Merge "ASoC: msm: qdsp6v2: Send processing parameters based on topology" (bd395d1e) · Commits · e / devices / android_kernel_sony_msm8994

include/sound/apr_audio-v2.h

+4 −0

Original line number	Diff line number	Diff line
		@@ -6922,10 +6922,14 @@ struct srs_trumedia_params {
		} __packed;
		/* SRS TruMedia end */

		#define AUDPROC_PARAM_ID_ENABLE 0x00010904
		#define ASM_STREAM_POSTPROC_TOPO_ID_SA_PLUS 0x1000FFFF
		/* DTS Eagle */
		#define AUDPROC_MODULE_ID_DTS_HPX_PREMIX 0x0001077C
		#define AUDPROC_MODULE_ID_DTS_HPX_POSTMIX 0x0001077B
		#define ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX 0x00010DED
		#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS 0x11000000
		#define ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER 0x11000001
		struct asm_dts_eagle_param {
		struct apr_hdr hdr;
		struct asm_stream_cmd_set_pp_params_v2 param;

include/sound/msm-audio-effects-q6-v2.h

+3 −0

Original line number	Diff line number	Diff line
		@@ -16,6 +16,9 @@

		#include <sound/audio_effects.h>

		bool msm_audio_effects_is_effmodule_supp_in_top(int effect_module,
		int topology);

		int msm_audio_effects_reverb_handler(struct audio_client *ac,
		struct reverb_params *reverb,
		long *values);

sound/soc/msm/qdsp6v2/msm-audio-effects-q6-v2.c

+38 −0

Original line number	Diff line number	Diff line
		@@ -16,6 +16,44 @@
		#include <sound/compress_params.h>
		#include <sound/msm-audio-effects-q6-v2.h>

		bool msm_audio_effects_is_effmodule_supp_in_top(int effect_module,
		int topology)
		{
		switch (effect_module) {
		case VIRTUALIZER_MODULE:
		case REVERB_MODULE:
		case BASS_BOOST_MODULE:
		case EQ_MODULE:
		switch (topology) {
		case ASM_STREAM_POSTPROC_TOPO_ID_SA_PLUS:
		case ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS:
		case ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER:
		return true;
		default:
		return false;
		}
		case DTS_EAGLE_MODULE:
		switch (topology) {
		case ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX:
		case ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS:
		case ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER:
		return true;
		default:
		return false;
		}
		case SOFT_VOLUME2_MODULE:
		switch (topology) {
		case ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS:
		case ASM_STREAM_POSTPROC_TOPO_ID_HPX_MASTER:
		return true;
		default:
		return false;
		}
		default:
		return false;
		}
		}

		int msm_audio_effects_virtualizer_handler(struct audio_client *ac,
		struct virtualizer_params *virtualizer,
		long *values)

sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c

+79 −25

Original line number	Diff line number	Diff line
		@@ -28,6 +28,7 @@
		#include <sound/control.h>
		#include <sound/q6asm-v2.h>
		#include <sound/pcm_params.h>
		#include <sound/audio_effects.h>
		#include <asm/dma.h>
		#include <linux/dma-mapping.h>
		#include <linux/msm_audio_ion.h>
		@@ -156,6 +157,7 @@ struct msm_compr_audio_effects {
		struct virtualizer_params virtualizer;
		struct reverb_params reverb;
		struct eq_params equalizer;
		struct soft_volume_params volume;
		};

		struct msm_compr_dec_params {
		@@ -666,6 +668,56 @@ static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream,
		return ret;
		}

		static int msm_compr_init_pp_params(struct snd_compr_stream *cstream,
		struct audio_client *ac)
		{
		int ret = 0;
		struct asm_softvolume_params softvol = {
		.period = SOFT_VOLUME_PERIOD,
		.step = SOFT_VOLUME_STEP,
		.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
		};

		switch (ac->topology) {
		case ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS: /* HPX + SA+ topology */

		ret = q6asm_set_softvolume_v2(ac, &softvol,
		SOFT_VOLUME_INSTANCE_1);
		if (ret < 0)
		pr_err("%s: Send SoftVolume Param failed ret=%d\n",
		__func__, ret);

		ret = q6asm_set_softvolume_v2(ac, &softvol,
		SOFT_VOLUME_INSTANCE_2);
		if (ret < 0)
		pr_err("%s: Send SoftVolume2 Param failed ret=%d\n",
		__func__, ret);

