEN to avoid sound clipping. */ if (list_entry_is_head(pos, &bus->stream_list, list) && - first_running->direction == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_is_capture(first_running->direction)) avs_hda_l1sen_enable(adev, false);
snd_hdac_stream_stop(hdac_stream(host_stream)); @@ -1602,7 +1602,7 @@ static int avs_component_hda_open(struct snd_soc_component *component, }
/* RESUME unsupported for de-coupled HD-Audio capture. */ - if (dir == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(dir)) hwparams.info &= ~SNDRV_PCM_INFO_RESUME;
return snd_soc_set_runtime_hwparams(substream, &hwparams);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 2 +- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 4 +-- sound/soc/intel/catpt/pcm.c | 4 +-- sound/soc/intel/keembay/kmb_platform.c | 28 ++++++++++---------- sound/soc/intel/skylake/skl-pcm.c | 14 +++++----- sound/soc/intel/skylake/skl-topology.c | 18 ++++++------- 9 files changed, 38 insertions(+), 38 deletions(-)
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 38116c7587174..0c6ce403148f8 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1333,7 +1333,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dev_dbg(dai->dev, "enter, dai-name=%s dir=%d\n", dai->name, stream); dev_dbg(dai->dev, "Stream name=%s\n", w->name);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { snd_soc_dapm_widget_for_each_sink_path(w, p) { if (p->connected && !p->connected(w, p->sink)) continue; diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index dc31c2c8f54c8..f02ee7f48a2a4 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -487,7 +487,7 @@ static inline int sst_calc_tstamp(struct intel_sst_drv *ctx, fw_tstamp->ring_buffer_counter); dev_dbg(ctx->dev, "mrfld hardware_counter %llu in bytes\n", fw_tstamp->hardware_counter); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) delay_bytes = (size_t) (fw_tstamp->ring_buffer_counter - fw_tstamp->hardware_counter); else diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index 3c7cee03a02e6..a5df4d3067d71 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -150,7 +150,7 @@ static int bdw_rt5650_fe_startup(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime;
/* Board supports stereo and quad configurations for capture */ - if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream)) return 0;
runtime->hw.channels_max = 4; diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index fcc3b95e57a4f..f520442bbb096 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -196,7 +196,7 @@ static int max_98373_trigger(struct snd_pcm_substream *substream, int cmd) int ret = 0;
/* set spk pin by playback only */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) return 0;
cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d258728d64cf5..5818c21173032 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -846,8 +846,8 @@ static int create_sdw_dailink(struct snd_soc_card *card,
WARN_ON(i != num_cpus || j != num_codecs);
- playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); - capture = (stream == SNDRV_PCM_STREAM_CAPTURE); + playback = snd_pcm_is_playback(stream); + capture = snd_pcm_is_capture(stream);
asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, cpus, num_cpus, platform_component, diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index 81a2f0339e055..c32c101e65b9c 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -83,11 +83,11 @@ catpt_get_stream_template(struct snd_pcm_substream *substream) /* account for capture in bidirectional dais */ switch (type) { case CATPT_STRM_TYPE_SYSTEM: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) type = CATPT_STRM_TYPE_CAPTURE; break; case CATPT_STRM_TYPE_BLUETOOTH_RENDER: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) type = CATPT_STRM_TYPE_BLUETOOTH_CAPTURE; break; default: diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 37ea2e1d2e922..eab7b8223b51b 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -170,7 +170,7 @@ static inline void kmb_i2s_disable_channels(struct kmb_i2s_info *kmb_i2s, u32 i;
/* Disable all channels regardless of configuration*/ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { for (i = 0; i < MAX_ISR; i++) writel(0, kmb_i2s->i2s_base + TER(i)); } else { @@ -184,7 +184,7 @@ static inline void kmb_i2s_clear_irqs(struct kmb_i2s_info *kmb_i2s, u32 stream) struct i2s_clk_config_data *config = &kmb_i2s->config; u32 i;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { for (i = 0; i < config->chan_nr / 2; i++) readl(kmb_i2s->i2s_base + TOR(i)); } else { @@ -199,7 +199,7 @@ static inline void kmb_i2s_irq_trigger(struct kmb_i2s_info *kmb_i2s, u32 i, irq; u32 flag;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) flag = TX_INT_FLAG; else flag = RX_INT_FLAG; @@ -270,7 +270,7 @@ static int kmb_pcm_trigger(struct snd_soc_component *component,
switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { kmb_i2s->tx_ptr = 0; kmb_i2s->tx_substream = substream; } else { @@ -279,7 +279,7 @@ static int kmb_pcm_trigger(struct snd_soc_component *component, } break; case SNDRV_PCM_TRIGGER_STOP: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) kmb_i2s->tx_substream = NULL; else kmb_i2s->rx_substream = NULL; @@ -378,7 +378,7 @@ static snd_pcm_uframes_t kmb_pcm_pointer(struct snd_soc_component *component, struct kmb_i2s_info *kmb_i2s = runtime->private_data; snd_pcm_uframes_t pos;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) pos = kmb_i2s->tx_ptr; else pos = kmb_i2s->rx_ptr; @@ -419,7 +419,7 @@ static inline void kmb_i2s_enable_dma(struct kmb_i2s_info *kmb_i2s, u32 stream)
dma_reg = readl(kmb_i2s->i2s_base + I2S_DMACR); /* Enable DMA handshake for stream */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) dma_reg |= I2S_DMAEN_TXBLOCK; else dma_reg |= I2S_DMAEN_RXBLOCK; @@ -433,7 +433,7 @@ static inline void kmb_i2s_disable_dma(struct kmb_i2s_info *kmb_i2s, u32 stream)
dma_reg = readl(kmb_i2s->i2s_base + I2S_DMACR); /* Disable DMA handshake for stream */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { dma_reg &= ~I2S_DMAEN_TXBLOCK; writel(1, kmb_i2s->i2s_base + I2S_RTXDMA); } else { @@ -451,7 +451,7 @@ static void kmb_i2s_start(struct kmb_i2s_info *kmb_i2s, /* I2S Programming sequence in Keem_Bay_VPU_DB_v1.1 */ writel(1, kmb_i2s->i2s_base + IER);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) writel(1, kmb_i2s->i2s_base + ITER); else writel(1, kmb_i2s->i2s_base + IRER); @@ -474,7 +474,7 @@ static void kmb_i2s_stop(struct kmb_i2s_info *kmb_i2s, /* I2S Programming sequence in Keem_Bay_VPU_DB_v1.1 */ kmb_i2s_clear_irqs(kmb_i2s, substream->stream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) writel(0, kmb_i2s->i2s_base + ITER); else writel(0, kmb_i2s->i2s_base + IRER); @@ -556,7 +556,7 @@ static void kmb_i2s_config(struct kmb_i2s_info *kmb_i2s, int stream) kmb_i2s_disable_channels(kmb_i2s, stream);
for (ch_reg = 0; ch_reg < config->chan_nr / 2; ch_reg++) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { writel(kmb_i2s->xfer_resolution, kmb_i2s->i2s_base + TCR(ch_reg));
@@ -678,7 +678,7 @@ static int kmb_dai_prepare(struct snd_pcm_substream *substream, { struct kmb_i2s_info *kmb_i2s = snd_soc_dai_get_drvdata(cpu_dai);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) writel(1, kmb_i2s->i2s_base + TXFFR); else writel(1, kmb_i2s->i2s_base + RXFFR); @@ -695,7 +695,7 @@ static int kmb_dai_startup(struct snd_pcm_substream *substream, if (kmb_i2s->use_pio) return 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dma_data = &kmb_i2s->play_dma_data; else dma_data = &kmb_i2s->capture_dma_data; @@ -713,7 +713,7 @@ static int kmb_dai_hw_free(struct snd_pcm_substream *substream, if (kmb_i2s->use_pio) kmb_i2s_clear_irqs(kmb_i2s, substream->stream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) writel(0, kmb_i2s->i2s_base + ITER); else writel(0, kmb_i2s->i2s_base + IRER); diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 613b27b8da134..2ffd511eedfe4 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -189,7 +189,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) snd_hdac_ext_stream_setup(stream, format_val);
stream_tag = hstream->stream_tag; - if (stream->hstream.direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream->hstream.direction)) { list_for_each_entry(link, &bus->hlink_list, list) { if (link->index == params->link_index) snd_hdac_ext_bus_link_set_stream_id(link, @@ -225,7 +225,7 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, * disable WALLCLOCK timestamps for capture streams * until we figure out how to handle digital inputs */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */ runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME; } @@ -319,7 +319,7 @@ static int skl_pcm_hw_params(struct snd_pcm_substream *substream, p_params.host_dma_id = dma_id; p_params.stream = substream->stream; p_params.format = params_format(params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) p_params.host_bps = dai->driver->playback.sig_bits; else p_params.host_bps = dai->driver->capture.sig_bits; @@ -574,7 +574,7 @@ static int skl_link_hw_params(struct snd_pcm_substream *substream, p_params.link_index = link->index; p_params.format = params_format(params);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) p_params.link_bps = codec_dai->driver->playback.sig_bits; else p_params.link_bps = codec_dai->driver->capture.sig_bits; @@ -645,7 +645,7 @@ static int skl_link_hw_free(struct snd_pcm_substream *substream, if (!link) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { stream_tag = hdac_stream(link_dev)->stream_tag; snd_hdac_ext_bus_link_clear_stream_id(link, stream_tag); } @@ -1193,7 +1193,7 @@ static snd_pcm_uframes_t skl_platform_soc_pointer( * or greater than period boundary. */
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { pos = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + (AZX_REG_VS_SDXDPIB_XINTERVAL * hdac_stream(hstream)->index)); @@ -1226,7 +1226,7 @@ static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, codec_nsecs = div_u64(codec_frames * 1000000000LL, substream->runtime->rate);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) return nsec + codec_nsecs;
return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 602ef43211221..cb51b98b92c9a 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -198,7 +198,7 @@ static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg, in_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].inputs[0].fmt; out_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].outputs[0].fmt;
- if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(params->stream)) { if (is_fe) { in_fixup = m_cfg->params_fixup; out_fixup = (~m_cfg->converter) & @@ -618,9 +618,9 @@ skl_tplg_get_pipe_config(struct skl_dev *skl, struct skl_module_cfg *mconfig) }
if ((pipe->conn_type == SKL_PIPE_CONN_TYPE_FE && - pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) || + snd_pcm_is_playback(pipe->direction)) || (pipe->conn_type == SKL_PIPE_CONN_TYPE_BE && - pipe->direction == SNDRV_PCM_STREAM_CAPTURE)) + snd_pcm_is_capture(pipe->direction))) in_fmt = true;
for (i = 0; i < pipe->nr_cfgs; i++) { @@ -1612,7 +1612,7 @@ int skl_tplg_update_pipe_params(struct device *dev, if (skl->nr_modules) return 0;
- if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(params->stream)) format = &mconfig->module->formats[mconfig->fmt_idx].inputs[0].fmt; else format = &mconfig->module->formats[mconfig->fmt_idx].outputs[0].fmt; @@ -1642,7 +1642,7 @@ int skl_tplg_update_pipe_params(struct device *dev, return -EINVAL; }
- if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(params->stream)) { res->ibs = (format->s_freq / 1000) * (format->channels) * (format->bit_depth >> 3); @@ -1666,7 +1666,7 @@ skl_tplg_fe_get_cpr_module(struct snd_soc_dai *dai, int stream) struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); struct snd_soc_dapm_path *p = NULL;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { snd_soc_dapm_widget_for_each_sink_path(w, p) { if (p->connect && p->sink->power && !is_skl_dsp_widget_type(p->sink, dai->dev)) @@ -1745,7 +1745,7 @@ skl_tplg_be_get_cpr_module(struct snd_soc_dai *dai, int stream) struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); struct skl_module_cfg *mconfig;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { mconfig = skl_get_mconfig_pb_cpr(dai, w); } else { mconfig = skl_get_mconfig_cap_cpr(dai, w); @@ -1813,7 +1813,7 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, goto err;
dev_dbg(skl->dev, "%s using pipe config: %d\n", __func__, pipe->cur_config_idx); - if (pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(pipe->direction)) pipe_fmt = &pipe->configs[pipe->cur_config_idx].out_fmt; else pipe_fmt = &pipe->configs[pipe->cur_config_idx].in_fmt; @@ -1903,7 +1903,7 @@ int skl_tplg_be_update_params(struct snd_soc_dai *dai, { struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, params->stream);
- if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(params->stream)) { return skl_tplg_be_set_src_pipe_params(dai, w, params); } else { return skl_tplg_be_set_sink_pipe_params(dai, w, params);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/soc-core.c | 2 +- sound/soc/soc-dai.c | 2 +- sound/soc/soc-dapm.c | 4 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- sound/soc/soc-pcm.c | 6 +++--- 5 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 20248a29d1674..286e872ae4547 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3456,7 +3456,7 @@ int snd_soc_get_stream_cpu(const struct snd_soc_dai_link *dai_link, int stream) * CPU : SNDRV_PCM_STREAM_PLAYBACK * Codec: SNDRV_PCM_STREAM_CAPTURE */ - if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) return SNDRV_PCM_STREAM_PLAYBACK;
return SNDRV_PCM_STREAM_CAPTURE; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..28a0a7a0993af 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -379,7 +379,7 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, */ if (dai->driver->ops && dai->driver->ops->mute_stream && - (direction == SNDRV_PCM_STREAM_PLAYBACK || + (snd_pcm_is_playback(direction) || !dai->driver->ops->no_capture_mute)) ret = dai->driver->ops->mute_stream(dai, mute, direction);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d7d6dbb9d9eae..0d60942f64113 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1312,7 +1312,7 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
snd_soc_dapm_mutex_lock(card);
