Merge "Revert "ASoC: mpq8092: Add the passthrough support in the compressed driver"" (082c624f) · Commits · e / devices / android_kernel_sony_msm8994

include/sound/apr_audio-v2.h

+0 −15

Original line number	Diff line number	Diff line
		@@ -22,7 +22,6 @@
		#define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323
		#define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324

		#define ADM_CMD_STREAM_DEVICE_MAP_ROUTINGS_V5 0x0001033D
		#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325

		/* Enumeration for an audio Rx matrix ID.*/
		@@ -30,7 +29,6 @@

		#define ADM_MATRIX_ID_AUDIO_TX 1

		#define ADM_COMPRESSED_AUDIO_OUT 2
		/* Enumeration for an audio Tx matrix ID.*/
		#define ADM_MATRIX_ID_AUDIOX 1

		@@ -2402,7 +2400,6 @@ struct afe_port_cmdrsp_get_param_v2 {

		#define NULL_COPP_TOPOLOGY 0x00010312
		#define DEFAULT_COPP_TOPOLOGY 0x00010be3
		#define COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY 0x0001076B
		#define DEFAULT_POPP_TOPOLOGY 0x00010be4
		#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71
		#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72
		@@ -2818,17 +2815,6 @@ struct asm_multi_channel_pcm_enc_cfg_v2 {
		#define ASM_MEDIA_FMT_AAC_AOT_PS 29
		#define ASM_MEDIA_FMT_AAC_AOT_BSAC 22

		#define AC3_DECODER 0x00010BF6
		#define EAC3_DECODER 0x00010C3C
		#define MP3 0x00010BE9
		#define DTS 0x00010D88
		#define DTS_LBR 0x00010DBB
		#define MPEG4_AAC 0x00010BEA
		#define ATRAC 0x00010D89
		#define WMA_V10PRO 0x00010BF3
		#define MAT 0x00010D8A
		#define MP2 0x00010DBE

		struct asm_aac_fmt_blk_v2 {
		struct apr_hdr hdr;
		struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
		@@ -3843,7 +3829,6 @@ struct asm_session_cmd_run_v2 {
		#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
		#define ASM_SESSION_CMD_GET_SESSIONTIME_V3 0x00010D9D
		#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5
		#define ADM_MULTI_CH_COPP_OPEN_PERF_MODE_BIT (1<<13)

		struct asm_session_cmd_rgstr_rx_underflow {
		struct apr_hdr hdr;

include/sound/q6adm-v2.h

+0 −1

Original line number	Diff line number	Diff line
		@@ -16,7 +16,6 @@
		#define ADM_PATH_PLAYBACK 0x1
		#define ADM_PATH_LIVE_REC 0x2
		#define ADM_PATH_NONLIVE_REC 0x3
		#define ADM_PATH_COMPRESSED_RX 0x5
		#include <linux/qdsp6v2/rtac.h>
		#include <sound/q6afe-v2.h>
		#include <sound/q6audio-v2.h>

include/sound/q6asm-v2.h

+0 −7

Original line number	Diff line number	Diff line
		@@ -44,11 +44,6 @@
		#define FORMAT_MULTI_CHANNEL_LINEAR_PCM 0x0012
		#define FORMAT_AC3 0x0013
		#define FORMAT_EAC3 0x0014
		#define FORMAT_DTS 0x0016
		#define FORMAT_ATRAC 0x0017
		#define FORMAT_MAT 0x0018
		#define FORMAT_AAC 0x0019
		#define FORMAT_DTS_LBR 0x001a
		#define FORMAT_MP2 0x0015

		#define ENCDEC_SBCBITRATE 0x0001
		@@ -222,8 +217,6 @@ int q6asm_stream_open_write_v2(struct audio_client *ac, uint32_t format,
		uint16_t bits_per_sample, int32_t stream_id,
		bool is_gapless_mode);

		int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format);

		int q6asm_open_read_write(struct audio_client *ac,
		uint32_t rd_format,
		uint32_t wr_format);

include/uapi/sound/compress_offload.h

+0 −1

Original line number	Diff line number	Diff line
		@@ -50,7 +50,6 @@ struct snd_compressed_buffer {
		struct snd_compr_params {
		struct snd_compressed_buffer buffer;
		struct snd_codec codec;
		__u32 compr_passthr;
		__u8 no_wake_mode;
		};

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

+14 −54

Original line number	Diff line number	Diff line
		@@ -92,7 +92,6 @@ struct msm_compr_audio {
		struct audio_client *audio_client;

		uint32_t codec;
		uint32_t compr_passthr;
		void buffer; / virtual address */
		uint32_t buffer_paddr; /* physical address */
		uint32_t app_pointer;
		@@ -143,8 +142,6 @@ struct msm_compr_audio_effects {
		struct eq_params equalizer;
		};

		u32 compr_codecs[] = {SND_AUDIOCODEC_AC3};

		struct msm_compr_dec_params {
		struct snd_dec_ddp ddp_params;
		};
		@@ -587,28 +584,6 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)

		prtd->gapless_state.stream_opened[ac->stream_id] = 1;
		pr_debug("%s be_id %d\n", __func__, soc_prtd->dai_link->be_id);

		if (prtd->compr_passthr == true) {
		ret = q6asm_open_write_compressed(ac, prtd->codec);
		if (ret < 0) {
		pr_err("%s: Session out open failed\n", __func__);
		return -ENOMEM;
		}
		msm_pcm_routing_reg_phy_compr_stream(
		soc_prtd->dai_link->be_id,
		ac->perf_mode,
		prtd->session_id,
		SNDRV_PCM_STREAM_PLAYBACK,
		prtd->compr_passthr);
		} else {
		ret = q6asm_stream_open_write_v2(ac,
		prtd->codec, bits_per_sample,
		ac->stream_id, true/gapless/);
		if (ret < 0) {
		pr_err("%s: Session out open failed\n", __func__);
		return -ENOMEM;
		}

		msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
		ac->perf_mode,
		prtd->session_id,
		@@ -627,7 +602,6 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
		if (ret < 0)
		pr_err("%s: Send SoftVolume Param failed ret=%d\n",
		__func__, ret);
		}

		ret = q6asm_set_io_mode(ac, (COMPRESSED_IO \| ASYNC_IO_MODE));
		if (ret < 0) {
		@@ -833,15 +807,6 @@ static int msm_compr_free(struct snd_compr_stream *cstream)
		return 0;
		}

		int validate_codec_compr(__u32 codec_id)
		{
		int i;
		for (i = 0; i < ARRAY_SIZE(compr_codecs); i++)
		if (compr_codecs[i] == codec_id)
		return 0;
		return -EINVAL;
		}

		/* compress stream operations */
		static int msm_compr_set_params(struct snd_compr_stream *cstream,
		struct snd_compr_params *params)
		@@ -857,8 +822,6 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		/* ToDo: remove duplicates */
		prtd->num_channels = prtd->codec_param.codec.ch_in;

		prtd->compr_passthr = params->compr_passthr;

		switch (prtd->codec_param.codec.sample_rate) {
		case SNDRV_PCM_RATE_8000:
		prtd->sample_rate = 8000;
		@@ -883,11 +846,8 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
		prtd->sample_rate = 48000;
		break;
		}
		if (prtd->compr_passthr && validate_codec_compr(params->codec.id)) {
		pr_err("%s codec not supported in passthrough, id =%d\n",
		__func__, params->codec.id);
		return -EINVAL;
		}

		pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate);

		switch (params->codec.id) {
		case SND_AUDIOCODEC_PCM: {