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authorMark Brown <broonie@linaro.org>2014-03-23 14:00:41 +0000
committerMark Brown <broonie@linaro.org>2014-03-23 14:00:41 +0000
commitd66fa86956149a211db3d7ae9e9f2536b65ccde4 (patch)
tree013c71ec06c6d9710a183854ce1a0fe33530a0db /sound/soc/intel/sst-mfld-dsp.h
parentebec909345bbb1e2d06cd0d94f65664edcc0f208 (diff)
parentdeeed33850c8a376addabbf971df433b2a1ba74c (diff)
Merge tag 'asoc-v3.15' into asoc-next
ASoC: Updates for v3.15 Quite a busy release for ASoC this time, more on janitorial work than exciting new features but welcome nontheless: - Lots of cleanups from Takashi for enumerations; the original API for these was error prone so he's refactored lots of code to use more modern APIs which avoid issues. - Elimination of the ASoC level wrappers for I2C and SPI moving us closer to converting to regmap completely and avoiding some randconfig hassle. - Provide both manually and transparently locked DAPM APIs rather than a mix of the two fixing some concurrency issues. - Start converting CODEC drivers to use separate bus interface drivers rather than having them all in one file helping avoid dependency issues. - DPCM support for Intel Haswell and Bay Trail platforms. - Lots of work on improvements for simple-card, DaVinci and the Renesas rcar drivers. - New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the CSR SiRF SoC. # gpg: Signature made Wed 12 Mar 2014 23:05:45 GMT using RSA key ID 7EA229BD # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Diffstat (limited to 'sound/soc/intel/sst-mfld-dsp.h')
-rw-r--r--sound/soc/intel/sst-mfld-dsp.h134
1 files changed, 134 insertions, 0 deletions
diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h
new file mode 100644
index 00000000000..3b63edc04b7
--- /dev/null
+++ b/sound/soc/intel/sst-mfld-dsp.h
@@ -0,0 +1,134 @@
+#ifndef __SST_MFLD_DSP_H__
+#define __SST_MFLD_DSP_H__
+/*
+ * sst_mfld_dsp.h - Intel SST Driver for audio engine
+ *
+ * Copyright (C) 2008-12 Intel Corporation
+ * Authors: Vinod Koul <vinod.koul@linux.intel.com>
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+enum sst_codec_types {
+ /* AUDIO/MUSIC CODEC Type Definitions */
+ SST_CODEC_TYPE_UNKNOWN = 0,
+ SST_CODEC_TYPE_PCM, /* Pass through Audio codec */
+ SST_CODEC_TYPE_MP3,
+ SST_CODEC_TYPE_MP24,
+ SST_CODEC_TYPE_AAC,
+ SST_CODEC_TYPE_AACP,
+ SST_CODEC_TYPE_eAACP,
+};
+
+enum stream_type {
+ SST_STREAM_TYPE_NONE = 0,
+ SST_STREAM_TYPE_MUSIC = 1,
+};
+
+struct snd_pcm_params {
+ u16 codec; /* codec type */
+ u8 num_chan; /* 1=Mono, 2=Stereo */
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u32 reserved; /* Bitrate in bits per second */
+ u32 sfreq; /* Sampling rate in Hz */
+ u8 use_offload_path;
+ u8 reserved2;
+ u16 reserved3;
+ u8 channel_map[8];
+} __packed;
+
+/* MP3 Music Parameters Message */
+struct snd_mp3_params {
+ u16 codec;
+ u8 num_chan; /* 1=Mono, 2=Stereo */
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u8 crc_check; /* crc_check - disable (0) or enable (1) */
+ u8 reserved1; /* unused*/
+ u16 reserved2; /* Unused */
+} __packed;
+
+#define AAC_BIT_STREAM_ADTS 0
+#define AAC_BIT_STREAM_ADIF 1
+#define AAC_BIT_STREAM_RAW 2
+
+/* AAC Music Parameters Message */
+struct snd_aac_params {
+ u16 codec;
+ u8 num_chan; /* 1=Mono, 2=Stereo*/
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */
+ u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */
+ u16 reser2;
+ u32 externalsr; /*sampling rate of basic AAC raw bit stream*/
+ u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/
+ u8 reser1;
+ u16 reser3;
+} __packed;
+
+/* WMA Music Parameters Message */
+struct snd_wma_params {
+ u16 codec;
+ u8 num_chan; /* 1=Mono, 2=Stereo */
+ u8 pcm_wd_sz; /* 16/24 - bit*/
+ u32 brate; /* Use the hard coded value. */
+ u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */
+ u32 channel_mask; /* Channel Mask */
+ u16 format_tag; /* Format Tag */
+ u16 block_align; /* packet size */
+ u16 wma_encode_opt;/* Encoder option */
+ u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */
+ u8 reserved; /* reserved */
+} __packed;
+
+/* Codec params struture */
+union snd_sst_codec_params {
+ struct snd_pcm_params pcm_params;
+ struct snd_mp3_params mp3_params;
+ struct snd_aac_params aac_params;
+ struct snd_wma_params wma_params;
+} __packed;
+
+/* Address and size info of a frame buffer */
+struct sst_address_info {
+ u32 addr; /* Address at IA */
+ u32 size; /* Size of the buffer */
+};
+
+struct snd_sst_alloc_params_ext {
+ struct sst_address_info ring_buf_info[8];
+ u8 sg_count;
+ u8 reserved;
+ u16 reserved2;
+ u32 frag_size; /*Number of samples after which period elapsed
+ message is sent valid only if path = 0*/
+} __packed;
+
+struct snd_sst_stream_params {
+ union snd_sst_codec_params uc;
+} __packed;
+
+struct snd_sst_params {
+ u32 stream_id;
+ u8 codec;
+ u8 ops;
+ u8 stream_type;
+ u8 device_type;
+ struct snd_sst_stream_params sparams;
+ struct snd_sst_alloc_params_ext aparams;
+};
+
+#endif /* __SST_MFLD_DSP_H__ */