diff options
Diffstat (limited to 'include/system/audio.h')
| -rw-r--r-- | include/system/audio.h | 827 |
1 files changed, 784 insertions, 43 deletions
diff --git a/include/system/audio.h b/include/system/audio.h index 8838e71..1c70240 100644 --- a/include/system/audio.h +++ b/include/system/audio.h @@ -20,6 +20,7 @@ #include <stdbool.h> #include <stdint.h> +#include <stdio.h> #include <sys/cdefs.h> #include <sys/types.h> @@ -34,11 +35,17 @@ __BEGIN_DECLS /* 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, @@ -46,7 +53,9 @@ typedef enum { AUDIO_STREAM_ALARM = 4, AUDIO_STREAM_NOTIFICATION = 5, AUDIO_STREAM_BLUETOOTH_SCO = 6, - AUDIO_STREAM_ENFORCED_AUDIBLE = 7, /* Sounds that cannot be muted by user and must be routed to speaker */ + AUDIO_STREAM_ENFORCED_AUDIBLE = 7, /* Sounds that cannot be muted by user + * and must be routed to speaker + */ AUDIO_STREAM_DTMF = 8, AUDIO_STREAM_TTS = 9, @@ -55,7 +64,60 @@ typedef enum { } 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/AudioAttributes.java + */ +typedef enum { + AUDIO_CONTENT_TYPE_UNKNOWN = 0, + AUDIO_CONTENT_TYPE_SPEECH = 1, + AUDIO_CONTENT_TYPE_MUSIC = 2, + AUDIO_CONTENT_TYPE_MOVIE = 3, + AUDIO_CONTENT_TYPE_SONIFICATION = 4, + + AUDIO_CONTENT_TYPE_CNT, + AUDIO_CONTENT_TYPE_MAX = AUDIO_CONTENT_TYPE_CNT - 1, +} audio_content_type_t; + +/* Do not change these values without updating their counterparts + * in frameworks/base/media/java/android/media/AudioAttributes.java + */ +typedef enum { + AUDIO_USAGE_UNKNOWN = 0, + AUDIO_USAGE_MEDIA = 1, + AUDIO_USAGE_VOICE_COMMUNICATION = 2, + AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING = 3, + AUDIO_USAGE_ALARM = 4, + AUDIO_USAGE_NOTIFICATION = 5, + AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE = 6, + AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST = 7, + AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT = 8, + AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED = 9, + AUDIO_USAGE_NOTIFICATION_EVENT = 10, + AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY = 11, + AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE = 12, + AUDIO_USAGE_ASSISTANCE_SONIFICATION = 13, + AUDIO_USAGE_GAME = 14, + + AUDIO_USAGE_CNT, + AUDIO_USAGE_MAX = AUDIO_USAGE_CNT - 1, +} audio_usage_t; + +typedef uint32_t audio_flags_mask_t; + +/* Do not change these values without updating their counterparts + * in frameworks/base/media/java/android/media/AudioAttributes.java + */ +enum { + AUDIO_FLAG_AUDIBILITY_ENFORCED = 0x1, + AUDIO_FLAG_SECURE = 0x2, + AUDIO_FLAG_SCO = 0x4, + AUDIO_FLAG_BEACON = 0x8, + AUDIO_FLAG_HW_AV_SYNC = 0x10 +}; + +/* Do not change these values without updating their counterparts + * in frameworks/base/media/java/android/media/MediaRecorder.java, + * frameworks/av/services/audiopolicy/AudioPolicyService.cpp, + * and system/media/audio_effects/include/audio_effects/audio_effects_conf.h! */ typedef enum { AUDIO_SOURCE_DEFAULT = 0, @@ -79,6 +141,16 @@ typedef enum { at the audio HAL. */ } audio_source_t; +/* Audio attributes */ +#define AUDIO_ATTRIBUTES_TAGS_MAX_SIZE 256 +typedef struct { + audio_content_type_t content_type; + audio_usage_t usage; + audio_source_t source; + audio_flags_mask_t flags; + char tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */ +} audio_attributes_t; + /* special audio session values * (XXX: should this be living in the audio effects land?) */ @@ -93,18 +165,35 @@ typedef enum { * (value must be 0) */ AUDIO_SESSION_OUTPUT_MIX = 0, + + /* application does not specify an explicit session ID to be used, + * and requests a new session ID to be allocated + * TODO use unique values for AUDIO_SESSION_OUTPUT_MIX and AUDIO_SESSION_ALLOCATE, + * after all uses have been updated from 0 to the appropriate symbol, and have been tested. + */ + AUDIO_SESSION_ALLOCATE = 0, } audio_session_t; +/* a unique ID allocated by AudioFlinger for use as a audio_io_handle_t or audio_session_t */ +typedef int audio_unique_id_t; + +#define AUDIO_UNIQUE_ID_ALLOCATE AUDIO_SESSION_ALLOCATE + /* Audio sub formats (see enum audio_format). */ /* PCM sub formats */ typedef enum { + /* All of these are in native byte order */ AUDIO_FORMAT_PCM_SUB_16_BIT = 0x1, /* DO NOT CHANGE - PCM signed 16 bits */ 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_24_BIT_PACKED = 0x6, /* PCM signed .23 fixed point packed in 3 bytes */ } 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... */ @@ -121,7 +210,16 @@ typedef enum { /* AAC sub format field definition: specify profile or bitrate for recording... */ typedef enum { - AUDIO_FORMAT_AAC_SUB_NONE = 0x0, + AUDIO_FORMAT_AAC_SUB_MAIN = 0x1, + AUDIO_FORMAT_AAC_SUB_LC = 0x2, + AUDIO_FORMAT_AAC_SUB_SSR = 0x4, + AUDIO_FORMAT_AAC_SUB_LTP = 0x8, + AUDIO_FORMAT_AAC_SUB_HE_V1 = 0x10, + AUDIO_FORMAT_AAC_SUB_SCALABLE = 0x20, + AUDIO_FORMAT_AAC_SUB_ERLC = 0x40, + AUDIO_FORMAT_AAC_SUB_LD = 0x80, + AUDIO_FORMAT_AAC_SUB_HE_V2 = 