diff options
| -rw-r--r-- | audio/include/system/audio.h | 172 |
1 files changed, 162 insertions, 10 deletions
diff --git a/audio/include/system/audio.h b/audio/include/system/audio.h index 3c8aa1a..676eed3 100644 --- a/audio/include/system/audio.h +++ b/audio/include/system/audio.h @@ -28,6 +28,19 @@ __BEGIN_DECLS +/* The macro FCC_2 highlights places where there are 2-channel assumptions. + * This is typically due to legacy implementation of stereo input or output. + * Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. + * Do not change this value. + */ +#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 + +/* The macro FCC_8 highlights places where there are 8-channel assumptions. + * This is typically due to audio mixer and resampler limitations. + * Do not change this value without verifying all locations that use it. + */ +#define FCC_8 8 // FCC_8 = Fixed Channel Count 8 + /* The enums were moved here mostly from * frameworks/base/include/media/AudioSystem.h */ @@ -64,6 +77,8 @@ typedef enum { AUDIO_STREAM_REROUTING = 11, /* For dynamic policy output mixes */ AUDIO_STREAM_PATCH = 12, /* For internal audio flinger tracks. Fixed volume */ AUDIO_STREAM_PUBLIC_CNT = AUDIO_STREAM_TTS + 1, + AUDIO_STREAM_FOR_POLICY_CNT = AUDIO_STREAM_PATCH, /* number of streams considered by + audio policy for volume and routing */ AUDIO_STREAM_CNT = AUDIO_STREAM_PATCH + 1, } audio_stream_type_t; @@ -120,6 +135,7 @@ enum { AUDIO_FLAG_HW_HOTWORD = 0x20, AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY = 0x40, AUDIO_FLAG_BYPASS_MUTE = 0x80, + AUDIO_FLAG_LOW_LATENCY = 0x100, }; /* Do not change these values without updating their counterparts @@ -184,11 +200,40 @@ typedef enum { * after all uses have been updated from 0 to the appropriate symbol, and have been tested. */ AUDIO_SESSION_ALLOCATE = 0, + + /* For use with AudioRecord::start(), this indicates no trigger session. + * It is also used with output tracks and patch tracks, which never have a session. + */ + AUDIO_SESSION_NONE = 0, } audio_session_t; -/* a unique ID allocated by AudioFlinger for use as a audio_io_handle_t or audio_session_t */ +/* a unique ID allocated by AudioFlinger for use as an audio_io_handle_t, audio_session_t, + * effect ID (int), audio_module_handle_t, and audio_patch_handle_t. + * Audio port IDs (audio_port_handle_t) are allocated by AudioPolicy + * in a different namespace than AudioFlinger unique IDs. + */ typedef int audio_unique_id_t; +/* Possible uses for an audio_unique_id_t */ +typedef enum { + AUDIO_UNIQUE_ID_USE_UNSPECIFIED = 0, + AUDIO_UNIQUE_ID_USE_SESSION = 1, // for allocated sessions, not special AUDIO_SESSION_* + AUDIO_UNIQUE_ID_USE_MODULE = 2, + AUDIO_UNIQUE_ID_USE_EFFECT = 3, + AUDIO_UNIQUE_ID_USE_PATCH = 4, + AUDIO_UNIQUE_ID_USE_OUTPUT = 5, + AUDIO_UNIQUE_ID_USE_INPUT = 6, + // 7 is available + AUDIO_UNIQUE_ID_USE_MAX = 