From 5f8de423f190bbb79a62f804151bc24824fa32d8 Mon Sep 17 00:00:00 2001 From: "Matt A. Tobin" Date: Fri, 2 Feb 2018 04:16:08 -0500 Subject: Add m-esr52 at 52.6.0 --- dom/system/gonk/android_audio/AudioSystem.h | 1134 ++++++++++++++++++++ dom/system/gonk/android_audio/AudioTrack.h | 489 +++++++++ dom/system/gonk/android_audio/EffectApi.h | 798 ++++++++++++++ dom/system/gonk/android_audio/IAudioFlinger.h | 184 ++++ .../gonk/android_audio/IAudioFlingerClient.h | 55 + dom/system/gonk/android_audio/IAudioRecord.h | 68 ++ dom/system/gonk/android_audio/IAudioTrack.h | 89 ++ dom/system/gonk/android_audio/IEffect.h | 60 ++ dom/system/gonk/android_audio/IEffectClient.h | 54 + 9 files changed, 2931 insertions(+) create mode 100644 dom/system/gonk/android_audio/AudioSystem.h create mode 100644 dom/system/gonk/android_audio/AudioTrack.h create mode 100644 dom/system/gonk/android_audio/EffectApi.h create mode 100644 dom/system/gonk/android_audio/IAudioFlinger.h create mode 100644 dom/system/gonk/android_audio/IAudioFlingerClient.h create mode 100644 dom/system/gonk/android_audio/IAudioRecord.h create mode 100644 dom/system/gonk/android_audio/IAudioTrack.h create mode 100644 dom/system/gonk/android_audio/IEffect.h create mode 100644 dom/system/gonk/android_audio/IEffectClient.h (limited to 'dom/system/gonk/android_audio') diff --git a/dom/system/gonk/android_audio/AudioSystem.h b/dom/system/gonk/android_audio/AudioSystem.h new file mode 100644 index 000000000..d5841eaaa --- /dev/null +++ b/dom/system/gonk/android_audio/AudioSystem.h @@ -0,0 +1,1134 @@ +/* + * Copyright (C) 2008 The Android Open Source Project + * + * 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 ANDROID_AUDIOSYSTEM_H_ +#define ANDROID_AUDIOSYSTEM_H_ + +#pragma GCC visibility push(default) + +#include +#include +#include "IAudioFlinger.h" + +#ifndef VANILLA_ANDROID +/* request to open a direct output with get_output() (by opposition to + * sharing an output with other AudioTracks) + */ +typedef enum { + AUDIO_POLICY_OUTPUT_FLAG_INDIRECT = 0x0, + AUDIO_POLICY_OUTPUT_FLAG_DIRECT = 0x1 +} audio_policy_output_flags_t; + +/* device categories used for audio_policy->set_force_use() */ +typedef enum { + AUDIO_POLICY_FORCE_NONE, + AUDIO_POLICY_FORCE_SPEAKER, + AUDIO_POLICY_FORCE_HEADPHONES, + AUDIO_POLICY_FORCE_BT_SCO, + AUDIO_POLICY_FORCE_BT_A2DP, + AUDIO_POLICY_FORCE_WIRED_ACCESSORY, + AUDIO_POLICY_FORCE_BT_CAR_DOCK, + AUDIO_POLICY_FORCE_BT_DESK_DOCK, + AUDIO_POLICY_FORCE_ANALOG_DOCK, + AUDIO_POLICY_FORCE_DIGITAL_DOCK, + AUDIO_POLICY_FORCE_NO_BT_A2DP, + AUDIO_POLICY_FORCE_CFG_CNT, + AUDIO_POLICY_FORCE_CFG_MAX = AUDIO_POLICY_FORCE_CFG_CNT - 1, + + AUDIO_POLICY_FORCE_DEFAULT = AUDIO_POLICY_FORCE_NONE, +} audio_policy_forced_cfg_t; + +/* usages used for audio_policy->set_force_use() */ +typedef enum { + AUDIO_POLICY_FORCE_FOR_COMMUNICATION, + AUDIO_POLICY_FORCE_FOR_MEDIA, + AUDIO_POLICY_FORCE_FOR_RECORD, + AUDIO_POLICY_FORCE_FOR_DOCK, + + AUDIO_POLICY_FORCE_USE_CNT, + AUDIO_POLICY_FORCE_USE_MAX = AUDIO_POLICY_FORCE_USE_CNT - 1, +} audio_policy_force_use_t; + +typedef enum { + AUDIO_STREAM_DEFAULT = -1, + AUDIO_STREAM_VOICE_CALL = 0, + AUDIO_STREAM_SYSTEM = 1, + AUDIO_STREAM_RING = 2, + AUDIO_STREAM_MUSIC = 3, + 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_DTMF = 8, + AUDIO_STREAM_TTS = 9, +#if ANDROID_VERSION < 19 + AUDIO_STREAM_FM = 10, +#endif + + AUDIO_STREAM_CNT, + AUDIO_STREAM_MAX = AUDIO_STREAM_CNT - 1, +} audio_stream_type_t; + +/* PCM sub formats */ +typedef enum { + 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_fmt_t; + +/* Audio format consists in 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 + * indicates options and parameters for each format. The sub format is mainly + * used for record to indicate for instance the requested bitrate or profile. + * It can also be used for certain formats to give informations not present in + * the encoded audio stream (e.g. octet alignement for AMR). + */ +typedef enum { + AUDIO_FORMAT_INVALID = 0xFFFFFFFFUL, + AUDIO_FORMAT_DEFAULT = 0, + AUDIO_FORMAT_PCM = 0x00000000UL, /* DO NOT CHANGE */ + AUDIO_FORMAT_MP3 = 0x01000000UL, + 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_VORBIS = 0x07000000UL, + AUDIO_FORMAT_MAIN_MASK = 0xFF000000UL, + AUDIO_FORMAT_SUB_MASK = 0x00FFFFFFUL, + + /* Aliases */ + AUDIO_FORMAT_PCM_16_BIT = (AUDIO_FORMAT_PCM | + AUDIO_FORMAT_PCM_SUB_16_BIT), + 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_t; + +typedef enum { + /* output channels */ + AUDIO_CHANNEL_OUT_FRONT_LEFT = 0x1, + AUDIO_CHANNEL_OUT_FRONT_RIGHT = 0x2, + AUDIO_CHANNEL_OUT_FRONT_CENTER = 0x4, + AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 0x8, + AUDIO_CHANNEL_OUT_BACK_LEFT = 0x10, + AUDIO_CHANNEL_OUT_BACK_RIGHT = 0x20, + AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40, + AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80, + AUDIO_CHANNEL_OUT_BACK_CENTER = 0x100, + AUDIO_CHANNEL_OUT_SIDE_LEFT = 0x200, + AUDIO_CHANNEL_OUT_SIDE_RIGHT = 0x400, + AUDIO_CHANNEL_OUT_TOP_CENTER = 0x800, + AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 0x1000, + AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 0x2000, + AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 0x4000, + AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 0x8000, + AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 0x10000, + AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 0x20000, + + AUDIO_CHANNEL_OUT_MONO = AUDIO_CHANNEL_OUT_FRONT_LEFT, + AUDIO_CHANNEL_OUT_STEREO = (AUDIO_CHANNEL_OUT_FRONT_LEFT | + AUDIO_CHANNEL_OUT_FRONT_RIGHT), + AUDIO_CHANNEL_OUT_QUAD = (AUDIO_CHANNEL_OUT_FRONT_LEFT | + 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_FRONT_RIGHT | + AUDIO_CHANNEL_OUT_FRONT_CENTER | + AUDIO_CHANNEL_OUT_BACK_CENTER), + 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), + // 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 | + AUDIO_CHANNEL_OUT_FRONT_CENTER | + AUDIO_CHANNEL_OUT_LOW_FREQUENCY | + AUDIO_CHANNEL_OUT_BACK_LEFT | + AUDIO_CHANNEL_OUT_BACK_RIGHT | + AUDIO_CHANNEL_OUT_SIDE_LEFT | + AUDIO_CHANNEL_OUT_SIDE_RIGHT), + AUDIO_CHANNEL_OUT_ALL = (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_FRONT_LEFT_OF_CENTER | + AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | + AUDIO_CHANNEL_OUT_BACK_CENTER| + AUDIO_CHANNEL_OUT_SIDE_LEFT| + AUDIO_CHANNEL_OUT_SIDE_RIGHT| + AUDIO_CHANNEL_OUT_TOP_CENTER| + AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT| + AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER| + AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT| + AUDIO_CHANNEL_OUT_TOP_BACK_LEFT| + AUDIO_CHANNEL_OUT_TOP_BACK_CENTER| + AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT), + + /* input channels */ + AUDIO_CHANNEL_IN_LEFT = 0x4, + AUDIO_CHANNEL_IN_RIGHT = 0x8, + AUDIO_CHANNEL_IN_FRONT = 0x10, + AUDIO_CHANNEL_IN_BACK = 0x20, + AUDIO_CHANNEL_IN_LEFT_PROCESSED = 0x40, + AUDIO_CHANNEL_IN_RIGHT_PROCESSED = 0x80, + AUDIO_CHANNEL_IN_FRONT_PROCESSED = 0x100, + AUDIO_CHANNEL_IN_BACK_PROCESSED = 0x200, + AUDIO_CHANNEL_IN_PRESSURE = 0x400, + AUDIO_CHANNEL_IN_X_AXIS = 0x800, + AUDIO_CHANNEL_IN_Y_AXIS = 0x1000, + AUDIO_CHANNEL_IN_Z_AXIS = 0x2000, + AUDIO_CHANNEL_IN_VOICE_UPLINK = 0x4000, + AUDIO_CHANNEL_IN_VOICE_DNLINK = 0x8000, + + AUDIO_CHANNEL_IN_MONO = AUDIO_CHANNEL_IN_FRONT, + AUDIO_CHANNEL_IN_STEREO = (AUDIO_CHANNEL_IN_LEFT | AUDIO_CHANNEL_IN_RIGHT), + AUDIO_CHANNEL_IN_ALL = (AUDIO_CHANNEL_IN_LEFT | + AUDIO_CHANNEL_IN_RIGHT | + AUDIO_CHANNEL_IN_FRONT | + AUDIO_CHANNEL_IN_BACK| + AUDIO_CHANNEL_IN_LEFT_PROCESSED | + AUDIO_CHANNEL_IN_RIGHT_PROCESSED | + AUDIO_CHANNEL_IN_FRONT_PROCESSED | + AUDIO_CHANNEL_IN_BACK_PROCESSED| + AUDIO_CHANNEL_IN_PRESSURE | + AUDIO_CHANNEL_IN_X_AXIS | + AUDIO_CHANNEL_IN_Y_AXIS | + AUDIO_CHANNEL_IN_Z_AXIS | + AUDIO_CHANNEL_IN_VOICE_UPLINK | + AUDIO_CHANNEL_IN_VOICE_DNLINK), +} audio_channels_t; + +#if ANDROID_VERSION >= 17 +typedef enum { + AUDIO_MODE_INVALID = -2, + AUDIO_MODE_CURRENT = -1, + AUDIO_MODE_NORMAL = 0, + AUDIO_MODE_RINGTONE = 1, + AUDIO_MODE_IN_CALL = 2, + AUDIO_MODE_IN_COMMUNICATION = 3, + + AUDIO_MODE_CNT, + AUDIO_MODE_MAX = AUDIO_MODE_CNT - 1, +} audio_mode_t; +#endif +#endif + +#if ANDROID_VERSION < 17 +typedef enum { + AUDIO_DEVICE_NONE = 0x0, + /* output devices */ + AUDIO_DEVICE_OUT_EARPIECE = 0x1, + AUDIO_DEVICE_OUT_SPEAKER = 0x2, + AUDIO_DEVICE_OUT_WIRED_HEADSET = 0x4, + AUDIO_DEVICE_OUT_WIRED_HEADPHONE = 0x8, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO = 0x10, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP = 0x80, + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, + AUDIO_DEVICE_OUT_AUX_DIGITAL = 0x400, + AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800, + AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000, + AUDIO_DEVICE_OUT_FM = 0x2000, + AUDIO_DEVICE_OUT_ANC_HEADSET = 0x4000, + AUDIO_DEVICE_OUT_ANC_HEADPHONE = 0x8000, + AUDIO_DEVICE_OUT_FM_TX = 0x10000, + AUDIO_DEVICE_OUT_DIRECTOUTPUT = 0x20000, + AUDIO_DEVICE_OUT_PROXY = 0x40000, + AUDIO_DEVICE_OUT_DEFAULT = 0x80000, + AUDIO_DEVICE_OUT_ALL = (AUDIO_DEVICE_OUT_EARPIECE | + AUDIO_DEVICE_OUT_SPEAKER | + AUDIO_DEVICE_OUT_WIRED_HEADSET | + AUDIO_DEVICE_OUT_WIRED_HEADPHONE | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT | + 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_ANLG_DOCK_HEADSET | + AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET | + AUDIO_DEVICE_OUT_FM | + AUDIO_DEVICE_OUT_ANC_HEADSET | + AUDIO_DEVICE_OUT_ANC_HEADPHONE | + AUDIO_DEVICE_OUT_FM_TX | + AUDIO_DEVICE_OUT_DIRECTOUTPUT | + AUDIO_DEVICE_OUT_PROXY | + AUDIO_DEVICE_OUT_DEFAULT), + AUDIO_DEVICE_OUT_ALL_A2DP = (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), + AUDIO_DEVICE_OUT_ALL_SCO = (AUDIO_DEVICE_OUT_BLUETOOTH_SCO | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), + /* input devices */ + AUDIO_DEVICE_IN_COMMUNICATION = 0x100000, + AUDIO_DEVICE_IN_AMBIENT = 0x200000, + AUDIO_DEVICE_IN_BUILTIN_MIC = 0x400000, + AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x800000, + AUDIO_DEVICE_IN_WIRED_HEADSET = 0x1000000, + AUDIO_DEVICE_IN_AUX_DIGITAL = 0x2000000, + AUDIO_DEVICE_IN_VOICE_CALL = 0x4000000, + AUDIO_DEVICE_IN_BACK_MIC = 0x8000000, + AUDIO_DEVICE_IN_ANC_HEADSET = 0x10000000, + AUDIO_DEVICE_IN_FM_RX = 0x20000000, + AUDIO_DEVICE_IN_FM_RX_A2DP = 0x40000000, + AUDIO_DEVICE_IN_DEFAULT = 0x80000000, + + AUDIO_DEVICE_IN_ALL = (AUDIO_DEVICE_IN_COMMUNICATION | + AUDIO_DEVICE_IN_AMBIENT | + 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_BACK_MIC | + AUDIO_DEVICE_IN_ANC_HEADSET | + AUDIO_DEVICE_IN_FM_RX | + AUDIO_DEVICE_IN_FM_RX_A2DP | + AUDIO_DEVICE_IN_DEFAULT), + AUDIO_DEVICE_IN_ALL_SCO = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, +} audio_devices_t; +#elif ANDROID_VERSION < 21 +enum { + AUDIO_DEVICE_NONE = 0x0, + /* reserved bits */ + AUDIO_DEVICE_BIT_IN = 0x80000000, + AUDIO_DEVICE_BIT_DEFAULT = 0x40000000, + /* output devices */ + AUDIO_DEVICE_OUT_EARPIECE = 0x1, + AUDIO_DEVICE_OUT_SPEAKER = 0x2, + AUDIO_DEVICE_OUT_WIRED_HEADSET = 0x4, + AUDIO_DEVICE_OUT_WIRED_HEADPHONE = 0x8, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO = 0x10, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP = 0x80, + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, + AUDIO_DEVICE_OUT_AUX_DIGITAL = 0x400, + AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET = 0x800, + AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000, + AUDIO_DEVICE_OUT_USB_ACCESSORY = 0x2000, + AUDIO_DEVICE_OUT_USB_DEVICE = 0x4000, + AUDIO_DEVICE_OUT_REMOTE_SUBMIX = 0x8000, + AUDIO_DEVICE_OUT_ANC_HEADSET = 0x10000, + AUDIO_DEVICE_OUT_ANC_HEADPHONE = 0x20000, + AUDIO_DEVICE_OUT_PROXY = 0x40000, + AUDIO_DEVICE_OUT_FM = 0x80000, + AUDIO_DEVICE_OUT_FM_TX = 0x100000, + AUDIO_DEVICE_OUT_DEFAULT = AUDIO_DEVICE_BIT_DEFAULT, + AUDIO_DEVICE_OUT_ALL = (AUDIO_DEVICE_OUT_EARPIECE | + AUDIO_DEVICE_OUT_SPEAKER | + AUDIO_DEVICE_OUT_WIRED_HEADSET | + AUDIO_DEVICE_OUT_WIRED_HEADPHONE | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT | + 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_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_ANC_HEADSET | + AUDIO_DEVICE_OUT_ANC_HEADPHONE | + AUDIO_DEVICE_OUT_PROXY | + AUDIO_DEVICE_OUT_FM | + AUDIO_DEVICE_OUT_FM_TX | + AUDIO_DEVICE_OUT_DEFAULT), + AUDIO_DEVICE_OUT_ALL_A2DP = (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), + AUDIO_DEVICE_OUT_ALL_SCO = (AUDIO_DEVICE_OUT_BLUETOOTH_SCO | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), + AUDIO_DEVICE_OUT_ALL_USB = (AUDIO_DEVICE_OUT_USB_ACCESSORY | + AUDIO_DEVICE_OUT_USB_DEVICE), + + /* input devices */ + AUDIO_DEVICE_IN_COMMUNICATION = AUDIO_DEVICE_BIT_IN | 0x1, + AUDIO_DEVICE_IN_AMBIENT = AUDIO_DEVICE_BIT_IN | 0x2, + AUDIO_DEVICE_IN_BUILTIN_MIC = AUDIO_DEVICE_BIT_IN | 0x4, + 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_VOICE_CALL = AUDIO_DEVICE_BIT_IN | 0x40, + 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, + AUDIO_DEVICE_IN_ANC_HEADSET = AUDIO_DEVICE_BIT_IN | 0x2000, + AUDIO_DEVICE_IN_PROXY = AUDIO_DEVICE_BIT_IN | 0x4000, + AUDIO_DEVICE_IN_FM_RX = AUDIO_DEVICE_BIT_IN | 0x8000, + AUDIO_DEVICE_IN_FM_RX_A2DP = AUDIO_DEVICE_BIT_IN | 0x10000, + AUDIO_DEVICE_IN_DEFAULT = AUDIO_DEVICE_BIT_IN | AUDIO_DEVICE_BIT_DEFAULT, + + AUDIO_DEVICE_IN_ALL = (AUDIO_DEVICE_IN_COMMUNICATION | + AUDIO_DEVICE_IN_AMBIENT | + 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_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_ANC_HEADSET | + AUDIO_DEVICE_IN_FM_RX | + AUDIO_DEVICE_IN_FM_RX_A2DP | + AUDIO_DEVICE_IN_PROXY | + AUDIO_DEVICE_IN_DEFAULT), + AUDIO_DEVICE_IN_ALL_SCO = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, +}; + +typedef uint32_t audio_devices_t; +#else +enum { + AUDIO_DEVICE_NONE = 0x0, + /* reserved bits */ + AUDIO_DEVICE_BIT_IN = 0x80000000, + AUDIO_DEVICE_BIT_DEFAULT = 0x40000000, + /* output devices */ + AUDIO_DEVICE_OUT_EARPIECE = 0x1, + AUDIO_DEVICE_OUT_SPEAKER = 0x2, + AUDIO_DEVICE_OUT_WIRED_HEADSET = 0x4, + AUDIO_DEVICE_OUT_WIRED_HEADPHONE = 0x8, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO = 0x10, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP = 0x80, + 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, + /* limited-output speaker device for acoustic safety */ + AUDIO_DEVICE_OUT_SPEAKER_SAFE = 0x400000, + AUDIO_DEVICE_OUT_DEFAULT = AUDIO_DEVICE_BIT_DEFAULT, + AUDIO_DEVICE_OUT_ALL = (AUDIO_DEVICE_OUT_EARPIECE | + AUDIO_DEVICE_OUT_SPEAKER | + AUDIO_DEVICE_OUT_WIRED_HEADSET | + AUDIO_DEVICE_OUT_WIRED_HEADPHONE | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT | + 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_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_SPEAKER_SAFE | + AUDIO_DEVICE_OUT_DEFAULT), + AUDIO_DEVICE_OUT_ALL_A2DP = (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | + AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), + AUDIO_DEVICE_OUT_ALL_SCO = (AUDIO_DEVICE_OUT_BLUETOOTH_SCO | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET | + AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), + AUDIO_DEVICE_OUT_ALL_USB = (AUDIO_DEVICE_OUT_USB_ACCESSORY | + AUDIO_DEVICE_OUT_USB_DEVICE), + /* input devices */ + AUDIO_DEVICE_IN_COMMUNICATION = AUDIO_DEVICE_BIT_IN | 0x1, + AUDIO_DEVICE_IN_AMBIENT = AUDIO_DEVICE_BIT_IN | 0x2, + AUDIO_DEVICE_IN_BUILTIN_MIC = AUDIO_DEVICE_BIT_IN | 0x4, + 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_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 | + AUDIO_DEVICE_IN_AMBIENT | + 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_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; +#endif + +static inline bool audio_is_output_device(uint32_t device) +{ +#if ANDROID_VERSION < 17 + if ((__builtin_popcount(device) == 1) && ((device & ~AUDIO_DEVICE_OUT_ALL) == 0)) + return true; + else + return false; +#else + if (((device & AUDIO_DEVICE_BIT_IN) == 0) && + (__builtin_popcount(device) == 1) && ((device & ~AUDIO_DEVICE_OUT_ALL) == 0)) + return true; + else + return false; +#endif +} + +/* device connection states used for audio_policy->set_device_connection_state() + * */ +typedef enum { + AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + + AUDIO_POLICY_DEVICE_STATE_CNT, + AUDIO_POLICY_DEVICE_STATE_MAX = AUDIO_POLICY_DEVICE_STATE_CNT - 1, +} audio_policy_dev_state_t; + +namespace android { + +typedef void (*audio_error_callback)(status_t err); +typedef int audio_io_handle_t; + +class IAudioPolicyService; +class String8; + +class AudioSystem +{ +public: + + enum stream_type { + DEFAULT =-1, + VOICE_CALL = 0, + SYSTEM = 1, + RING = 2, + MUSIC = 3, + ALARM = 4, + NOTIFICATION = 5, + BLUETOOTH_SCO = 6, + ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker + DTMF = 8, + TTS = 9, + FM = 10, + NUM_STREAM_TYPES + }; + + // Audio sub formats (see AudioSystem::audio_format). + enum pcm_sub_format { + PCM_SUB_16_BIT = 0x1, // must be 1 for backward compatibility + PCM_SUB_8_BIT = 0x2, // must be 2 for backward compatibility + }; + + // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify + // bit rate, stereo mode, version... + enum mp3_sub_format { + //TODO + }; + + // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned, + // encoding mode for recording... + enum amr_sub_format { + //TODO + }; + + // AAC sub format field definition: specify profile or bitrate for recording... + enum aac_sub_format { + //TODO + }; + + // VORBIS sub format field definition: specify quality for recording... + enum vorbis_sub_format { + //TODO + }; + + // Audio format consists in 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 indicates options and parameters + // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate + // or profile. It can also be used for certain formats to give informations not present in the encoded + // audio stream (e.g. octet alignement for AMR). + enum audio_format { + INVALID_FORMAT = -1, + FORMAT_DEFAULT = 0, + PCM = 0x00000000, // must be 0 for backward compatibility + MP3 = 0x01000000, + AMR_NB = 0x02000000, + AMR_WB = 0x03000000, + AAC = 0x04000000, + HE_AAC_V1 = 0x05000000, + HE_AAC_V2 = 0x06000000, + VORBIS = 0x07000000, + EVRC = 0x08000000, + QCELP = 0x09000000, + VOIP_PCM_INPUT = 0x0A000000, + MAIN_FORMAT_MASK = 0xFF000000, + SUB_FORMAT_MASK = 0x00FFFFFF, + // Aliases + PCM_16_BIT = (PCM|PCM_SUB_16_BIT), + PCM_8_BIT = (PCM|PCM_SUB_8_BIT) + }; + + + // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java + enum audio_channels { + // output channels + CHANNEL_OUT_FRONT_LEFT = 0x4, + CHANNEL_OUT_FRONT_RIGHT = 0x8, + CHANNEL_OUT_FRONT_CENTER = 0x10, + CHANNEL_OUT_LOW_FREQUENCY = 0x20, + CHANNEL_OUT_BACK_LEFT = 0x40, + CHANNEL_OUT_BACK_RIGHT = 0x80, + CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100, + CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200, + CHANNEL_OUT_BACK_CENTER = 0x400, + CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT, + CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT), + CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | + CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), + CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | + CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER), + CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | + CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), + CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | + CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | + CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER), + CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | + CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | + CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER), + + // input channels + CHANNEL_IN_LEFT = 0x4, + CHANNEL_IN_RIGHT = 0x8, + CHANNEL_IN_FRONT = 0x10, + CHANNEL_IN_BACK = 0x20, + CHANNEL_IN_LEFT_PROCESSED = 0x40, + CHANNEL_IN_RIGHT_PROCESSED = 0x80, + CHANNEL_IN_FRONT_PROCESSED = 0x100, + CHANNEL_IN_BACK_PROCESSED = 0x200, + CHANNEL_IN_PRESSURE = 0x400, + CHANNEL_IN_X_AXIS = 0x800, + CHANNEL_IN_Y_AXIS = 0x1000, + CHANNEL_IN_Z_AXIS = 0x2000, + CHANNEL_IN_VOICE_UPLINK = 0x4000, + CHANNEL_IN_VOICE_DNLINK = 0x8000, + CHANNEL_IN_MONO = CHANNEL_IN_FRONT, + CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT), + CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK| + CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED| + CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS | + CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK) + }; + + enum audio_mode { + MODE_INVALID = -2, + MODE_CURRENT = -1, + MODE_NORMAL = 0, + MODE_RINGTONE, + MODE_IN_CALL, + MODE_IN_COMMUNICATION, + NUM_MODES // not a valid entry, denotes end-of-list + }; + + enum audio_in_acoustics { + AGC_ENABLE = 0x0001, + AGC_DISABLE = 0, + NS_ENABLE = 0x0002, + NS_DISABLE = 0, + TX_IIR_ENABLE = 0x0004, + TX_DISABLE = 0 + }; + + // special audio session values + enum audio_sessions { + SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream + // (value must be less than 0) + SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can + // be moved by audio policy manager to another output stream + // (value must be 0) + }; + + /* These are static methods to control the system-wide AudioFlinger + * only privileged processes can have access to them + */ + + // mute/unmute microphone + static status_t muteMicrophone(bool state); + static status_t isMicrophoneMuted(bool *state); + + // set/get master volume + static status_t setMasterVolume(float value); + static status_t getMasterVolume(float* volume); + // mute/unmute audio outputs + static status_t setMasterMute(bool mute); + static status_t getMasterMute(bool* mute); + + // set/get stream volume on specified output + static status_t setStreamVolume(int stream, float value, int output); + static status_t getStreamVolume(int stream, float* volume, int output); + + // mute/unmute stream + static status_t setStreamMute(int stream, bool mute); + static status_t getStreamMute(int stream, bool* mute); + + // set audio mode in audio hardware (see AudioSystem::audio_mode) + static status_t setMode(int mode); + + // returns true in *state if tracks are active on the specified stream + static status_t isStreamActive(int stream, bool *state); + + // set/get audio hardware parameters. The function accepts a list of parameters + // key value pairs in the form: key1=value1;key2=value2;... + // Some keys are reserved for standard parameters (See AudioParameter class). + static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); + static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); + + static void setErrorCallback(audio_error_callback cb); + + // helper function to obtain AudioFlinger service handle + static const sp& get_audio_flinger(); + + static float linearToLog(int volume); + static int logToLinear(float volume); + + static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT); + static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT); + static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT); + + static bool routedToA2dpOutput(int streamType); + + static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, + size_t* buffSize); + + static status_t setVoiceVolume(float volume); + + // return the number of audio frames written by AudioFlinger to audio HAL and + // audio dsp to DAC since the output on which the specificed stream is playing + // has exited standby. + // returned status (from utils/Errors.h) can be: + // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data + // - INVALID_OPERATION: Not supported on current hardware platform + // - BAD_VALUE: invalid parameter + // NOTE: this feature is not supported on all hardware platforms and it is + // necessary to check returned status before using the returned values. + static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT); + + static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); + + static int newAudioSessionId(); + // + // AudioPolicyService interface + // + + enum audio_devices { + // output devices + DEVICE_OUT_EARPIECE = 0x1, + DEVICE_OUT_SPEAKER = 0x2, + DEVICE_OUT_WIRED_HEADSET = 0x4, + DEVICE_OUT_WIRED_HEADPHONE = 0x8, + DEVICE_OUT_BLUETOOTH_SCO = 