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { invalidate_paths_ep(w, SND_SOC_DAPM_DIR_OUT); paths = is_connected_output_ep(w, &widgets, custom_stop_condition); @@ -4539,7 +4539,7 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
void snd_soc_dapm_stream_stop(struct snd_soc_pcm_runtime *rtd, int stream) { - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { if (snd_soc_runtime_ignore_pmdown_time(rtd)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index a63e942fdc0b7..4265e8052d6bf 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -297,7 +297,7 @@ static int dmaengine_copy(struct snd_soc_component *component, int (*process)(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, unsigned long bytes) = pcm->config->process; - bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool is_playback = snd_pcm_is_playback(substream); void *dma_ptr = runtime->dma_area + hwoff + channel * (runtime->dma_bytes / runtime->channels);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5520944ac9ddc..8f501178195e9 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -905,7 +905,7 @@ static int __soc_pcm_prepare(struct snd_soc_pcm_runtime *rtd, goto out;
/* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && rtd->pop_wait) { rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); @@ -1517,11 +1517,11 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
switch (widget->id) { case snd_soc_dapm_dai_in: - if (stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(stream)) continue; break; case snd_soc_dapm_dai_out: - if (stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(stream)) continue; break; default:
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/spear/spdif_in.c | 6 +++--- sound/soc/spear/spdif_out.c | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 4ad8b1fc713a7..fb1b54019194a 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -68,7 +68,7 @@ static void spdif_in_shutdown(struct snd_pcm_substream *substream, { struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
- if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream)) return;
writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); @@ -98,7 +98,7 @@ static int spdif_in_hw_params(struct snd_pcm_substream *substream, struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); u32 format;
- if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream)) return -EINVAL;
format = params_format(params); @@ -114,7 +114,7 @@ static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, u32 ctrl; int ret = 0;
- if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream)) return -EINVAL;
switch (cmd) { diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 469373d1bb418..a95a9b9e61e3b 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -63,7 +63,7 @@ static int spdif_out_startup(struct snd_pcm_substream *substream, struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); int ret;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -EINVAL;
ret = clk_enable(host->clk); @@ -81,7 +81,7 @@ static void spdif_out_shutdown(struct snd_pcm_substream *substream, { struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return;
clk_disable(host->clk); @@ -109,7 +109,7 @@ static int spdif_out_hw_params(struct snd_pcm_substream *substream, struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); u32 rate, core_freq;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -EINVAL;
rate = params_rate(params); @@ -155,7 +155,7 @@ static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd, u32 ctrl; int ret = 0;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -EINVAL;
switch (cmd) {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sunxi/sun4i-codec.c | 8 ++++---- sound/soc/sunxi/sun4i-i2s.c | 4 ++-- sound/soc/sunxi/sun4i-spdif.c | 4 ++-- sound/soc/sunxi/sun50i-dmic.c | 4 ++-- 4 files changed, 10 insertions(+), 10 deletions(-)
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index a2618ed650b00..edde5cb84ab61 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -285,7 +285,7 @@ static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) sun4i_codec_start_playback(scodec); else sun4i_codec_start_capture(scodec); @@ -294,7 +294,7 @@ static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) sun4i_codec_stop_playback(scodec); else sun4i_codec_stop_capture(scodec); @@ -385,7 +385,7 @@ static int sun4i_codec_prepare_playback(struct snd_pcm_substream *substream, static int sun4i_codec_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return sun4i_codec_prepare_playback(substream, dai);
return sun4i_codec_prepare_capture(substream, dai); @@ -569,7 +569,7 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, if (hwrate < 0) return hwrate;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return sun4i_codec_hw_params_playback(scodec, params, hwrate);
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 5f8d979585b69..0b1bce325f04e 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1033,7 +1033,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) sun4i_i2s_start_playback(i2s); else sun4i_i2s_start_capture(i2s); @@ -1042,7 +1042,7 @@ static int sun4i_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) sun4i_i2s_stop_playback(i2s); else sun4i_i2s_stop_capture(i2s); diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index f41c309558579..989cebb55798f 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -248,7 +248,7 @@ static int sun4i_spdif_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0));
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -EINVAL;
sun4i_spdif_configure(host); @@ -364,7 +364,7 @@ static int sun4i_spdif_trigger(struct snd_pcm_substream *substream, int cmd, int ret = 0; struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(dai);
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -EINVAL;
switch (cmd) { diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index 884394ddaf86b..1b662b40b7b6c 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -90,7 +90,7 @@ static int sun50i_dmic_startup(struct snd_pcm_substream *substream, struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0));
/* only support capture */ - if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream)) return -EINVAL;
regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, @@ -205,7 +205,7 @@ static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, int ret = 0; struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai);
- if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + if (!snd_pcm_is_capture(substream)) return -EINVAL;
switch (cmd) {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/tegra/tegra20_ac97.c | 4 ++-- sound/soc/tegra/tegra20_i2s.c | 4 ++-- sound/soc/tegra/tegra210_admaif.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 6 +++--- sound/soc/tegra/tegra30_i2s.c | 6 +++--- sound/soc/tegra/tegra_pcm.c | 2 +- 6 files changed, 12 insertions(+), 12 deletions(-)
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 8011afe93c96e..0314402361558 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -182,7 +182,7 @@ static int tegra20_ac97_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tegra20_ac97_start_playback(ac97); else tegra20_ac97_start_capture(ac97); @@ -190,7 +190,7 @@ static int tegra20_ac97_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tegra20_ac97_stop_playback(ac97); else tegra20_ac97_stop_capture(ac97); diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index f11618e8f13ee..330400d0e530c 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -232,7 +232,7 @@ static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tegra20_i2s_start_playback(i2s); else tegra20_i2s_start_capture(i2s); @@ -240,7 +240,7 @@ static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tegra20_i2s_stop_playback(i2s); else tegra20_i2s_stop_capture(i2s); diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index 9f9334e480490..e79711ea65dc9 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -299,7 +299,7 @@ static int tegra_admaif_hw_params(struct snd_pcm_substream *substream, cif_conf.client_ch = channels; cif_conf.audio_ch = channels;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { path = ADMAIF_TX_PATH; reg = CH_TX_REG(TEGRA_ADMAIF_CH_ACIF_TX_CTRL, dai->id); } else { diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index e93ceb7afb4c4..74a0ada1e98c7 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -95,7 +95,7 @@ static int tegra210_i2s_sw_reset(struct snd_soc_component *compnt, unsigned int cif_ctrl, stream_ctrl, i2s_ctrl, val; int err;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { reset_reg = TEGRA210_I2S_RX_SOFT_RESET; cif_reg = TEGRA210_I2S_RX_CIF_CTRL; stream_reg = TEGRA210_I2S_RX_CTRL; @@ -673,12 +673,12 @@ static int tegra210_i2s_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params);
/* For playback I2S RX-CIF and for capture TX-CIF is used */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) path = I2S_RX_PATH; else path = I2S_TX_PATH;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { unsigned int max_th;
/* FIFO threshold in terms of frames */ diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index a8ff51d12edb5..edcb7095bf0ac 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -188,7 +188,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, cif_conf.truncate = 0; cif_conf.mono_conv = 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX; reg = TEGRA30_I2S_CIF_RX_CTRL; } else { @@ -244,7 +244,7 @@ static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tegra30_i2s_start_playback(i2s); else tegra30_i2s_start_capture(i2s); @@ -252,7 +252,7 @@ static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tegra30_i2s_stop_playback(i2s); else tegra30_i2s_stop_capture(i2s); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 05d59e03b1c5e..1ed74f6b6431a 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -164,7 +164,7 @@ int tegra_pcm_hw_params(struct snd_soc_component *component, return ret; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.dst_addr = dmap->addr; slave_config.dst_maxburst = 8;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/ux500/mop500_ab8500.c | 4 ++-- sound/soc/ux500/ux500_msp_dai.c | 6 +++--- sound/soc/ux500/ux500_msp_i2s.c | 4 ++-- sound/soc/ux500/ux500_pcm.c | 2 +- 4 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 710b6744e0136..102d4922d6e62 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -203,7 +203,7 @@ static void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) dev_dbg(dev, "%s: Enter\n", __func__);
/* Reset slots configuration to default(s) */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tx_slots = DEF_TX_SLOTS; else rx_slots = DEF_RX_SLOTS; @@ -291,7 +291,7 @@ static int mop500_ab8500_hw_params(struct snd_pcm_substream *substream,
/* Setup TDM-slots */
- is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + is_playback = snd_pcm_is_playback(substream); switch (channels) { case 1: slots = 16; diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 3fd13e8dd1107..1a02d66d45cd1 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -312,7 +312,7 @@ static int setup_msp_config(struct snd_pcm_substream *substream, msp_config->tx_fifo_config = TX_FIFO_ENABLE; msp_config->rx_fifo_config = RX_FIFO_ENABLE; msp_config->def_elem_len = 1; - msp_config->direction = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + msp_config->direction = snd_pcm_is_playback(substream) ? MSP_DIR_TX : MSP_DIR_RX; msp_config->data_size = MSP_DATA_BITS_32; msp_config->frame_freq = runtime->rate; @@ -423,7 +423,7 @@ static void ux500_msp_dai_shutdown(struct snd_pcm_substream *substream, { int ret; struct ux500_msp_i2s_drvdata *drvdata = dev_get_drvdata(dai->dev); - bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool is_playback = snd_pcm_is_playback(substream);
dev_dbg(dai->dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, snd_pcm_stream_str(substream)); @@ -511,7 +511,7 @@ static int ux500_msp_dai_hw_params(struct snd_pcm_substream *substream,
case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_DSP_A: - mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + mask = snd_pcm_is_playback(substream) ? drvdata->tx_mask : drvdata->rx_mask;
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index fbfeefa418ca7..36819fbd66781 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -565,7 +565,7 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(direction)) enable_bit = TX_ENABLE; else enable_bit = RX_ENABLE; @@ -576,7 +576,7 @@ int ux500_msp_i2s_trigger(struct ux500_msp *msp, int cmd, int direction) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(direction)) disable_msp_tx(msp); else disable_msp_rx(msp); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index b7f38873d2d8a..65a2125e36594 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -50,7 +50,7 @@ static int ux500_pcm_prepare_slave_config(struct snd_pcm_substream *substream, slave_config->src_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) slave_config->dst_addr = dma_addr; else slave_config->src_addr = dma_addr;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/cirrus/ep93xx-i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 522de4b802939..cb652b273b7e7 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -121,7 +121,7 @@ static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream) }
/* Enable fifo */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) base_reg = EP93XX_I2S_TX0EN; else base_reg = EP93XX_I2S_RX0EN; @@ -129,7 +129,7 @@ static void ep93xx_i2s_enable(struct ep93xx_i2s_info *info, int stream)
/* Enable TX IRQs (FIFO empty or underflow) */ if (IS_ENABLED(CONFIG_SND_EP93XX_SOC_I2S_WATCHDOG) && - stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_is_playback(stream)) ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCTRL, EP93XX_I2S_TXCTRL_TXEMPTY_LVL | EP93XX_I2S_TXCTRL_TXUFIE); @@ -141,11 +141,11 @@ static void ep93xx_i2s_disable(struct ep93xx_i2s_info *info, int stream)
/* Disable IRQs */ if (IS_ENABLED(CONFIG_SND_EP93XX_SOC_I2S_WATCHDOG) && - stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_is_playback(stream)) ep93xx_i2s_write_reg(info, EP93XX_I2S_TXCTRL, 0);
