539 lines
22 KiB
C
539 lines
22 KiB
C
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/*
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* Copyright (C) 2008 The Android Open Source Project
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* Copyright (c) 2009-2011, The Linux Foundation. All rights reserved.
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef ANDROID_AUDIOSYSTEM_H_
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#define ANDROID_AUDIOSYSTEM_H_
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#include <utils/RefBase.h>
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#include <utils/threads.h>
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#include <media/IAudioFlinger.h>
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namespace android {
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typedef void (*audio_error_callback)(status_t err);
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typedef int audio_io_handle_t;
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class IAudioPolicyService;
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class String8;
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class AudioSystem
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{
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public:
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enum stream_type {
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DEFAULT =-1,
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VOICE_CALL = 0,
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SYSTEM = 1,
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RING = 2,
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MUSIC = 3,
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ALARM = 4,
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NOTIFICATION = 5,
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BLUETOOTH_SCO = 6,
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ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker
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DTMF = 8,
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TTS = 9,
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FM = 10,
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NUM_STREAM_TYPES
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};
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// Audio sub formats (see AudioSystem::audio_format).
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enum pcm_sub_format {
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PCM_SUB_16_BIT = 0x1, // must be 1 for backward compatibility
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PCM_SUB_8_BIT = 0x2, // must be 2 for backward compatibility
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};
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// MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify
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// bit rate, stereo mode, version...
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enum mp3_sub_format {
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//TODO
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};
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// AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned,
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// encoding mode for recording...
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enum amr_sub_format {
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//TODO
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};
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// AAC sub format field definition: specify profile or bitrate for recording...
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enum aac_sub_format {
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//TODO
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};
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// VORBIS sub format field definition: specify quality for recording...
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enum vorbis_sub_format {
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//TODO
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};
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// Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits).
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// The main format indicates the main codec type. The sub format field indicates options and parameters
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// for each format. The sub format is mainly used for record to indicate for instance the requested bitrate
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// or profile. It can also be used for certain formats to give informations not present in the encoded
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// audio stream (e.g. octet alignement for AMR).
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enum audio_format {
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INVALID_FORMAT = -1,
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FORMAT_DEFAULT = 0,
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PCM = 0x00000000, // must be 0 for backward compatibility
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MP3 = 0x01000000,
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AMR_NB = 0x02000000,
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AMR_WB = 0x03000000,
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AAC = 0x04000000,
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HE_AAC_V1 = 0x05000000,
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HE_AAC_V2 = 0x06000000,
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VORBIS = 0x07000000,
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EVRC = 0x08000000,
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QCELP = 0x09000000,
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VOIP_PCM_INPUT = 0x0A000000,
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MAIN_FORMAT_MASK = 0xFF000000,
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SUB_FORMAT_MASK = 0x00FFFFFF,
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// Aliases
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PCM_16_BIT = (PCM|PCM_SUB_16_BIT),
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PCM_8_BIT = (PCM|PCM_SUB_8_BIT)
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};
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// Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java
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enum audio_channels {
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// output channels
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CHANNEL_OUT_FRONT_LEFT = 0x4,
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CHANNEL_OUT_FRONT_RIGHT = 0x8,
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CHANNEL_OUT_FRONT_CENTER = 0x10,
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CHANNEL_OUT_LOW_FREQUENCY = 0x20,
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CHANNEL_OUT_BACK_LEFT = 0x40,
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CHANNEL_OUT_BACK_RIGHT = 0x80,
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CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100,
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CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,
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CHANNEL_OUT_BACK_CENTER = 0x400,
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CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT,
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CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT),
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CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
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CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
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CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
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CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER),
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CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
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CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT),
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CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
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CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
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CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),
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CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
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CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