0x10, + DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, + DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, + DEVICE_OUT_BLUETOOTH_A2DP = 0x80, + DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, + DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, + DEVICE_OUT_AUX_DIGITAL = 0x400, + DEVICE_OUT_DEFAULT = 0x8000, + DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | + DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | + DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | + DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_DEFAULT), + DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | + DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), + + // input devices + DEVICE_IN_COMMUNICATION = 0x10000, + DEVICE_IN_AMBIENT = 0x20000, + DEVICE_IN_BUILTIN_MIC = 0x40000, + DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000, + DEVICE_IN_WIRED_HEADSET = 0x100000, + DEVICE_IN_AUX_DIGITAL = 0x200000, + DEVICE_IN_VOICE_CALL = 0x400000, + DEVICE_IN_BACK_MIC = 0x800000, + DEVICE_IN_DEFAULT = 0x80000000, + + DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC | + DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL | + DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT) + }; + + // device connection states used for setDeviceConnectionState() + enum device_connection_state { + DEVICE_STATE_UNAVAILABLE, + DEVICE_STATE_AVAILABLE, + NUM_DEVICE_STATES + }; + + // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks) + enum output_flags { + OUTPUT_FLAG_INDIRECT = 0x0, + OUTPUT_FLAG_DIRECT = 0x1 + }; + + // device categories used for setForceUse() + enum forced_config { + FORCE_NONE, + FORCE_SPEAKER, + FORCE_HEADPHONES, + FORCE_BT_SCO, + FORCE_BT_A2DP, + FORCE_WIRED_ACCESSORY, + FORCE_BT_CAR_DOCK, + FORCE_BT_DESK_DOCK, + FORCE_ANALOG_DOCK, + FORCE_DIGITAL_DOCK, + FORCE_NO_BT_A2DP, + NUM_FORCE_CONFIG, + FORCE_DEFAULT = FORCE_NONE + }; + + // usages used for setForceUse() + enum force_use { + FOR_COMMUNICATION, + FOR_MEDIA, + FOR_RECORD, + FOR_DOCK, + NUM_FORCE_USE + }; + + // types of io configuration change events received with ioConfigChanged() + enum io_config_event { + OUTPUT_OPENED, + OUTPUT_CLOSED, + OUTPUT_CONFIG_CHANGED, + INPUT_OPENED, + INPUT_CLOSED, + INPUT_CONFIG_CHANGED, + STREAM_CONFIG_CHANGED, + NUM_CONFIG_EVENTS + }; + + // audio output descritor used to cache output configurations in client process to avoid frequent calls + // through IAudioFlinger + class OutputDescriptor { + public: + OutputDescriptor() + : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {} + + uint32_t samplingRate; + int32_t format; + int32_t channels; + size_t frameCount; + uint32_t latency; + }; + + // + // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) + // + static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address); + static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address); + static status_t setPhoneState(int state); +#if ANDROID_VERSION >= 17 + static status_t setPhoneState(audio_mode_t state); +#endif + static status_t setRingerMode(uint32_t mode, uint32_t mask); +#ifdef VANILLA_ANDROID + static status_t setForceUse(force_use usage, forced_config config); + static forced_config getForceUse(force_use usage); + static audio_io_handle_t getOutput(stream_type stream, + uint32_t samplingRate = 0, + uint32_t format = FORMAT_DEFAULT, + uint32_t channels = CHANNEL_OUT_STEREO, + output_flags flags = OUTPUT_FLAG_INDIRECT); + static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address); + static status_t setFmVolume(float volume); + static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); +#else + static status_t setForceUse(force_use usage, forced_config config) __attribute__((weak)); + static forced_config getForceUse(force_use usage) __attribute__((weak)); + static audio_io_handle_t getOutput(stream_type stream, + uint32_t samplingRate = 0, + uint32_t format = FORMAT_DEFAULT, + uint32_t channels = CHANNEL_OUT_STEREO, + output_flags flags = OUTPUT_FLAG_INDIRECT) __attribute__((weak)); + + static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) __attribute__((weak)); + static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) __attribute__((weak)); + static audio_io_handle_t getOutput(audio_stream_type_t stream, + uint32_t samplingRate = 0, + uint32_t format = AUDIO_FORMAT_DEFAULT, + uint32_t channels = AUDIO_CHANNEL_OUT_STEREO, + audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT) __attribute__((weak)); + static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address) __attribute__((weak)); + static status_t setFmVolume(float volume) __attribute__((weak)); + static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address) __attribute__((weak)); + +#endif + static status_t startOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0); + static status_t stopOutput(audio_io_handle_t output, + AudioSystem::stream_type stream, + int session = 0); + static void releaseOutput(audio_io_handle_t output); + static audio_io_handle_t getInput(int inputSource, + uint32_t samplingRate = 0, + uint32_t format = FORMAT_DEFAULT, + uint32_t channels = CHANNEL_IN_MONO, + audio_in_acoustics acoustics = (audio_in_acoustics)0); + static status_t startInput(audio_io_handle_t input); + static status_t stopInput(audio_io_handle_t input); + static void releaseInput(audio_io_handle_t input); + static status_t initStreamVolume(stream_type stream, + int indexMin, + int indexMax); + static status_t initStreamVolume(audio_stream_type_t stream, + int indexMin, + int indexMax); + static status_t setStreamVolumeIndex(stream_type stream, int index); + static status_t setStreamVolumeIndex(audio_stream_type_t stream, int index); +#if ANDROID_VERSION >= 17 + static status_t setStreamVolumeIndex(audio_stream_type_t stream, + int index, + audio_devices_t device); + static status_t getStreamVolumeIndex(audio_stream_type_t stream, + int *index, + audio_devices_t device); +#endif + static status_t getStreamVolumeIndex(stream_type stream, int *index); + static status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index); + + static uint32_t getStrategyForStream(stream_type stream); +#if ANDROID_VERSION >= 17 + static audio_devices_t getDevicesForStream(audio_stream_type_t stream); +#endif + + static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc); + static status_t registerEffect(effect_descriptor_t *desc, + audio_io_handle_t output, + uint32_t strategy, + int session, + int id); + static status_t unregisterEffect(int id); + + static const sp& get_audio_policy_service(); + + // ---------------------------------------------------------------------------- + + static uint32_t popCount(uint32_t u); + static bool isOutputDevice(audio_devices device); + static bool isInputDevice(audio_devices device); + static bool isA2dpDevice(audio_devices device); + static bool isBluetoothScoDevice(audio_devices device); + static bool isSeperatedStream(stream_type stream); + static bool isLowVisibility(stream_type stream); + static bool isOutputChannel(uint32_t channel); + static bool isInputChannel(uint32_t channel); + static bool isValidFormat(uint32_t format); + static bool isLinearPCM(uint32_t format); + static bool isModeInCall(); + +#if ANDROID_VERSION >= 21 + class AudioPortCallback : public RefBase + { + public: + + AudioPortCallback() {} + virtual ~AudioPortCallback() {} + + virtual void onAudioPortListUpdate() = 0; + virtual void onAudioPatchListUpdate() = 0; + virtual void onServiceDied() = 0; + + }; + + static void setAudioPortCallback(sp callBack); +#endif + +private: + + class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient + { + public: + AudioFlingerClient() { + } + + // DeathRecipient + virtual void binderDied(const wp& who); + + // IAudioFlingerClient + + // indicate a change in the configuration of an output or input: keeps the cached + // values for output/input parameters upto date in client process + virtual void ioConfigChanged(int event, int ioHandle, void *param2); + }; + + class AudioPolicyServiceClient: public IBinder::DeathRecipient + { + public: + AudioPolicyServiceClient() { + } + + // DeathRecipient + virtual void binderDied(const wp& who); + }; + + static sp gAudioFlingerClient; + static sp gAudioPolicyServiceClient; + friend class AudioFlingerClient; + friend class AudioPolicyServiceClient; + + static Mutex gLock; + static sp gAudioFlinger; + static audio_error_callback gAudioErrorCallback; + + static size_t gInBuffSize; + // previous parameters for recording buffer size queries + static uint32_t gPrevInSamplingRate; + static int gPrevInFormat; + static int gPrevInChannelCount; + + static sp gAudioPolicyService; + + // mapping between stream types and outputs + static DefaultKeyedVector gStreamOutputMap; + // list of output descritor containing cached parameters (sampling rate, framecount, channel count...) + static DefaultKeyedVector gOutputs; +}; + +class AudioParameter { + +public: + AudioParameter() {} + AudioParameter(const String8& keyValuePairs); + virtual ~AudioParameter(); + + // reserved parameter keys for changing standard parameters with setParameters() function. + // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input + // configuration changes and act accordingly. + // keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices + // keySamplingRate: to change sampling rate routing, value is an int + // keyFormat: to change audio format, value is an int in AudioSystem::audio_format + // keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels + // keyFrameCount: to change audio output frame count, value is an int + // keyInputSource: to change audio input source, value is an int in audio_source + // (defined in media/mediarecorder.h) + static const char *keyRouting; + static const char *keySamplingRate; + static const char *keyFormat; + static const char *keyChannels; + static const char *keyFrameCount; + static const char *keyInputSource; + + String8 toString(); + + status_t add(const String8& key, const String8& value); + status_t addInt(const String8& key, const int value); + status_t addFloat(const String8& key, const float value); + + status_t remove(const String8& key); + + status_t get(const String8& key, String8& value); + status_t getInt(const String8& key, int& value); + status_t getFloat(const String8& key, float& value); + status_t getAt(size_t index, String8& key, String8& value); + + size_t size() { return mParameters.size(); } + +private: + String8 mKeyValuePairs; + KeyedVector mParameters; +}; + +}; // namespace android + +#pragma GCC visibility pop + +#endif /*ANDROID_AUDIOSYSTEM_H_*/ diff --git a/dom/system/gonk/android_audio/AudioTrack.h b/dom/system/gonk/android_audio/AudioTrack.h new file mode 100644 index 000000000..6f8c6bb28 --- /dev/null +++ b/dom/system/gonk/android_audio/AudioTrack.h @@ -0,0 +1,489 @@ +/* + * Copyright (C) 2007 The Android Open Source Project + * + * 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 ANDROID_AUDIOTRACK_H +#define ANDROID_AUDIOTRACK_H + +#include +#include + +#include "IAudioFlinger.h" +#include "IAudioTrack.h" +#include "AudioSystem.h" + +#include +#include +#include +#include +#include + + +namespace android { + +// ---------------------------------------------------------------------------- + +class audio_track_cblk_t; + +// ---------------------------------------------------------------------------- + +class AudioTrack +{ +public: + enum channel_index { + MONO = 0, + LEFT = 0, + RIGHT = 1 + }; + + /* Events used by AudioTrack callback function (audio_track_cblk_t). + */ + enum event_type { + EVENT_MORE_DATA = 0, // Request to write more data to PCM buffer. + EVENT_UNDERRUN = 1, // PCM buffer underrun occured. + EVENT_LOOP_END = 2, // Sample loop end was reached; playback restarted from loop start if loop count was not 0. + EVENT_MARKER = 3, // Playback head is at the specified marker position (See setMarkerPosition()). + EVENT_NEW_POS = 4, // Playback head is at a new position (See setPositionUpdatePeriod()). + EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer. + }; + + /* Create