/* Disable fifo */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) base_reg = EP93XX_I2S_TX0EN; else base_reg = EP93XX_I2S_RX0EN; @@ -328,7 +328,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ep93xx_i2s_write_reg(info, EP93XX_I2S_TXWRDLEN, word_len); else ep93xx_i2s_write_reg(info, EP93XX_I2S_RXWRDLEN, word_len);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/google/chv3-i2s.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/google/chv3-i2s.c b/sound/soc/google/chv3-i2s.c index 08e558f24af86..1bda70bcf57b3 100644 --- a/sound/soc/google/chv3-i2s.c +++ b/sound/soc/google/chv3-i2s.c @@ -142,7 +142,7 @@ static int chv3_dma_open(struct snd_soc_component *component, if (res) return res;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) i2s->rx_substream = substream; else i2s->tx_substream = substream; @@ -155,7 +155,7 @@ static int chv3_dma_close(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct chv3_i2s_dev *i2s = snd_soc_dai_get_drvdata(snd_soc_rtd_to_cpu(rtd, 0));
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) chv3_i2s_wr(i2s, I2S_RX_ENABLE, 0); else chv3_i2s_wr(i2s, I2S_TX_ENABLE, 0); @@ -208,7 +208,7 @@ static int chv3_dma_prepare(struct snd_soc_component *component, period_bytes = snd_pcm_lib_period_bytes(substream); period_size = substream->runtime->period_size;
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream->pstr->stream)) { chv3_i2s_wr(i2s, I2S_SOFT_RESET, I2S_SOFT_RESET_RX_BIT); chv3_i2s_wr(i2s, I2S_RX_BASE_ADDR, substream->dma_buffer.addr); chv3_i2s_wr(i2s, I2S_RX_BUFFER_SIZE, buffer_bytes); @@ -237,7 +237,7 @@ static snd_pcm_uframes_t chv3_dma_pointer(struct snd_soc_component *component, frame_bytes = substream->runtime->frame_bits * 8; buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream->pstr->stream)) { idx_bytes = chv3_i2s_rd(i2s, I2S_RX_PRODUCER_IDX); } else { idx_bytes = chv3_i2s_rd(i2s, I2S_TX_CONSUMER_IDX); @@ -259,7 +259,7 @@ static int chv3_dma_ack(struct snd_soc_component *component, bytes = frames_to_bytes(runtime, runtime->control->appl_ptr); idx = bytes & (snd_pcm_lib_buffer_bytes(substream) - 1);
- if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream->pstr->stream)) chv3_i2s_wr(i2s, I2S_RX_CONSUMER_IDX, idx); else chv3_i2s_wr(i2s, I2S_TX_PRODUCER_IDX, idx);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/jz4740/jz4740-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 5176195316158..6bce5c08ee45d 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -103,7 +103,7 @@ static int jz4740_i2s_startup(struct snd_pcm_substream *substream, * because it does not disturb other active substreams. */ if (!i2s->soc_info->shared_fifo_flush) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_set_bits(i2s->regmap, JZ_REG_AIC_CTRL, JZ_AIC_CTRL_TFLUSH); else regmap_set_bits(i2s->regmap, JZ_REG_AIC_CTRL, JZ_AIC_CTRL_RFLUSH); @@ -148,7 +148,7 @@ static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); uint32_t mask;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) mask = JZ_AIC_CTRL_ENABLE_PLAYBACK | JZ_AIC_CTRL_ENABLE_TX_DMA; else mask = JZ_AIC_CTRL_ENABLE_CAPTURE | JZ_AIC_CTRL_ENABLE_RX_DMA; @@ -278,7 +278,7 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ctrl &= ~JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE; ctrl |= FIELD_PREP(JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE, sample_size);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/xilinx/xlnx_formatter_pcm.c | 12 ++++++------ sound/soc/xilinx/xlnx_spdif.c | 4 ++-- 2 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 158fc21a86c10..7076d0befb4be 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -271,7 +271,7 @@ static void xlnx_formatter_disable_irqs(void __iomem *mmio_base, int stream)
val = readl(mmio_base + XLNX_AUD_CTRL); val &= ~AUD_CTRL_IOC_IRQ_MASK; - if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) val &= ~AUD_CTRL_TOUT_IRQ_MASK;
writel(val, mmio_base + XLNX_AUD_CTRL); @@ -334,10 +334,10 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component, struct snd_pcm_runtime *runtime = substream->runtime; struct xlnx_pcm_drv_data *adata = dev_get_drvdata(component->dev);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && !adata->mm2s_presence) return -ENODEV; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + else if (snd_pcm_is_capture(substream) && !adata->s2mm_presence) return -ENODEV;
@@ -345,7 +345,7 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component, if (!stream_data) return -ENOMEM;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ch_count_mask = CFG_MM2S_CH_MASK; ch_count_shift = CFG_MM2S_CH_SHIFT; data_xfer_mode = CFG_MM2S_XFER_MASK; @@ -466,7 +466,7 @@ static int xlnx_formatter_pcm_hw_params(struct snd_soc_component *component, if (active_ch > stream_data->ch_limit) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && adata->sysclk) { unsigned int mclk_fs = adata->sysclk / params_rate(params);
@@ -479,7 +479,7 @@ static int xlnx_formatter_pcm_hw_params(struct snd_soc_component *component, writel(mclk_fs, stream_data->mmio + XLNX_AUD_FS_MULTIPLIER); }
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + if (snd_pcm_is_capture(substream) && stream_data->xfer_mode == AES_TO_PCM) { val = readl(stream_data->mmio + XLNX_AUD_STS); if (val & AUD_STS_CH_STS_MASK) { diff --git a/sound/soc/xilinx/xlnx_spdif.c b/sound/soc/xilinx/xlnx_spdif.c index d52d5fc7b5b81..bf79639081ecc 100644 --- a/sound/soc/xilinx/xlnx_spdif.c +++ b/sound/soc/xilinx/xlnx_spdif.c @@ -84,7 +84,7 @@ static int xlnx_spdif_startup(struct snd_pcm_substream *substream, val |= XSPDIF_FIFO_FLUSH_MASK; writel(val, ctx->base + XSPDIF_CONTROL_REG);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { writel(XSPDIF_CH_STS_MASK, ctx->base + XSPDIF_IRQ_ENABLE_REG); writel(XSPDIF_GLOBAL_IRQ_ENABLE, @@ -179,7 +179,7 @@ static int xlnx_spdif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val |= XSPDIF_CORE_ENABLE_MASK; writel(val, ctx->base + XSPDIF_CONTROL_REG); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) ret = rx_stream_detect(dai); break; case SNDRV_PCM_TRIGGER_STOP:
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41.c | 2 +- sound/soc/codecs/cs35l45.c | 2 +- sound/soc/codecs/cs35l56.c | 4 ++-- sound/soc/codecs/cs4234.c | 4 ++-- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4271.c | 6 +++--- sound/soc/codecs/cs42l42-sdw.c | 2 +- sound/soc/codecs/cs42l42.c | 4 ++-- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs42xx8.c | 4 ++-- 12 files changed, 18 insertions(+), 18 deletions(-)
diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 7a01b1d9fc9d7..133b38108ebfe 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -512,7 +512,7 @@ static int cs35l35_hw_params(struct snd_pcm_substream *substream, * You can pull more Monitor data from the SDOUT pin than going to SDIN * Just make sure your SCLK is fast enough to fill the frame */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (params_width(params)) { case 8: audin_format = CS35L35_SDIN_DEPTH_8; diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index cbea79bd89808..a3c49b5760c2e 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -891,7 +891,7 @@ static int cs35l36_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(cs35l36->regmap, CS35L36_ASP_FRAME_CTRL, CS35L36_ASP_RX_WIDTH_MASK, asp_width << CS35L36_ASP_RX_WIDTH_SHIFT); diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 1688c2c688f06..1ef19481c12d9 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -777,7 +777,7 @@ static int cs35l41_pcm_hw_params(struct snd_pcm_substream *substream, CS35L41_GLOBAL_FS_MASK, cs35l41_fs_rates[i].fs_cfg << CS35L41_GLOBAL_FS_SHIFT);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(cs35l41->regmap, CS35L41_SP_FORMAT, CS35L41_ASP_WIDTH_RX_MASK, asp_wl << CS35L41_ASP_WIDTH_RX_SHIFT); diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 2392c6effed85..114b4ffbfeede 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -741,7 +741,7 @@ static int cs35l45_asp_hw_params(struct snd_pcm_substream *substream, else asp_width = params_width(params);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(cs35l45->regmap, CS35L45_ASP_CONTROL2, CS35L45_ASP_WIDTH_RX_MASK, asp_width << CS35L45_ASP_WIDTH_RX_SHIFT); diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 84c34f5b1a516..5ead5d568d05b 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -468,7 +468,7 @@ static int cs35l56_asp_dai_hw_params(struct snd_pcm_substream *substream, freq_id << CS35L56_ASP_BCLK_FREQ_SHIFT); }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(cs35l56->base.regmap, CS35L56_ASP1_CONTROL2, CS35L56_ASP_RX_WIDTH_MASK, asp_width << CS35L56_ASP_RX_WIDTH_SHIFT); @@ -557,7 +557,7 @@ static int cs35l56_sdw_dai_hw_params(struct snd_pcm_substream *substream, sconfig.frame_rate = params_rate(params); sconfig.bps = snd_pcm_format_width(params_format(params));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { sconfig.direction = SDW_DATA_DIR_RX; pconfig.num = CS35L56_SDW1_PLAYBACK_PORT; pconfig.ch_mask = cs35l56->rx_mask; diff --git a/sound/soc/codecs/cs4234.c b/sound/soc/codecs/cs4234.c index 69287ba7e9558..1095cce0ecff3 100644 --- a/sound/soc/codecs/cs4234.c +++ b/sound/soc/codecs/cs4234.c @@ -410,7 +410,7 @@ static int cs4234_dai_hw_params(struct snd_pcm_substream *sub, dev_err(component->dev, "Unsupported sample width\n"); return -EINVAL; } - if (sub->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(sub)) regmap_update_bits(cs4234->regmap, CS4234_SAMPLE_WIDTH, CS4234_SDOUTX_SW_MASK, sample_width << CS4234_SDOUTX_SW_SHIFT); @@ -477,7 +477,7 @@ static int cs4234_dai_startup(struct snd_pcm_substream *sub, struct snd_soc_dai * Note: SNDRV_PCM_HW_PARAM_SAMPLE_BITS constrains the physical * width, which we don't care about, so constrain the format. */ - if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(sub)) { ret = snd_pcm_hw_constraint_mask64( sub->runtime, SNDRV_PCM_HW_PARAM_FORMAT, diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 78ffb7fa7fc5f..2a0121287b753 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -407,7 +407,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, struct cs4265_private *cs4265 = snd_soc_component_get_drvdata(component); int index;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + if (snd_pcm_is_capture(substream) && ((cs4265->format & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_RIGHT_J)) return -EINVAL; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e864188ae5eb9..b31d06ddb463b 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -351,9 +351,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, * registers every time. */
- if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if ((snd_pcm_is_playback(substream) && !snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_CAPTURE)) || - (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + (snd_pcm_is_capture(substream) && !snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_PLAYBACK))) { ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN, @@ -408,7 +408,7 @@ static int cs4271_mute_stream(struct snd_soc_dai *dai, int mute, int stream) int val_a = 0; int val_b = 0;
- if (stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(stream)) return 0;
if (mute) { diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 29891c1f6bece..92a110a841e13 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -78,7 +78,7 @@ static int cs42l42_sdw_dai_hw_params(struct snd_pcm_substream *substream,
snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) port_config.num = CS42L42_SDW_PLAYBACK_PORT; else port_config.num = CS42L42_SDW_CAPTURE_PORT; diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 60d366e53526f..a06a0ed8ded49 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1033,7 +1033,7 @@ int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
if (mute) { /* Mute the headphone */ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | CS42L42_HP_ANA_BMUTE_MASK, @@ -1106,7 +1106,7 @@ int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) } cs42l42->stream_use |= 1 << stream;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 92674314227c4..e2345e0cfdebc 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -426,7 +426,7 @@ static int cs42l43_asp_hw_params(struct snd_pcm_substream *substream, CS42L43_ASP_NUM_BCLKS_PER_FSYNC_MASK, frame << CS42L43_ASP_NUM_BCLKS_PER_FSYNC_SHIFT);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { reg = CS42L43_ASP_TX_CH1_CTRL; slots = priv->tx_slots; } else { diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 9c44b6283b8f9..7f619ee811836 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -262,7 +262,7 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); u32 ratio[2]; u32 rate[2]; u32 fm[2]; @@ -350,7 +350,7 @@ static int cs42xx8_hw_free(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream);
/* Clear stored rate */ cs42xx8->rate[tx] = 0;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/rt1017-sdca-sdw.c | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1316-sdw.c | 2 +- sound/soc/codecs/rt1318-sdw.c | 2 +- sound/soc/codecs/rt1320-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 4 ++-- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt712-sdca.c | 2 +- sound/soc/codecs/rt722-sdca.c | 2 +- 11 files changed, 12 insertions(+), 12 deletions(-)
diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c index 7c8103a0d562a..986f3ab407a29 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.c +++ b/sound/soc/codecs/rt1017-sdca-sdw.c @@ -593,7 +593,7 @@ static int rt1017_sdca_pcm_hw_params(struct snd_pcm_substream *substream,
/* SoundWire specific configuration */ /* port 1 for playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { direction = SDW_DATA_DIR_RX; port = 1; } else { diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 563df483a466c..aa7cbd8af2f5d 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -550,7 +550,7 @@ static int rt1308_sdw_hw_params(struct snd_pcm_substream *substream, snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
/* port 1 for playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) port_config.num = 1; else return -EINVAL; diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 22f1ed4e03f1a..01a8ad0c4e943 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -529,7 +529,7 @@ static int rt1316_sdw_hw_params(struct snd_pcm_substream *substream, snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
/* port 1 for playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) port_config.num = 1; else port_config.num = 2; diff --git a/sound/soc/codecs/rt1318-sdw.c b/sound/soc/codecs/rt1318-sdw.c index 319f71f5e60d3..60b29188aa642 100644 --- a/sound/soc/codecs/rt1318-sdw.c +++ b/sound/soc/codecs/rt1318-sdw.c @@ -584,7 +584,7 @@ static int rt1318_sdw_hw_params(struct snd_pcm_substream *substream,