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CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER),
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// input channels
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CHANNEL_IN_LEFT = 0x4,
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CHANNEL_IN_RIGHT = 0x8,
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CHANNEL_IN_FRONT = 0x10,
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CHANNEL_IN_BACK = 0x20,
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CHANNEL_IN_LEFT_PROCESSED = 0x40,
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CHANNEL_IN_RIGHT_PROCESSED = 0x80,
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CHANNEL_IN_FRONT_PROCESSED = 0x100,
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CHANNEL_IN_BACK_PROCESSED = 0x200,
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CHANNEL_IN_PRESSURE = 0x400,
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CHANNEL_IN_X_AXIS = 0x800,
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CHANNEL_IN_Y_AXIS = 0x1000,
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CHANNEL_IN_Z_AXIS = 0x2000,
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CHANNEL_IN_VOICE_UPLINK = 0x4000,
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CHANNEL_IN_VOICE_DNLINK = 0x8000,
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CHANNEL_IN_MONO = CHANNEL_IN_FRONT,
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CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT),
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CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK|
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CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED|
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CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS |
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CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)
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};
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enum audio_mode {
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MODE_INVALID = -2,
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MODE_CURRENT = -1,
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MODE_NORMAL = 0,
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MODE_RINGTONE,
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MODE_IN_CALL,
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MODE_IN_COMMUNICATION,
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NUM_MODES // not a valid entry, denotes end-of-list
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};
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enum audio_in_acoustics {
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AGC_ENABLE = 0x0001,
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AGC_DISABLE = 0,
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NS_ENABLE = 0x0002,
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NS_DISABLE = 0,
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TX_IIR_ENABLE = 0x0004,
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TX_DISABLE = 0
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};
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// special audio session values
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enum audio_sessions {
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SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream
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// (value must be less than 0)
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SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can
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// be moved by audio policy manager to another output stream
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// (value must be 0)
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};
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/* These are static methods to control the system-wide AudioFlinger
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* only privileged processes can have access to them
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*/
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// mute/unmute microphone
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static status_t muteMicrophone(bool state);
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static status_t isMicrophoneMuted(bool *state);
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// set/get master volume
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static status_t setMasterVolume(float value);
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static status_t getMasterVolume(float* volume);
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// mute/unmute audio outputs
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static status_t setMasterMute(bool mute);
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static status_t getMasterMute(bool* mute);
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// set/get stream volume on specified output
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static status_t setStreamVolume(int stream, float value, int output);
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static status_t getStreamVolume(int stream, float* volume, int output);
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// mute/unmute stream
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static status_t setStreamMute(int stream, bool mute);
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static status_t getStreamMute(int stream, bool* mute);
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// set audio mode in audio hardware (see AudioSystem::audio_mode)
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static status_t setMode(int mode);
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// returns true in *state if tracks are active on the specified stream
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static status_t isStreamActive(int stream, bool *state);
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// set/get audio hardware parameters. The function accepts a list of parameters
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// key value pairs in the form: key1=value1;key2=value2;...
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// Some keys are reserved for standard parameters (See AudioParameter class).
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static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
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static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
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static void setErrorCallback(audio_error_callback cb);
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// helper function to obtain AudioFlinger service handle
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static const sp<IAudioFlinger>& get_audio_flinger();
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static float linearToLog(int volume);
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static int logToLinear(float volume);
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static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT);
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static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT);
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static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT);
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static bool routedToA2dpOutput(int streamType);
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static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
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size_t* buffSize);
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static status_t setVoiceVolume(float volume);
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static status_t setFmVolume(float volume);
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// return the number of audio frames written by AudioFlinger to audio HAL and
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// audio dsp to DAC since the output on which the specificed stream is playing
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// has exited standby.
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// returned status (from utils/Errors.h) can be:
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// - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
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// - INVALID_OPERATION: Not supported on current hardware platform
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// - BAD_VALUE: invalid parameter
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// NOTE: this feature is not supported on all hardware platforms and it is
