[alsa-devel] [PATCH 1/2] ASoC: sirf: Add audio usp interface driver

[Mark Brown]

On Tue, Jul 01, 2014 at 10:18:53AM +0800, Rongjun Ying wrote:

A few *really* trivial things below, otherwise this looks good:

> +	if (usp->daifmt_format == SND_SOC_DAIFMT_I2S)
> +		regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL,
> +			USP_I2S_SYNC_CHG, USP_I2S_SYNC_CHG);
> +	else if (usp->daifmt_format == SND_SOC_DAIFMT_DSP_A) {
> +		regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL,
> +			USP_I2S_SYNC_CHG, 0);
> +		frame_len = data_len * params_channels(params);
> +		data_len = frame_len;
> +	}

Make this a switch statement for legibility, and if you do need to use {
} make sure they're used consistently for all branches of an if ().

> +	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;

I'm not sure the extra variable adds much here.

> +static struct snd_soc_dai_driver sirf_usp_pcm_dai = {
> +	.probe = sirf_usp_pcm_dai_probe,
> +	.name		= "sirf-usp-pcm",
> +	.id			= 0,

Weird indentation here.
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