/* SoundWire specific configuration */ /* port 1 for playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { direction = SDW_DATA_DIR_RX; port = 1; } else { diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c index 2916fa77b7915..563eb935751c7 100644 --- a/sound/soc/codecs/rt1320-sdw.c +++ b/sound/soc/codecs/rt1320-sdw.c @@ -1967,7 +1967,7 @@ static int rt1320_sdw_hw_params(struct snd_pcm_substream *substream, /* SoundWire specific configuration */ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (dai->id == RT1320_AIF1) port_config.num = 1; else diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 5edf11e136b43..88258390afb7d 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -124,7 +124,7 @@ static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, /* SoundWire specific configuration */ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) port_config.num = 1; else port_config.num = 2; @@ -204,7 +204,7 @@ static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, osr_c = RT5682_ADC_OSR_D_2; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK, RT5682_SDW_REF_1_MASK, val_p); regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1, diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 434b926f96c83..575bb6772c89d 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -918,7 +918,7 @@ static int rt700_pcm_hw_params(struct snd_pcm_substream *substream, snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
/* This code assumes port 1 for playback and port 2 for capture */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) port_config.num = 1; else port_config.num = 2; diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index dd6ccf17afd43..a8b29df666645 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -1351,7 +1351,7 @@ static int rt711_sdca_pcm_hw_params(struct snd_pcm_substream *substream, /* SoundWire specific configuration */ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { port_config.num = 3; } else { if (dai->id == RT711_AIF1) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5446f9506a167..49c595f2ae4c1 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -1006,7 +1006,7 @@ static int rt711_pcm_hw_params(struct snd_pcm_substream *substream, /* SoundWire specific configuration */ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { port_config.num = 3; } else { if (dai->id == RT711_AIF1) diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c index e210c574bb74a..b36d4a61121e7 100644 --- a/sound/soc/codecs/rt712-sdca.c +++ b/sound/soc/codecs/rt712-sdca.c @@ -1437,7 +1437,7 @@ static int rt712_sdca_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL;
/* SoundWire specific configuration */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { direction = SDW_DATA_DIR_RX; if (dai->id == RT712_AIF1) port = 1; diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index e5bd9ef812de1..5e791a808b654 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -1183,7 +1183,7 @@ static int rt722_sdca_pcm_hw_params(struct snd_pcm_substream *substream, * RT722_AIF2 with port = 3 for speaker playback * RT722_AIF3 with port = 6 for digital-mic capture */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { direction = SDW_DATA_DIR_RX; if (dai->id == RT722_AIF1) port = 1;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/wm2200.c | 4 ++-- sound/soc/codecs/wm5100.c | 4 ++-- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8960.c | 4 ++-- sound/soc/codecs/wm8961.c | 4 ++-- sound/soc/codecs/wm8994.c | 6 +++--- sound/soc/codecs/wm8995.c | 4 ++-- sound/soc/codecs/wm8996.c | 4 ++-- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 14 files changed, 24 insertions(+), 24 deletions(-)
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 841247173d98e..b755be789c4c7 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1749,7 +1749,7 @@ static int wm2200_hw_params(struct snd_pcm_substream *substream,
lrclk = bclk_rates[bclk] / params_rate(params); dev_dbg(component->dev, "Setting %dHz LRCLK\n", bclk_rates[bclk] / lrclk); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(substream) || wm2200->symmetric_rates) snd_soc_component_update_bits(component, WM2200_AUDIO_IF_1_7, WM2200_AIF1RX_BCPF_MASK, lrclk); @@ -1758,7 +1758,7 @@ static int wm2200_hw_params(struct snd_pcm_substream *substream, WM2200_AIF1TX_BCPF_MASK, lrclk);
i = (wl << WM2200_AIF1TX_WL_SHIFT) | wl; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_soc_component_update_bits(component, WM2200_AUDIO_IF_1_9, WM2200_AIF1RX_WL_MASK | WM2200_AIF1RX_SLOT_LEN_MASK, i); diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 11bbc94a282c7..b55c90f00ac5e 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1478,7 +1478,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream,
lrclk = bclk_rates[bclk] / params_rate(params); dev_dbg(component->dev, "Setting %dHz LRCLK\n", bclk_rates[bclk] / lrclk); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(substream) || wm5100->aif_symmetric[dai->id]) snd_soc_component_update_bits(component, base + 7, WM5100_AIF1RX_BCPF_MASK, lrclk); @@ -1487,7 +1487,7 @@ static int wm5100_hw_params(struct snd_pcm_substream *substream, WM5100_AIF1TX_BCPF_MASK, lrclk);
i = (wl << WM5100_AIF1TX_WL_SHIFT) | fl; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_soc_component_update_bits(component, base + 9, WM5100_AIF1RX_WL_MASK | WM5100_AIF1RX_SLOT_LEN_MASK, i); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 66bd281095e1c..05178509bdfab 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -931,7 +931,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, /* The sloping stopband filter is recommended for use with * lower sample rates to improve performance. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (params_rate(params) < 24000) wm8350_set_bits(wm8350, WM8350_DAC_MUTE_VOLUME, WM8350_DAC_SB_FILT); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 73a8edc797fb2..f00c3c1e62332 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -576,7 +576,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, dev_dbg(component->dev, "Running at %dfs with %dHz clock\n", wm8580_sysclk_ratios[i], wm8580->sysclk[dai->driver->id]);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (ratio) { case 128: case 192: diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e44fdf97796f3..f084143c8171e 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -649,7 +649,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream,
snd_soc_component_write(component, WM8900_REG_AUDIO1, reg);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { reg = snd_soc_component_read(component, WM8900_REG_DACCTRL);
if (params_rate(params) <= 24000) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 8a532f7d750c8..cda6c4d8e129b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -408,7 +408,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, return ret;
/* LoutR control */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE + if (snd_pcm_is_capture(substream) && params_channels(params) == 2) iface |= (1 << 9);
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 00858b9c95686..9ba1edd5ba9a9 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -833,7 +833,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); u16 iface = snd_soc_component_read(component, WM8960_IFACE1) & 0xfff3; - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); int i;
wm8960->bclk = snd_soc_params_to_bclk(params); @@ -891,7 +891,7 @@ static int wm8960_hw_free(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream);
wm8960->is_stream_in_use[tx] = false;
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index d1c731e25777b..6b0e1f76358ec 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -531,12 +531,12 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream, /* Select a CLK_SYS/fs ratio equal to or higher than required */ target = wm8961->sysclk / fs;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && target < 64) { + if (snd_pcm_is_playback(substream) && target < 64) { dev_err(component->dev, "SYSCLK must be at least 64*fs for DAC\n"); return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && target < 256) { + if (snd_pcm_is_capture(substream) && target < 256) { dev_err(component->dev, "SYSCLK must be at least 256*fs for ADC\n"); return -EINVAL; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a99908582a50a..d4adbefae32be 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2934,7 +2934,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, aif2_reg = WM8994_AIF1_CONTROL_2; bclk_reg = WM8994_AIF1_BCLK; rate_reg = WM8994_AIF1_RATE; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(substream) || wm8994->lrclk_shared[0]) { lrclk_reg = WM8994_AIF1DAC_LRCLK; } else { @@ -2947,7 +2947,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, aif2_reg = WM8994_AIF2_CONTROL_2; bclk_reg = WM8994_AIF2_BCLK; rate_reg = WM8994_AIF2_RATE; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(substream) || wm8994->lrclk_shared[1]) { lrclk_reg = WM8994_AIF2DAC_LRCLK; } else { @@ -3069,7 +3069,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, rate_reg, WM8994_AIF1_SR_MASK | WM8994_AIF1CLK_RATE_MASK, rate_val);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { switch (dai->id) { case 1: wm8994->dac_rates[0] = params_rate(params); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 1f9a9b6369350..2c2074b9a6bdf 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1563,7 +1563,7 @@ static int wm8995_hw_params(struct snd_pcm_substream *substream, aif1_reg = WM8995_AIF1_CONTROL_1; bclk_reg = WM8995_AIF1_BCLK; rate_reg = WM8995_AIF1_RATE; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK /* || + if (snd_pcm_is_playback(substream) /* || wm8995->lrclk_shared[0] */) { lrclk_reg = WM8995_AIF1DAC_LRCLK; } else { @@ -1575,7 +1575,7 @@ static int wm8995_hw_params(struct snd_pcm_substream *substream, aif1_reg = WM8995_AIF2_CONTROL_1; bclk_reg = WM8995_AIF2_BCLK; rate_reg = WM8995_AIF2_RATE; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK /* || + if (snd_pcm_is_playback(substream) /* || wm8995->lrclk_shared[1] */) { lrclk_reg = WM8995_AIF2DAC_LRCLK; } else { diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 5c06cea09bd18..d9c0bd6b09925 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1740,7 +1740,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream,
switch (dai->id) { case 0: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(substream) || (snd_soc_component_read(component, WM8996_GPIO_1)) & WM8996_GP1_FN_MASK) { aifdata_reg = WM8996_AIF1RX_DATA_CONFIGURATION; lrclk_reg = WM8996_AIF1_RX_LRCLK_1; @@ -1751,7 +1751,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, dsp_shift = 0; break; case 1: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + if (snd_pcm_is_playback(substream) || (snd_soc_component_read(component, WM8996_GPIO_2)) & WM8996_GP2_FN_MASK) { aifdata_reg = WM8996_AIF2RX_DATA_CONFIGURATION; lrclk_reg = WM8996_AIF2_RX_LRCLK_1; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 5c6aebe29cf13..97f3c9c7a4413 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -236,7 +236,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x1, 0x1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e63921de0c37a..d3a190c06ea9c 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -529,7 +529,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x1, 0x1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; @@ -546,7 +546,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x1, 0x1); snd_soc_component_update_bits(component, AC97_PCI_SID, 0x8000, 0x8000);
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -ENODEV;
return snd_soc_component_write(component, AC97_PCM_SURR_DAC_RATE, runtime->rate); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 64b69316e4c70..bf2824be4f0d4 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1032,7 +1032,7 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x0001, 0x0001);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; @@ -1049,7 +1049,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x0001, 0x0001); snd_soc_component_update_bits(component, AC97_PCI_SID, 0x8000, 0x8000);
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -ENODEV;
return snd_soc_component_write(component, AC97_PCM_SURR_DAC_RATE, runtime->rate);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 0e013edfe63d7..aceeeae518d29 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -35,7 +35,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
- int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + int reg = snd_pcm_is_playback(substream) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/cpcap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 4f9dabd9d78a6..e8309458eb86e 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1271,7 +1271,7 @@ static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, if (err) return err;
- if (direction == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(direction)) { mask = 0x0000; mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0); mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT1);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/es8311.c | 4 ++-- sound/soc/codecs/es8326.c | 4 ++-- sound/soc/codecs/es8328.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/es8311.c b/sound/soc/codecs/es8311.c index f557e33c26ad9..6a839bd5ea3b9 100644 --- a/sound/soc/codecs/es8311.c +++ b/sound/soc/codecs/es8311.c @@ -452,7 +452,7 @@ static int es8311_mute(struct snd_soc_dai *dai, int mute, int direction) struct snd_soc_component *component = dai->component; struct es8311_priv *es8311 = snd_soc_component_get_drvdata(component);
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { unsigned int mask = ES8311_DAC1_DAC_DSMMUTE | ES8311_DAC1_DAC_DEMMUTE; unsigned int val = mute ? mask : 0; @@ -508,7 +508,7 @@ static int es8311_hw_params(struct snd_pcm_substream *substream, } unsigned int width = (unsigned int)par_width;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { snd_soc_component_update_bits(component, ES8311_SDP_IN, ES8311_SDP_WL_MASK, wl << ES8311_SDP_WL_SHIFT); diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 60877116c0ef6..fd3e89cc02862 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -605,7 +605,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) unsigned int offset_l, offset_r;