Buffer on the stack and pass it to obtainBuffer() + * and releaseBuffer(). + */ + + class Buffer + { + public: + enum { + MUTE = 0x00000001 + }; + uint32_t flags; + int channelCount; + int format; + size_t frameCount; + size_t size; + union { + void* raw; + short* i16; + int8_t* i8; + }; + }; + + + /* As a convenience, if a callback is supplied, a handler thread + * is automatically created with the appropriate priority. This thread + * invokes the callback when a new buffer becomes availlable or an underrun condition occurs. + * Parameters: + * + * event: type of event notified (see enum AudioTrack::event_type). + * user: Pointer to context for use by the callback receiver. + * info: Pointer to optional parameter according to event type: + * - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write + * more bytes than indicated by 'size' field and update 'size' if less bytes are + * written. + * - EVENT_UNDERRUN: unused. + * - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining. + * - EVENT_MARKER: pointer to an uin32_t containing the marker position in frames. + * - EVENT_NEW_POS: pointer to an uin32_t containing the new position in frames. + * - EVENT_BUFFER_END: unused. + */ + + typedef void (*callback_t)(int event, void* user, void *info); + + /* Returns the minimum frame count required for the successful creation of + * an AudioTrack object. + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful operation + * - NO_INIT: audio server or audio hardware not initialized + */ + + static status_t getMinFrameCount(int* frameCount, + int streamType =-1, + uint32_t sampleRate = 0); + + /* Constructs an uninitialized AudioTrack. No connection with + * AudioFlinger takes place. + */ + AudioTrack(); + + /* Creates an audio track and registers it with AudioFlinger. + * Once created, the track needs to be started before it can be used. + * Unspecified values are set to the audio hardware's current + * values. + * + * Parameters: + * + * streamType: Select the type of audio stream this track is attached to + * (e.g. AudioSystem::MUSIC). + * sampleRate: Track sampling rate in Hz. + * format: Audio format (e.g AudioSystem::PCM_16_BIT for signed + * 16 bits per sample). + * channels: Channel mask: see AudioSystem::audio_channels. + * frameCount: Total size of track PCM buffer in frames. This defines the + * latency of the track. + * flags: Reserved for future use. + * cbf: Callback function. If not null, this function is called periodically + * to request new PCM data. + * notificationFrames: The callback function is called each time notificationFrames PCM + * frames have been comsumed from track input buffer. + * user Context for use by the callback receiver. + */ + + AudioTrack( int streamType, + uint32_t sampleRate = 0, + int format = 0, + int channels = 0, + int frameCount = 0, + uint32_t flags = 0, + callback_t cbf = 0, + void* user = 0, + int notificationFrames = 0, + int sessionId = 0); + + /* Creates an audio track and registers it with AudioFlinger. With this constructor, + * The PCM data to be rendered by AudioTrack is passed in a shared memory buffer + * identified by the argument sharedBuffer. This prototype is for static buffer playback. + * PCM data must be present into memory before the AudioTrack is started. + * The Write() and Flush() methods are not supported in this case. + * It is recommented to pass a callback function to be notified of playback end by an + * EVENT_UNDERRUN event. + */ + + AudioTrack( int streamType, + uint32_t sampleRate = 0, + int format = 0, + int channels = 0, + const sp& sharedBuffer = 0, + uint32_t flags = 0, + callback_t cbf = 0, + void* user = 0, + int notificationFrames = 0, + int sessionId = 0); + + /* Terminates the AudioTrack and unregisters it from AudioFlinger. + * Also destroys all resources assotiated with the AudioTrack. + */ + ~AudioTrack(); + + + /* Initialize an uninitialized AudioTrack. + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful intialization + * - INVALID_OPERATION: AudioTrack is already intitialized + * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) + * - NO_INIT: audio server or audio hardware not initialized + * */ + status_t set(int streamType =-1, + uint32_t sampleRate = 0, + int format = 0, + int channels = 0, + int frameCount = 0, + uint32_t flags = 0, + callback_t cbf = 0, + void* user = 0, + int notificationFrames = 0, + const sp& sharedBuffer = 0, + bool threadCanCallJava = false, + int sessionId = 0); + + + /* Result of constructing the AudioTrack. This must be checked + * before using any AudioTrack API (except for set()), using + * an uninitialized AudioTrack produces undefined results. + * See set() method above for possible return codes. + */ + status_t initCheck() const; + + /* Returns this track's latency in milliseconds. + * This includes the latency due to AudioTrack buffer size, AudioMixer (if any) + * and audio hardware driver. + */ + uint32_t latency() const; + + /* getters, see constructor */ + + int streamType() const; + int format() const; + int channelCount() const; + uint32_t frameCount() const; + int frameSize() const; + sp& sharedBuffer(); + + + /* After it's created the track is not active. Call start() to + * make it active. If set, the callback will start being called. + */ + void start(); + + /* Stop a track. If set, the callback will cease being called and + * obtainBuffer returns STOPPED. Note that obtainBuffer() still works + * and will fill up buffers until the pool is exhausted. + */ + void stop(); + bool stopped() const; + + /* flush a stopped track. All pending buffers are discarded. + * This function has no effect if the track is not stoped. + */ + void flush(); + + /* Pause a track. If set, the callback will cease being called and + * obtainBuffer returns STOPPED. Note that obtainBuffer() still works + * and will fill up buffers until the pool is exhausted. + */ + void pause(); + + /* mute or unmutes this track. + * While mutted, the callback, if set, is still called. + */ + void mute(bool); + bool muted() const; + + + /* set volume for this track, mostly used for games' sound effects + * left and right volumes. Levels must be <= 1.0. + */ + status_t setVolume(float left, float right); + void getVolume(float* left, float* right); + + /* set the send level for this track. An auxiliary effect should be attached + * to the track with attachEffect(). Level must be <= 1.0. + */ + status_t setAuxEffectSendLevel(float level); + void getAuxEffectSendLevel(float* level); + + /* set sample rate for this track, mostly used for games' sound effects + */ + status_t setSampleRate(int sampleRate); + uint32_t getSampleRate(); + + /* Enables looping and sets the start and end points of looping. + * + * Parameters: + * + * loopStart: loop start expressed as the number of PCM frames played since AudioTrack start. + * loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start. + * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any pending or + * active loop. loopCount = -1 means infinite looping. + * + * For proper operation the following condition must be respected: + * (loopEnd-loopStart) <= framecount() + */ + status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount); + status_t getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount); + + + /* Sets marker position. When playback reaches the number of frames specified, a callback with event + * type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker notification + * callback. + * If the AudioTrack has been opened with no callback function associated, the operation will fail. + * + * Parameters: + * + * marker: marker position expressed in frames. + * + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful operation + * - INVALID_OPERATION: the AudioTrack has no callback installed. + */ + status_t setMarkerPosition(uint32_t marker); + status_t getMarkerPosition(uint32_t *marker); + + + /* Sets position update period. Every time the number of frames specified has been played, + * a callback with event type EVENT_NEW_POS is called. + * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification + * callback. + * If the AudioTrack has been opened with no callback function associated, the operation will fail. + * + * Parameters: + * + * updatePeriod: position update notification period expressed in frames. + * + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful operation + * - INVALID_OPERATION: the AudioTrack has no callback installed. + */ + status_t setPositionUpdatePeriod(uint32_t updatePeriod); + status_t getPositionUpdatePeriod(uint32_t *updatePeriod); + + + /* Sets playback head position within AudioTrack buffer. The new position is specified + * in number of frames. + * This method must be called with the AudioTrack in paused or stopped state. + * Note that the actual position set is modulo the AudioTrack buffer size in frames. + * Therefore using this method makes sense only when playing a "static" audio buffer + * as opposed to streaming. + * The getPosition() method on the other hand returns the total number of frames played since + * playback start. + * + * Parameters: + * + * position: New playback head position within AudioTrack buffer. + * + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful operation + * - INVALID_OPERATION: the AudioTrack is not stopped. + * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer + */ + status_t setPosition(uint32_t position); + status_t getPosition(uint32_t *position); + + /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids + * rewriting the buffer before restarting playback after a stop. + * This method must be called with the AudioTrack in paused or stopped state. + * + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful operation + * - INVALID_OPERATION: the AudioTrack is not stopped. + */ + status_t reload(); + + /* returns a handle on the audio output used by this AudioTrack. + * + * Parameters: + * none. + * + * Returned value: + * handle on audio hardware output + */ + audio_io_handle_t getOutput(); + + /* returns the unique ID associated to this track. + * + * Parameters: + * none. + * + * Returned value: + * AudioTrack ID. + */ + int getSessionId(); + + + /* Attach track auxiliary output to specified effect. Used effectId = 0 + * to detach track from effect. + * + * Parameters: + * + * effectId: effectId obtained from AudioEffect::id(). + * + * Returned status (from utils/Errors.h) can be: + * - NO_ERROR: successful operation + * - INVALID_OPERATION: the effect is not an auxiliary effect. + * - BAD_VALUE: The specified effect ID is invalid + */ + status_t attachAuxEffect(int effectId); + + /* obtains a buffer of "frameCount" frames. The buffer must be + * filled entirely. If the track is stopped, obtainBuffer() returns + * STOPPED instead of NO_ERROR as long as there are buffers availlable, + * at which point NO_MORE_BUFFERS is returned. + * Buffers will be returned until the pool (buffercount()) + * is exhausted, at which point obtainBuffer() will either block + * or return WOULD_BLOCK depending on the value of the "blocking" + * parameter. + */ + + enum { + NO_MORE_BUFFERS = 0x80000001, + STOPPED = 1 + }; + + status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount); + void releaseBuffer(Buffer* audioBuffer); + + + /* As a convenience we provide a write() interface to the audio buffer. + * This is implemented on top of lockBuffer/unlockBuffer. For best + * performance + * + */ + ssize_t write(const void* buffer, size_t size); + + /* + * Dumps the state of an audio track. + */ + status_t dump(int fd, const Vector& args) const; + +private: + /* copying audio tracks is not allowed */ + AudioTrack(const AudioTrack& other); + AudioTrack& operator = (const AudioTrack& other); + + /* a small internal class to handle the callback */ + class AudioTrackThread : public Thread + { + public: + AudioTrackThread(AudioTrack& receiver, bool bCanCallJava = false); + private: + friend class AudioTrack; + virtual bool threadLoop(); + virtual