initial audio HAL implementation for mako

alsa_sound is imported from codeaurora at:

c1217338f349fe746e0933fcf9b1b288b532808d

[remote "quic"]
        url = git://git-android.quicinc.com/platform/hardware/alsa_sound.git
        review = review-android.quicinc.com
        projectname = platform/hardware/alsa_sound
        fetch = +refs/heads/*:refs/remotes/quic/*

Change-Id: Ic985cc3a1088c3957b6e2ac5537e2c36caaf7212
Signed-off-by: Iliyan Malchev <malchev@google.com>
diff --git a/libalsa-intf/alsa_audio.h b/libalsa-intf/alsa_audio.h
new file mode 100644
index 0000000..83482dd
--- /dev/null
+++ b/libalsa-intf/alsa_audio.h
@@ -0,0 +1,289 @@
+/*
+** Copyright 2010, The Android Open-Source Project
+** Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved.
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+**     http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#ifndef _AUDIO_H_
+#define _AUDIO_H_
+
+#include <sound/asound.h>
+#define PCM_ERROR_MAX 128
+
+struct pcm {
+    int fd;
+    int timer_fd;
+    unsigned rate;
+    unsigned channels;
+    unsigned flags;
+    unsigned format;
+    unsigned running:1;
+    int underruns;
+    unsigned buffer_size;
+    unsigned period_size;
+    unsigned period_cnt;
+    char error[PCM_ERROR_MAX];
+    struct snd_pcm_hw_params *hw_p;
+    struct snd_pcm_sw_params *sw_p;
+    struct snd_pcm_sync_ptr *sync_ptr;
+    struct snd_pcm_channel_info ch[2];
+    void *addr;
+    int card_no;
+    int device_no;
+    int start;
+};
+
+enum decoder_alias {
+    FORMAT_MP3,
+    FORMAT_AC3_PASS_THROUGH = 2,
+};
+
+#define FORMAT(v) SNDRV_PCM_FORMAT_##v
+
+#define PCM_OUT        0x00000000
+#define PCM_IN         0x10000000
+
+#define PCM_STEREO     0x00000000
+#define PCM_MONO       0x01000000
+#define PCM_QUAD       0x02000000
+#define PCM_5POINT1    0x04000000
+
+#define PCM_44100HZ    0x00000000
+#define PCM_48000HZ    0x00100000
+#define PCM_8000HZ     0x00200000
+#define PCM_RATE_MASK  0x00F00000
+
+#define PCM_MMAP       0x00010000
+#define PCM_NMMAP      0x00000000
+
+#define DEBUG_ON       0x00000001
+#define DEBUG_OFF      0x00000000
+
+#define PCM_PERIOD_CNT_MIN 2
+#define PCM_PERIOD_CNT_SHIFT 16
+#define PCM_PERIOD_CNT_MASK (0xF << PCM_PERIOD_CNT_SHIFT)
+#define PCM_PERIOD_SZ_MIN 128
+#define PCM_PERIOD_SZ_SHIFT 12
+#define PCM_PERIOD_SZ_MASK (0xF << PCM_PERIOD_SZ_SHIFT)
+
+#define TIMEOUT_INFINITE  -1
+
+/* Acquire/release a pcm channel.
+ * Returns non-zero on error
+ */
+
+struct mixer_ctl {
+    struct mixer *mixer;
+    struct snd_ctl_elem_info *info;
+    char **ename;
+};
+
+#define __snd_alloca(ptr,type) do { *ptr = (type *) alloca(sizeof(type)); memset(*ptr, 0, sizeof(type)); } while (0)
+#define snd_ctl_elem_id_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_id)
+#define snd_ctl_card_info_alloca(ptr) __snd_alloca(ptr, snd_ctl_card_info)
+#define snd_ctl_event_alloca(ptr) __snd_alloca(ptr, snd_ctl_event)
+#define snd_ctl_elem_list_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_list)
+#define snd_ctl_elem_info_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_info)
+#define snd_ctl_elem_value_alloca(ptr) __snd_alloca(ptr, snd_ctl_elem_value)
+
+
+enum snd_pcm_stream_t {
+        /** Playback stream */
+        SND_PCM_STREAM_PLAYBACK = 0,
+        /** Capture stream */
+        SND_PCM_STREAM_CAPTURE,
+        SND_PCM_STREAM_LAST = SND_PCM_STREAM_CAPTURE
+};
+
+enum _snd_ctl_elem_iface {
+        /** Card level */
+        SND_CTL_ELEM_IFACE_CARD = 0,
+        /** Hardware dependent device */
+        SND_CTL_ELEM_IFACE_HWDEP,
+        /** Mixer */
+        SND_CTL_ELEM_IFACE_MIXER,
+        /** PCM */
+        SND_CTL_ELEM_IFACE_PCM,
+        /** RawMidi */
+        SND_CTL_ELEM_IFACE_RAWMIDI,
+        /** Timer */
+        SND_CTL_ELEM_IFACE_TIMER,
+        /** Sequencer */
+        SND_CTL_ELEM_IFACE_SEQUENCER,
+        SND_CTL_ELEM_IFACE_LAST = SND_CTL_ELEM_IFACE_SEQUENCER
+};
+
+struct mixer {
+    int fd;
+    struct snd_ctl_elem_info *info;
+    struct mixer_ctl *ctl;
+    unsigned count;
+};
+
+int get_format(const char* name);
+const char *get_format_name(int format);
+const char *get_format_desc(int format);
+struct pcm *pcm_open(unsigned flags, char *device);
+int pcm_close(struct pcm *pcm);
+int pcm_ready(struct pcm *pcm);
+int mmap_buffer(struct pcm *pcm);
+u_int8_t *dst_address(struct pcm *pcm);
+int sync_ptr(struct pcm *pcm);
+
+void param_init(struct snd_pcm_hw_params *p);
+void param_set_mask(struct snd_pcm_hw_params *p, int n, unsigned bit);