if (mute) { - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { regmap_write(es8326->regmap, ES8326_HP_CAL, ES8326_HP_OFF); regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, ES8326_MUTE_MASK, ES8326_MUTE); @@ -627,7 +627,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x00); diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index f3c97da798dc8..0c371da80c7e3 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -483,7 +483,7 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, int wl; int ratio;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) reg = ES8328_DACCONTROL2; else reg = ES8328_ADCCONTROL5; @@ -535,7 +535,7 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { snd_soc_component_update_bits(component, ES8328_DACCONTROL1, ES8328_DACCONTROL1_DACWL_MASK, wl << ES8328_DACCONTROL1_DACWL_SHIFT);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/ad193x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 1d3c4d94b4ae9..34c309c0d96ff 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -316,7 +316,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, int word_len = 0, master_rate = 0; struct snd_soc_component *component = dai->component; struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(component); - bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool is_playback = snd_pcm_is_playback(substream); u8 dacc0;
dev_dbg(dai->dev, "%s() rate=%u format=%#x width=%u channels=%u\n",
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/ak4613.c | 4 ++-- sound/soc/codecs/ak4619.c | 2 +- sound/soc/codecs/ak4641.c | 2 +- sound/soc/codecs/ak4642.c | 4 ++-- 4 files changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index de9e431855559..3d73c659545d1 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -434,7 +434,7 @@ static void ak4613_hw_constraints(struct ak4613_priv *priv, unsigned int mask; unsigned int mode; unsigned int fs; - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int is_play = snd_pcm_is_playback(substream); int sdti_num; int i;
@@ -739,7 +739,7 @@ static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, (cmd != SNDRV_PCM_TRIGGER_RESUME)) return 0;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return 0;
priv->component = component; diff --git a/sound/soc/codecs/ak4619.c b/sound/soc/codecs/ak4619.c index 8f2442482f725..1d63e5c447e1b 100644 --- a/sound/soc/codecs/ak4619.c +++ b/sound/soc/codecs/ak4619.c @@ -538,7 +538,7 @@ static int ak4619_dai_hw_params(struct snd_pcm_substream *substream, unsigned int width; unsigned int rate; unsigned int fs; - bool is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool is_play = snd_pcm_is_playback(substream); u8 dai_ctrl = 0; u8 clk_mode = 0;
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index ec33e7d73c6c2..179d391083c57 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -344,7 +344,7 @@ static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4641_MODE2, (0x3 << 5), mode2);
/* Update de-emphasis filter for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ak4641->playback_fs = rate; ak4641_set_deemph(component); } diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index fe035d2fc9131..9cf3bc9387b13 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -277,7 +277,7 @@ static const struct reg_default ak4648_reg[] = { static int ak4642_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int is_play = snd_pcm_is_playback(substream); struct snd_soc_component *component = dai->component;
if (is_play) { @@ -320,7 +320,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int is_play = snd_pcm_is_playback(substream); struct snd_soc_component *component = dai->component;
if (is_play) {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/jz4725b.c | 2 +- sound/soc/codecs/jz4760.c | 8 ++++---- sound/soc/codecs/jz4770.c | 8 ++++---- 3 files changed, 9 insertions(+), 9 deletions(-)
diff --git a/sound/soc/codecs/jz4725b.c b/sound/soc/codecs/jz4725b.c index 39cebaa167beb..7add57e5c4bac 100644 --- a/sound/soc/codecs/jz4725b.c +++ b/sound/soc/codecs/jz4725b.c @@ -476,7 +476,7 @@ static int jz4725b_codec_hw_params(struct snd_pcm_substream *substream, if (rate == ARRAY_SIZE(jz4725b_codec_sample_rates)) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(icdc->regmap, JZ4725B_CODEC_REG_CR2, REG_CR2_DAC_ADWL_MASK, diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index 6217e611259fe..f39bb0dc93e02 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -205,7 +205,7 @@ static int jz4760_codec_startup(struct snd_pcm_substream *substream, * DMA transfer going during playback when all audible outputs have * been disabled. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ret = snd_soc_dapm_force_enable_pin(dapm, "SYSCLK"); return ret; } @@ -216,7 +216,7 @@ static void jz4760_codec_shutdown(struct snd_pcm_substream *substream, struct snd_soc_component *codec = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(codec);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_soc_dapm_disable_pin(dapm, "SYSCLK"); }
@@ -231,7 +231,7 @@ static int jz4760_codec_pcm_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) snd_soc_component_force_bias_level(codec, SND_SOC_BIAS_ON); break; case SNDRV_PCM_TRIGGER_STOP: @@ -693,7 +693,7 @@ static int jz4760_codec_hw_params(struct snd_pcm_substream *substream, if (rate == ARRAY_SIZE(jz4760_codec_sample_rates)) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(codec->regmap, JZ4760_CODEC_REG_AICR, REG_AICR_DAC_ADWL_MASK, FIELD_PREP(REG_AICR_DAC_ADWL_MASK, bit_width)); diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index acb9eaa7ea1c5..740b4e926c76c 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -224,7 +224,7 @@ static int jz4770_codec_startup(struct snd_pcm_substream *substream, * DMA transfer going during playback when all audible outputs have * been disabled. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_soc_dapm_force_enable_pin(dapm, "SYSCLK");
return 0; @@ -236,7 +236,7 @@ static void jz4770_codec_shutdown(struct snd_pcm_substream *substream, struct snd_soc_component *codec = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(codec);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) snd_soc_dapm_disable_pin(dapm, "SYSCLK"); }
@@ -251,7 +251,7 @@ static int jz4770_codec_pcm_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) snd_soc_component_force_bias_level(codec, SND_SOC_BIAS_ON); break; @@ -730,7 +730,7 @@ static int jz4770_codec_hw_params(struct snd_pcm_substream *substream, if (rate == ARRAY_SIZE(jz4770_codec_sample_rates)) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { regmap_update_bits(codec->regmap, JZ4770_CODEC_REG_AICR_DAC, REG_AICR_DAC_ADWL_MASK, bit_width << REG_AICR_DAC_ADWL_OFFSET);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/mt6351.c | 4 ++-- sound/soc/codecs/mt6358.c | 4 ++-- sound/soc/codecs/mt6359.c | 12 ++++++------ 3 files changed, 10 insertions(+), 10 deletions(-)
diff --git a/sound/soc/codecs/mt6351.c b/sound/soc/codecs/mt6351.c index 2a5e963fb2b57..7bc8a6fd88262 100644 --- a/sound/soc/codecs/mt6351.c +++ b/sound/soc/codecs/mt6351.c @@ -270,9 +270,9 @@ static int mt6351_codec_dai_hw_params(struct snd_pcm_substream *substream, dev_dbg(priv->dev, "%s(), substream->stream %d, rate %d\n", __func__, substream->stream, rate);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) priv->dl_rate = rate; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) priv->ul_rate = rate;
return 0; diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 9247b90d1b99e..a0b34508f78f4 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -2363,9 +2363,9 @@ static int mt6358_codec_dai_hw_params(struct snd_pcm_substream *substream, rate, substream->number);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) priv->dl_rate = rate; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) priv->ul_rate = rate;
return 0; diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 0b76a55664b03..8a302607316cc 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -2653,9 +2653,9 @@ static int mt6359_codec_dai_hw_params(struct snd_pcm_substream *substream, dev_dbg(priv->dev, "%s(), id %d, substream->stream %d, rate %d, number %d\n", __func__, id, substream->stream, rate, substream->number);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) priv->dl_rate[id] = rate; - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) priv->ul_rate[id] = rate;
return 0; @@ -2668,9 +2668,9 @@ static int mt6359_codec_dai_startup(struct snd_pcm_substream *substream, struct mt6359_priv *priv = snd_soc_component_get_drvdata(cmpnt);
dev_dbg(priv->dev, "%s stream %d\n", __func__, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) mt6359_set_playback_gpio(priv); - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) mt6359_set_capture_gpio(priv);
return 0; @@ -2683,9 +2683,9 @@ static void mt6359_codec_dai_shutdown(struct snd_pcm_substream *substream, struct mt6359_priv *priv = snd_soc_component_get_drvdata(cmpnt);
dev_dbg(priv->dev, "%s stream %d\n", __func__, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) mt6359_reset_playback_gpio(priv); - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) mt6359_reset_capture_gpio(priv); }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/sta529.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index f7718491c8996..e750536b1f740 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -232,7 +232,7 @@ static int sta529_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { snd_soc_component_update_bits(component, STA529_S2PCFG1, PDATA_LEN_MSK, pdata << 6); snd_soc_component_update_bits(component, STA529_S2PCFG1, BCLK_TO_FS_MSK,
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/rk3308_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/rk3308_codec.c b/sound/soc/codecs/rk3308_codec.c index 8b51e87a17115..7a36cffc4f3be 100644 --- a/sound/soc/codecs/rk3308_codec.c +++ b/sound/soc/codecs/rk3308_codec.c @@ -674,7 +674,7 @@ static int rk3308_codec_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct rk3308_codec_priv *rk3308 = snd_soc_component_get_drvdata(component);
- return (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + return (snd_pcm_is_playback(substream)) ? rk3308_codec_dac_dig_config(rk3308, params) : rk3308_codec_adc_dig_config(rk3308, params); }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/wl1273.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 737ca82cf9764..2ea7264a18d43 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -298,13 +298,13 @@ static int wl1273_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_CHANNELS, 1); break; case WL1273_MODE_FM_RX: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { pr_err("Cannot play in RX mode.\n"); return -EINVAL; } break; case WL1273_MODE_FM_TX: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { pr_err("Cannot capture in TX mode.\n"); return -EINVAL; } @@ -348,13 +348,13 @@ static int wl1273_hw_params(struct snd_pcm_substream *substream, }
if (wl1273->mode == WL1273_MODE_FM_TX && - substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + snd_pcm_is_capture(substream)) { pr_err("Only playback supported with TX.\n"); return -EINVAL; }
if (wl1273->mode == WL1273_MODE_FM_RX && - substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_pcm_is_playback(substream)) { pr_err("Only capture supported with RX.\n"); return -EINVAL; }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/mc13783.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 086ac97e83866..77f9215699dbd 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -138,7 +138,7 @@ static int mc13783_pcm_hw_params_sync(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return mc13783_pcm_hw_params_dac(substream, params, dai); else return mc13783_pcm_hw_params_codec(substream, params, dai);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/nau8821.c | 4 ++-- sound/soc/codecs/nau8824.c | 4 ++-- sound/soc/codecs/nau8825.c | 4 ++-- 3 files changed, 6 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index de5c4db05c8f8..f887c192b87cf 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -803,7 +803,7 @@ nau8821_get_osr(struct nau8821 *nau8821, int stream) { unsigned int osr;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { regmap_read(nau8821->regmap, NAU8821_R2C_DAC_CTRL1, &osr); osr &= NAU8821_DAC_OVERSAMPLE_MASK; if (osr >= ARRAY_SIZE(osr_dac_sel)) @@ -854,7 +854,7 @@ static int nau8821_hw_params(struct snd_pcm_substream *substream, return -EINVAL; if (nau8821->fs * osr->osr > CLK_DA_AD_MAX) return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, NAU8821_CLK_DAC_SRC_MASK, osr->clk_src << NAU8821_CLK_DAC_SRC_SFT); diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d5..21cda3b473bad 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -1038,7 +1038,7 @@ nau8824_get_osr(struct nau8824 *nau8824, int stream) { unsigned int osr;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { regmap_read(nau8824->regmap, NAU8824_REG_DAC_FILTER_CTRL_1, &osr); osr &= NAU8824_DAC_OVERSAMPLE_MASK; @@ -1094,7 +1094,7 @@ static int nau8824_hw_params(struct snd_pcm_substream *substream, goto error; if (nau8824->fs * osr->osr > CLK_DA_AD_MAX) goto error; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(nau8824->regmap, NAU8824_REG_CLK_DIVIDER, NAU8824_CLK_DAC_SRC_MASK, osr->clk_src << NAU8824_CLK_DAC_SRC_SFT); diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index bde25bc6909d5..951406fa5d32f 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1238,7 +1238,7 @@ nau8825_get_osr(struct nau8825 *nau8825, int stream) { unsigned int osr;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream)) { regmap_read(nau8825->regmap, NAU8825_REG_DAC_CTRL1, &osr); osr &= NAU8825_DAC_OVERSAMPLE_MASK; @@ -1294,7 +1294,7 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, goto error; if (params_rate(params) * osr->osr > CLK_DA_AD_MAX) goto error; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_DAC_SRC_MASK, osr->clk_src << NAU8825_CLK_DAC_SRC_SFT);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/peb2466.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b9..7c5c575300671 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -796,7 +796,7 @@ static int peb2466_dai_startup(struct snd_pcm_substream *substream, unsigned int max_ch; int ret;
