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Let's use it. Signed-off-by: Kuninori Morimoto --- 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; }