0x100, + AUDIO_FORMAT_AAC_SUB_ELD = 0x200, } audio_format_aac_sub_fmt_t; /* VORBIS sub format field definition: specify quality for recording... */ @@ -129,7 +227,7 @@ typedef enum { 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 @@ -146,24 +244,65 @@ typedef enum { AUDIO_FORMAT_AMR_NB = 0x02000000UL, AUDIO_FORMAT_AMR_WB = 0x03000000UL, AUDIO_FORMAT_AAC = 0x04000000UL, - AUDIO_FORMAT_HE_AAC_V1 = 0x05000000UL, - AUDIO_FORMAT_HE_AAC_V2 = 0x06000000UL, + AUDIO_FORMAT_HE_AAC_V1 = 0x05000000UL, /* Deprecated, Use AUDIO_FORMAT_AAC_HE_V1*/ + AUDIO_FORMAT_HE_AAC_V2 = 0x06000000UL, /* Deprecated, Use AUDIO_FORMAT_AAC_HE_V2*/ AUDIO_FORMAT_VORBIS = 0x07000000UL, + AUDIO_FORMAT_OPUS = 0x08000000UL, + AUDIO_FORMAT_AC3 = 0x09000000UL, + AUDIO_FORMAT_E_AC3 = 0x0A000000UL, AUDIO_FORMAT_MAIN_MASK = 0xFF000000UL, 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_PCM_24_BIT_PACKED = (AUDIO_FORMAT_PCM | + AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED), + AUDIO_FORMAT_AAC_MAIN = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_MAIN), + AUDIO_FORMAT_AAC_LC = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_LC), + AUDIO_FORMAT_AAC_SSR = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_SSR), + AUDIO_FORMAT_AAC_LTP = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_LTP), + AUDIO_FORMAT_AAC_HE_V1 = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_HE_V1), + AUDIO_FORMAT_AAC_SCALABLE = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_SCALABLE), + AUDIO_FORMAT_AAC_ERLC = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_ERLC), + AUDIO_FORMAT_AAC_LD = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_LD), + AUDIO_FORMAT_AAC_HE_V2 = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_HE_V2), + AUDIO_FORMAT_AAC_ELD = (AUDIO_FORMAT_AAC | + AUDIO_FORMAT_AAC_SUB_ELD), } audio_format_t; +/* For the channel mask for position assignment representation */ enum { + +/* These can be a complete audio_channel_mask_t. */ + + AUDIO_CHANNEL_NONE = 0x0, + AUDIO_CHANNEL_INVALID = 0xC0000000, + +/* These can be the bits portion of an audio_channel_mask_t + * with representation AUDIO_CHANNEL_REPRESENTATION_POSITION. + * Using these bits as a complete audio_channel_mask_t is deprecated. + */ + /* output channels */ AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1, AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2, @@ -184,6 +323,8 @@ enum { AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000, AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000, +/* TODO: should these be considered complete channel masks, or only bits? */ + AUDIO_CHANNEL_OUT_MONO = AUDIO_CHANNEL_OUT_FRONT_LEFT, AUDIO_CHANNEL_OUT_STEREO = (AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT), @@ -191,16 +332,26 @@ enum { AUDIO_CHANNEL_OUT_FRONT_RIGHT | AUDIO_CHANNEL_OUT_BACK_LEFT | AUDIO_CHANNEL_OUT_BACK_RIGHT), - AUDIO_CHANNEL_OUT_SURROUND = (AUDIO_CHANNEL_OUT_FRONT_LEFT | + AUDIO_CHANNEL_OUT_QUAD_BACK = AUDIO_CHANNEL_OUT_QUAD, + /* like AUDIO_CHANNEL_OUT_QUAD_BACK with *_SIDE_* instead of *_BACK_* */ + AUDIO_CHANNEL_OUT_QUAD_SIDE = (AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT | - AUDIO_CHANNEL_OUT_FRONT_CENTER | - AUDIO_CHANNEL_OUT_BACK_CENTER), + AUDIO_CHANNEL_OUT_SIDE_LEFT | + AUDIO_CHANNEL_OUT_SIDE_RIGHT), AUDIO_CHANNEL_OUT_5POINT1 = (AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT | AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_BACK_LEFT | AUDIO_CHANNEL_OUT_BACK_RIGHT), + AUDIO_CHANNEL_OUT_5POINT1_BACK = AUDIO_CHANNEL_OUT_5POINT1, + /* like AUDIO_CHANNEL_OUT_5POINT1_BACK with *_SIDE_* instead of *_BACK_* */ + AUDIO_CHANNEL_OUT_5POINT1_SIDE = (AUDIO_CHANNEL_OUT_FRONT_LEFT | + AUDIO_CHANNEL_OUT_FRONT_RIGHT | + AUDIO_CHANNEL_OUT_FRONT_CENTER | + AUDIO_CHANNEL_OUT_LOW_FREQUENCY | + AUDIO_CHANNEL_OUT_SIDE_LEFT | + AUDIO_CHANNEL_OUT_SIDE_RIGHT), // matches the correct AudioFormat.CHANNEL_OUT_7POINT1_SURROUND definition for 7.1 AUDIO_CHANNEL_OUT_7POINT1 = (AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT | @@ -229,6 +380,8 @@ enum { AUDIO_CHANNEL_OUT_TOP_BACK_CENTER| AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT), +/* These are bits only, not complete values */ + /* input channels */ AUDIO_CHANNEL_IN_LEFT = 0x4, AUDIO_CHANNEL_IN_RIGHT = 0x8, @@ -245,6 +398,8 @@ enum { AUDIO_CHANNEL_IN_VOICE_UPLINK = 0x4000, AUDIO_CHANNEL_IN_VOICE_DNLINK = 0x8000, +/* TODO: should these be considered complete channel masks, or only bits, or deprecated? */ + AUDIO_CHANNEL_IN_MONO = AUDIO_CHANNEL_IN_FRONT, AUDIO_CHANNEL_IN_STEREO = (AUDIO_CHANNEL_IN_LEFT | AUDIO_CHANNEL_IN_RIGHT), AUDIO_CHANNEL_IN_FRONT_BACK = (AUDIO_CHANNEL_IN_FRONT | AUDIO_CHANNEL_IN_BACK), @@ -264,8 +419,111 @@ enum { AUDIO_CHANNEL_IN_VOICE_DNLINK), }; +/* A channel mask per se only defines the presence or absence of a channel, not the order. + * But see AUDIO_INTERLEAVE_* below for the platform convention of