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+/*
+ * 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_
+
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+#include <media/IAudioFlinger.h>
+
+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,
+ 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,
+ 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,
+ 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<IAudioFlinger>& 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,
+ 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);
+ 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 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);
+
+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 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 changeing 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
+ static const char *keyRouting;
+ static const char *keySamplingRate;
+ static const char *keyFormat;
+ static const char *keyChannels;
+ static const char *keyFrameCount;
+
+ 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_*/