- max_ch = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + max_ch = snd_pcm_is_playback(substream) ? peb2466->max_chan_playback : peb2466->max_chan_capture;
/*
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/sma1303.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/sma1303.c b/sound/soc/codecs/sma1303.c index 980c48cbc3482..4da80cbab7946 100644 --- a/sound/soc/codecs/sma1303.c +++ b/sound/soc/codecs/sma1303.c @@ -997,7 +997,7 @@ static int sma1303_dai_hw_params_amp(struct snd_pcm_substream *substream, __func__, params_rate(params), params_width(params), params_channels(params));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (sma1303->sys_clk_id == SMA1303_PLL_CLKIN_BCLK) { if (sma1303->last_bclk != bclk) { sma1303_setup_pll(component, bclk); @@ -1195,7 +1195,7 @@ static int sma1303_dai_mute(struct snd_soc_dai *dai, int mute, int stream) struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); int ret = 0;
- if (stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(stream)) return ret;
if (mute) {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/adav80x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index c8c0fc9282116..21d20cb42d200 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -521,7 +521,7 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream, if (rate * 256 != adav80x->sysclk) return -EINVAL;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { adav80x_set_playback_pcm_format(component, dai, params); adav80x_set_dac_clock(component, rate); } else {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/twl4030.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 9c50ac356c895..a36b34b92adab 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1629,7 +1629,7 @@ static void twl4030_tdm_enable(struct snd_soc_component *component, int directio
reg = twl4030_read(component, TWL4030_REG_OPTION);
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(direction)) mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN; else mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; @@ -1913,7 +1913,7 @@ static void twl4030_voice_enable(struct snd_soc_component *component, int direct
reg = twl4030_read(component, TWL4030_REG_OPTION);
- if (direction == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(direction)) mask = TWL4030_ARXL1_VRX_EN; else mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/uda1380.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 4f8fdd574585b..7ef7b5fc927f2 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -556,7 +556,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, uda1380_write(component, UDA1380_PM, R02_PON_PLL | pm); }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) clk |= R00_EN_DAC | R00_EN_INT; else clk |= R00_EN_ADC | R00_EN_DEC; @@ -577,7 +577,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, uda1380_write(component, UDA1380_PM, ~R02_PON_PLL & pm); }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) clk &= ~(R00_EN_DAC | R00_EN_INT); else clk &= ~(R00_EN_ADC | R00_EN_DEC);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd934x.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 373a31ddccb2d..f66f4134af8f1 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1732,7 +1732,7 @@ static int wcd9335_slim_set_hw_params(struct wcd9335_codec *wcd, i = 0; list_for_each_entry(ch, slim_ch_list, list) { cfg->chs[i++] = ch->ch_num; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { /* write to interface device */ ret = regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_RX_PORT_MULTI_CHNL_0(ch->port), diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 291d0c80a6fcf..82894ec51a53a 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1732,7 +1732,7 @@ static int wcd934x_slim_set_hw_params(struct wcd934x_codec *wcd, i = 0; list_for_each_entry(ch, slim_ch_list, list) { cfg->chs[i++] = ch->ch_num; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(direction)) { /* write to interface device */ ret = regmap_write(wcd->if_regmap, WCD934X_SLIM_PGD_RX_PORT_MULTI_CHNL_0(ch->port),
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/zl38060.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/zl38060.c b/sound/soc/codecs/zl38060.c index 28c92d90299e9..8650c9ca0cefd 100644 --- a/sound/soc/codecs/zl38060.c +++ b/sound/soc/codecs/zl38060.c @@ -271,7 +271,7 @@ static int zl38_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct zl38_codec_priv *priv = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); unsigned int fsrate; int err;
@@ -317,7 +317,7 @@ static int zl38_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct zl38_codec_priv *priv = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream);
priv->is_stream_in_use[tx] = false;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adau17x1.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df7f75..8f97eaca0be59 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -469,7 +469,7 @@ static int adau1701_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(adau1701->regmap, ADAU1701_DSPCTRL, ADAU1701_DSPCTRL_SR_MASK, val);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return adau1701_set_playback_pcm_format(component, params); else return adau1701_set_capture_pcm_format(component, params); diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index f2932713b4de9..2d3579bdc91d1 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -208,7 +208,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol, break; }
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) reg = ADAU17X1_SERIAL_INPUT_ROUTE; else reg = ADAU17X1_SERIAL_OUTPUT_ROUTE; @@ -237,7 +237,7 @@ static int adau17x1_dsp_mux_enum_get(struct snd_kcontrol *kcontrol, unsigned int reg, val; int ret;
- if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) reg = ADAU17X1_SERIAL_INPUT_ROUTE; else reg = ADAU17X1_SERIAL_OUTPUT_ROUTE;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/hdac_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 29c88de5508b8..d5b72afd82709 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -220,7 +220,7 @@ static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, unsigned int maxbps; unsigned int bits;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) maxbps = dai->driver->playback.sig_bits; else maxbps = dai->driver->capture.sig_bits;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/max98363.c | 2 +- sound/soc/codecs/max98373-sdw.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index 950105e5bffdc..5ea6b36941355 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -221,7 +221,7 @@ static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream, if (!max98363->slave) return -EINVAL;
- if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + if (!snd_pcm_is_playback(substream)) return -EINVAL;
direction = SDW_DATA_DIR_RX; diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 26860882fd91a..531c67023f922 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -534,7 +534,7 @@ static int max98373_sdw_dai_hw_params(struct snd_pcm_substream *substream,
snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { port_config.num = 1;
if (max98373->slot) {
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7aa89e34657ea..5b2544e05db7e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1061,7 +1061,7 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return -EFAULT; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) stereo = SGTL5000_DAC_STEREO; else stereo = SGTL5000_ADC_STEREO;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/stac9766.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 2f9f10a4dfed9..3511776af8f95 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -171,7 +171,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, /* enable variable rate audio, disable SPDIF output */ snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x5, 0x1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/tscs42xx.c | 4 ++-- sound/soc/codecs/tscs454.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index f8a3d1b40990c..dd87a37b97854 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1178,12 +1178,12 @@ static int tscs42xx_mute_stream(struct snd_soc_dai *dai, int mute, int stream) int ret;
if (mute) - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) ret = dac_mute(component); else ret = adc_mute(component); else - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) ret = dac_unmute(component); else ret = adc_unmute(component); diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index 850e5de9271ed..b8ee1281db489 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3221,7 +3221,7 @@ static int tscs454_hw_params(struct snd_pcm_substream *substream, }
set_aif_status_active(&tscs454->aifs_status, aif->id, - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + snd_pcm_is_playback(substream));
dev_dbg(component->dev, "Set aif %d active. Streams status is 0x%x\n", aif->id, tscs454->aifs_status.streams); @@ -3241,7 +3241,7 @@ static int tscs454_hw_free(struct snd_pcm_substream *substream, struct aif *aif = &tscs454->aifs[dai->id];
return aif_free(component, aif, - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + snd_pcm_is_playback(substream)); }
static int tscs454_prepare(struct snd_pcm_substream *substream,
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/idt821034.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/idt821034.c b/sound/soc/codecs/idt821034.c index cb7a68c799f8f..82279f6d3ce84 100644 --- a/sound/soc/codecs/idt821034.c +++ b/sound/soc/codecs/idt821034.c @@ -839,7 +839,7 @@ static int idt821034_dai_startup(struct snd_pcm_substream *substream, unsigned int max_ch = 0; int ret;
- max_ch = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + max_ch = snd_pcm_is_playback(substream) ? idt821034->max_ch_playback : idt821034->max_ch_capture;
/*
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/sdw-mockup.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/sdw-mockup.c b/sound/soc/codecs/sdw-mockup.c index 574c08b14f0c2..24fabd392d00b 100644 --- a/sound/soc/codecs/sdw-mockup.c +++ b/sound/soc/codecs/sdw-mockup.c @@ -72,7 +72,7 @@ static int sdw_mockup_pcm_hw_params(struct snd_pcm_substream *substream, /* SoundWire specific configuration */ snd_sdw_params_to_config(substream, params, &stream_config, &port_config);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) port_config.num = 1; else port_config.num = 8;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/hdmi-codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 74caae52e1273..f8a4a1b62698e 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -428,7 +428,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); bool has_capture = !hcp->hcd.no_i2s_capture; bool has_playback = !hcp->hcd.no_i2s_playback; int ret = 0; @@ -474,7 +474,7 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); - bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool tx = snd_pcm_is_playback(substream); bool has_capture = !hcp->hcd.no_i2s_capture; bool has_playback = !hcp->hcd.no_i2s_playback;
@@ -699,7 +699,7 @@ static int hdmi_codec_mute(struct snd_soc_dai *dai, int mute, int direction) * snd_soc_dai_digital_mute() */ if (hcp->hcd.ops->mute_stream && - (direction == SNDRV_PCM_STREAM_PLAYBACK || + (snd_pcm_is_playback(direction) || !hcp->hcd.ops->no_capture_mute)) return hcp->hcd.ops->mute_stream(dai->dev->parent, hcp->hcd.data,
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/tlv320aic23.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c47aa4d4162dd..06dab9d9b7576 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -342,7 +342,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, u32 sample_rate_dac = aic23->requested_dac; u32 sample_rate = params_rate(params);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { aic23->requested_dac = sample_rate_dac = sample_rate; if (!sample_rate_adc) sample_rate_adc = sample_rate; @@ -398,7 +398,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, udelay(50); snd_soc_component_write(component, TLV320AIC23_ACTIVE, 0x0); } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) aic23->requested_dac = 0; else aic23->requested_adc = 0;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/codecs/framer-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/soc/codecs/framer-codec.c b/sound/soc/codecs/framer-codec.c index 6f57a3aeecc89..10ad78e87a7cd 100644 --- a/sound/soc/codecs/framer-codec.c +++ b/sound/soc/codecs/framer-codec.c @@ -192,7 +192,7 @@ static int framer_dai_startup(struct snd_pcm_substream *substream, u64 format; int ret;
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { format = framer_formats(framer->max_chan_capture); hw_rule_channels_by_format = framer_dai_hw_rule_capture_channels_by_format; hw_rule_format_by_channels = framer_dai_hw_rule_capture_format_by_channels;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/samsung/i2s.c | 8 ++++---- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/spdif.c | 2 +- 3 files changed, 7 insertions(+), 7 deletions(-)
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1bcabb114e29f..fdf494a49dd92 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -744,13 +744,13 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, val |= MOD_DC1_EN; break; case 2: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) i2s->dma_playback.addr_width = 4; else i2s->dma_capture.addr_width = 4; break; case 1: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) i2s->dma_playback.addr_width = 2; else i2s->dma_capture.addr_width = 2; @@ -936,7 +936,7 @@ static int i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct samsung_i2s_priv *priv = snd_soc_dai_get_drvdata(dai); - int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + int capture = snd_pcm_is_capture(substream); struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct i2s_dai *i2s = to_info(snd_soc_rtd_to_cpu(rtd, 0)); unsigned long flags; @@ -1026,7 +1026,7 @@ i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
WARN_ON(!pm_runtime_active(dai->dev));
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) delay = FIC_RXCOUNT(reg); else if (is_secondary(i2s)) delay = FICS_TXCOUNT(readl(priv->addr + I2SFICS)); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 573b2dee7f07c..a9bcc2adb4403 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -228,7 +228,7 @@ static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: spin_lock_irqsave(&pcm->lock, flags);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) s3c_pcm_snd_rxctrl(pcm, 1); else s3c_pcm_snd_txctrl(pcm, 1); @@ -241,7 +241,7 @@ static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: spin_lock_irqsave(&pcm->lock, flags);