order. + * + * audio_channel_mask_t is an opaque type and its internal layout should not + * be assumed as it may change in the future. + * Instead, always use the functions declared in this header to examine. + * + * These are the current representations: + * + * AUDIO_CHANNEL_REPRESENTATION_POSITION + * is a channel mask representation for position assignment. + * Each low-order bit corresponds to the spatial position of a transducer (output), + * or interpretation of channel (input). + * The user of a channel mask needs to know the context of whether it is for output or input. + * The constants AUDIO_CHANNEL_OUT_* or AUDIO_CHANNEL_IN_* apply to the bits portion. + * It is not permitted for no bits to be set. + * + * AUDIO_CHANNEL_REPRESENTATION_INDEX + * is a channel mask representation for index assignment. + * Each low-order bit corresponds to a selected channel. + * There is no platform interpretation of the various bits. + * There is no concept of output or input. + * It is not permitted for no bits to be set. + * + * All other representations are reserved for future use. + * + * Warning: current representation distinguishes between input and output, but this will not the be + * case in future revisions of the platform. Wherever there is an ambiguity between input and output + * that is currently resolved by checking the channel mask, the implementer should look for ways to + * fix it with additional information outside of the mask. + */ typedef uint32_t audio_channel_mask_t; +/* Maximum number of channels for all representations */ +#define AUDIO_CHANNEL_COUNT_MAX 30 + +/* log(2) of maximum number of representations, not part of public API */ +#define AUDIO_CHANNEL_REPRESENTATION_LOG2 2 + +/* Representations */ +typedef enum { + AUDIO_CHANNEL_REPRESENTATION_POSITION = 0, // must be zero for compatibility + // 1 is reserved for future use + AUDIO_CHANNEL_REPRESENTATION_INDEX = 2, + // 3 is reserved for future use +} audio_channel_representation_t; + +/* The return value is undefined if the channel mask is invalid. */ +static inline uint32_t audio_channel_mask_get_bits(audio_channel_mask_t channel) +{ + return channel & ((1 << AUDIO_CHANNEL_COUNT_MAX) - 1); +} + +/* The return value is undefined if the channel mask is invalid. */ +static inline audio_channel_representation_t audio_channel_mask_get_representation( + audio_channel_mask_t channel) +{ + // The right shift should be sufficient, but also "and" for safety in case mask is not 32 bits + return (audio_channel_representation_t) + ((channel >> AUDIO_CHANNEL_COUNT_MAX) & ((1 << AUDIO_CHANNEL_REPRESENTATION_LOG2) - 1)); +} + +/* Returns true if the channel mask is valid, + * or returns false for AUDIO_CHANNEL_NONE, AUDIO_CHANNEL_INVALID, and other invalid values. + * This function is unable to determine whether a channel mask for position assignment + * is invalid because an output mask has an invalid output bit set, + * or because an input mask has an invalid input bit set. + * All other APIs that take a channel mask assume that it is valid. + */ +static inline bool audio_channel_mask_is_valid(audio_channel_mask_t channel) +{ + uint32_t bits = audio_channel_mask_get_bits(channel); + audio_channel_representation_t representation = audio_channel_mask_get_representation(channel); + switch (representation) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + break; + default: + bits = 0; + break; + } + return bits != 0; +} + +/* Not part of public API */ +static inline audio_channel_mask_t audio_channel_mask_from_representation_and_bits( + audio_channel_representation_t representation, uint32_t bits) +{ + return (audio_channel_mask_t) ((representation << AUDIO_CHANNEL_COUNT_MAX) | bits); +} + +/* 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]. + * + * For multi-channel beyond stereo, the platform convention is that channels + * are interleaved in order from least significant channel mask bit + * to most significant channel mask bit, with unused bits skipped. + * Any exceptions to this convention will be noted at the appropriate API. + */ +enum { + AUDIO_INTERLEAVE_LEFT = 0, + AUDIO_INTERLEAVE_RIGHT = 1, +}; + typedef enum { AUDIO_MODE_INVALID = -2, AUDIO_MODE_CURRENT = -1, @@ -278,7 +536,9 @@ typedef enum { 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, @@ -304,11 +564,27 @@ enum { AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, AUDIO_DEVICE_OUT_AUX_DIGITAL = 0x400, + AUDIO_DEVICE_OUT_HDMI = AUDIO_DEVICE_OUT_AUX_DIGITAL, + /* 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, + /* Telephony voice TX path */ + AUDIO_DEVICE_OUT_TELEPHONY_TX = 0x10000, + /* Analog jack with line impedance detected */ + AUDIO_DEVICE_OUT_LINE = 0x20000, + /* HDMI Audio Return Channel */ + AUDIO_DEVICE_OUT_HDMI_ARC = 0x40000, + /* S/PDIF out */ + AUDIO_DEVICE_OUT_SPDIF = 0x80000, + /* FM transmitter out */ + AUDIO_DEVICE_OUT_FM = 0x100000, + /* Line