		ret = msm_compr_set_volume(cstream, 0, 0);
		if (ret < 0)
		pr_err("%s : Set Volume failed : %d", __func__, ret);

		case ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX:

		msm_dts_eagle_init_pre(ac);

		break;
		default:
		ret = q6asm_set_softvolume_v2(ac, &softvol,
		SOFT_VOLUME_INSTANCE_1);
		if (ret < 0)
		pr_err("%s: Send SoftVolume Param failed ret=%d\n",
		__func__, ret);

		ret = msm_compr_set_volume(cstream, 0, 0);
		if (ret < 0)
		pr_err("%s : Set Volume failed : %d", __func__, ret);

		break;
		}
		return ret;
		}

		static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
		{
		struct snd_compr_runtime *runtime = cstream->runtime;
		@@ -675,17 +727,12 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
		int dir = IN, ret = 0;
		struct audio_client *ac = prtd->audio_client;
		uint32_t stream_index;
		struct asm_softvolume_params softvol = {
		.period = SOFT_VOLUME_PERIOD,
		.step = SOFT_VOLUME_STEP,
		.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
		};

		pr_debug("%s: stream_id %d\n", __func__, ac->stream_id);
		if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE)
		bits_per_sample = 24;
		if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S32_LE)
		bits_per_sample = 32;

		pr_debug("%s: stream_id %d\n", __func__, ac->stream_id);
		ret = q6asm_stream_open_write_v2(ac,
		prtd->codec, bits_per_sample,
		ac->stream_id,
		@@ -712,20 +759,7 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
		return ret;
		}

		/*
		* Setting the master volume gain to 0 while
		* configuring ASM session. This is to address
		* DSP pop noise issue where. This change is
		* there from begining may be DSP limitation
		*/
		ret = msm_compr_set_volume(cstream, 0, 0);
		if (ret < 0)
		pr_err("%s : Set Volume failed : %d", __func__, ret);

		ret = q6asm_set_softvolume(ac, &softvol);
		if (ret < 0)
		pr_err("%s: Send SoftVolume Param failed ret=%d\n",
		__func__, ret);
		msm_compr_init_pp_params(cstream, ac);

		ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO \| ASYNC_IO_MODE));
		if (ret < 0) {
		@@ -1902,24 +1936,33 @@ static int msm_compr_audio_effects_config_put(struct snd_kcontrol *kcontrol,
		switch (effects_module) {
		case VIRTUALIZER_MODULE:
		pr_debug("%s: VIRTUALIZER_MODULE\n", __func__);
		msm_audio_effects_virtualizer_handler(prtd->audio_client,
		if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
		prtd->audio_client->topology))
		msm_audio_effects_virtualizer_handler(
		prtd->audio_client,
		&(audio_effects->virtualizer),
		values);
		break;
		case REVERB_MODULE:
		pr_debug("%s: REVERB_MODULE\n", __func__);
		if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
		prtd->audio_client->topology))
		msm_audio_effects_reverb_handler(prtd->audio_client,
		&(audio_effects->reverb),
		values);
		break;
		case BASS_BOOST_MODULE:
		pr_debug("%s: BASS_BOOST_MODULE\n", __func__);
		if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
		prtd->audio_client->topology))
		msm_audio_effects_bass_boost_handler(prtd->audio_client,
		&(audio_effects->bass_boost),
		values);
		break;
		case EQ_MODULE:
		pr_debug("%s: EQ_MODULE\n", __func__);
		if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
		prtd->audio_client->topology))
		msm_audio_effects_popless_eq_handler(prtd->audio_client,
		&(audio_effects->equalizer),
		values);
		@@ -1928,6 +1971,17 @@ static int msm_compr_audio_effects_config_put(struct snd_kcontrol *kcontrol,
		pr_debug("%s: DTS_EAGLE_MODULE\n", __func__);
		msm_dts_eagle_handler_pre(prtd->audio_client, values);
		break;
		case SOFT_VOLUME_MODULE:
		pr_debug("%s: SOFT_VOLUME_MODULE\n", __func__);
		break;
		case SOFT_VOLUME2_MODULE:
		pr_debug("%s: SOFT_VOLUME2_MODULE\n", __func__);
		if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
		prtd->audio_client->topology))
		msm_audio_effects_volume_handler_v2(prtd->audio_client,
		&(audio_effects->volume),
		values, SOFT_VOLUME_INSTANCE_2);
		break;
		default:
		pr_err("%s Invalid effects config module\n", __func__);
		return -EINVAL;