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) s3c_pcm_snd_rxctrl(pcm, 0); else s3c_pcm_snd_txctrl(pcm, 0); diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index f44e3180e8d3d..d5eaeacefd230 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -187,7 +187,7 @@ static int spdif_hw_params(struct snd_pcm_substream *substream,
dev_dbg(spdif->dev, "Entered %s\n", __func__);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dma_data = spdif->dma_playback; else { dev_err(spdif->dev, "Capture is not supported\n");
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/kirkwood/kirkwood-dma.c | 10 +++++----- sound/soc/kirkwood/kirkwood-i2s.c | 6 +++--- 2 files changed, 8 insertions(+), 8 deletions(-)
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 036b42058272f..1a3749b50d0be 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -140,7 +140,7 @@ static int kirkwood_dma_open(struct snd_soc_component *component, writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (priv->substream_play) return -EBUSY; priv->substream_play = substream; @@ -161,7 +161,7 @@ static int kirkwood_dma_close(struct snd_soc_component *component, if (!priv) return 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) priv->substream_play = NULL; else priv->substream_rec = NULL; @@ -185,7 +185,7 @@ static int kirkwood_dma_hw_params(struct snd_soc_component *component, if (!dram) return 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) kirkwood_dma_conf_mbus_windows(priv->io, KIRKWOOD_PLAYBACK_WIN, addr, dram); else @@ -206,7 +206,7 @@ static int kirkwood_dma_prepare(struct snd_soc_component *component, size = (size>>2)-1; count = snd_pcm_lib_period_bytes(substream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { writel(count, priv->io + KIRKWOOD_PLAY_BYTE_INT_COUNT); writel(runtime->dma_addr, priv->io + KIRKWOOD_PLAY_BUF_ADDR); writel(size, priv->io + KIRKWOOD_PLAY_BUF_SIZE); @@ -227,7 +227,7 @@ static snd_pcm_uframes_t kirkwood_dma_pointer( struct kirkwood_dma_data *priv = kirkwood_priv(substream); snd_pcm_uframes_t count;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) count = bytes_to_frames(substream->runtime, readl(priv->io + KIRKWOOD_PLAY_BYTE_COUNT)); else diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d1eb90310afa2..5d43924bc1caf 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -258,7 +258,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int i2s_reg; unsigned long i2s_value;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { i2s_reg = KIRKWOOD_I2S_PLAYCTL; } else { i2s_reg = KIRKWOOD_I2S_RECCTL; @@ -314,7 +314,7 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (params_channels(params) == 1) ctl_play |= KIRKWOOD_PLAYCTL_MONO_BOTH; else @@ -501,7 +501,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return kirkwood_i2s_play_trigger(substream, cmd, dai); else return kirkwood_i2s_rec_trigger(substream, cmd, dai);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/loongson/loongson_dma.c | 2 +- sound/soc/loongson/loongson_i2s.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c index 4fcc2868160bb..3a7018cae33e4 100644 --- a/sound/soc/loongson/loongson_dma.c +++ b/sound/soc/loongson/loongson_dma.c @@ -176,7 +176,7 @@ static int loongson_pcm_hw_params(struct snd_soc_component *component, desc->daddr = prtd->dma_data->dev_addr;
desc->cmd = BIT(0); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) desc->cmd |= BIT(12);
desc->length = period_len >> 2; diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c index d45228a3a558b..6b3a4d57a4b46 100644 --- a/sound/soc/loongson/loongson_i2s.c +++ b/sound/soc/loongson/loongson_i2s.c @@ -31,7 +31,7 @@ static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN); @@ -43,7 +43,7 @@ static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, 0); else @@ -95,7 +95,7 @@ static int loongson_i2s_hw_params(struct snd_pcm_substream *substream, regmap_read(i2s->regmap, LS_I2S_CFG, &val); val |= (bits << 24); val |= (bclk_ratio << 8); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) val |= (bits << 16); else val |= bits;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 4 ++-- sound/soc/mediatek/common/mtk-btcvsd.c | 18 +++++++++--------- sound/soc/mediatek/common/mtk-dsp-sof-common.c | 4 ++-- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 10 +++++----- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 2 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 2 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 2 +- sound/soc/mediatek/mt8186/mt8186-dai-src.c | 4 ++-- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 2 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 2 +- sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 2 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 2 +- 13 files changed, 29 insertions(+), 29 deletions(-)
diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index 3044d9ab3d4d9..71223feefa1af 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -61,7 +61,7 @@ int mtk_afe_fe_startup(struct snd_pcm_substream *substream, * This easily leads to overrun when avail_min is period_size. * One more period can hold the possible unread buffer. */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { int periods_max = mtk_afe_hardware->periods_max;
ret = snd_pcm_hw_constraint_minmax(runtime, @@ -268,7 +268,7 @@ int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, int id = snd_soc_rtd_to_cpu(rtd, 0)->id; int pbuf_size;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { if (afe->get_memif_pbuf_size) { pbuf_size = afe->get_memif_pbuf_size(substream); mtk_memif_set_pbuf_size(afe, id, pbuf_size); diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index c12d170fa1de6..a896f0c01e600 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -647,7 +647,7 @@ static int wait_for_bt_irq(struct mtk_btcvsd_snd *bt,
while (max_timeout_trial && !bt_stream->wait_flag) { t1 = sched_clock(); - if (bt_stream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(bt_stream->stream)) { ret = wait_event_interruptible_timeout(bt->tx_wait, bt_stream->wait_flag, nsecs_to_jiffies(timeout_limit)); @@ -850,7 +850,7 @@ static ssize_t mtk_btcvsd_snd_write(struct mtk_btcvsd_snd *bt, static struct mtk_btcvsd_snd_stream *get_bt_stream (struct mtk_btcvsd_snd *bt, struct snd_pcm_substream *substream) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return bt->tx; else return bt->rx; @@ -879,7 +879,7 @@ static int mtk_pcm_btcvsd_open(struct snd_soc_component *component,
snd_soc_set_runtime_hwparams(substream, &mtk_btcvsd_hardware);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ret = mtk_btcvsd_snd_tx_init(bt); bt->tx->substream = substream; } else { @@ -909,7 +909,7 @@ static int mtk_pcm_btcvsd_hw_params(struct snd_soc_component *component, { struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(substream) && params_buffer_bytes(hw_params) % bt->tx->packet_size != 0) { dev_warn(bt->dev, "%s(), error, buffer size %d not valid\n", __func__, @@ -926,7 +926,7 @@ static int mtk_pcm_btcvsd_hw_free(struct snd_soc_component *component, { struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) btcvsd_tx_clean_buffer(bt);
return 0; @@ -958,7 +958,7 @@ static int mtk_pcm_btcvsd_trigger(struct snd_soc_component *component, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - hw_packet_ptr = stream == SNDRV_PCM_STREAM_PLAYBACK ? + hw_packet_ptr = snd_pcm_is_playback(stream) ? bt_stream->packet_r : bt_stream->packet_w; bt_stream->prev_packet_idx = hw_packet_ptr; bt_stream->prev_frame = 0; @@ -987,7 +987,7 @@ static snd_pcm_uframes_t mtk_pcm_btcvsd_pointer( spinlock_t *lock; /* spinlock for bt stream control */ unsigned long flags;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { lock = &bt->tx_lock; bt_stream = bt->tx; } else { @@ -996,7 +996,7 @@ static snd_pcm_uframes_t mtk_pcm_btcvsd_pointer( }
spin_lock_irqsave(lock, flags); - hw_packet_ptr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + hw_packet_ptr = snd_pcm_is_playback(substream) ? bt->tx->packet_r : bt->rx->packet_w;
/* get packet diff from last time */ @@ -1030,7 +1030,7 @@ static int mtk_pcm_btcvsd_copy(struct snd_soc_component *component, { struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(component);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return mtk_btcvsd_snd_write(bt, buf, count); else return mtk_btcvsd_snd_read(bt, buf, count); diff --git a/sound/soc/mediatek/common/mtk-dsp-sof-common.c b/sound/soc/mediatek/common/mtk-dsp-sof-common.c index bca758dca2c9a..050a72b5fc21e 100644 --- a/sound/soc/mediatek/common/mtk-dsp-sof-common.c +++ b/sound/soc/mediatek/common/mtk-dsp-sof-common.c @@ -200,13 +200,13 @@ int mtk_sof_card_late_probe(struct snd_soc_card *card) struct snd_soc_dapm_widget *widget = snd_soc_dai_get_widget(cpu_dai, conn->stream_dir);
memset(&route, 0, sizeof(route)); - if (conn->stream_dir == SNDRV_PCM_STREAM_CAPTURE && widget) { + if (snd_pcm_is_capture(conn->stream_dir) && widget) { snd_soc_dapm_widget_for_each_sink_path(widget, p) { route.source = conn->sof_dma; route.sink = p->sink->name; snd_soc_dapm_add_routes(&card->dapm, &route, 1); } - } else if (conn->stream_dir == SNDRV_PCM_STREAM_PLAYBACK && widget) { + } else if (snd_pcm_is_playback(conn->stream_dir) && widget) { snd_soc_dapm_widget_for_each_source_path(widget, p) { route.source = p->source->name; route.sink = conn->sof_dma; diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 6a17deb874df7..6bad411dcf243 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -160,7 +160,7 @@ static void mt2701_afe_i2s_shutdown(struct snd_pcm_substream *substream, mt2701_afe_i2s_path_disable(afe, i2s_path, substream->stream);
/* need to disable i2s-out path when disable i2s-in */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) mt2701_afe_i2s_path_disable(afe, i2s_path, !substream->stream);
exit: @@ -192,7 +192,7 @@ static int mt2701_i2s_path_enable(struct mtk_base_afe *afe, ASYS_I2S_CON_I2S_MODE | ASYS_I2S_CON_WIDE_MODE_SET(w_len);
- if (stream_dir == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(stream_dir)) { mask |= ASYS_I2S_IN_PHASE_FIX; val |= ASYS_I2S_IN_PHASE_FIX; reg = ASMI_TIMING_CON1; @@ -250,7 +250,7 @@ static int mt2701_afe_i2s_prepare(struct snd_pcm_substream *substream, i2s_path->occupied[substream->stream] = 1;
/* need to enable i2s-out path when enable i2s-in */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) mt2701_i2s_path_enable(afe, i2s_path, !substream->stream, substream->runtime->rate);
@@ -368,7 +368,7 @@ static int mt2701_simple_fe_startup(struct snd_pcm_substream *substream, int stream_dir = substream->stream;
/* can't run single DL & DLM at the same time */ - if (stream_dir == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(stream_dir)) { memif_tmp = &afe->memif[MT2701_MEMIF_DLM]; if (memif_tmp->substream) { dev_warn(afe->dev, "memif is not available"); @@ -387,7 +387,7 @@ static int mt2701_simple_fe_hw_params(struct snd_pcm_substream *substream, int stream_dir = substream->stream;
/* single DL use PAIR_INTERLEAVE */ - if (stream_dir == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream_dir)) regmap_update_bits(afe->regmap, AFE_MEMIF_PBUF_SIZE, AFE_MEMIF_PBUF_SIZE_DLM_MASK, diff --git a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c index 78f3ad758c120..baf5a46781071 100644 --- a/sound/soc/mediatek/mt6797/mt6797-dai-adda.c +++ b/sound/soc/mediatek/mt6797/mt6797-dai-adda.c @@ -158,7 +158,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", __func__, dai->id, substream->stream, rate);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { unsigned int dl_src2_con0 = 0; unsigned int dl_src2_con1 = 0;
diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index be69bcea2a786..8317fc1cc5e25 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -276,7 +276,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", __func__, dai->id, substream->stream, rate);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { unsigned int dl_src2_con0 = 0; unsigned int dl_src2_con1 = 0;
diff --git a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c index bafbef96a42da..08776982e44ff 100644 --- a/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c +++ b/sound/soc/mediatek/mt8186/mt8186-afe-pcm.c @@ -184,7 +184,7 @@ static int mt8186_fe_trigger(struct snd_pcm_substream *substream, int cmd, * for small latency record * ul memif need read some data before irq enable */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + if (snd_pcm_is_capture(substream) && ((runtime->period_size * 1000) / rate <= 10)) udelay(300);
@@ -219,7 +219,7 @@ static int mt8186_fe_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: if (afe_priv->xrun_assert[id] > 0) { - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { int avail = snd_pcm_capture_avail(runtime); /* alsa can trigger stop/start when occur xrun */ if (avail >= runtime->buffer_size) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index dbd157d1a1ea2..5243e263c105d 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -565,7 +565,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, dev_dbg(afe->dev, "%s(), id %d, stream %d, rate %d\n", __func__, id, substream->stream, rate);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { unsigned int dl_src2_con0; unsigned int dl_src2_con1;
diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-src.c b/sound/soc/mediatek/mt8186/mt8186-dai-src.c index e475f4591aef5..5071fd69281c7 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-src.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-src.c @@ -560,7 +560,7 @@ static int mtk_dai_src_hw_params(struct snd_pcm_substream *substream, __func__, id, substream->stream, rate);
/* rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { src_priv->dl_rate = rate; if (id == MT8186_DAI_SRC_1) { sft = GENERAL1_ASRCIN_MODE_SFT; @@ -596,7 +596,7 @@ static int mtk_dai_src_hw_free(struct snd_pcm_substream *substream, dev_dbg(afe->dev, "%s(), id %d, stream %d\n", __func__, id, substream->stream);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) src_priv->dl_rate = 0; else src_priv->ul_rate = 0; diff --git a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c index ccb6c1f3adc7d..2058e0aa63993 100644 --- a/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c +++ b/sound/soc/mediatek/mt8188/mt8188-afe-pcm.c @@ -397,7 +397,7 @@ static int mt8188_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, fs << irq_data->irq_fs_shift);