status_t readyToRun(); + virtual void onFirstRef(); + AudioTrack& mReceiver; + Mutex mLock; + }; + + bool processAudioBuffer(const sp& thread); + status_t createTrack(int streamType, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + const sp& sharedBuffer, + audio_io_handle_t output, + bool enforceFrameCount); + + sp mAudioTrack; + sp mCblkMemory; + sp mAudioTrackThread; + + float mVolume[2]; + float mSendLevel; + uint32_t mFrameCount; + + audio_track_cblk_t* mCblk; + uint8_t mStreamType; + uint8_t mFormat; + uint8_t mChannelCount; + uint8_t mMuted; + uint32_t mChannels; + status_t mStatus; + uint32_t mLatency; + + volatile int32_t mActive; + + callback_t mCbf; + void* mUserData; + uint32_t mNotificationFramesReq; // requested number of frames between each notification callback + uint32_t mNotificationFramesAct; // actual number of frames between each notification callback + sp mSharedBuffer; + int mLoopCount; + uint32_t mRemainingFrames; + uint32_t mMarkerPosition; + bool mMarkerReached; + uint32_t mNewPosition; + uint32_t mUpdatePeriod; + uint32_t mFlags; + int mSessionId; + int mAuxEffectId; + uint32_t mPadding[8]; +}; + + +}; // namespace android + +#endif // ANDROID_AUDIOTRACK_H diff --git a/dom/system/gonk/android_audio/EffectApi.h b/dom/system/gonk/android_audio/EffectApi.h new file mode 100644 index 000000000..729545d0c --- /dev/null +++ b/dom/system/gonk/android_audio/EffectApi.h @@ -0,0 +1,798 @@ +/* + * Copyright (C) 2010 The Android Open Source Project + * + * 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 ANDROID_EFFECTAPI_H_ +#define ANDROID_EFFECTAPI_H_ + +#include +#include +#include + +#if __cplusplus +extern "C" { +#endif + +///////////////////////////////////////////////// +// Effect control interface +///////////////////////////////////////////////// + +// The effect control interface is exposed by each effect engine implementation. It consists of +// a set of functions controlling the configuration, activation and process of the engine. +// The functions are grouped in a structure of type effect_interface_s: +// struct effect_interface_s { +// effect_process_t process; +// effect_command_t command; +// }; + + +// effect_interface_t: Effect control interface handle. +// The effect_interface_t serves two purposes regarding the implementation of the effect engine: +// - 1 it is the address of a pointer to an effect_interface_s structure where the functions +// of the effect control API for a particular effect are located. +// - 2 it is the address of the context of a particular effect instance. +// A typical implementation in the effect library would define a structure as follows: +// struct effect_module_s { +// const struct effect_interface_s *itfe; +// effect_config_t config; +// effect_context_t context; +// } +// The implementation of EffectCreate() function would then allocate a structure of this +// type and return its address as effect_interface_t +typedef struct effect_interface_s **effect_interface_t; + + +// Effect API version 1.0 +#define EFFECT_API_VERSION 0x0100 // Format 0xMMmm MM: Major version, mm: minor version + +// Maximum length of character strings in structures defines by this API. +#define EFFECT_STRING_LEN_MAX 64 + +// +//--- Effect descriptor structure effect_descriptor_t +// + +// Unique effect ID (can be generated from the following site: +// http://www.itu.int/ITU-T/asn1/uuid.html) +// This format is used for both "type" and "uuid" fields of the effect descriptor structure. +// - When used for effect type and the engine is implementing and effect corresponding to a standard +// OpenSL ES interface, this ID must be the one defined in OpenSLES_IID.h for that interface. +// - When used as uuid, it should be a unique UUID for this particular implementation. +typedef struct effect_uuid_s { + uint32_t timeLow; + uint16_t timeMid; + uint16_t timeHiAndVersion; + uint16_t clockSeq; + uint8_t node[6]; +} effect_uuid_t; + +// NULL UUID definition (matches SL_IID_NULL_) +#define EFFECT_UUID_INITIALIZER { 0xec7178ec, 0xe5e1, 0x4432, 0xa3f4, \ + { 0x46, 0x57, 0xe6, 0x79, 0x52, 0x10 } } +static const effect_uuid_t EFFECT_UUID_NULL_ = EFFECT_UUID_INITIALIZER; +const effect_uuid_t * const EFFECT_UUID_NULL = &EFFECT_UUID_NULL_; +const char * const EFFECT_UUID_NULL_STR = "ec7178ec-e5e1-4432-a3f4-4657e6795210"; + +// The effect descriptor contains necessary information to facilitate the enumeration of the effect +// engines present in a library. +typedef struct effect_descriptor_s { + effect_uuid_t type; // UUID of to the OpenSL ES interface implemented by this effect + effect_uuid_t uuid; // UUID for this particular implementation + uint16_t apiVersion; // Version of the effect API implemented: matches EFFECT_API_VERSION + uint32_t flags; // effect engine capabilities/requirements flags (see below) + uint16_t cpuLoad; // CPU load indication (see below) + uint16_t memoryUsage; // Data Memory usage (see below) + char name[EFFECT_STRING_LEN_MAX]; // human readable effect name + char implementor[EFFECT_STRING_LEN_MAX]; // human readable effect implementor name +} effect_descriptor_t; + +// CPU load and memory usage indication: each effect implementation must provide an indication of +// its CPU and memory usage for the audio effect framework to limit the number of effects +// instantiated at a given time on a given platform. +// The CPU load is expressed in 0.1 MIPS units as estimated on an ARM9E core (ARMv5TE) with 0 WS. +// The memory usage is expressed in KB and includes only dynamically allocated memory + +// Definitions for flags field of effect descriptor. +// +---------------------------+-----------+----------------------------------- +// | description | bits | values +// +---------------------------+-----------+----------------------------------- +// | connection mode | 0..1 | 0 insert: after track process +// | | | 1 auxiliary: connect to track auxiliary +// | | | output and use send level +// | | | 2 replace: replaces track process function; +// | | | must implement SRC, volume and mono to stereo. +// | | | 3 reserved +// +---------------------------+-----------+----------------------------------- +// | insertion preference | 2..4 | 0 none +// | | | 1 first of the chain +// | | | 2 last of the chain +// | | | 3 exclusive (only effect in the insert chain) +// | | | 4..7 reserved +// +---------------------------+-----------+----------------------------------- +// | Volume management | 5..6 | 0 none +// | | | 1 implements volume control +// | | | 2 requires volume indication +// | | | 3 reserved +// +---------------------------+-----------+----------------------------------- +// | Device indication | 7..8 | 0 none +// | | | 1 requires device updates +// | | | 2..3 reserved +// +---------------------------+-----------+----------------------------------- +// | Sample input mode | 9..10 | 0 direct: process() function or EFFECT_CMD_CONFIGURE +// | | | command must specify a buffer descriptor +// | | | 1 provider: process() function uses the +// | | | bufferProvider indicated by the +// | | | EFFECT_CMD_CONFIGURE command to request input. +// | | | buffers. +// | | | 2 both: both input modes are supported +// | | | 3 reserved +// +---------------------------+-----------+----------------------------------- +// | Sample output mode | 11..12 | 0 direct: process() function or EFFECT_CMD_CONFIGURE +// | | | command must specify a buffer descriptor +// | | | 1 provider: process() function uses the +// | | | bufferProvider indicated by the +// | | | EFFECT_CMD_CONFIGURE command to request output +// | | | buffers. +// | | | 2 both: both output modes are supported +// | | | 3 reserved +// +---------------------------+-----------+----------------------------------- +// | Hardware acceleration | 13..15 | 0 No hardware acceleration +// | | | 1 non tunneled hw acceleration: the process() function +// | | | reads the samples, send them to HW accelerated +// | | | effect processor, reads back the processed samples +// | | | and returns them to the output buffer. +// | | | 2 tunneled hw acceleration: the process() function is +// | | | transparent. The effect interface is only used to +// | | | control the effect engine. This mode is relevant for +// | | | global effects actually applied by the audio +// | | | hardware on the output stream. +// +---------------------------+-----------+----------------------------------- +// | Audio Mode indication | 16..17 | 0 none +// | | | 1 requires audio mode updates +// | | | 2..3 reserved +// +---------------------------+-----------+----------------------------------- + +// Insert mode +#define EFFECT_FLAG_TYPE_MASK 0x00000003 +#define EFFECT_FLAG_TYPE_INSERT 0x00000000 +#define EFFECT_FLAG_TYPE_AUXILIARY 0x00000001 +#define EFFECT_FLAG_TYPE_REPLACE 0x00000002 + +// Insert preference +#define EFFECT_FLAG_INSERT_MASK 0x0000001C +#define EFFECT_FLAG_INSERT_ANY 0x00000000 +#define EFFECT_FLAG_INSERT_FIRST 0x00000004 +#define EFFECT_FLAG_INSERT_LAST 0x00000008 +#define EFFECT_FLAG_INSERT_EXCLUSIVE 0x0000000C + + +// Volume control +#define EFFECT_FLAG_VOLUME_MASK 0x00000060 +#define EFFECT_FLAG_VOLUME_CTRL 0x00000020 +#define EFFECT_FLAG_VOLUME_IND 0x00000040 +#define EFFECT_FLAG_VOLUME_NONE 0x00000000 + +// Device indication +#define EFFECT_FLAG_DEVICE_MASK 0x00000180 +#define EFFECT_FLAG_DEVICE_IND 0x00000080 +#define EFFECT_FLAG_DEVICE_NONE 0x00000000 + +// Sample input modes +#define EFFECT_FLAG_INPUT_MASK 0x00000600 +#define EFFECT_FLAG_INPUT_DIRECT 0x00000000 +#define EFFECT_FLAG_INPUT_PROVIDER 0x00000200 +#define EFFECT_FLAG_INPUT_BOTH 0x00000400 + +// Sample output modes +#define EFFECT_FLAG_OUTPUT_MASK 0x00001800 +#define EFFECT_FLAG_OUTPUT_DIRECT 0x00000000 +#define EFFECT_FLAG_OUTPUT_PROVIDER 0x00000800 +#define EFFECT_FLAG_OUTPUT_BOTH 0x00001000 + +// Hardware acceleration mode +#define EFFECT_FLAG_HW_ACC_MASK 0x00006000 +#define EFFECT_FLAG_HW_ACC_SIMPLE 0x00002000 +#define EFFECT_FLAG_HW_ACC_TUNNEL 0x00004000 + +// Audio mode indication +#define EFFECT_FLAG_AUDIO_MODE_MASK 0x00018000 +#define EFFECT_FLAG_AUDIO_MODE_IND 0x00008000 +#define EFFECT_FLAG_AUDIO_MODE_NONE 0x00000000 + +// Forward definition of type audio_buffer_t +typedef struct audio_buffer_s audio_buffer_t; + +//////////////////////////////////////////////////////////////////////////////// +// +// Function: process +// +// Description: Effect process function. Takes input samples as specified +// (count and location) in input buffer descriptor and output processed +// samples as specified in output buffer descriptor. If the buffer descriptor +// is not specified the function must use either the buffer or the +// buffer provider function installed by the EFFECT_CMD_CONFIGURE command. +// The effect framework will call the process() function after the EFFECT_CMD_ENABLE +// command is received and until the EFFECT_CMD_DISABLE is received. When the engine +// receives the EFFECT_CMD_DISABLE command it should turn off the effect gracefully +// and when done indicate that it is OK to stop calling the process() function by +// returning the -ENODATA status. +// +// NOTE: the process() function implementation should be "real-time safe" that is +// it should not perform blocking calls: malloc/free, sleep, read/write/open/close, +// pthread_cond_wait/pthread_mutex_lock... +// +// Input: +// effect_interface_t: handle to the effect interface this function +// is called on. +// inBuffer: buffer descriptor indicating where to read samples to process. +// If NULL, use the configuration passed by EFFECT_CMD_CONFIGURE command. +// +// inBuffer: buffer descriptor indicating where to write processed samples. +// If NULL, use the configuration passed by EFFECT_CMD_CONFIGURE command. +// +// Output: +// returned value: 0 successful operation +// -ENODATA the engine has finished the disable phase and the framework +// can stop calling process() +// -EINVAL invalid interface handle or +// invalid input/output buffer description +//////////////////////////////////////////////////////////////////////////////// +typedef