out for av devices */ + AUDIO_DEVICE_OUT_AUX_LINE = 0x200000, AUDIO_DEVICE_OUT_DEFAULT = AUDIO_DEVICE_BIT_DEFAULT, AUDIO_DEVICE_OUT_ALL = (AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | @@ -320,12 +596,18 @@ enum { AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | - AUDIO_DEVICE_OUT_AUX_DIGITAL | + AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET | AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET | AUDIO_DEVICE_OUT_USB_ACCESSORY | AUDIO_DEVICE_OUT_USB_DEVICE | AUDIO_DEVICE_OUT_REMOTE_SUBMIX | + AUDIO_DEVICE_OUT_TELEPHONY_TX | + AUDIO_DEVICE_OUT_LINE | + AUDIO_DEVICE_OUT_HDMI_ARC | + AUDIO_DEVICE_OUT_SPDIF | + AUDIO_DEVICE_OUT_FM | + AUDIO_DEVICE_OUT_AUX_LINE | AUDIO_DEVICE_OUT_DEFAULT), AUDIO_DEVICE_OUT_ALL_A2DP = (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | @@ -343,13 +625,26 @@ enum { AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET = AUDIO_DEVICE_BIT_IN | 0x8, AUDIO_DEVICE_IN_WIRED_HEADSET = AUDIO_DEVICE_BIT_IN | 0x10, AUDIO_DEVICE_IN_AUX_DIGITAL = AUDIO_DEVICE_BIT_IN | 0x20, + AUDIO_DEVICE_IN_HDMI = AUDIO_DEVICE_IN_AUX_DIGITAL, + /* Telephony voice RX path */ AUDIO_DEVICE_IN_VOICE_CALL = AUDIO_DEVICE_BIT_IN | 0x40, + AUDIO_DEVICE_IN_TELEPHONY_RX = AUDIO_DEVICE_IN_VOICE_CALL, AUDIO_DEVICE_IN_BACK_MIC = AUDIO_DEVICE_BIT_IN | 0x80, AUDIO_DEVICE_IN_REMOTE_SUBMIX = AUDIO_DEVICE_BIT_IN | 0x100, AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET = AUDIO_DEVICE_BIT_IN | 0x200, AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET = AUDIO_DEVICE_BIT_IN | 0x400, AUDIO_DEVICE_IN_USB_ACCESSORY = AUDIO_DEVICE_BIT_IN | 0x800, AUDIO_DEVICE_IN_USB_DEVICE = AUDIO_DEVICE_BIT_IN | 0x1000, + /* FM tuner input */ + AUDIO_DEVICE_IN_FM_TUNER = AUDIO_DEVICE_BIT_IN | 0x2000, + /* TV tuner input */ + AUDIO_DEVICE_IN_TV_TUNER = AUDIO_DEVICE_BIT_IN | 0x4000, + /* Analog jack with line impedance detected */ + AUDIO_DEVICE_IN_LINE = AUDIO_DEVICE_BIT_IN | 0x8000, + /* S/PDIF in */ + AUDIO_DEVICE_IN_SPDIF = AUDIO_DEVICE_BIT_IN | 0x10000, + AUDIO_DEVICE_IN_BLUETOOTH_A2DP = AUDIO_DEVICE_BIT_IN | 0x20000, + AUDIO_DEVICE_IN_LOOPBACK = AUDIO_DEVICE_BIT_IN | 0x40000, AUDIO_DEVICE_IN_DEFAULT = AUDIO_DEVICE_BIT_IN | AUDIO_DEVICE_BIT_DEFAULT, AUDIO_DEVICE_IN_ALL = (AUDIO_DEVICE_IN_COMMUNICATION | @@ -357,16 +652,24 @@ enum { AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET | AUDIO_DEVICE_IN_WIRED_HEADSET | - AUDIO_DEVICE_IN_AUX_DIGITAL | - AUDIO_DEVICE_IN_VOICE_CALL | + AUDIO_DEVICE_IN_HDMI | + AUDIO_DEVICE_IN_TELEPHONY_RX | AUDIO_DEVICE_IN_BACK_MIC | AUDIO_DEVICE_IN_REMOTE_SUBMIX | AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET | AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET | AUDIO_DEVICE_IN_USB_ACCESSORY | AUDIO_DEVICE_IN_USB_DEVICE | + AUDIO_DEVICE_IN_FM_TUNER | + AUDIO_DEVICE_IN_TV_TUNER | + AUDIO_DEVICE_IN_LINE | + AUDIO_DEVICE_IN_SPDIF | + AUDIO_DEVICE_IN_BLUETOOTH_A2DP | + AUDIO_DEVICE_IN_LOOPBACK | AUDIO_DEVICE_IN_DEFAULT), AUDIO_DEVICE_IN_ALL_SCO = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, + AUDIO_DEVICE_IN_ALL_USB = (AUDIO_DEVICE_IN_USB_ACCESSORY | + AUDIO_DEVICE_IN_USB_DEVICE), }; typedef uint32_t audio_devices_t; @@ -394,7 +697,8 @@ typedef enum { AUDIO_OUTPUT_FLAG_DEEP_BUFFER = 0x8, // use deep audio buffers AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD = 0x10, // offload playback of compressed // streams to hardware codec - AUDIO_OUTPUT_FLAG_NON_BLOCKING = 0x20 // use non-blocking write + AUDIO_OUTPUT_FLAG_NON_BLOCKING = 0x20, // use non-blocking write + AUDIO_OUTPUT_FLAG_HW_AV_SYNC = 0x40 // output uses a hardware A/V synchronization source } audio_output_flags_t; /* The audio input flags are analogous to audio output flags. @@ -444,6 +748,264 @@ static const audio_offload_info_t AUDIO_INFO_INITIALIZER = { is_streaming: false }; +/* common audio stream configuration parameters + * You should memset() the entire structure to zero before use to + * ensure forward compatibility + */ +struct audio_config { + uint32_t sample_rate; + audio_channel_mask_t channel_mask; + audio_format_t format; + audio_offload_info_t offload_info; + size_t frame_count; +}; +typedef struct audio_config audio_config_t; + +static const audio_config_t AUDIO_CONFIG_INITIALIZER = { + sample_rate: 0, + channel_mask: AUDIO_CHANNEL_NONE, + format: AUDIO_FORMAT_DEFAULT, + offload_info: { + version: AUDIO_OFFLOAD_INFO_VERSION_CURRENT, + size: sizeof(audio_offload_info_t), + sample_rate: 0, + channel_mask: 0, + format: AUDIO_FORMAT_DEFAULT, + stream_type: AUDIO_STREAM_VOICE_CALL, + bit_rate: 0, + duration_us: 0, + has_video: false, + is_streaming: false + }, + frame_count: 0, +}; + + +/* audio hw module handle functions or structures referencing a module */ +typedef int audio_module_handle_t; + +/****************************** + * Volume control + *****************************/ + +/* If the audio hardware supports gain control on some audio