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authorVinod Koul <vinod.koul@linux.intel.com>2012-08-16 17:10:42 +0530
committerMark Brown <broonie@opensource.wolfsonmicro.com>2012-08-20 20:50:39 +0100
commitc514a9119a982a6c7a9fd29ee62c0ba8a8e4c7d1 (patch)
treeb5a94c730f16b0809b8425be19b67fd276da6593 /sound/soc/mid-x86
parent1245b7005de02d5bfa0c321df925f5b6c83c99e1 (diff)
ASoC: mid-x86 - add support for compressed streams
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com> Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com> Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound/soc/mid-x86')
-rw-r--r--sound/soc/mid-x86/mfld_machine.c9
-rw-r--r--sound/soc/mid-x86/sst_dsp.h134
-rw-r--r--sound/soc/mid-x86/sst_platform.c204
-rw-r--r--sound/soc/mid-x86/sst_platform.h26
4 files changed, 371 insertions, 2 deletions
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
index 2937e54da49..2cc7782714b 100644
--- a/sound/soc/mid-x86/mfld_machine.c
+++ b/sound/soc/mid-x86/mfld_machine.c
@@ -318,6 +318,15 @@ static struct snd_soc_dai_link mfld_msic_dailink[] = {
.platform_name = "sst-platform",
.init = NULL,
},
+ {
+ .name = "Medfield Compress",
+ .stream_name = "Speaker",
+ .cpu_dai_name = "Compress-cpu-dai",
+ .codec_dai_name = "SN95031 Speaker",
+ .codec_name = "sn95031",
+ .platform_name = "sst-platform",
+ .init = NULL,
+ },
};
/* SoC card */
diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/mid-x86/sst_dsp.h
new file mode 100644
index 00000000000..0fce1de284f
--- /dev/null
+++ b/sound/soc/mid-x86/sst_dsp.h
@@ -0,0 +1,134 @@
+#ifndef __SST_DSP_H__
+#define __SST_DSP_H__
+/*
+ * sst_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_DSP_H__ */
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index d34563b12c3..a263cbed862 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -1,7 +1,7 @@
/*
* sst_platform.c - Intel MID Platform driver
*
- * Copyright (C) 2010 Intel Corp
+ * Copyright (C) 2010-2012 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@@ -32,6 +32,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/compress_driver.h>
#include "sst_platform.h"
static struct sst_device *sst;
@@ -152,6 +153,16 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
},
},
+{
+ .name = "Compress-cpu-dai",
+ .compress_dai = 1,
+ .playback = {
+ .channels_min = SST_STEREO,
+ .channels_max = SST_STEREO,
+ .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+},
};
/* helper functions */
@@ -463,8 +474,199 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
}
return retval;
}
+
+/* compress stream operations */
+static void sst_compr_fragment_elapsed(void *arg)
+{
+ struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg;
+
+ pr_debug("fragment elapsed by driver\n");
+ if (cstream)
+ snd_compr_fragment_elapsed(cstream);
+}
+
+static int sst_platform_compr_open(struct snd_compr_stream *cstream)
+{
+
+ int ret_val = 0;
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct sst_runtime_stream *stream;
+
+ stream = kzalloc(sizeof(*stream), GFP_KERNEL);
+ if (!stream)
+ return -ENOMEM;
+
+ spin_lock_init(&stream->status_lock);
+
+ /* get the sst ops */
+ if (!sst || !try_module_get(sst->dev->driver->owner)) {
+ pr_err("no device available to run\n");
+ ret_val = -ENODEV;
+ goto out_ops;
+ }
+ stream->compr_ops = sst->compr_ops;
+
+ stream->id = 0;
+ sst_set_stream_status(stream, SST_PLATFORM_INIT);
+ runtime->private_data = stream;
+ return 0;
+out_ops:
+ kfree(stream);
+ return ret_val;
+}
+
+static int sst_platform_compr_free(struct snd_compr_stream *cstream)
+{
+ struct sst_runtime_stream *stream;
+ int ret_val = 0, str_id;
+
+ stream = cstream->runtime->private_data;
+ /*need to check*/
+ str_id = stream->id;
+ if (str_id)
+ ret_val = stream->compr_ops->close(str_id);
+ module_put(sst->dev->driver->owner);
+ kfree(stream);
+ pr_debug("%s: %d\n", __func__, ret_val);
+ return 0;
+}
+
+static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params)
+{
+ struct sst_runtime_stream *stream;
+ int retval;
+ struct snd_sst_params str_params;
+ struct sst_compress_cb cb;
+
+ stream = cstream->runtime->private_data;
+ /* construct fw structure for this*/
+ memset(&str_params, 0, sizeof(str_params));
+
+ str_params.ops = STREAM_OPS_PLAYBACK;
+ str_params.stream_type = SST_STREAM_TYPE_MUSIC;
+ str_params.device_type = SND_SST_DEVICE_COMPRESS;
+
+ switch (params->codec.id) {
+ case SND_AUDIOCODEC_MP3: {
+ str_params.codec = SST_CODEC_TYPE_MP3;
+ str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3;
+ str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in;
+ str_params.sparams.uc.mp3_params.pcm_wd_sz = 16;
+ break;
+ }
+
+ case SND_AUDIOCODEC_AAC: {
+ str_params.codec = SST_CODEC_TYPE_AAC;
+ str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC;
+ str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in;