int32_t (*effect_process_t)(effect_interface_t self, + audio_buffer_t *inBuffer, + audio_buffer_t *outBuffer); + +//////////////////////////////////////////////////////////////////////////////// +// +// Function: command +// +// Description: Send a command and receive a response to/from effect engine. +// +// Input: +// effect_interface_t: handle to the effect interface this function +// is called on. +// cmdCode: command code: the command can be a standardized command defined in +// effect_command_e (see below) or a proprietary command. +// cmdSize: size of command in bytes +// pCmdData: pointer to command data +// pReplyData: pointer to reply data +// +// Input/Output: +// replySize: maximum size of reply data as input +// actual size of reply data as output +// +// Output: +// returned value: 0 successful operation +// -EINVAL invalid interface handle or +// invalid command/reply size or format according to command code +// The return code should be restricted to indicate problems related to the this +// API specification. Status related to the execution of a particular command should be +// indicated as part of the reply field. +// +// *pReplyData updated with command response +// +//////////////////////////////////////////////////////////////////////////////// +typedef int32_t (*effect_command_t)(effect_interface_t self, + uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t *replySize, + void *pReplyData); + + +// Effect control interface definition +struct effect_interface_s { + effect_process_t process; + effect_command_t command; +}; + + +// +//--- Standardized command codes for command() function +// +enum effect_command_e { + EFFECT_CMD_INIT, // initialize effect engine + EFFECT_CMD_CONFIGURE, // configure effect engine (see effect_config_t) + EFFECT_CMD_RESET, // reset effect engine + EFFECT_CMD_ENABLE, // enable effect process + EFFECT_CMD_DISABLE, // disable effect process + EFFECT_CMD_SET_PARAM, // set parameter immediately (see effect_param_t) + EFFECT_CMD_SET_PARAM_DEFERRED, // set parameter deferred + EFFECT_CMD_SET_PARAM_COMMIT, // commit previous set parameter deferred + EFFECT_CMD_GET_PARAM, // get parameter + EFFECT_CMD_SET_DEVICE, // set audio device (see audio_device_e) + EFFECT_CMD_SET_VOLUME, // set volume + EFFECT_CMD_SET_AUDIO_MODE, // set the audio mode (normal, ring, ...) + EFFECT_CMD_FIRST_PROPRIETARY = 0x10000 // first proprietary command code +}; + +//================================================================================================== +// command: EFFECT_CMD_INIT +//-------------------------------------------------------------------------------------------------- +// description: +// Initialize effect engine: All configurations return to default +//-------------------------------------------------------------------------------------------------- +// command format: +// size: 0 +// data: N/A +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: sizeof(int) +// data: status +//================================================================================================== +// command: EFFECT_CMD_CONFIGURE +//-------------------------------------------------------------------------------------------------- +// description: +// Apply new audio parameters configurations for input and output buffers +//-------------------------------------------------------------------------------------------------- +// command format: +// size: sizeof(effect_config_t) +// data: effect_config_t +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: sizeof(int) +// data: status +//================================================================================================== +// command: EFFECT_CMD_RESET +//-------------------------------------------------------------------------------------------------- +// description: +// Reset the effect engine. Keep configuration but resets state and buffer content +//-------------------------------------------------------------------------------------------------- +// command format: +// size: 0 +// data: N/A +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: 0 +// data: N/A +//================================================================================================== +// command: EFFECT_CMD_ENABLE +//-------------------------------------------------------------------------------------------------- +// description: +// Enable the process. Called by the framework before the first call to process() +//-------------------------------------------------------------------------------------------------- +// command format: +// size: 0 +// data: N/A +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: sizeof(int) +// data: status +//================================================================================================== +// command: EFFECT_CMD_DISABLE +//-------------------------------------------------------------------------------------------------- +// description: +// Disable the process. Called by the framework after the last call to process() +//-------------------------------------------------------------------------------------------------- +// command format: +// size: 0 +// data: N/A +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: sizeof(int) +// data: status +//================================================================================================== +// command: EFFECT_CMD_SET_PARAM +//-------------------------------------------------------------------------------------------------- +// description: +// Set a parameter and apply it immediately +//-------------------------------------------------------------------------------------------------- +// command format: +// size: sizeof(effect_param_t) + size of param and value +// data: effect_param_t + param + value. See effect_param_t definition below for value offset +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: sizeof(int) +// data: status +//================================================================================================== +// command: EFFECT_CMD_SET_PARAM_DEFERRED +//-------------------------------------------------------------------------------------------------- +// description: +// Set a parameter but apply it only when receiving EFFECT_CMD_SET_PARAM_COMMIT command +//-------------------------------------------------------------------------------------------------- +// command format: +// size: sizeof(effect_param_t) + size of param and value +// data: effect_param_t + param + value. See effect_param_t definition below for value offset +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: 0 +// data: N/A +//================================================================================================== +// command: EFFECT_CMD_SET_PARAM_COMMIT +//-------------------------------------------------------------------------------------------------- +// description: +// Apply all previously received EFFECT_CMD_SET_PARAM_DEFERRED commands +//-------------------------------------------------------------------------------------------------- +// command format: +// size: 0 +// data: N/A +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: sizeof(int) +// data: status +//================================================================================================== +// command: EFFECT_CMD_GET_PARAM +//-------------------------------------------------------------------------------------------------- +// description: +// Get a parameter value +//-------------------------------------------------------------------------------------------------- +// command format: +// size: sizeof(effect_param_t) + size of param +// data: effect_param_t + param +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: sizeof(effect_param_t) + size of param and value +// data: effect_param_t + param + value. See effect_param_t definition below for value offset +//================================================================================================== +// command: EFFECT_CMD_SET_DEVICE +//-------------------------------------------------------------------------------------------------- +// description: +// Set the rendering device the audio output path is connected to. See audio_device_e for device +// values. +// The effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to receive this +// command when the device changes +//-------------------------------------------------------------------------------------------------- +// command format: +// size: sizeof(uint32_t) +// data: audio_device_e +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: 0 +// data: N/A +//================================================================================================== +// command: EFFECT_CMD_SET_VOLUME +//-------------------------------------------------------------------------------------------------- +// description: +// Set and get volume. Used by audio framework to delegate volume control to effect engine. +// The effect implementation must set EFFECT_FLAG_VOLUME_IND or EFFECT_FLAG_VOLUME_CTRL flag in +// its descriptor to receive this command before every call to process() function +// If EFFECT_FLAG_VOLUME_CTRL flag is set in the effect descriptor, the effect engine must return +// the volume that should be applied before the effect is processed. The overall volume (the volume +// actually applied by the effect engine multiplied by the returned value) should match the value +// indicated in the command. +//-------------------------------------------------------------------------------------------------- +// command format: +// size: n * sizeof(uint32_t) +// data: volume for each channel defined in effect_config_t for output buffer expressed in +// 8.24 fixed point format +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: n * sizeof(uint32_t) / 0 +// data: - if EFFECT_FLAG_VOLUME_CTRL is set in effect descriptor: +// volume for each channel defined in effect_config_t for output buffer expressed in +// 8.24 fixed point format +// - if EFFECT_FLAG_VOLUME_CTRL is not set in effect descriptor: +// N/A +// It is legal to receive a null pointer as pReplyData in which case the effect framework has +// delegated volume control to another effect +//================================================================================================== +// command: EFFECT_CMD_SET_AUDIO_MODE +//-------------------------------------------------------------------------------------------------- +// description: +// Set the audio mode. The effect implementation must set EFFECT_FLAG_AUDIO_MODE_IND flag in its +// descriptor to receive this command when the audio mode changes. +//-------------------------------------------------------------------------------------------------- +// command format: +// size: sizeof(uint32_t) +// data: audio_mode_e +//-------------------------------------------------------------------------------------------------- +// reply format: +// size: 0 +// data: N/A +//================================================================================================== +// command: EFFECT_CMD_FIRST_PROPRIETARY +//-------------------------------------------------------------------------------------------------- +// description: +// All proprietary effect commands must use command codes above this value. The size and format of +// command and response fields is free in this case +//================================================================================================== + + +// Audio buffer descriptor used by process(), bufferProvider() functions and buffer_config_t +// structure. Multi-channel audio is always interleaved. The channel order is from LSB to MSB with +// regard to the channel mask definition in audio_channels_e e.g : +// Stereo: left, right +// 5 point 1: front left, front right, front center, low frequency, back left, back right +// The buffer size is expressed in frame count, a frame being composed of samples for all +// channels at a given time. Frame size for