paths, + * the platform can expose them in the audio_policy.conf file. The audio HAL + * will then implement gain control functions that will use the following data + * structures. */ + +/* Type of gain control exposed by an audio port */ +#define AUDIO_GAIN_MODE_JOINT 0x1 /* supports joint channel gain control */ +#define AUDIO_GAIN_MODE_CHANNELS 0x2 /* supports separate channel gain control */ +#define AUDIO_GAIN_MODE_RAMP 0x4 /* supports gain ramps */ + +typedef uint32_t audio_gain_mode_t; + + +/* An audio_gain struct is a representation of a gain stage. + * A gain stage is always attached to an audio port. */ +struct audio_gain { + audio_gain_mode_t mode; /* e.g. AUDIO_GAIN_MODE_JOINT */ + audio_channel_mask_t channel_mask; /* channels which gain an be controlled. + N/A if AUDIO_GAIN_MODE_CHANNELS is not supported */ + int min_value; /* minimum gain value in millibels */ + int max_value; /* maximum gain value in millibels */ + int default_value; /* default gain value in millibels */ + unsigned int step_value; /* gain step in millibels */ + unsigned int min_ramp_ms; /* minimum ramp duration in ms */ + unsigned int max_ramp_ms; /* maximum ramp duration in ms */ +}; + +/* The gain configuration structure is used to get or set the gain values of a + * given port */ +struct audio_gain_config { + int index; /* index of the corresponding audio_gain in the + audio_port gains[] table */ + audio_gain_mode_t mode; /* mode requested for this command */ + audio_channel_mask_t channel_mask; /* channels which gain value follows. + N/A in joint mode */ + int values[sizeof(audio_channel_mask_t) * 8]; /* gain values in millibels + for each channel ordered from LSb to MSb in + channel mask. The number of values is 1 in joint + mode or popcount(channel_mask) */ + unsigned int ramp_duration_ms; /* ramp duration in ms */ +}; + +/****************************** + * Routing control + *****************************/ + +/* Types defined here are used to describe an audio source or sink at internal + * framework interfaces (audio policy, patch panel) or at the audio HAL. + * Sink and sources are grouped in a concept of “audio port” representing an + * audio end point at the edge of the system managed by the module exposing + * the interface. */ + +/* Audio port role: either source or sink */ +typedef enum { + AUDIO_PORT_ROLE_NONE, + AUDIO_PORT_ROLE_SOURCE, + AUDIO_PORT_ROLE_SINK, +} audio_port_role_t; + +/* Audio port type indicates if it is a session (e.g AudioTrack), + * a mix (e.g PlaybackThread output) or a physical device + * (e.g AUDIO_DEVICE_OUT_SPEAKER) */ +typedef enum { + AUDIO_PORT_TYPE_NONE, + AUDIO_PORT_TYPE_DEVICE, + AUDIO_PORT_TYPE_MIX, + AUDIO_PORT_TYPE_SESSION, +} audio_port_type_t; + +/* Each port has a unique ID or handle allocated by policy manager */ +typedef int audio_port_handle_t; +#define AUDIO_PORT_HANDLE_NONE 0 + + +/* maximum audio device address length */ +#define AUDIO_DEVICE_MAX_ADDRESS_LEN 32 + +/* extension for audio port configuration structure when the audio port is a + * hardware device */ +struct audio_port_config_device_ext { + audio_module_handle_t hw_module; /* module the device is attached to */ + audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */ + char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; /* device address. "" if N/A */ +}; + +/* extension for audio port configuration structure when the audio port is a + * sub mix */ +struct audio_port_config_mix_ext { + audio_module_handle_t hw_module; /* module the stream is attached to */ + audio_io_handle_t handle; /* I/O handle of the input/output stream */ + union { + //TODO: change use case for output streams: use strategy and mixer attributes + audio_stream_type_t stream; + audio_source_t source; + } usecase; +}; + +/* extension for audio port configuration structure when the audio port is an + * audio session */ +struct audio_port_config_session_ext { + audio_session_t session; /* audio session */ +}; + +/* Flags indicating which fields are to be considered in struct audio_port_config */ +#define AUDIO_PORT_CONFIG_SAMPLE_RATE 0x1 +#define AUDIO_PORT_CONFIG_CHANNEL_MASK 0x2 +#define AUDIO_PORT_CONFIG_FORMAT 0x4 +#define AUDIO_PORT_CONFIG_GAIN 0x8 +#define AUDIO_PORT_CONFIG_ALL (AUDIO_PORT_CONFIG_SAMPLE_RATE | \ + AUDIO_PORT_CONFIG_CHANNEL_MASK | \ + AUDIO_PORT_CONFIG_FORMAT | \ + AUDIO_PORT_CONFIG_GAIN) + +/* audio port configuration structure used to specify a particular configuration of + * an audio port */ +struct audio_port_config { + audio_port_handle_t id; /* port unique ID */ + audio_port_role_t role; /* sink or source */ + audio_port_type_t type; /* device, mix ... */ + unsigned int config_mask; /* e.g AUDIO_PORT_CONFIG_ALL */ + unsigned int sample_rate; /* sampling rate in Hz */ + audio_channel_mask_t channel_mask; /* channel mask if applicable */ + audio_format_t format; /* format if applicable */ + struct