Merge "ASoc: msm: Support SEC_MI2S_TX Port ASM Loopback" (831b748d) · Commits · e / devices / android_kernel_xiaomi_markw

sound/soc/msm/apq8096-auto.c

+52 −11

Original line number	Diff line number	Diff line
		@@ -58,6 +58,7 @@ static int msm_quat_mi2s_rx_ch = 2;
		static int msm_sec_mi2s_tx_bit_format = SNDRV_PCM_FORMAT_S16_LE;
		static int msm_tert_mi2s_tx_bit_format = SNDRV_PCM_FORMAT_S16_LE;
		static int msm_quat_mi2s_rx_bit_format = SNDRV_PCM_FORMAT_S16_LE;
		static int msm_sec_mi2s_rate = SAMPLING_RATE_48KHZ;

		/* TDM default channels */
		static int msm_sec_tdm_tx_0_ch = 2; /* STEREO MIC */
		@@ -386,6 +387,8 @@ static char const *ec_ref_bit_format_text[] = {"0", "S16_LE", "S24_LE"};
		static const char *const ec_ref_rate_text[] = {"0", "8000", "16000",
		"32000", "44100", "48000", "96000", "192000", "384000"};

		static const char *const mi2s_rate_text[] = {"32000", "44100", "48000"};

		static struct afe_clk_set sec_mi2s_tx_clk = {
		AFE_API_VERSION_I2S_CONFIG,
		Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT,
		@@ -695,6 +698,37 @@ static int msm_sec_mi2s_tx_bit_format_put(struct snd_kcontrol *kcontrol,
		return 0;
		}

		static int msm_sec_mi2s_rate_get(struct snd_kcontrol *kcontrol,
		struct snd_ctl_elem_value *ucontrol)
		{
		ucontrol->value.integer.value[0] = msm_sec_mi2s_rate;
		pr_debug("%s: msm_sec_mi2s_rate = %d\n", __func__, msm_sec_mi2s_rate);
		return 0;
		}

		static int msm_sec_mi2s_rate_put(struct snd_kcontrol *kcontrol,
		struct snd_ctl_elem_value *ucontrol)
		{
		switch (ucontrol->value.integer.value[0]) {
		case 0:
		msm_sec_mi2s_rate = SAMPLING_RATE_32KHZ;
		break;
		case 1:
		msm_sec_mi2s_rate = SAMPLING_RATE_44P1KHZ;
		break;
		case 2:
		msm_sec_mi2s_rate = SAMPLING_RATE_48KHZ;
		break;
		default:
		msm_sec_mi2s_rate = SAMPLING_RATE_48KHZ;
		break;
		}
		pr_debug("%s: msm_sec_mi2s_rate = %d\n",
		__func__, msm_sec_mi2s_rate);
		return 0;
		}


		static int msm_sec_tdm_tx_0_ch_get(struct snd_kcontrol *kcontrol,
		struct snd_ctl_elem_value *ucontrol)
		{
		@@ -1983,11 +2017,14 @@ static int msm_mi2s_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
		struct snd_interval *channels = hw_param_interval(params,
		SNDRV_PCM_HW_PARAM_CHANNELS);

		rate->min = rate->max = SAMPLING_RATE_48KHZ;

		switch (cpu_dai->id) {
		case 0: /MSM_PRIM_MI2S/
		break;
		case 1: /MSM_SEC_MI2S/
		pr_debug("%s: channel:%d\n", __func__, msm_sec_mi2s_tx_ch);
		rate->min = rate->max = msm_sec_mi2s_rate;
		channels->min = channels->max = msm_sec_mi2s_tx_ch;
		param_set_mask(params, SNDRV_PCM_HW_PARAM_FORMAT,
		msm_sec_mi2s_tx_bit_format);
		@@ -2005,7 +2042,6 @@ static int msm_mi2s_tx_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
		__func__, cpu_dai->id);
		return -EINVAL;
		}
		rate->min = rate->max = SAMPLING_RATE_48KHZ;

		pr_debug("%s: dai id = 0x%x channels = %d rate = %d format = 0x%x\n",
		__func__, cpu_dai->id, channels->max, rate->max,
		@@ -2631,6 +2667,7 @@ static const struct soc_enum msm_snd_enum[] = {
		SOC_ENUM_SINGLE_EXT(9, ec_ref_ch_text),
		SOC_ENUM_SINGLE_EXT(3, ec_ref_bit_format_text),
		SOC_ENUM_SINGLE_EXT(9, ec_ref_rate_text),
		SOC_ENUM_SINGLE_EXT(3, mi2s_rate_text),
		};

		static const struct snd_kcontrol_new msm_snd_controls[] = {
		@@ -2758,6 +2795,8 @@ static const struct snd_kcontrol_new msm_snd_controls[] = {
		SOC_ENUM_EXT("SEC_MI2S_TX Bit Format", msm_snd_enum[7],
		msm_sec_mi2s_tx_bit_format_get,
		msm_sec_mi2s_tx_bit_format_put),
		SOC_ENUM_EXT("SEC_MI2S_TX SampleRate", msm_snd_enum[11],
		msm_sec_mi2s_rate_get, msm_sec_mi2s_rate_put),
		SOC_ENUM_EXT("EC Reference Channels", msm_snd_enum[8],
		msm_ec_ref_ch_get, msm_ec_ref_ch_put),
		SOC_ENUM_EXT("EC Reference Bit Format", msm_snd_enum[9],
		@@ -3059,20 +3098,22 @@ static struct snd_soc_dai_link apq8096_common_dai_links[] = {
		.ops = &apq8096_ll_ops,
		},
		{
		.name = "Listen 1 Audio Service",
		.stream_name = "Listen 1 Audio Service",
		.cpu_dai_name = "LSM1",
		.platform_name = "msm-lsm-client",
		.name = "MSM8996 Media20",
		.stream_name = "MultiMedia20",
		.cpu_dai_name = "MultiMedia20",
		.platform_name = "msm-pcm-loopback",
		.dynamic = 1,
		.dpcm_playback = 1,
		.dpcm_capture = 1,
		.trigger = {SND_SOC_DPCM_TRIGGER_POST,
		SND_SOC_DPCM_TRIGGER_POST},
		.no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
		.ignore_suspend = 1,
		.ignore_pmdown_time = 1,
		.codec_dai_name = "snd-soc-dummy-dai",
		.codec_name = "snd-soc-dummy",
		.be_id = MSM_FRONTEND_DAI_LSM1,
		.ignore_suspend = 1,
		.no_host_mode = SND_SOC_DAI_LINK_NO_HOST,
		/* this dainlink has playback support */
		.ignore_pmdown_time = 1,
		.be_id = MSM_FRONTEND_DAI_MULTIMEDIA20,
		},
		/* Multiple Tunnel instances */
		{