unspecified format (AUDIO_FORMAT_OTHER) is 8 bit by +// definition +struct audio_buffer_s { + size_t frameCount; // number of frames in buffer + union { + void* raw; // raw pointer to start of buffer + int32_t* s32; // pointer to signed 32 bit data at start of buffer + int16_t* s16; // pointer to signed 16 bit data at start of buffer + uint8_t* u8; // pointer to unsigned 8 bit data at start of buffer + }; +}; + +// The buffer_provider_s structure contains functions that can be used +// by the effect engine process() function to query and release input +// or output audio buffer. +// The getBuffer() function is called to retrieve a buffer where data +// should read from or written to by process() function. +// The releaseBuffer() function MUST be called when the buffer retrieved +// with getBuffer() is not needed anymore. +// The process function should use the buffer provider mechanism to retrieve +// input or output buffer if the inBuffer or outBuffer passed as argument is NULL +// and the buffer configuration (buffer_config_t) given by the EFFECT_CMD_CONFIGURE +// command did not specify an audio buffer. + +typedef int32_t (* buffer_function_t)(void *cookie, audio_buffer_t *buffer); + +typedef struct buffer_provider_s { + buffer_function_t getBuffer; // retrieve next buffer + buffer_function_t releaseBuffer; // release used buffer + void *cookie; // for use by client of buffer provider functions +} buffer_provider_t; + + +// The buffer_config_s structure specifies the input or output audio format +// to be used by the effect engine. It is part of the effect_config_t +// structure that defines both input and output buffer configurations and is +// passed by the EFFECT_CMD_CONFIGURE command. +typedef struct buffer_config_s { + audio_buffer_t buffer; // buffer for use by process() function if not passed explicitly + uint32_t samplingRate; // sampling rate + uint32_t channels; // channel mask (see audio_channels_e) + buffer_provider_t bufferProvider; // buffer provider + uint8_t format; // Audio format (see audio_format_e) + uint8_t accessMode; // read/write or accumulate in buffer (effect_buffer_access_e) + uint16_t mask; // indicates which of the above fields is valid +} buffer_config_t; + +// Sample format +enum audio_format_e { + SAMPLE_FORMAT_PCM_S15, // PCM signed 16 bits + SAMPLE_FORMAT_PCM_U8, // PCM unsigned 8 bits + SAMPLE_FORMAT_PCM_S7_24, // PCM signed 7.24 fixed point representation + SAMPLE_FORMAT_OTHER // other format (e.g. compressed) +}; + +// Channel mask +enum audio_channels_e { + CHANNEL_FRONT_LEFT = 0x1, // front left channel + CHANNEL_FRONT_RIGHT = 0x2, // front right channel + CHANNEL_FRONT_CENTER = 0x4, // front center channel + CHANNEL_LOW_FREQUENCY = 0x8, // low frequency channel + CHANNEL_BACK_LEFT = 0x10, // back left channel + CHANNEL_BACK_RIGHT = 0x20, // back right channel + CHANNEL_FRONT_LEFT_OF_CENTER = 0x40, // front left of center channel + CHANNEL_FRONT_RIGHT_OF_CENTER = 0x80, // front right of center channel + CHANNEL_BACK_CENTER = 0x100, // back center channel + CHANNEL_MONO = CHANNEL_FRONT_LEFT, + CHANNEL_STEREO = (CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT), + CHANNEL_QUAD = (CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT), + CHANNEL_SURROUND = (CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_FRONT_CENTER | CHANNEL_BACK_CENTER), + CHANNEL_5POINT1 = (CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY | CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT), + CHANNEL_7POINT1 = (CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | + CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY | CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT | + CHANNEL_FRONT_LEFT_OF_CENTER | CHANNEL_FRONT_RIGHT_OF_CENTER), +}; + +// Render device +enum audio_device_e { + DEVICE_EARPIECE = 0x1, // earpiece + DEVICE_SPEAKER = 0x2, // speaker + DEVICE_WIRED_HEADSET = 0x4, // wired headset, with microphone + DEVICE_WIRED_HEADPHONE = 0x8, // wired headphone, without microphone + DEVICE_BLUETOOTH_SCO = 0x10, // generic bluetooth SCO + DEVICE_BLUETOOTH_SCO_HEADSET = 0x20, // bluetooth SCO headset + DEVICE_BLUETOOTH_SCO_CARKIT = 0x40, // bluetooth SCO car kit + DEVICE_BLUETOOTH_A2DP = 0x80, // generic bluetooth A2DP + DEVICE_BLUETOOTH_A2DP_HEADPHONES = 0x100, // bluetooth A2DP headphones + DEVICE_BLUETOOTH_A2DP_SPEAKER = 0x200, // bluetooth A2DP speakers + DEVICE_AUX_DIGITAL = 0x400, // digital output + DEVICE_EXTERNAL_SPEAKER = 0x800 // external speaker (stereo and High quality) +}; + +#if ANDROID_VERSION < 17 +// Audio mode +enum audio_mode_e { + AUDIO_MODE_NORMAL, // device idle + AUDIO_MODE_RINGTONE, // device ringing + AUDIO_MODE_IN_CALL // audio call connected (VoIP or telephony) +}; +#endif + +// Values for "accessMode" field of buffer_config_t: +// overwrite, read only, accumulate (read/modify/write) +enum effect_buffer_access_e { + EFFECT_BUFFER_ACCESS_WRITE, + EFFECT_BUFFER_ACCESS_READ, + EFFECT_BUFFER_ACCESS_ACCUMULATE + +}; + +// Values for bit field "mask" in buffer_config_t. If a bit is set, the corresponding field +// in buffer_config_t must be taken into account when executing the EFFECT_CMD_CONFIGURE command +#define EFFECT_CONFIG_BUFFER 0x0001 // buffer field must be taken into account +#define EFFECT_CONFIG_SMP_RATE 0x0002 // samplingRate field must be taken into account +#define EFFECT_CONFIG_CHANNELS 0x0004 // channels field must be taken into account +#define EFFECT_CONFIG_FORMAT 0x0008 // format field must be taken into account +#define EFFECT_CONFIG_ACC_MODE 0x0010 // accessMode field must be taken into account +#define EFFECT_CONFIG_PROVIDER 0x0020 // bufferProvider field must be taken into account +#define EFFECT_CONFIG_ALL (EFFECT_CONFIG_BUFFER | EFFECT_CONFIG_SMP_RATE | \ + EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT | \ + EFFECT_CONFIG_ACC_MODE | EFFECT_CONFIG_PROVIDER) + + +// effect_config_s structure describes the format of the pCmdData argument of EFFECT_CMD_CONFIGURE +// command to configure audio parameters and buffers for effect engine input and output. +typedef struct effect_config_s { + buffer_config_t inputCfg; + buffer_config_t outputCfg;; +} effect_config_t; + + +// effect_param_s structure describes the format of the pCmdData argument of EFFECT_CMD_SET_PARAM +// command and pCmdData and pReplyData of EFFECT_CMD_GET_PARAM command. +// psize and vsize represent the actual size of parameter and value. +// +// NOTE: the start of value field inside the data field is always on a 32 bit boundary: +// +// +-----------+ +// | status | sizeof(int) +// +-----------+ +// | psize | sizeof(int) +// +-----------+ +// | vsize | sizeof(int) +// +-----------+ +// | | | | +// ~ parameter ~ > psize | +// | | | > ((psize - 1)/sizeof(int) + 1) * sizeof(int) +// +-----------+ | +// | padding | | +// +-----------+ +// | | | +// ~ value ~ > vsize +// | | | +// +-----------+ + +typedef struct effect_param_s { + int32_t status; // Transaction status (unused for command, used for reply) + uint32_t psize; // Parameter size + uint32_t vsize; // Value size + char data[]; // Start of Parameter + Value data +} effect_param_t; + + +///////////////////////////////////////////////// +// Effect library interface +///////////////////////////////////////////////// + +// An effect library is required to implement and expose the following functions +// to enable effect enumeration and instantiation. The name of these functions must be as +// specified here as the effect framework will get the function address with dlsym(): +// +// - effect_QueryNumberEffects_t EffectQueryNumberEffects; +// - effect_QueryEffect_t EffectQueryEffect; +// - effect_CreateEffect_t EffectCreate; +// - effect_ReleaseEffect_t EffectRelease; + + +//////////////////////////////////////////////////////////////////////////////// +// +// Function: EffectQueryNumberEffects +// +// Description: Returns the number of different effects exposed by the +// library. Each effect must have a unique effect uuid (see +// effect_descriptor_t). This function together with EffectQueryEffect() +// is used to enumerate all effects present in the library. +// +// Input/Output: +// pNumEffects: address where the number of effects should be returned. +// +// Output: +// returned value: 0 successful operation. +// -ENODEV library failed to initialize +// -EINVAL invalid pNumEffects +// *pNumEffects: updated with number of effects in library +// +//////////////////////////////////////////////////////////////////////////////// +typedef int32_t (*effect_QueryNumberEffects_t)(uint32_t *pNumEffects); + +//////////////////////////////////////////////////////////////////////////////// +// +// Function: EffectQueryEffect +// +// Description: Returns the descriptor of the effect engine which index is +// given as first argument. +// See effect_descriptor_t for details on effect descriptors. +// This function together with EffectQueryNumberEffects() is used to enumerate all +// effects present in the library. The enumeration sequence is: +// EffectQueryNumberEffects(&num_effects); +// for (i = 0; i < num_effects; i++) +// EffectQueryEffect(i,...); +// +// Input/Output: +// index: index of the effect +// pDescriptor: address where to return the effect descriptor. +// +// Output: +// returned value: 0 successful operation. +// -ENODEV library failed to initialize +// -EINVAL invalid pDescriptor or index +// -ENOSYS effect list has changed since last execution of +// EffectQueryNumberEffects() +// -ENOENT no more effect available +// *pDescriptor: updated with the effect descriptor. +// +//////////////////////////////////////////////////////////////////////////////// +typedef int32_t (*effect_QueryEffect_t)(uint32_t index, + effect_descriptor_t *pDescriptor); + +//////////////////////////////////////////////////////////////////////////////// +// +// Function: EffectCreate +// +// Description: Creates an effect engine of the specified type and returns an +// effect control interface on this engine. The function will allocate the +// resources for an instance of the requested effect engine and return +// a handle on the effect control interface. +// +// Input: +// uuid: pointer to the effect uuid. +// sessionId: audio session to which this effect instance will be attached. All effects +// created with the same session ID are connected in series and process the same signal +// stream. Knowing that two effects are part of the same effect chain can help the +// library implement some kind of optimizations. +// ioId: identifies the output or input stream this effect is directed to at audio HAL. +// For future use especially with tunneled HW accelerated effects +// +// Input/Output: +// pInterface: address where to return the effect interface. +// +// Output: +// returned value: 0 successful operation. +// -ENODEV library failed to initialize +// -EINVAL invalid pEffectUuid or pInterface +// -ENOENT no effect with this uuid found +// *pInterface: updated with the effect interface handle. +// +//////////////////////////////////////////////////////////////////////////////// +typedef int32_t (*effect_CreateEffect_t)(effect_uuid_t *uuid, + int32_t sessionId, + int32_t ioId, + effect_interface_t *pInterface); + +//////////////////////////////////////////////////////////////////////////////// +// +// Function: EffectRelease +// +// Description: Releases the effect engine whose handle is given as argument. +// All resources allocated to this particular instance of the effect are +// released. +// +// Input: +// interface: handle on the effect interface to be released. +// +// Output: +// returned value: 0 successful operation. +// -ENODEV library failed to initialize +// -EINVAL invalid interface handle +// +//////////////////////////////////////////////////////////////////////////////// +typedef int32_t (*effect_ReleaseEffect_t)(effect_interface_t interface); + + +#if __cplusplus +} // extern "C" +#endif + + +#endif /*ANDROID_EFFECTAPI_H_*/ diff --git a/dom/system/gonk/android_audio/IAudioFlinger.h