Merge "Miscellaneous updates to <system/audio.h>"
diff --git a/include/system/audio.h b/include/system/audio.h
index 59e5e41..2424baf 100644
--- a/include/system/audio.h
+++ b/include/system/audio.h
@@ -34,11 +34,17 @@
/* device address used to refer to the standard remote submix */
#define AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS "0"
+/* AudioFlinger and AudioPolicy services use I/O handles to identify audio sources and sinks */
typedef int audio_io_handle_t;
+#define AUDIO_IO_HANDLE_NONE 0
/* Audio stream types */
typedef enum {
+ /* These values must kept in sync with
+ * frameworks/base/media/java/android/media/AudioSystem.java
+ */
AUDIO_STREAM_DEFAULT = -1,
+ AUDIO_STREAM_MIN = 0,
AUDIO_STREAM_VOICE_CALL = 0,
AUDIO_STREAM_SYSTEM = 1,
AUDIO_STREAM_RING = 2,
@@ -55,7 +61,9 @@
} audio_stream_type_t;
/* Do not change these values without updating their counterparts
- * in media/java/android/media/MediaRecorder.java!
+ * in frameworks/base/media/java/android/media/MediaRecorder.java,
+ * frameworks/av/services/audioflinger/AudioPolicyService.cpp,
+ * and system/media/audio_effects/include/audio_effects/audio_effects_conf.h!
*/
typedef enum {
AUDIO_SOURCE_DEFAULT = 0,
@@ -110,8 +118,11 @@
AUDIO_FORMAT_PCM_SUB_8_BIT = 0x2, /* DO NOT CHANGE - PCM unsigned 8 bits */
AUDIO_FORMAT_PCM_SUB_32_BIT = 0x3, /* PCM signed .31 fixed point */
AUDIO_FORMAT_PCM_SUB_8_24_BIT = 0x4, /* PCM signed 7.24 fixed point */
+ AUDIO_FORMAT_PCM_SUB_FLOAT = 0x5, /* PCM single-precision floating point */
} audio_format_pcm_sub_fmt_t;
+/* The audio_format_*_sub_fmt_t declarations are not currently used */
+
/* MP3 sub format field definition : can use 11 LSBs in the same way as MP3
* frame header to specify bit rate, stereo mode, version...
*/
@@ -136,7 +147,7 @@
AUDIO_FORMAT_VORBIS_SUB_NONE = 0x0,
} audio_format_vorbis_sub_fmt_t;
-/* Audio format consists in a main format field (upper 8 bits) and a sub format
+/* Audio format consists of a main format field (upper 8 bits) and a sub format
* field (lower 24 bits).
*
* The main format indicates the main codec type. The sub format field
@@ -160,14 +171,18 @@
AUDIO_FORMAT_SUB_MASK = 0x00FFFFFFUL,
/* Aliases */
+ /* note != AudioFormat.ENCODING_PCM_16BIT */
AUDIO_FORMAT_PCM_16_BIT = (AUDIO_FORMAT_PCM |
AUDIO_FORMAT_PCM_SUB_16_BIT),
+ /* note != AudioFormat.ENCODING_PCM_8BIT */
AUDIO_FORMAT_PCM_8_BIT = (AUDIO_FORMAT_PCM |
AUDIO_FORMAT_PCM_SUB_8_BIT),
AUDIO_FORMAT_PCM_32_BIT = (AUDIO_FORMAT_PCM |
AUDIO_FORMAT_PCM_SUB_32_BIT),
AUDIO_FORMAT_PCM_8_24_BIT = (AUDIO_FORMAT_PCM |
AUDIO_FORMAT_PCM_SUB_8_24_BIT),
+ AUDIO_FORMAT_PCM_FLOAT = (AUDIO_FORMAT_PCM |
+ AUDIO_FORMAT_PCM_SUB_FLOAT),
} audio_format_t;
enum {
@@ -273,6 +288,15 @@
typedef uint32_t audio_channel_mask_t;
+/* Expresses the convention when stereo audio samples are stored interleaved
+ * in an array. This should improve readability by allowing code to use
+ * symbolic indices instead of hard-coded [0] and [1].
+ */
+enum {
+ AUDIO_INTERLEAVE_LEFT = 0,
+ AUDIO_INTERLEAVE_RIGHT = 1,
+};
+
typedef enum {
AUDIO_MODE_INVALID = -2,
AUDIO_MODE_CURRENT = -1,
@@ -285,7 +309,9 @@
AUDIO_MODE_MAX = AUDIO_MODE_CNT - 1,
} audio_mode_t;
+/* This enum is deprecated */
typedef enum {
+ AUDIO_IN_ACOUSTICS_NONE = 0,
AUDIO_IN_ACOUSTICS_AGC_ENABLE = 0x0001,
AUDIO_IN_ACOUSTICS_AGC_DISABLE = 0,
AUDIO_IN_ACOUSTICS_NS_ENABLE = 0x0002,
@@ -311,9 +337,12 @@
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
AUDIO_DEVICE_OUT_AUX_DIGITAL = 0x400,
+ /* uses an analog connection (multiplexed over the USB connector pins for instance) */
AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800,
AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000,
+ /* USB accessory mode: your Android device is a USB device and the dock is a USB host */
AUDIO_DEVICE_OUT_USB_ACCESSORY = 0x2000,
+ /* USB host mode: your Android device is a USB host and the dock is a USB device */
AUDIO_DEVICE_OUT_USB_DEVICE = 0x4000,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX = 0x8000,
AUDIO_DEVICE_OUT_DEFAULT = AUDIO_DEVICE_BIT_DEFAULT,
@@ -529,7 +558,7 @@
*/
static inline audio_channel_mask_t audio_channel_out_mask_from_count(uint32_t channel_count)
{
- switch(channel_count) {
+ switch (channel_count) {
case 1:
return AUDIO_CHANNEL_OUT_MONO;
case 2:
@@ -569,7 +598,7 @@
switch (format & AUDIO_FORMAT_MAIN_MASK) {
case AUDIO_FORMAT_PCM:
if (format != AUDIO_FORMAT_PCM_16_BIT &&
- format != AUDIO_FORMAT_PCM_8_BIT) {
+ format != AUDIO_FORMAT_PCM_8_BIT && format != AUDIO_FORMAT_PCM_FLOAT) {
return false;
}
case AUDIO_FORMAT_MP3:
@@ -605,6 +634,9 @@
case AUDIO_FORMAT_PCM_8_BIT:
size = sizeof(uint8_t);
break;
+ case AUDIO_FORMAT_PCM_FLOAT:
+ size = sizeof(float);
+ break;
default:
break;
}