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// necessary to check returned status before using the returned values.
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static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT);
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static unsigned int getInputFramesLost(audio_io_handle_t ioHandle);
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static int newAudioSessionId();
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//
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// AudioPolicyService interface
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//
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enum audio_devices {
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// output devices
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DEVICE_OUT_EARPIECE = 0x1,
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DEVICE_OUT_SPEAKER = 0x2,
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DEVICE_OUT_WIRED_HEADSET = 0x4,
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DEVICE_OUT_WIRED_HEADPHONE = 0x8,
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DEVICE_OUT_BLUETOOTH_SCO = 0x10,
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DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20,
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DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40,
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DEVICE_OUT_BLUETOOTH_A2DP = 0x80,
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DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
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DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200,
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DEVICE_OUT_AUX_DIGITAL = 0x400,
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DEVICE_OUT_FM = 0x800,
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DEVICE_OUT_ANC_HEADSET = 0x1000,
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DEVICE_OUT_ANC_HEADPHONE = 0x2000,
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DEVICE_OUT_FM_TX = 0x4000,
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DEVICE_OUT_DEFAULT = 0x8000,
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// Since no free bits available in output device , using free bits from input device list
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DEVICE_OUT_DIRECTOUTPUT = 0x8000000,
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DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET |
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DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_FM | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
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DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
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DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_FM |
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DEVICE_OUT_ANC_HEADSET | DEVICE_OUT_ANC_HEADPHONE | DEVICE_OUT_FM_TX | DEVICE_OUT_DIRECTOUTPUT |
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DEVICE_OUT_DEFAULT),
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DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
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DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
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// input devices
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DEVICE_IN_COMMUNICATION = 0x10000,
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DEVICE_IN_AMBIENT = 0x20000,
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DEVICE_IN_BUILTIN_MIC = 0x40000,
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DEVICE_IN_VOICE_CALL = 0x80000,
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DEVICE_IN_BACK_MIC = 0x100000,
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DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x200000,
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DEVICE_IN_WIRED_HEADSET = 0x400000,
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DEVICE_IN_AUX_DIGITAL = 0x800000,
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DEVICE_IN_ANC_HEADSET = 0x1000000,
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DEVICE_IN_FM_RX = 0x2000000,
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DEVICE_IN_FM_RX_A2DP = 0x4000000,
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DEVICE_IN_DEFAULT = 0x80000000,
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DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |
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DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |
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DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_FM_RX | DEVICE_IN_FM_RX_A2DP | DEVICE_IN_ANC_HEADSET | DEVICE_IN_DEFAULT)
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};
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// device connection states used for setDeviceConnectionState()
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enum device_connection_state {
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DEVICE_STATE_UNAVAILABLE,
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DEVICE_STATE_AVAILABLE,
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NUM_DEVICE_STATES
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};
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// request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)
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enum output_flags {
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OUTPUT_FLAG_INDIRECT = 0x0,
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OUTPUT_FLAG_DIRECT = 0x1,
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OUTPUT_FLAG_SESSION = 0x2
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};
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// device categories used for setForceUse()
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enum forced_config {
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FORCE_NONE,
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FORCE_SPEAKER,
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FORCE_HEADPHONES,
|
||
|
|
FORCE_BT_SCO,
|
||
|
|
FORCE_BT_A2DP,
|
||
|
|
FORCE_WIRED_ACCESSORY,
|
||
|
|
FORCE_BT_CAR_DOCK,
|
||
|
|
FORCE_BT_DESK_DOCK,
|
||
|
|
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,
|
||
|
|
A2DP_OUTPUT_STATE,
|
||
|
|
EFFECT_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);
|
||
|
|
static status_t setRingerMode(uint32_t mode, uint32_t mask);
|
||
|
|
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 audio_io_handle_t getSession(stream_type stream,
|
||
|
|
uint32_t format = FORMAT_DEFAULT,
|
||
|
|
output_flags flags = OUTPUT_FLAG_DIRECT,
|
||
|
|
int32_t sessionId = -1);
|
||
|
|
static void closeSession(audio_io_handle_t output);
|
||
|
|
static status_t pauseSession(audio_io_handle_t output, stream_type stream);
|
||
|
|
static status_t resumeSession(audio_io_handle_t output, stream_type stream);
|
||
|
|
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 setStreamVolumeIndex(stream_type stream, int index);
|
||
|
|
static status_t getStreamVolumeIndex(stream_type stream, int *index);
|
||
|
|
|
||
|
|
static uint32_t getStrategyForStream(stream_type stream);
|
||
|
|
|
||
|
|
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<IAudioPolicyService>& 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 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();
|
||
|
|
|
||
|
|
private:
|
||
|
|
|
||
|
|
class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient
|
||
|
|
{
|
||
|
|
public:
|
||
|
|
AudioFlingerClient() {
|
||
|
|
}
|
||
|
|
|
||
|
|
// DeathRecipient
|
||
|
|
virtual void binderDied(const wp<IBinder>& 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<IBinder>& who);
|
||
|
|
};
|
||
|
|
|
||
|
|
static sp<AudioFlingerClient> gAudioFlingerClient;
|
||
|
|
static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient;
|
||
|
|
friend class AudioFlingerClient;
|
||
|
|
friend class AudioPolicyServiceClient;
|
||
|
|
|
||
|
|
static Mutex gLock;
|
||
|
|
static sp<IAudioFlinger> 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 int gPhoneState;
|
||
|
|
|
||
|
|
static sp<IAudioPolicyService> gAudioPolicyService;
|
||
|
|
|
||
|
|
// mapping between stream types and outputs
|
||
|
|
static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
|
||
|
|
// list of output descritor containing cached parameters (sampling rate, framecount, channel count...)
|
||
|
|
static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> 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 <String8, String8> mParameters;
|
||
|
|
};
|
||
|
|
|
||
|
|
}; // namespace android
|
||
|
|
|
||
|
|
#endif /*ANDROID_AUDIOSYSTEM_H_*/
|