b/dom/system/gonk/android_audio/IAudioFlinger.h new file mode 100644 index 000000000..b10d3ab93 --- /dev/null +++ b/dom/system/gonk/android_audio/IAudioFlinger.h @@ -0,0 +1,184 @@ +/* + * Copyright (C) 2007 The Android Open Source Project + * + * 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 ANDROID_IAUDIOFLINGER_H +#define ANDROID_IAUDIOFLINGER_H + +#include +#include +#include + +#include +#include +#include +#include "IAudioTrack.h" +#include "IAudioRecord.h" +#include "IAudioFlingerClient.h" +#include "EffectApi.h" +#include "IEffect.h" +#include "IEffectClient.h" +#include + +namespace android { + +// ---------------------------------------------------------------------------- + +class IAudioFlinger : public IInterface +{ +public: + DECLARE_META_INTERFACE(AudioFlinger); + + /* create an audio track and registers it with AudioFlinger. + * return null if the track cannot be created. + */ + virtual sp createTrack( + pid_t pid, + int streamType, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + const sp& sharedBuffer, + int output, + int *sessionId, + status_t *status) = 0; + + virtual sp openRecord( + pid_t pid, + int input, + uint32_t sampleRate, + int format, + int channelCount, + int frameCount, + uint32_t flags, + int *sessionId, + status_t *status) = 0; + + /* query the audio hardware state. This state never changes, + * and therefore can be cached. + */ + virtual uint32_t sampleRate(int output) const = 0; + virtual int channelCount(int output) const = 0; + virtual int format(int output) const = 0; + virtual size_t frameCount(int output) const = 0; + virtual uint32_t latency(int output) const = 0; + + /* set/get the audio hardware state. This will probably be used by + * the preference panel, mostly. + */ + virtual status_t setMasterVolume(float value) = 0; + virtual status_t setMasterMute(bool muted) = 0; + + virtual float masterVolume() const = 0; + virtual bool masterMute() const = 0; + + /* set/get stream type state. This will probably be used by + * the preference panel, mostly. + */ + virtual status_t setStreamVolume(int stream, float value, int output) = 0; + virtual status_t setStreamMute(int stream, bool muted) = 0; + + virtual float streamVolume(int stream, int output) const = 0; + virtual bool streamMute(int stream) const = 0; + + // set audio mode + virtual status_t setMode(int mode) = 0; + + // mic mute/state + virtual status_t setMicMute(bool state) = 0; + virtual bool getMicMute() const = 0; + + // is any track active on this stream? + virtual bool isStreamActive(int stream) const = 0; + + virtual status_t setParameters(int ioHandle, const String8& keyValuePairs) = 0; + virtual String8 getParameters(int ioHandle, const String8& keys) = 0; + + // register a current process for audio output change notifications + virtual void registerClient(const sp& client) = 0; + + // retrieve the audio recording buffer size + virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0; + + virtual int openOutput(uint32_t *pDevices, + uint32_t *pSamplingRate, + uint32_t *pFormat, + uint32_t *pChannels, + uint32_t *pLatencyMs, + uint32_t flags) = 0; + virtual int openDuplicateOutput(int output1, int output2) = 0; + virtual status_t closeOutput(int output) = 0; + virtual status_t suspendOutput(int output) = 0; + virtual status_t restoreOutput(int output) = 0; + + virtual int openInput(uint32_t *pDevices, + uint32_t *pSamplingRate, + uint32_t *pFormat, + uint32_t *pChannels, + uint32_t acoustics) = 0; + virtual status_t closeInput(int input) = 0; + + virtual status_t setStreamOutput(uint32_t stream, int output) = 0; + + virtual status_t setVoiceVolume(float volume) = 0; + + virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) = 0; + + virtual unsigned int getInputFramesLost(int ioHandle) = 0; + + virtual int newAudioSessionId() = 0; + + virtual status_t loadEffectLibrary(const char *libPath, int *handle) = 0; + + virtual status_t unloadEffectLibrary(int handle) = 0; + + virtual status_t queryNumberEffects(uint32_t *numEffects) = 0; + + virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) = 0; + + virtual status_t getEffectDescriptor(effect_uuid_t *pEffectUUID, effect_descriptor_t *pDescriptor) = 0; + + virtual sp createEffect(pid_t pid, + effect_descriptor_t *pDesc, + const sp& client, + int32_t priority, + int output, + int sessionId, + status_t *status, + int *id, + int *enabled) = 0; + + virtual status_t moveEffects(int session, int srcOutput, int dstOutput) = 0; +}; + + +// ---------------------------------------------------------------------------- + +class BnAudioFlinger : public BnInterface +{ +public: + virtual status_t onTransact( uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags = 0); +}; + +// ---------------------------------------------------------------------------- + +}; // namespace android + +#endif // ANDROID_IAUDIOFLINGER_H diff --git a/dom/system/gonk/android_audio/IAudioFlingerClient.h b/dom/system/gonk/android_audio/IAudioFlingerClient.h new file mode 100644 index 000000000..aa0cdcff1 --- /dev/null +++ b/dom/system/gonk/android_audio/IAudioFlingerClient.h @@ -0,0 +1,55 @@ +/* + * Copyright (C) 2009 The Android Open Source Project + * + * 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 ANDROID_IAUDIOFLINGERCLIENT_H +#define ANDROID_IAUDIOFLINGERCLIENT_H + + +#include +#include +#include + +namespace android { + +// ---------------------------------------------------------------------------- + +class IAudioFlingerClient : public IInterface +{ +public: + DECLARE_META_INTERFACE(AudioFlingerClient); + + // Notifies a change of audio input/output configuration. + virtual void ioConfigChanged(int event, int ioHandle, void *param2) = 0; + +}; + + +// ---------------------------------------------------------------------------- + +class BnAudioFlingerClient : public BnInterface +{ +public: + virtual status_t onTransact( uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags = 0); +}; + +// ---------------------------------------------------------------------------- + +}; // namespace android + +#endif // ANDROID_IAUDIOFLINGERCLIENT_H diff --git a/dom/system/gonk/android_audio/IAudioRecord.h b/dom/system/gonk/android_audio/IAudioRecord.h new file mode 100644 index 000000000..46735def2 --- /dev/null +++ b/dom/system/gonk/android_audio/IAudioRecord.h @@ -0,0 +1,68 @@ +/* + * Copyright (C) 2007 The Android Open Source Project + * + * 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 IAUDIORECORD_H_ +#define IAUDIORECORD_H_ + +#include +#include + +#include +#include +#include +#include + + +namespace android { + +// ---------------------------------------------------------------------------- + +class IAudioRecord : public IInterface +{ +public: + DECLARE_META_INTERFACE(AudioRecord); + + /* After it's created the track is not active. Call start() to + * make it active. If set, the callback will start being called. + */ + virtual status_t start() = 0; + + /* Stop a track. If set, the callback will cease being called and + * obtainBuffer will return an error. Buffers that are already released + * will be processed, unless flush() is called. + */ + virtual void stop() = 0; + + /* get this tracks control block */ + virtual sp getCblk() const = 0; +}; + +// ---------------------------------------------------------------------------- + +class BnAudioRecord : public BnInterface +{ +public: + virtual status_t onTransact( uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags = 0); +}; + +// ---------------------------------------------------------------------------- + +}; // namespace android + +#endif /*IAUDIORECORD_H_*/ diff --git a/dom/system/gonk/android_audio/IAudioTrack.h b/dom/system/gonk/android_audio/IAudioTrack.h new file mode 100644 index 000000000..47d530be5 --- /dev/null +++ b/dom/system/gonk/android_audio/IAudioTrack.h @@ -0,0 +1,89 @@ +/* + * Copyright (C) 2007 The Android Open Source Project + * + * 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 ANDROID_IAUDIOTRACK_H +#define ANDROID_IAUDIOTRACK_H + +#include +#include + +#include +#include +#include +#include + + +namespace android { + +// ---------------------------------------------------------------------------- + +class IAudioTrack : public IInterface +{ +public: + DECLARE_META_INTERFACE(AudioTrack); + + /* After it's created the track is not active. Call start() to + * make it active. If set, the callback will start being called. + */ + virtual status_t start() = 0; + + /* Stop a track. If set, the callback will cease being called and + * obtainBuffer will return an error. Buffers that are already released + * will be processed, unless flush() is called. + */ + virtual void stop() = 0; + + /* flush a stopped track. All pending buffers are discarded. + * This function has no effect if the track is not stoped. + */ + virtual void flush() = 0; + + /* mute or unmutes this track. + * While mutted, the callback, if set, is still called. + */ + virtual void mute(bool) = 0; + + /* Pause a track. If set, the callback will cease being called and + * obtainBuffer will return an error. Buffers that are already released + * will be processed, unless flush() is called. + */ + virtual void pause() = 0; + + /* Attach track auxiliary output to specified effect. Use effectId = 0 + * to detach track from effect. + */ + virtual status_t attachAuxEffect(int effectId) = 0; + + /* get this tracks control block */ + virtual sp getCblk() const = 0; +}; + +// ---------------------------------------------------------------------------- + +class BnAudioTrack : public BnInterface +{ +public: + virtual status_t onTransact( uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags = 0); +}; + +// ---------------------------------------------------------------------------- + +}; // namespace android + +#endif // ANDROID_IAUDIOTRACK_H diff --git a/dom/system/gonk/android_audio/IEffect.h b/dom/system/gonk/android_audio/IEffect.h new file mode 100644 index 000000000..ff04869e0 --- /dev/null +++ b/dom/system/gonk/android_audio/IEffect.h @@ -0,0 +1,60 @@ +/* + * Copyright (C) 2010 The Android Open Source Project + * + * 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 ANDROID_IEFFECT_H +#define ANDROID_IEFFECT_H + +#include +#include +#include +#include + +namespace android { + +class IEffect: public IInterface +{ +public: + DECLARE_META_INTERFACE(Effect); + + virtual status_t enable() = 0; + + virtual status_t disable() = 0; + + virtual status_t command(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t *pReplySize, + void *pReplyData) = 0; + + virtual void disconnect() = 0; + + virtual sp getCblk() const = 0; +}; + +// ---------------------------------------------------------------------------- + +class BnEffect: public BnInterface +{ +public: + virtual status_t onTransact( uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags = 0); +}; + +}; // namespace android + +#endif // ANDROID_IEFFECT_H diff --git a/dom/system/gonk/android_audio/IEffectClient.h b/dom/system/gonk/android_audio/IEffectClient.h new file mode 100644 index 000000000..2f78c98f1 --- /dev/null +++ b/dom/system/gonk/android_audio/IEffectClient.h @@ -0,0 +1,54 @@ +/* + * Copyright (C) 2010 The Android Open Source Project + * + * 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 ANDROID_IEFFECTCLIENT_H +#define ANDROID_IEFFECTCLIENT_H + +#include +#include +#include +#include + +namespace android { + +class IEffectClient: public IInterface +{ +public: + DECLARE_META_INTERFACE(EffectClient); + + virtual void controlStatusChanged(bool controlGranted) = 0; + virtual void enableStatusChanged(bool enabled) = 0; + virtual void commandExecuted(uint32_t cmdCode, + uint32_t cmdSize, + void *pCmdData, + uint32_t replySize, + void *pReplyData) = 0; +}; + +// ---------------------------------------------------------------------------- + +class BnEffectClient: public BnInterface +{ +public: + virtual status_t onTransact( uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags = 0); +}; + +}; // namespace android + +#endif // ANDROID_IEFFECTCLIENT_H -- cgit v1.2.3