8, // must be a power-of-two + AUDIO_UNIQUE_ID_USE_MASK = AUDIO_UNIQUE_ID_USE_MAX - 1 +} audio_unique_id_use_t; + +/* Return the use of an audio_unique_id_t */ +static inline audio_unique_id_use_t audio_unique_id_get_use(audio_unique_id_t id) +{ + return (audio_unique_id_use_t) (id & AUDIO_UNIQUE_ID_USE_MASK); +} + #define AUDIO_UNIQUE_ID_ALLOCATE AUDIO_SESSION_ALLOCATE /* Audio sub formats (see enum audio_format). */ @@ -265,6 +310,8 @@ typedef enum { AUDIO_FORMAT_E_AC3 = 0x0A000000UL, AUDIO_FORMAT_DTS = 0x0B000000UL, AUDIO_FORMAT_DTS_HD = 0x0C000000UL, + // IEC61937 is encoded audio wrapped in 16-bit PCM. + AUDIO_FORMAT_IEC61937 = 0x0D000000UL, AUDIO_FORMAT_EVRC = 0x10000000UL, AUDIO_FORMAT_QCELP = 0x11000000UL, AUDIO_FORMAT_WMA = 0x12000000UL, @@ -283,6 +330,10 @@ typedef enum { AUDIO_FORMAT_DSD = 0x1F000000UL, AUDIO_FORMAT_SBC = 0x20000000UL, AUDIO_FORMAT_APTX = 0x21000000UL, + AUDIO_FORMAT_APTX_HD = 0x22000000UL, + AUDIO_FORMAT_AAC_LATM = 0x23000000UL, + + AUDIO_FORMAT_DOLBY_TRUEHD = 0x0E000000UL, AUDIO_FORMAT_MAIN_MASK = 0xFF000000UL, AUDIO_FORMAT_SUB_MASK = 0x00FFFFFFUL, @@ -341,6 +392,12 @@ typedef enum { AUDIO_FORMAT_AAC_SUB_HE_V2), AUDIO_FORMAT_AAC_ADTS_ELD = (AUDIO_FORMAT_AAC_ADTS | AUDIO_FORMAT_AAC_SUB_ELD), + AUDIO_FORMAT_AAC_LATM_LC = (AUDIO_FORMAT_AAC_LATM |\ + AUDIO_FORMAT_AAC_SUB_LC), + AUDIO_FORMAT_AAC_LATM_HE_V1 = (AUDIO_FORMAT_AAC_LATM |\ + AUDIO_FORMAT_AAC_SUB_HE_V1), + AUDIO_FORMAT_AAC_LATM_HE_V2 = (AUDIO_FORMAT_AAC_LATM |\ + AUDIO_FORMAT_AAC_SUB_HE_V2), /*Offload PCM formats*/ AUDIO_FORMAT_PCM_16_BIT_OFFLOAD = (AUDIO_FORMAT_PCM_OFFLOAD | AUDIO_FORMAT_PCM_SUB_16_BIT), @@ -687,7 +744,9 @@ enum { /* limited-output speaker device for acoustic safety */ AUDIO_DEVICE_OUT_SPEAKER_SAFE = 0x400000, AUDIO_DEVICE_OUT_IP = 0x800000, - AUDIO_DEVICE_OUT_PROXY = 0x1000000, + /* audio bus implemented by the audio system (e.g an MOST stereo channel) */ + AUDIO_DEVICE_OUT_BUS = 0x1000000, + AUDIO_DEVICE_OUT_PROXY = 0x2000000, AUDIO_DEVICE_OUT_DEFAULT = AUDIO_DEVICE_BIT_DEFAULT, AUDIO_DEVICE_OUT_ALL = (AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER | @@ -713,6 +772,7 @@ enum { AUDIO_DEVICE_OUT_AUX_LINE | AUDIO_DEVICE_OUT_SPEAKER_SAFE | AUDIO_DEVICE_OUT_IP | + AUDIO_DEVICE_OUT_BUS | AUDIO_DEVICE_OUT_PROXY | AUDIO_DEVICE_OUT_DEFAULT), AUDIO_DEVICE_OUT_ALL_A2DP = (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | @@ -751,7 +811,9 @@ enum { AUDIO_DEVICE_IN_BLUETOOTH_A2DP = AUDIO_DEVICE_BIT_IN | 0x20000, AUDIO_DEVICE_IN_LOOPBACK = AUDIO_DEVICE_BIT_IN | 0x40000, AUDIO_DEVICE_IN_IP = AUDIO_DEVICE_BIT_IN | 0x80000, - AUDIO_DEVICE_IN_PROXY = AUDIO_DEVICE_BIT_IN | 0x1000000, + /* audio bus implemented by the audio system (e.g an MOST stereo channel) */ + AUDIO_DEVICE_IN_BUS = AUDIO_DEVICE_BIT_IN | 0x100000, + AUDIO_DEVICE_IN_PROXY = AUDIO_DEVICE_BIT_IN | 0x2000000, AUDIO_DEVICE_IN_DEFAULT = AUDIO_DEVICE_BIT_IN | AUDIO_DEVICE_BIT_DEFAULT, AUDIO_DEVICE_IN_ALL = (AUDIO_DEVICE_IN_COMMUNICATION | @@ -775,6 +837,7 @@ enum { AUDIO_DEVICE_IN_LOOPBACK | AUDIO_DEVICE_IN_PROXY | AUDIO_DEVICE_IN_IP | + AUDIO_DEVICE_IN_BUS | 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 | @@ -833,7 +896,6 @@ typedef enum { AUDIO_INPUT_FLAG_HW_HOTWORD = 0x2, // prefer an input that captures from hw hotword source AUDIO_INPUT_FLAG_RAW = 0x4, // minimize signal processing AUDIO_INPUT_FLAG_SYNC = 0x8, // synchronize I/O streams - AUDIO_INPUT_FLAG_TIMESTAMP = 0x10, // timestamp metadata mode } audio_input_flags_t; @@ -858,6 +920,45 @@ typedef struct { audio_usage_t usage; } audio_offload_info_t; +/* Information about BT SBC encoder configuration + * This data is used between audio HAL module and + * BT IPC library to configure DSP encoder + */ +typedef struct { + uint32_t subband; /* 4, 8 */ + uint32_t blk_len; /* 4, 8, 12, 16 */ + uint16_t sampling_rate; /*44.1khz,48khz*/ + uint8_t channels; /*0(Mono),1(Dual_mono),2(Stereo),3(JS)*/ + uint8_t alloc; /*0(Loudness),1(SNR)*/ + uint8_t min_bitpool; /* 2 */ + uint8_t max_bitpool; /*53(44.1khz),51 (48khz) */ + uint32_t bitrate; /* 320kbps to 512kbps */ +} audio_sbc_encoder_config; + + +/* Information about BT APTX encoder configuration + * This data is used between audio HAL module and + * BT IPC library to configure DSP encoder + */ +typedef struct { + uint16_t sampling_rate; + uint8_t channels; + uint32_t bitrate; +} audio_aptx_encoder_config; + + +/* Information about BT AAC encoder configuration + * This data is used between audio HAL module and + * BT IPC library to configure DSP encoder + */ +typedef struct { + uint32_t enc_mode; /* LC, SBR, PS */ + uint16_t format_flag; /* RAW, ADTS */ + uint16_t channels; /* 1-Mono, 2-Stereo */ + uint32_t sampling_rate; + uint32_t bitrate; +} audio_aac_encoder_config; + #define AUDIO_MAKE_OFFLOAD_INFO_VERSION(maj,min) \ ((((maj) & 0xff) << 8) | ((min) & 0xff)) @@ -912,9 +1013,24 @@ static const audio_config_t AUDIO_CONFIG_INITIALIZER = { frame_count: 0, }; +struct audio_config_base { + uint32_t sample_rate; + audio_channel_mask_t channel_mask; + audio_format_t format; +}; + +typedef struct audio_config_base audio_config_base_t; + +static const audio_config_base_t AUDIO_CONFIG_BASE_INITIALIZER = { + sample_rate: 0, + channel_mask: AUDIO_CHANNEL_NONE, + format: AUDIO_FORMAT_DEFAULT +}; /* audio hw module handle functions or structures referencing a module */ -typedef int audio_module_handle_t; +typedef enum { + AUDIO_MODULE_HANDLE_NONE = 0, +} audio_module_handle_t; /****************************** * Volume control @@ -1121,8 +1237,9 @@ struct audio_port { * 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 +typedef enum { + AUDIO_PATCH_HANDLE_NONE = 0, +} audio_patch_handle_t; #define AUDIO_PATCH_PORTS_MAX 16 @@ -1444,6 +1561,7 @@ static inline bool