audio_gain_config gain; /* gain to apply if applicable */ + union { + struct audio_port_config_device_ext device; /* device specific info */ + struct audio_port_config_mix_ext mix; /* mix specific info */ + struct audio_port_config_session_ext session; /* session specific info */ + } ext; +}; + + +/* max number of sampling rates in audio port */ +#define AUDIO_PORT_MAX_SAMPLING_RATES 16 +/* max number of channel masks in audio port */ +#define AUDIO_PORT_MAX_CHANNEL_MASKS 16 +/* max number of audio formats in audio port */ +#define AUDIO_PORT_MAX_FORMATS 16 +/* max number of gain controls in audio port */ +#define AUDIO_PORT_MAX_GAINS 16 + +/* extension for audio port structure when the audio port is a hardware device */ +struct audio_port_device_ext { + audio_module_handle_t hw_module; /* module the device is attached to */ + audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */ + char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; +}; + +/* Latency class of the audio mix */ +typedef enum { + AUDIO_LATENCY_LOW, + AUDIO_LATENCY_NORMAL, +} audio_mix_latency_class_t; + +/* extension for audio port structure when the audio port is a sub mix */ +struct audio_port_mix_ext { + audio_module_handle_t hw_module; /* module the stream is attached to */ + audio_io_handle_t handle; /* I/O handle of the input.output stream */ + audio_mix_latency_class_t latency_class; /* latency class */ + // other attributes: routing strategies +}; + +/* extension for audio port structure when the audio port is an audio session */ +struct audio_port_session_ext { + audio_session_t session; /* audio session */ +}; + + +struct audio_port { + audio_port_handle_t id; /* port unique ID */ + audio_port_role_t role; /* sink or source */ + audio_port_type_t type; /* device, mix ... */ + unsigned int num_sample_rates; /* number of sampling rates in following array */ + unsigned int sample_rates[AUDIO_PORT_MAX_SAMPLING_RATES]; + unsigned int num_channel_masks; /* number of channel masks in following array */ + audio_channel_mask_t channel_masks[AUDIO_PORT_MAX_CHANNEL_MASKS]; + unsigned int num_formats; /* number of formats in following array */ + audio_format_t formats[AUDIO_PORT_MAX_FORMATS]; + unsigned int num_gains; /* number of gains in following array */ + struct audio_gain gains[AUDIO_PORT_MAX_GAINS]; + struct audio_port_config active_config; /* current audio port configuration */ + union { + struct audio_port_device_ext device; + struct audio_port_mix_ext mix; + struct audio_port_session_ext session; + } ext; +}; + +/* An audio patch represents a connection between one or more source ports and + * one or more sink ports. Patches are connected and disconnected by audio policy manager or by + * applications via framework APIs. + * Each patch is identified by a handle at the interface used to create that patch. For instance, + * when a patch is created by the audio HAL, the HAL allocates and returns a handle. + * This handle is unique to a given audio HAL hardware module. + * But the same patch receives another system wide unique handle allocated by the framework. + * This unique handle is used for all transactions inside the framework. + */ +typedef int audio_patch_handle_t; +#define AUDIO_PATCH_HANDLE_NONE 0 + +#define AUDIO_PATCH_PORTS_MAX 16 + +struct audio_patch { + audio_patch_handle_t id; /* patch unique ID */ + unsigned int num_sources; /* number of sources in following array */ + struct audio_port_config sources[AUDIO_PATCH_PORTS_MAX]; + unsigned int num_sinks; /* number of sinks in following array */ + struct audio_port_config sinks[AUDIO_PATCH_PORTS_MAX]; +}; + + + +/* a HW synchronization source returned by the audio HAL */ +typedef uint32_t audio_hw_sync_t; + +/* an invalid HW synchronization source indicating an error */ +#define AUDIO_HW_SYNC_INVALID 0 + static inline bool audio_is_output_device(audio_devices_t device) { if (((device & AUDIO_DEVICE_BIT_IN) == 0) && @@ -468,8 +1030,17 @@ static inline bool audio_is_output_devices(audio_devices_t device) return (device & AUDIO_DEVICE_BIT_IN) == 0; } +static inline bool audio_is_a2dp_in_device(audio_devices_t device) +{ + if ((device & AUDIO_DEVICE_BIT_IN) != 0) { + device &= ~AUDIO_DEVICE_BIT_IN; + if ((popcount(device) == 1) && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP)) + return true; + } + return false; +} -static inline bool audio_is_a2dp_device(audio_devices_t device) +static inline bool audio_is_a2dp_out_device(audio_devices_t device) { if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_A2DP)) return true; @@ -477,6 +1048,12 @@ static inline bool audio_is_a2dp_device(audio_devices_t device) return false; } +// Deprecated - use audio_is_a2dp_out_device() instead +static inline bool audio_is_a2dp_device(audio_devices_t device) +{ + return audio_is_a2dp_out_device(device); +} + static inline bool audio_is_bluetooth_sco_device(audio_devices_t