+ str_params.sparams.uc.aac_params.pcm_wd_sz = 16;
+ if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS)
+ str_params.sparams.uc.aac_params.bs_format =
+ AAC_BIT_STREAM_ADTS;
+ else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW)
+ str_params.sparams.uc.aac_params.bs_format =
+ AAC_BIT_STREAM_RAW;
+ else {
+ pr_err("Undefined format%d\n", params->codec.format);
+ return -EINVAL;
+ }
+ str_params.sparams.uc.aac_params.externalsr =
+ params->codec.sample_rate;
+ break;
+ }
+
+ default:
+ pr_err("codec not supported, id =%d\n", params->codec.id);
+ return -EINVAL;
+ }
+
+ str_params.aparams.ring_buf_info[0].addr =
+ virt_to_phys(cstream->runtime->buffer);
+ str_params.aparams.ring_buf_info[0].size =
+ cstream->runtime->buffer_size;
+ str_params.aparams.sg_count = 1;
+ str_params.aparams.frag_size = cstream->runtime->fragment_size;
+
+ cb.param = cstream;
+ cb.compr_cb = sst_compr_fragment_elapsed;
+
+ retval = stream->compr_ops->open(&str_params, &cb);
+ if (retval < 0) {
+ pr_err("stream allocation failed %d\n", retval);
+ return retval;
+ }
+
+ stream->id = retval;
+ return 0;
+}
+
+static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
+{
+ struct sst_runtime_stream *stream =
+ cstream->runtime->private_data;
+
+ return stream->compr_ops->control(cmd, stream->id);
+}
+
+static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *tstamp)
+{
+ struct sst_runtime_stream *stream;
+
+ stream = cstream->runtime->private_data;
+ stream->compr_ops->tstamp(stream->id, tstamp);
+ tstamp->byte_offset = tstamp->copied_total %
+ (u32)cstream->runtime->buffer_size;
+ pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset);
+ return 0;
+}
+
+static int sst_platform_compr_ack(struct snd_compr_stream *cstream,
+ size_t bytes)
+{
+ struct sst_runtime_stream *stream;
+
+ stream = cstream->runtime->private_data;
+ stream->compr_ops->ack(stream->id, (unsigned long)bytes);
+ stream->bytes_written += bytes;
+
+ return 0;
+}
+
+static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_caps *caps)
+{
+ struct sst_runtime_stream *stream =
+ cstream->runtime->private_data;
+
+ return stream->compr_ops->get_caps(caps);
+}
+
+static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_codec_caps *codec)
+{
+ struct sst_runtime_stream *stream =
+ cstream->runtime->private_data;
+
+ return stream->compr_ops->get_codec_caps(codec);
+}
+
+static struct snd_compr_ops sst_platform_compr_ops = {
+
+ .open = sst_platform_compr_open,
+ .free = sst_platform_compr_free,
+ .set_params = sst_platform_compr_set_params,
+ .trigger = sst_platform_compr_trigger,
+ .pointer = sst_platform_compr_pointer,
+ .ack = sst_platform_compr_ack,
+ .get_caps = sst_platform_compr_get_caps,
+ .get_codec_caps = sst_platform_compr_get_codec_caps,
+};
+
static struct snd_soc_platform_driver sst_soc_platform_drv = {
.ops = &sst_platform_ops,
+ .compr_ops = &sst_platform_compr_ops,
.pcm_new = sst_pcm_new,
.pcm_free = sst_pcm_free,
};
diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h
index f04f4f72daa..d61c5d514ff 100644
--- a/sound/soc/mid-x86/sst_platform.h
+++ b/sound/soc/mid-x86/sst_platform.h
@@ -27,6 +27,8 @@
#ifndef __SST_PLATFORMDRV_H__
#define __SST_PLATFORMDRV_H__
+#include "sst_dsp.h"
+
#define SST_MONO 1
#define SST_STEREO 2
#define SST_MAX_CAP 5
@@ -42,7 +44,6 @@
#define SST_MIN_PERIODS 2
#define SST_MAX_PERIODS (1024*2)
#define SST_FIFO_SIZE 0
-#define SST_CODEC_TYPE_PCM 1
struct pcm_stream_info {
int str_id;
@@ -83,6 +84,7 @@ enum sst_audio_device_type {
SND_SST_DEVICE_VIBRA,
SND_SST_DEVICE_HAPTIC,
SND_SST_DEVICE_CAPTURE,
+ SND_SST_DEVICE_COMPRESS,
};
/* PCM Parameters */
@@ -107,6 +109,24 @@ struct sst_stream_params {
struct sst_pcm_params sparams;
};
+struct sst_compress_cb {
+ void *param;
+ void (*compr_cb)(void *param);
+};
+
+struct compress_sst_ops {
+ const char *name;
+ int (*open) (struct snd_sst_params *str_params,
+ struct sst_compress_cb *cb);
+ int (*control) (unsigned int cmd, unsigned int str_id);
+ int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp);
+ int (*ack) (unsigned int str_id, unsigned long bytes);
+ int (*close) (unsigned int str_id);
+ int (*get_caps) (struct snd_compr_caps *caps);
+ int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
+
+};
+
struct sst_ops {
int (*open) (struct sst_stream_params *str_param);
int (*device_control) (int cmd, void *arg);
@@ -115,8 +135,11 @@ struct sst_ops {
struct sst_runtime_stream {
int stream_status;
+ unsigned int id;
+ size_t bytes_written;
struct pcm_stream_info stream_info;
struct sst_ops *ops;
+ struct compress_sst_ops *compr_ops;
spinlock_t status_lock;
};
@@ -124,6 +147,7 @@ struct sst_device {
char *name;
struct device *dev;
struct sst_ops *ops;
+ struct compress_sst_ops *compr_ops;
};
int sst_register_dsp(struct sst_device *sst);