Initial Contribution
msm-2.6.38: tag AU_LINUX_ANDROID_GINGERBREAD.02.03.04.00.142
Signed-off-by: Bryan Huntsman <bryanh@codeaurora.org>
diff --git a/sound/soc/msm/msm-pcm.h b/sound/soc/msm/msm-pcm.h
new file mode 100644
index 0000000..e7ddd30
--- /dev/null
+++ b/sound/soc/msm/msm-pcm.h
@@ -0,0 +1,200 @@
+/* sound/soc/msm/msm-pcm.h
+ *
+ * Copyright (C) 2008 Google, Inc.
+ * Copyright (C) 2008 HTC Corporation
+ * Copyright (c) 2008-2009, Code Aurora Forum. All rights reserved.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * 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, you can find it at http://www.fsf.org.
+ */
+
+#ifndef _MSM_PCM_H
+#define _MSM_PCM_H
+
+
+#include <mach/qdsp5/qdsp5audppcmdi.h>
+#include <mach/qdsp5/qdsp5audppmsg.h>
+#include <mach/qdsp5/qdsp5audreccmdi.h>
+#include <mach/qdsp5/qdsp5audrecmsg.h>
+#include <mach/qdsp5/qdsp5audpreproccmdi.h>
+#include <mach/qdsp5/qdsp5audpreprocmsg.h>
+
+#include <../arch/arm/mach-msm/qdsp5/adsp.h>
+#include <../arch/arm/mach-msm/qdsp5/audmgr.h>
+
+
+#define FRAME_NUM (8)
+#define FRAME_SIZE (2052 * 2)
+#define MONO_DATA_SIZE (2048)
+#define STEREO_DATA_SIZE (MONO_DATA_SIZE * 2)
+#define CAPTURE_DMASZ (FRAME_SIZE * FRAME_NUM)
+
+#define BUFSZ (960 * 5)
+#define PLAYBACK_DMASZ (BUFSZ * 2)
+
+#define MSM_PLAYBACK_DEFAULT_VOLUME 0 /* 0dB */
+#define MSM_PLAYBACK_DEFAULT_PAN 0
+
+#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+#define USE_CHANNELS_MIN 1
+#define USE_CHANNELS_MAX 2
+/* Support unconventional sample rates 12000, 24000 as well */
+#define USE_RATE \
+ (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT)
+#define USE_RATE_MIN 8000
+#define USE_RATE_MAX 48000
+#define MAX_BUFFER_PLAYBACK_SIZE \
+ (4800*4)
+/* 2048 frames (Mono), 1024 frames (Stereo) */
+#define CAPTURE_SIZE 4096
+#define MAX_BUFFER_CAPTURE_SIZE (4096*4)
+#define MAX_PERIOD_SIZE BUFSZ
+#define USE_PERIODS_MAX 1024
+#define USE_PERIODS_MIN 1
+
+
+#define MAX_DB (16)
+#define MIN_DB (-50)
+#define PCMPLAYBACK_DECODERID 5
+
+/* 0xFFFFFFFF Indicates not to be used for audio data copy */
+#define BUF_INVALID_LEN 0xFFFFFFFF
+
+extern int copy_count;
+extern int intcnt;
+
+struct msm_volume {
+ bool update;
+ int volume; /* Volume parameter, in dB Scale */
+ int pan;
+};
+
+struct buffer {
+ void *data;
+ unsigned size;
+ unsigned used;
+ unsigned addr;
+};
+
+struct buffer_rec {
+ void *data;
+ unsigned int size;
+ unsigned int read;
+ unsigned int addr;
+};
+
+struct audio_locks {
+ struct mutex lock;
+ struct mutex write_lock;
+ struct mutex read_lock;
+ spinlock_t read_dsp_lock;
+ spinlock_t write_dsp_lock;
+ spinlock_t mixer_lock;
+ wait_queue_head_t read_wait;
+ wait_queue_head_t write_wait;
+ wait_queue_head_t eos_wait;
+};
+
+extern struct audio_locks the_locks;
+
+struct msm_audio_event_callbacks {
+ /* event is called from interrupt context when a message
+ * arrives from the DSP.
+ */
+ void (*playback)(void *);
+ void (*capture)(void *);
+};
+
+
+struct msm_audio {
+ struct buffer out[2];
+ struct buffer_rec in[8];
+
+ uint8_t out_head;
+ uint8_t out_tail;
+ uint8_t out_needed; /* number of buffers the dsp is waiting for */
+ atomic_t out_bytes;
+
+ /* configuration to use on next enable */
+ uint32_t out_sample_rate;
+ uint32_t out_channel_mode;
+ uint32_t out_weight;
+ uint32_t out_buffer_size;
+
+ struct audmgr audmgr;
+ struct snd_pcm_substream *playback_substream;
+ struct snd_pcm_substream *capture_substream;
+
+ /* data allocated for various buffers */
+ char *data;
+ dma_addr_t phys;
+
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int pcm_buf_pos; /* position in buffer */
+
+ struct msm_adsp_module *audpre;
+ struct msm_adsp_module *audrec;
+
+ /* configuration to use on next enable */
+ uint32_t samp_rate;
+ uint32_t channel_mode;
+ uint32_t buffer_size; /* 2048 for mono, 4096 for stereo */
+ uint32_t type; /* 0 for PCM ,1 for AAC */
+ uint32_t dsp_cnt;
+ uint32_t in_head; /* next buffer dsp will write */
+ uint32_t in_tail; /* next buffer read() will read */
+ uint32_t in_count; /* number of buffers available to read() */
+
+ unsigned short samp_rate_index;
+
+ /* audpre settings */
+ audpreproc_cmd_cfg_agc_params tx_agc_cfg;
+ audpreproc_cmd_cfg_ns_params ns_cfg;
+ /* For different sample rate, the coeff might be different. *
+ * All the coeff should be passed from user space */
+ audpreproc_cmd_cfg_iir_tuning_filter_params iir_cfg;
+
+ struct msm_audio_event_callbacks *ops;
+
+ int dir;
+ int opened;
+ int enabled;
+ int running;
+ int stopped; /* set when stopped, cleared on flush */
+ int eos_ack;
+};
+
+
+
+/* platform data */
+extern int audio_dsp_out_enable(struct msm_audio *prtd, int yes);
+extern struct snd_soc_platform_driver msm_soc_platform;
+
+int audrec_encoder_config(struct msm_audio *prtd);
+extern void alsa_get_dsp_frames(struct msm_audio *prtd);
+extern int alsa_rec_dsp_enable(struct msm_audio *prtd, int enable);
+extern int alsa_audrec_disable(struct msm_audio *prtd);
+extern int alsa_audio_configure(struct msm_audio *prtd);
+extern int alsa_audio_disable(struct msm_audio *prtd);
+extern int alsa_adsp_configure(struct msm_audio *prtd);
+extern int alsa_buffer_read(struct msm_audio *prtd, void __user *buf,
+ size_t count, loff_t *pos);
+ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf,
+ size_t count, loff_t *pos);
+int msm_audio_volume_update(unsigned id,
+ int volume, int pan);
+extern struct audio_locks the_locks;
+extern struct msm_volume msm_vol_ctl;
+
+#endif /*_MSM_PCM_H*/