audio_is_valid_format(audio_format_t format) case AUDIO_FORMAT_AMR_WB: case AUDIO_FORMAT_AAC: case AUDIO_FORMAT_AAC_ADTS: + case AUDIO_FORMAT_AAC_LATM: case AUDIO_FORMAT_HE_AAC_V1: case AUDIO_FORMAT_HE_AAC_V2: case AUDIO_FORMAT_VORBIS: @@ -1452,6 +1570,7 @@ static inline bool audio_is_valid_format(audio_format_t format) case AUDIO_FORMAT_E_AC3: case AUDIO_FORMAT_DTS: case AUDIO_FORMAT_DTS_HD: + case AUDIO_FORMAT_IEC61937: case AUDIO_FORMAT_QCELP: case AUDIO_FORMAT_EVRC: case AUDIO_FORMAT_EVRCB: @@ -1466,21 +1585,51 @@ static inline bool audio_is_valid_format(audio_format_t format) case AUDIO_FORMAT_WMA: case AUDIO_FORMAT_WMA_PRO: case AUDIO_FORMAT_APTX: + case AUDIO_FORMAT_DSD: return true; case AUDIO_FORMAT_PCM_OFFLOAD: if (format != AUDIO_FORMAT_PCM_16_BIT_OFFLOAD && format != AUDIO_FORMAT_PCM_24_BIT_OFFLOAD) { return false; } + case AUDIO_FORMAT_DOLBY_TRUEHD: return true; default: return false; } } +/** + * Extract the primary format, eg. PCM, AC3, etc. + */ +static inline audio_format_t audio_get_main_format(audio_format_t format) +{ + return (audio_format_t)(format & AUDIO_FORMAT_MAIN_MASK); +} + +/** + * Is the data plain PCM samples that can be scaled and mixed? + */ static inline bool audio_is_linear_pcm(audio_format_t format) { - return ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM); + return (audio_get_main_format(format) == AUDIO_FORMAT_PCM); +} + +/** + * For this format, is the number of PCM audio frames directly proportional + * to the number of data bytes? + * + * In other words, is the format transported as PCM audio samples, + * but not necessarily scalable or mixable. + * This returns true for real PCM, but also for AUDIO_FORMAT_IEC61937, + * which is transported as 16 bit PCM audio, but where the encoded data + * cannot be mixed or scaled. + */ +static inline bool audio_has_proportional_frames(audio_format_t format) +{ + audio_format_t mainFormat = audio_get_main_format(format); + return (mainFormat == AUDIO_FORMAT_PCM + || mainFormat == AUDIO_FORMAT_IEC61937); } static inline size_t audio_bytes_per_sample(audio_format_t format) @@ -1497,6 +1646,7 @@ static inline size_t audio_bytes_per_sample(audio_format_t format) size = sizeof(uint8_t) * 3; break; case AUDIO_FORMAT_PCM_16_BIT: + case AUDIO_FORMAT_IEC61937: case AUDIO_FORMAT_PCM_16_BIT_OFFLOAD: size = sizeof(int16_t); break; @@ -1534,14 +1684,16 @@ static inline bool audio_device_is_digital(audio_devices_t device) { return (~AUDIO_DEVICE_BIT_IN & device & (AUDIO_DEVICE_IN_ALL_USB | AUDIO_DEVICE_IN_HDMI | AUDIO_DEVICE_IN_SPDIF | - AUDIO_DEVICE_IN_IP)) != 0; + AUDIO_DEVICE_IN_IP | + AUDIO_DEVICE_IN_BUS)) != 0; } else { // output return (device & (AUDIO_DEVICE_OUT_ALL_USB | AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_HDMI_ARC | AUDIO_DEVICE_OUT_SPDIF | - AUDIO_DEVICE_OUT_IP)) != 0; + AUDIO_DEVICE_OUT_IP | + AUDIO_DEVICE_OUT_BUS)) != 0; } } |