device) { if ((device & AUDIO_DEVICE_BIT_IN) == 0) { @@ -491,12 +1068,25 @@ static inline bool audio_is_bluetooth_sco_device(audio_devices_t device) return false; } +static inline bool audio_is_usb_out_device(audio_devices_t device) +{ + return ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_USB)); +} + +static inline bool audio_is_usb_in_device(audio_devices_t device) +{ + if ((device & AUDIO_DEVICE_BIT_IN) != 0) { + device &= ~AUDIO_DEVICE_BIT_IN; + if (popcount(device) == 1 && (device & AUDIO_DEVICE_IN_ALL_USB) != 0) + return true; + } + return false; +} + +/* OBSOLETE - use audio_is_usb_out_device() instead. */ static inline bool audio_is_usb_device(audio_devices_t device) { - if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_USB)) - return true; - else - return false; + return audio_is_usb_out_device(device); } static inline bool audio_is_remote_submix_device(audio_devices_t device) @@ -508,75 +1098,200 @@ static inline bool audio_is_remote_submix_device(audio_devices_t device) return false; } +/* Returns true if: + * representation is valid, and + * there is at least one channel bit set which _could_ correspond to an input channel, and + * there are no channel bits set which could _not_ correspond to an input channel. + * Otherwise returns false. + */ static inline bool audio_is_input_channel(audio_channel_mask_t channel) { - if ((channel & ~AUDIO_CHANNEL_IN_ALL) == 0) - return channel != 0; - else + uint32_t bits = audio_channel_mask_get_bits(channel); + switch (audio_channel_mask_get_representation(channel)) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + if (bits & ~AUDIO_CHANNEL_IN_ALL) { + bits = 0; + } + // fall through + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + return bits != 0; + default: return false; + } } +/* Returns true if: + * representation is valid, and + * there is at least one channel bit set which _could_ correspond to an output channel, and + * there are no channel bits set which could _not_ correspond to an output channel. + * Otherwise returns false. + */ static inline bool audio_is_output_channel(audio_channel_mask_t channel) { - if ((channel & ~AUDIO_CHANNEL_OUT_ALL) == 0) - return channel != 0; - else + uint32_t bits = audio_channel_mask_get_bits(channel); + switch (audio_channel_mask_get_representation(channel)) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + if (bits & ~AUDIO_CHANNEL_OUT_ALL) { + bits = 0; + } + // fall through + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + return bits != 0; + default: return false; + } +} + +/* Returns the number of channels from an input channel mask, + * used in the context of audio input or recording. + * If a channel bit is set which could _not_ correspond to an input channel, + * it is excluded from the count. + * Returns zero if the representation is invalid. + */ +static inline uint32_t audio_channel_count_from_in_mask(audio_channel_mask_t channel) +{ + uint32_t bits = audio_channel_mask_get_bits(channel); + switch (audio_channel_mask_get_representation(channel)) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + // TODO: We can now merge with from_out_mask and remove anding + bits &= AUDIO_CHANNEL_IN_ALL; + // fall through + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + return popcount(bits); + default: + return 0; + } } -/* Derive an output channel mask from a channel count. +/* Returns the number of channels from an output channel mask, + * used in the context of audio output or playback. + * If a channel bit is set which could _not_ correspond to an output channel, + * it is excluded from the count. + * Returns zero if the representation is invalid. + */ +static inline uint32_t audio_channel_count_from_out_mask(audio_channel_mask_t channel) +{ + uint32_t bits = audio_channel_mask_get_bits(channel); + switch (audio_channel_mask_get_representation(channel)) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + // TODO: We can now merge with from_in_mask and remove anding + bits &= AUDIO_CHANNEL_OUT_ALL; + // fall through + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + return popcount(bits); + default: + return 0; + } +} + +/* Derive an output channel mask for position assignment from a channel count. * This is to be used when the content channel mask is unknown. The 1, 2, 4, 5, 6, 7 and 8 channel * cases are mapped to the standard game/home-theater layouts, but note that 4 is mapped to quad, * and not stereo + FC + mono surround. A channel count of 3 is arbitrarily mapped to stereo + FC * for continuity with stereo. - * Returns the matching channel mask, or 0 if the number of channels exceeds that of the - * configurations for which a default channel mask is defined. + * Returns the matching channel mask, + * or AUDIO_CHANNEL_NONE if the channel count is zero, + * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the + * configurations for which a default output