const struct pollfd& pfd = mFDs[mInputDeviceIndex];
if (pfd.revents & POLLIN) {
readSize = read(pfd.fd, mInputBufferData,sizeof(struct input_event) * INPUT_BUFFER_SIZE);
mInputBufferCount = readSize / sizeof(struct input_event);
mInputBufferIndex = 0;
}
if (mInputBufferIndex < mInputBufferCount) {
const struct input_event& iev = mInputBufferData[mInputBufferIndex++];
const device_t* device = mDevices[mInputDeviceIndex];
struct input_event {
struct timeval time;
__u16 type;
__u16 code;
__s32 value;
};
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
LOCAL_CFLAGS := \
-fPIC -D_POSIX_SOURCE \
-DALSA_CONFIG_DIR=\"/system/usr/share/alsa\" \
-DALSA_PLUGIN_DIR=\"/system/usr/lib/alsa-lib\" \
-DALSA_DEVICE_DIRECTORY=\"/dev/snd/\"
LOCAL_C_INCLUDES := \
external/alsa-lib/include
LOCAL_SRC_FILES := \
asound_ioctl.c
LOCAL_MODULE_TAGS := eng
LOCAL_MODULE := asound_ioctl
LOCAL_SHARED_LIBRARIES := \
libaudio \
libasound \
libc
include $(BUILD_EXECUTABLE)
#include <linux/ioctl.h>
struct rt56xx_cmd {
size_t number;
int *buf;
};
enum {
RT_READ_CODEC_REG_IOCTL = _IOR('R', 0x01, struct rt56xx_cmd),
RT_WRITE_CODEC_REG_IOCTL = _IOW('R', 0x01, struct rt56xx_cmd),
RT_READ_ALL_CODEC_REG_IOCTL = _IOR('R', 0x02, struct rt56xx_cmd),
RT_READ_CODEC_INDEX_IOCTL = _IOR('R', 0x03, struct rt56xx_cmd),
RT_WRITE_CODEC_INDEX_IOCTL = _IOW('R', 0x03, struct rt56xx_cmd),
RT_READ_CODEC_DSP_IOCTL = _IOR('R', 0x04, struct rt56xx_cmd),
RT_WRITE_CODEC_DSP_IOCTL = _IOW('R', 0x04, struct rt56xx_cmd),
RT_SET_CODEC_HWEQ_IOCTL = _IOW('R', 0x05, struct rt56xx_cmd),
RT_GET_CODEC_HWEQ_IOCTL = _IOR('R', 0x05, struct rt56xx_cmd),
RT_SET_CODEC_SPK_VOL_IOCTL = _IOW('R', 0x06, struct rt56xx_cmd),
RT_GET_CODEC_SPK_VOL_IOCTL = _IOR('R', 0x06, struct rt56xx_cmd),
RT_SET_CODEC_MIC_GAIN_IOCTL = _IOW('R', 0x07, struct rt56xx_cmd),
RT_GET_CODEC_MIC_GAIN_IOCTL = _IOR('R', 0x07, struct rt56xx_cmd),
RT_SET_CODEC_3D_SPK_IOCTL = _IOW('R', 0x08, struct rt56xx_cmd),
RT_GET_CODEC_3D_SPK_IOCTL = _IOR('R', 0x08, struct rt56xx_cmd),
RT_SET_CODEC_MP3PLUS_IOCTL = _IOW('R', 0x09, struct rt56xx_cmd),
RT_GET_CODEC_MP3PLUS_IOCTL = _IOR('R', 0x09, struct rt56xx_cmd),
RT_SET_CODEC_3D_HEADPHONE_IOCTL = _IOW('R', 0x0a, struct rt56xx_cmd),
RT_GET_CODEC_3D_HEADPHONE_IOCTL = _IOR('R', 0x0a, struct rt56xx_cmd),
RT_SET_CODEC_BASS_BACK_IOCTL = _IOW('R', 0x0b, struct rt56xx_cmd),
RT_GET_CODEC_BASS_BACK_IOCTL = _IOR('R', 0x0b, struct rt56xx_cmd),
RT_SET_CODEC_DIPOLE_SPK_IOCTL = _IOW('R', 0x0c, struct rt56xx_cmd),
RT_GET_CODEC_DIPOLE_SPK_IOCTL = _IOR('R', 0x0c, struct rt56xx_cmd),
RT_SET_CODEC_DRC_AGC_ENABLE_IOCTL = _IOW('R', 0x0d, struct rt56xx_cmd),
RT_GET_CODEC_DRC_AGC_ENABLE_IOCTL = _IOR('R', 0x0d, struct rt56xx_cmd),
RT_SET_CODEC_DSP_MODE_IOCTL = _IOW('R', 0x0e, struct rt56xx_cmd),
RT_GET_CODEC_DSP_MODE_IOCTL = _IOR('R', 0x0e, struct rt56xx_cmd),
RT_SET_CODEC_WNR_ENABLE_IOCTL = _IOW('R', 0x0f, struct rt56xx_cmd),
....
$ cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 32: [ 1] : control 33: : timer 48: [ 1- 0]: digital audio playback 56: [ 1- 0]: digital audio capture 64: [ 2] : control 80: [ 2- 0]: digital audio playback 96: [ 3] : control 100: [ 3- 0]: hardware dependent 112: [ 3- 0]: digital audio playback 120: [ 3- 0]: digital audio capture第一行的數字就是 device node 的 minor number
bash-4.2# ls -l /dev/snd crw-rw-rw- 1 1000 1005 14, 4 Jan 1 1970 audio crw-rw-rw- 1 1000 1005 14, 20 Jan 1 1970 audio1 crw-rw-rw- 1 1000 1005 14, 36 Jan 1 1970 audio2 crw-rw-rw- 1 1000 1005 14, 52 Jan 1 1970 audio3 crw-rw-rw- 1 1000 1005 116, 0 Jan 1 1970 controlC0 crw-rw-rw- 1 1000 1005 116, 32 Jan 1 1970 controlC1 crw-rw-rw- 1 1000 1005 116, 64 Jan 1 1970 controlC2 crw-rw-rw- 1 1000 1005 116, 96 Jan 1 1970 controlC3 crw-rw---- 1 1000 1005 14, 3 Jan 1 1970 dsp crw-rw---- 1 1000 1005 14, 19 Jan 1 1970 dsp1 crw-rw-rw- 1 1000 1005 14, 35 Jan 1 1970 dsp2 crw-rw-rw- 1 1000 1005 14, 51 Jan 1 1970 dsp3 crw-rw-rw- 1 1000 1005 116, 100 Jan 1 1970 hwC3D0 crw-rw---- 1 1000 1005 14, 0 Jan 1 1970 mixer crw-rw-rw- 1 1000 1005 14, 16 Jan 1 1970 mixer1 crw-rw-rw- 1 1000 1005 14, 32 Jan 1 1970 mixer2 crw-rw-rw- 1 1000 1005 14, 48 Jan 1 1970 mixer3 crw-rw-rw- 1 1000 1005 116, 24 Jan 1 1970 pcmC0D0c crw-rw-rw- 1 1000 1005 116, 16 Jan 1 1970 pcmC0D0p crw-rw-rw- 1 1000 1005 116, 56 Jan 1 1970 pcmC1D0c crw-rw-rw- 1 1000 1005 116, 48 Jan 1 1970 pcmC1D0p crw-rw-rw- 1 1000 1005 116, 80 Jan 1 1970 pcmC2D0p crw-rw-rw- 1 1000 1005 116, 120 Jan 1 1970 pcmC3D0c crw-rw-rw- 1 1000 1005 116, 112 Jan 1 1970 pcmC3D0p crw-rw-rw- 1 1000 1005 116, 33 Jan 1 1970 timer可以看到 major, minor number.
bash-4.2# alsa_aplay -l **** List of PLAYBACK Hardware Devices **** card 0: ssfinneraudi [SSF internal audio], device 0: SSF internal audio [SSF internal audio] Subdevices: 0/1 Subdevice #0: subdevice #0 card 1: csrbtcodec [ccs-btcodec], device 0: btcodec hifi btcodec-usppcm-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: ssochdmi [ssoc-hdmi], device 0: HDMI ssoc-hdmi-audio-codec-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: ssocrt5625 [ssoc-rt5625], device 0: RT5625 rt5625-aif1-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0
I2S
===
I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
Rx lines are used for audio transmission, whilst the bit clock (BCLK) and
left/right clock (LRC) synchronise the link. I2S is flexible in that either the
controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock
usually varies depending on the sample rate and the master system clock
(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
ADC and DAC LRCLKs, this allows for simultaneous capture and playback at
different sample rates.
I2S has several different operating modes:-
o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC
transition.
o Left Justified - MSB is transmitted on transition of LRC.
o Right Justified - MSB is transmitted sample size BCLKs before LRC
transition.
Machine DAI Configuration
-------------------------
The machine DAI configuration glues all the codec and CPU DAIs together. It can
also be used to set up the DAI system clock and for any machine related DAI
initialisation e.g. the machine audio map can be connected to the codec audio
map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c
for examples.
struct snd_soc_dai_link is used to set up each DAI in your machine. e.g.
/* corgi digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link corgi_dai = {
.name = "WM8731",
.stream_name = "WM8731",
.cpu_dai_name = "pxa-is2-dai",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8713-codec.0-001a",
.init = corgi_wm8731_init,
.ops = &corgi_ops,
};
struct snd_soc_card then sets up the machine with its DAIs. e.g.
/* corgi audio machine driver */
static struct snd_soc_card snd_soc_corgi = {
.name = "Corgi",
.dai_link = &corgi_dai,
.num_links = 1,
};
/* * DAI hardware clock masters. * * This is wrt the codec, the inverse is true for the interface * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ #define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ #define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ #define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
snd_soc_dai_ops 中的 set_sysclk:
struct snd_soc_dai_ops {
/*
* DAI clocking configuration, all optional.
* Called by soc_card drivers, normally in their hw_params.
*/
int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir);
int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
/*
* DAI format configuration
* Called by soc_card drivers, normally in their hw_params.
*/
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
* DAI digital mute - optional.
* Called by soc-core to minimise any pops.
*/
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
/*
* ALSA PCM audio operations - all optional.
* Called by soc-core during audio PCM operations.
*/
int (*startup)(struct snd_pcm_substream *, struct snd_soc_dai *);
void (*shutdown)(struct snd_pcm_substream *, struct snd_soc_dai *);
int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *, struct snd_soc_dai *);
int (*hw_free)(struct snd_pcm_substream *, struct snd_soc_dai *);
int (*prepare)(struct snd_pcm_substream *, struct snd_soc_dai *);
int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *);
/*
* For hardware based FIFO caused delay reporting.
* Optional.
*/
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, struct snd_soc_dai *);
};
structure 間的關係
snd_soc_card
|
+ dai_link --- snd_soc_dai_link
|
+ ops ---------------- snd_soc_ops
| |
| + hw_params
+ stream name
|
+ cpu dai name
|
+ codec dai name
|
+ codec name
|
+ platform name