+void param_set_min(struct snd_pcm_hw_params *p, int n, unsigned val);
+void param_set_int(struct snd_pcm_hw_params *p, int n, unsigned val);
+void param_set_max(struct snd_pcm_hw_params *p, int n, unsigned val);
+int param_set_hw_refine(struct pcm *pcm, struct snd_pcm_hw_params *params);
+int param_set_hw_params(struct pcm *pcm, struct snd_pcm_hw_params *params);
+int param_set_sw_params(struct pcm *pcm, struct snd_pcm_sw_params *sparams);
+void param_dump(struct snd_pcm_hw_params *p);
+int pcm_prepare(struct pcm *pcm);
+long pcm_avail(struct pcm *pcm);
+
+/* Returns a human readable reason for the last error. */
+const char *pcm_error(struct pcm *pcm);
+
+/* Returns the buffer size (int bytes) that should be used for pcm_write.
+ * This will be 1/2 of the actual fifo size.
+ */
+int pcm_buffer_size(struct snd_pcm_hw_params *params);
+int pcm_period_size(struct snd_pcm_hw_params *params);
+
+/* Write data to the fifo.
+ * Will start playback on the first write or on a write that
+ * occurs after a fifo underrun.
+ */
+int pcm_write(struct pcm *pcm, void *data, unsigned count);
+int pcm_read(struct pcm *pcm, void *data, unsigned count);
+
+struct mixer;
+struct mixer_ctl;
+
+struct mixer *mixer_open(const char *device);
+void mixer_close(struct mixer *mixer);
+void mixer_dump(struct mixer *mixer);
+
+struct mixer_ctl *mixer_get_control(struct mixer *mixer,
+                                    const char *name, unsigned index);
+struct mixer_ctl *mixer_get_nth_control(struct mixer *mixer, unsigned n);
+
+int mixer_ctl_set(struct mixer_ctl *ctl, unsigned percent);
+int mixer_ctl_select(struct mixer_ctl *ctl, const char *value);
+void mixer_ctl_get(struct mixer_ctl *ctl, unsigned *value);
+int mixer_ctl_set_value(struct mixer_ctl *ctl, int count, char ** argv);
+
+
+#define MAX_NUM_CODECS 32
+
+/* compressed audio support */
+struct snd_compr_caps {
+        __u32 num_codecs;
+        __u32 min_fragment_size;
+        __u32 max_fragment_size;
+        __u32 min_fragments;
+        __u32 max_fragments;
+        __u32 codecs[MAX_NUM_CODECS];
+        __u32 reserved[11];
+};
+
+struct snd_enc_wma {
+        __u32 super_block_align; /* WMA Type-specific data */
+	__u32 bits_per_sample;
+	__u32 channelmask;
+	__u32 encodeopt;
+};
+
+struct snd_enc_vorbis {
+        int quality;
+        __u32 managed;
+        __u32 max_bit_rate;
+        __u32 min_bit_rate;
+        __u32 downmix;
+};
+
+struct snd_enc_real {
+        __u32 quant_bits;
+        __u32 start_region;
+        __u32 num_regions;
+};
+
+struct snd_enc_flac {
+        __u32 num;
+        __u32 gain;
+};
+
+struct snd_enc_generic {
+        __u32 bw;       /* encoder bandwidth */
+        int reserved[15];
+};
+
+union snd_codec_options {
+        struct snd_enc_wma wma;
+        struct snd_enc_vorbis vorbis;
+        struct snd_enc_real real;
+        struct snd_enc_flac flac;
+        struct snd_enc_generic generic;
+};
+
+struct snd_codec {
+        __u32 id;
+        __u32 ch_in;
+        __u32 ch_out;
+        __u32 sample_rate;
+        __u32 bit_rate;
+        __u32 rate_control;
+        __u32 profile;
+        __u32 level;
+        __u32 ch_mode;
+        __u32 format;
+        __u32 align;
+        union snd_codec_options options;
+        __u32 reserved[3];
+};
+
+struct snd_compressed_buffer {
+        size_t fragment_size;
+        int fragments;
+};
+
+/* */
+struct snd_compr_params {
+        struct snd_compressed_buffer buffer;
+        struct snd_codec codec;
+};
+
+struct snd_compr_tstamp {
+        size_t copied_bytes;
+        size_t copied_total;
+        size_t decoded;
+        size_t rendered;
+        __u32 sampling_rate;
+        uint64_t timestamp;
+};
+
+#define SNDRV_COMPRESS_GET_CAPS         _IOWR('C', 0x00, struct snd_compr_caps *)
+#define SNDRV_COMPRESS_GET_CODEC_CAPS   _IOWR('C', 0x01, struct snd_compr_codec_caps *)
+#define SNDRV_COMPRESS_SET_PARAMS       _IOW('C', 0x02, struct snd_compr_params *)
+#define SNDRV_COMPRESS_GET_PARAMS       _IOR('C', 0x03, struct snd_compr_params *)
+#define SNDRV_COMPRESS_TSTAMP           _IOR('C', 0x10, struct snd_compr_tstamp *)
+#define SNDRV_COMPRESS_AVAIL            _IOR('C', 0x11, struct snd_compr_avail *)
+#define SNDRV_COMPRESS_PAUSE            _IO('C', 0x20)
+#define SNDRV_COMPRESS_RESUME           _IO('C', 0x21)
+#define SNDRV_COMPRESS_START            _IO('C', 0x22)
+#define SNDRV_COMPRESS_STOP             _IO('C', 0x23)
+#define SNDRV_COMPRESS_DRAIN            _IO('C', 0x24)
+
+#endif