channel mask is defined. */ static inline audio_channel_mask_t audio_channel_out_mask_from_count(uint32_t channel_count) { - switch(channel_count) { + uint32_t bits; + switch (channel_count) { + case 0: + return AUDIO_CHANNEL_NONE; case 1: - return AUDIO_CHANNEL_OUT_MONO; + bits = AUDIO_CHANNEL_OUT_MONO; + break; case 2: - return AUDIO_CHANNEL_OUT_STEREO; + bits = AUDIO_CHANNEL_OUT_STEREO; + break; case 3: - return (AUDIO_CHANNEL_OUT_STEREO | AUDIO_CHANNEL_OUT_FRONT_CENTER); + bits = AUDIO_CHANNEL_OUT_STEREO | AUDIO_CHANNEL_OUT_FRONT_CENTER; + break; case 4: // 4.0 - return AUDIO_CHANNEL_OUT_QUAD; + bits = AUDIO_CHANNEL_OUT_QUAD; + break; case 5: // 5.0 - return (AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER); + bits = AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER; + break; case 6: // 5.1 - return AUDIO_CHANNEL_OUT_5POINT1; + bits = AUDIO_CHANNEL_OUT_5POINT1; + break; case 7: // 6.1 - return (AUDIO_CHANNEL_OUT_5POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER); + bits = AUDIO_CHANNEL_OUT_5POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER; + break; case 8: - return AUDIO_CHANNEL_OUT_7POINT1; + bits = AUDIO_CHANNEL_OUT_7POINT1; + break; default: - return 0; + return AUDIO_CHANNEL_INVALID; } + return audio_channel_mask_from_representation_and_bits( + AUDIO_CHANNEL_REPRESENTATION_POSITION, bits); } -/* Similar to above, but for input. Currently handles only mono and stereo. */ +/* Derive an input channel mask for position assignment from a channel count. + * Currently handles only mono and stereo. + * Returns the matching channel mask, + * or AUDIO_CHANNEL_NONE if the channel count is zero, + * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the + * configurations for which a default input channel mask is defined. + */ static inline audio_channel_mask_t audio_channel_in_mask_from_count(uint32_t channel_count) { + uint32_t bits; switch (channel_count) { + case 0: + return AUDIO_CHANNEL_NONE; case 1: - return AUDIO_CHANNEL_IN_MONO; + bits = AUDIO_CHANNEL_IN_MONO; + break; case 2: - return AUDIO_CHANNEL_IN_STEREO; + bits = AUDIO_CHANNEL_IN_STEREO; + break; default: - return 0; + return AUDIO_CHANNEL_INVALID; + } + return audio_channel_mask_from_representation_and_bits( + AUDIO_CHANNEL_REPRESENTATION_POSITION, bits); +} + +/* Derive a channel mask for index assignment from a channel count. + * Returns the matching channel mask, + * or AUDIO_CHANNEL_NONE if the channel count is zero, + * or AUDIO_CHANNEL_INVALID if the channel count exceeds AUDIO_CHANNEL_COUNT_MAX. + */ +static inline audio_channel_mask_t audio_channel_mask_for_index_assignment_from_count( + uint32_t channel_count) +{ + if (channel_count == 0) { + return AUDIO_CHANNEL_NONE; } + if (channel_count > AUDIO_CHANNEL_COUNT_MAX) { + return AUDIO_CHANNEL_INVALID; + } + uint32_t bits = (1 << channel_count) - 1; + return audio_channel_mask_from_representation_and_bits( + AUDIO_CHANNEL_REPRESENTATION_INDEX, bits); } static inline bool audio_is_valid_format(audio_format_t format) { switch (format & AUDIO_FORMAT_MAIN_MASK) { case AUDIO_FORMAT_PCM: - if (format != AUDIO_FORMAT_PCM_16_BIT && - format != AUDIO_FORMAT_PCM_8_BIT) { + switch (format) { + case AUDIO_FORMAT_PCM_16_BIT: + case AUDIO_FORMAT_PCM_8_BIT: + case AUDIO_FORMAT_PCM_32_BIT: + case AUDIO_FORMAT_PCM_8_24_BIT: + case AUDIO_FORMAT_PCM_FLOAT: + case AUDIO_FORMAT_PCM_24_BIT_PACKED: + return true; + default: return false; } + /* not reached */ case AUDIO_FORMAT_MP3: case AUDIO_FORMAT_AMR_NB: case AUDIO_FORMAT_AMR_WB: @@ -584,6 +1299,9 @@ static inline bool audio_is_valid_format(audio_format_t format) case AUDIO_FORMAT_HE_AAC_V1: case AUDIO_FORMAT_HE_AAC_V2: case AUDIO_FORMAT_VORBIS: + case AUDIO_FORMAT_OPUS: + case AUDIO_FORMAT_AC3: + case AUDIO_FORMAT_E_AC3: return true; default: return false; @@ -604,18 +1322,41 @@ static inline size_t audio_bytes_per_sample(audio_format_t format) case AUDIO_FORMAT_PCM_8_24_BIT: size = sizeof(int32_t); break; + case AUDIO_FORMAT_PCM_24_BIT_PACKED: + size = sizeof(uint8_t) * 3; + break; case AUDIO_FORMAT_PCM_16_BIT: size = sizeof(int16_t); break; case AUDIO_FORMAT_PCM_8_BIT: size = sizeof(uint8_t); break; + case AUDIO_FORMAT_PCM_FLOAT: + size = sizeof(float); + break; default: break; } return size; } +/* converts device address to string sent to audio HAL via set_parameters */ +static char *audio_device_address_to_parameter(audio_devices_t device, const char *address) +{ + const size_t kSize = AUDIO_DEVICE_MAX_ADDRESS_LEN + sizeof("a2dp_sink_address="); + char param[kSize]; + + if (device & AUDIO_DEVICE_OUT_ALL_A2DP) + snprintf(param, kSize, "%s=%s", "a2dp_sink_address", address); + else if (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX) + snprintf(param, kSize, "%s=%s", "mix", address); + else + snprintf(param, kSize, "%s", address); + + return strdup(param); +} + + __END_DECLS #endif // ANDROID_AUDIO_CORE_H |