/* delay for uplink */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { u32 sample_delay;
sample_delay = ((MEMIF_AXI_MINLEN + 1) * 64 + diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c index 8a17d1935c48f..58e1795bfad92 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-adda.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-adda.c @@ -414,7 +414,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream,
adda_priv->hires_required = (rate > ADDA_HIRES_THRES);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ret = mtk_dai_da_configure(afe, rate, id); else ret = mtk_dai_ad_configure(afe, rate, id); diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c index 99de85b876435..5fae5e877d449 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-adda.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-adda.c @@ -1067,7 +1067,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, unsigned int rate = params_rate(params); int id = dai->id;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { unsigned int dl_src2_con0 = 0; unsigned int dl_src2_con1 = 0;
diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c index 8da1587128ccf..18c63f7fc5407 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-adda.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-adda.c @@ -638,7 +638,7 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, else adda_priv->hires_required = 0;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) ret = mtk_dai_da_configure(afe, rate, dai->id); else ret = mtk_dai_ad_configure(afe, rate, dai->id);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/rockchip/rockchip_i2s.c | 6 +++--- sound/soc/rockchip/rockchip_i2s_tdm.c | 20 ++++++++++---------- sound/soc/rockchip/rockchip_pdm.c | 6 +++--- 3 files changed, 16 insertions(+), 16 deletions(-)
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b378f870b3ad2..09e4806071839 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -413,7 +413,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) regmap_update_bits(i2s->regmap, I2S_RXCR, I2S_RXCR_VDW_MASK | I2S_RXCR_CSR_MASK, val); @@ -471,7 +471,7 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) ret = rockchip_snd_rxctrl(i2s, 1); else ret = rockchip_snd_txctrl(i2s, 1); @@ -482,7 +482,7 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { if (!i2s->tx_start) i2s_pinctrl_select_bclk_off(i2s); ret = rockchip_snd_rxctrl(i2s, 0); diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index ee517d7b5b7bb..25d1a516962f6 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -287,7 +287,7 @@ static void rockchip_snd_txrxctrl(struct snd_pcm_substream *substream,
spin_lock_irqsave(&i2s_tdm->lock, flags); if (on) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) rockchip_enable_tde(i2s_tdm->regmap); else rockchip_enable_rde(i2s_tdm->regmap); @@ -301,7 +301,7 @@ static void rockchip_snd_txrxctrl(struct snd_pcm_substream *substream, I2S_XFER_RXS_START); } } else { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) rockchip_disable_tde(i2s_tdm->regmap); else rockchip_disable_rde(i2s_tdm->regmap); @@ -488,7 +488,7 @@ static void rockchip_i2s_tdm_xfer_pause(struct snd_pcm_substream *substream, int stream;
stream = SNDRV_PCM_STREAM_LAST - substream->stream; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) rockchip_disable_tde(i2s_tdm->regmap); else rockchip_disable_rde(i2s_tdm->regmap); @@ -502,7 +502,7 @@ static void rockchip_i2s_tdm_xfer_resume(struct snd_pcm_substream *substream, int stream;
stream = SNDRV_PCM_STREAM_LAST - substream->stream; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(stream)) rockchip_enable_tde(i2s_tdm->regmap); else rockchip_enable_rde(i2s_tdm->regmap); @@ -557,7 +557,7 @@ static int rockchip_i2s_io_multiplex(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (snd_pcm_is_capture(substream)) { struct snd_pcm_str *playback_str = &substream->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK];
@@ -629,7 +629,7 @@ static int rockchip_i2s_trcm_mode(struct snd_pcm_substream *substream, I2S_CKR_TSD_MASK | I2S_CKR_RSD_MASK, I2S_CKR_TSD(div_lrck) | I2S_CKR_RSD(div_lrck));
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) regmap_update_bits(i2s_tdm->regmap, I2S_TXCR, I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK, fmt); @@ -661,7 +661,7 @@ static int rockchip_i2s_tdm_hw_params(struct snd_pcm_substream *substream, mclk = i2s_tdm->mclk_tx; } else if (i2s_tdm->clk_trcm == TRCM_RX) { mclk = i2s_tdm->mclk_rx; - } else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + } else if (snd_pcm_is_playback(substream)) { mclk = i2s_tdm->mclk_tx; } else { mclk = i2s_tdm->mclk_rx; @@ -719,7 +719,7 @@ static int rockchip_i2s_tdm_hw_params(struct snd_pcm_substream *substream,
if (i2s_tdm->clk_trcm) { rockchip_i2s_trcm_mode(substream, dai, div_bclk, div_lrck, val); - } else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + } else if (snd_pcm_is_playback(substream)) { regmap_update_bits(i2s_tdm->regmap, I2S_CLKDIV, I2S_CLKDIV_TXM_MASK, I2S_CLKDIV_TXM(div_bclk)); @@ -755,7 +755,7 @@ static int rockchip_i2s_tdm_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (i2s_tdm->clk_trcm) rockchip_snd_txrxctrl(substream, dai, 1); - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) rockchip_snd_rxctrl(i2s_tdm, 1); else rockchip_snd_txctrl(i2s_tdm, 1); @@ -765,7 +765,7 @@ static int rockchip_i2s_tdm_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (i2s_tdm->clk_trcm) rockchip_snd_txrxctrl(substream, dai, 0); - else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + else if (snd_pcm_is_capture(substream)) rockchip_snd_rxctrl(i2s_tdm, 0); else rockchip_snd_txctrl(i2s_tdm, 0); diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index d16a4a67a6a2c..2e97a9e842a89 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -204,7 +204,7 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, bool change; int ret;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) return 0;
samplerate = params_rate(params); @@ -351,13 +351,13 @@ static int rockchip_pdm_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) rockchip_pdm_rxctrl(pdm, 1); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) rockchip_pdm_rxctrl(pdm, 0); break; default:
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/starfive/jh7110_tdm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/sound/soc/starfive/jh7110_tdm.c b/sound/soc/starfive/jh7110_tdm.c index 1e0ff67207471..c2b307558bfb5 100644 --- a/sound/soc/starfive/jh7110_tdm.c +++ b/sound/soc/starfive/jh7110_tdm.c @@ -146,7 +146,7 @@ static inline void jh7110_tdm_writel(struct jh7110_tdm_dev *tdm, u16 reg, u32 va static void jh7110_tdm_save_context(struct jh7110_tdm_dev *tdm, struct snd_pcm_substream *substream) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) tdm->saved_pcmtxcr = jh7110_tdm_readl(tdm, TDM_PCMTXCR); else tdm->saved_pcmrxcr = jh7110_tdm_readl(tdm, TDM_PCMRXCR); @@ -161,7 +161,7 @@ static void jh7110_tdm_start(struct jh7110_tdm_dev *tdm, jh7110_tdm_writel(tdm, TDM_PCMGBCR, data | PCMGBCR_ENABLE);
/* restore context */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) jh7110_tdm_writel(tdm, TDM_PCMTXCR, tdm->saved_pcmtxcr | PCMTXCR_TXEN); else jh7110_tdm_writel(tdm, TDM_PCMRXCR, tdm->saved_pcmrxcr | PCMRXCR_RXEN); @@ -172,7 +172,7 @@ static void jh7110_tdm_stop(struct jh7110_tdm_dev *tdm, { unsigned int val;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { val = jh7110_tdm_readl(tdm, TDM_PCMTXCR); val &= ~PCMTXCR_TXEN; jh7110_tdm_writel(tdm, TDM_PCMTXCR, val); @@ -237,7 +237,7 @@ static int jh7110_tdm_config(struct jh7110_tdm_dev *tdm, (tdm->tx.sl << SL_BIT) | (tdm->tx.lrj << LRJ_BIT);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) jh7110_tdm_writel(tdm, TDM_PCMTXCR, datatx); else jh7110_tdm_writel(tdm, TDM_PCMRXCR, datarx); @@ -380,7 +380,7 @@ static int jh7110_tdm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; }
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { tdm->tx.wl = chan_wl; tdm->tx.sl = chan_sl; tdm->tx.sscale = chan_nr;
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/uniphier/aio-cpu.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/uniphier/aio-cpu.c b/sound/soc/uniphier/aio-cpu.c index 470f129166a4c..8207e8fed2a80 100644 --- a/sound/soc/uniphier/aio-cpu.c +++ b/sound/soc/uniphier/aio-cpu.c @@ -64,12 +64,12 @@ static struct uniphier_aio_sub *find_volume(struct uniphier_aio_chip *chip, static bool match_spec(const struct uniphier_aio_spec *spec, const char *name, int dir) { - if (dir == SNDRV_PCM_STREAM_PLAYBACK && + if (snd_pcm_is_playback(dir) && spec->swm.dir != PORT_DIR_OUTPUT) { return false; }
- if (dir == SNDRV_PCM_STREAM_CAPTURE && + if (snd_pcm_is_capture(dir) && spec->swm.dir != PORT_DIR_INPUT) { return false; }
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/hisilicon/hi6210-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 250ae3781d140..37ffec8ff721f 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -421,7 +421,7 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream,
dma_data->maxburst = 2;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (snd_pcm_is_playback(substream)) dma_data->addr = i2s->base_phys + HII2S_ST_DL_CHANNEL; else dma_data->addr = i2s->base_phys + HII2S_STEREO_UPLINK_CHANNEL; @@ -478,14 +478,14 @@ static int hi6210_i2s_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) hi6210_i2s_rxctrl(cpu_dai, 1); else hi6210_i2s_txctrl(cpu_dai, 1); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) hi6210_i2s_rxctrl(cpu_dai, 0); else hi6210_i2s_txctrl(cpu_dai, 0);
We can use snd_pcm_is_playback/capture(). Let's use it.
Signed-off-by: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/sdw_utils/soc_sdw_maxim.c | 2 +- sound/soc/sdw_utils/soc_sdw_utils.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/soc/sdw_utils/soc_sdw_maxim.c b/sound/soc/sdw_utils/soc_sdw_maxim.c index cdcd8df37e1d3..714dadd75eedd 100644 --- a/sound/soc/sdw_utils/soc_sdw_maxim.c +++ b/sound/soc/sdw_utils/soc_sdw_maxim.c @@ -54,7 +54,7 @@ static int asoc_sdw_mx8373_enable_spk_pin(struct snd_pcm_substream *substream, b int j;
/* set spk pin by playback only */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (snd_pcm_is_capture(substream)) return 0;
cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 6183629d1754c..2b8f058450b25 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -750,7 +750,7 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream, return 0;
/* Identical data will be sent to all codecs in playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (snd_pcm_is_playback(substream)) { ch_mask = GENMASK(ch - 1, 0); step = 0; } else {
On Mon, Aug 05, 2024 at 12:33:38AM +0000, Kuninori Morimoto wrote:
Many drivers are using below code to know the Sound direction.
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
This patch-set add snd_pcm_is_playback/capture() macro to handle it.
Acked-by: Mark Brown broonie@kernel.org
On 05. 08. 24 2:33, Kuninori Morimoto wrote:
Hi ALSA-ML Cc Staging-ML, USB-ML
Many drivers are using below code to know the Sound direction.
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
This patch-set add snd_pcm_is_playback/capture() macro to handle it.
Thank you for your help !!
NAK from my side (overdesign, no improved readability). The defines (SNDRV_PCM_STREAM_*) are enough to check the stream type value correctly.
Jaroslav
On Mon, Aug 05, 2024 at 04:04:39PM +0200, Jaroslav Kysela wrote:
On 05. 08. 24 2:33, Kuninori Morimoto wrote:
Many drivers are using below code to know the Sound direction.
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
This patch-set add snd_pcm_is_playback/capture() macro to handle it.
NAK from my side (overdesign, no improved readability). The defines (SNDRV_PCM_STREAM_*) are enough to check the stream type value correctly.
I have to say I do remember this being a little bit of a confusing idiom when I first stated looking at ALSA stuff, especially for capture only cases.
Hi Takashi, Mark, Jaroslav
This is the reply for very old patch (almost 2 month ago).
Many drivers are using below code to know the Sound direction.
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
This patch-set add snd_pcm_is_playback/capture() macro to handle it.
NAK from my side (overdesign, no improved readability). The defines (SNDRV_PCM_STREAM_*) are enough to check the stream type value correctly.
I have to say I do remember this being a little bit of a confusing idiom when I first stated looking at ALSA stuff, especially for capture only cases.
This patch-set got both Ack and Nack. I wonder can I re-post this after merge-window again ? I'm asking because this is very huge patch-set.
Thank you for your help !!
Best regards --- Kuninori Morimoto
On Fri, 13 Sep 2024 03:35:28 +0200, Kuninori Morimoto wrote:
Hi Takashi, Mark, Jaroslav
This is the reply for very old patch (almost 2 month ago).
Many drivers are using below code to know the Sound direction.
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
This patch-set add snd_pcm_is_playback/capture() macro to handle it.
NAK from my side (overdesign, no improved readability). The defines (SNDRV_PCM_STREAM_*) are enough to check the stream type value correctly.
I have to say I do remember this being a little bit of a confusing idiom when I first stated looking at ALSA stuff, especially for capture only cases.
This patch-set got both Ack and Nack. I wonder can I re-post this after merge-window again ? I'm asking because this is very huge patch-set.
If we get a NACK for this kind of cleanups, it's rather negative. Unless its' a mandatory preliminary change for other upcoming stuff, I don't think it's worth to work on this further.
thanks,
Takashi
Hi Takashi
This patch-set got both Ack and Nack. I wonder can I re-post this after merge-window again ? I'm asking because this is very huge patch-set.
If we get a NACK for this kind of cleanups, it's rather negative. Unless its' a mandatory preliminary change for other upcoming stuff, I don't think it's worth to work on this further.
Hmm...
Thank you for your help !!
Best regards --- Kuninori Morimoto
participants (4)
-
Jaroslav Kysela
-
Kuninori Morimoto
-
Mark Brown
-
Takashi Iwai