From c2f1f07084818942352c6bbfb36af9b6b330eb4e Mon Sep 17 00:00:00 2001 From: Eric Laurent Date: Fri, 17 Jul 2009 12:17:14 -0700 Subject: Fix issue 1795088 Improve audio routing code Initial commit for review. Integrated comments after patch set 1 review. Fixed lockup in AudioFlinger::ThreadBase::exit() Fixed lockup when playing tone with AudioPlocyService startTone() --- include/media/AudioRecord.h | 57 +++-- include/media/AudioSystem.h | 398 ++++++++++++++++++++++++++----- include/media/AudioTrack.h | 25 +- include/media/IAudioFlinger.h | 56 +++-- include/media/IAudioFlingerClient.h | 6 +- include/media/IAudioPolicyService.h | 90 +++++++ include/private/media/AudioTrackShared.h | 13 +- 7 files changed, 520 insertions(+), 125 deletions(-) create mode 100644 include/media/IAudioPolicyService.h (limited to 'include') diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h index 83ff508..503cb31 100644 --- a/include/media/AudioRecord.h +++ b/include/media/AudioRecord.h @@ -39,21 +39,10 @@ class AudioRecord { public: - // input sources values must always be defined in the range - // [AudioRecord::DEFAULT_INPUT, AudioRecord::NUM_INPUT_SOURCES[ - enum input_source { - DEFAULT_INPUT =-1, - MIC_INPUT = 0, - VOICE_UPLINK_INPUT = 1, - VOICE_DOWNLINK_INPUT = 2, - VOICE_CALL_INPUT = 3, - NUM_INPUT_SOURCES - }; - static const int DEFAULT_SAMPLE_RATE = 8000; /* Events used by AudioRecord callback function (callback_t). - * + * * to keep in sync with frameworks/base/media/java/android/media/AudioRecord.java */ enum event_type { @@ -61,7 +50,7 @@ public: EVENT_OVERRUN = 1, // PCM buffer overrun occured. EVENT_MARKER = 2, // Record head is at the specified marker position // (See setMarkerPosition()). - EVENT_NEW_POS = 3, // Record head is at a new position + EVENT_NEW_POS = 3, // Record head is at a new position // (See setPositionUpdatePeriod()). }; @@ -123,11 +112,11 @@ public: * * Parameters: * - * inputSource: Select the audio input to record to (e.g. AudioRecord::MIC_INPUT). + * inputSource: Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT). * sampleRate: Track sampling rate in Hz. - * format: PCM sample format (e.g AudioSystem::PCM_16_BIT for signed + * format: Audio format (e.g AudioSystem::PCM_16_BIT for signed * 16 bits per sample). - * channelCount: Number of PCM channels (e.g 2 for stereo). + * channels: Channel mask: see AudioSystem::audio_channels. * frameCount: Total size of track PCM buffer in frames. This defines the * latency of the track. * flags: A bitmask of acoustic values from enum record_flags. It enables @@ -148,7 +137,7 @@ public: AudioRecord(int inputSource, uint32_t sampleRate = 0, int format = 0, - int channelCount = 0, + uint32_t channels = AudioSystem::CHANNEL_IN_MONO, int frameCount = 0, uint32_t flags = 0, callback_t cbf = 0, @@ -166,14 +155,14 @@ public: * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful intialization * - INVALID_OPERATION: AudioRecord is already intitialized or record device is already in use - * - BAD_VALUE: invalid parameter (channelCount, format, sampleRate...) + * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) * - NO_INIT: audio server or audio hardware not initialized * - PERMISSION_DENIED: recording is not allowed for the requesting process * */ status_t set(int inputSource = 0, uint32_t sampleRate = 0, int format = 0, - int channelCount = 0, + uint32_t channels = AudioSystem::CHANNEL_IN_MONO, int frameCount = 0, uint32_t flags = 0, callback_t cbf = 0, @@ -199,6 +188,7 @@ public: int format() const; int channelCount() const; + int channels() const; uint32_t frameCount() const; int frameSize() const; int inputSource() const; @@ -222,8 +212,8 @@ public: /* Sets marker position. When record 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 AudioRecord has been opened with no callback function associated, + * with marker == 0 cancels marker notification callback. + * If the AudioRecord has been opened with no callback function associated, * the operation will fail. * * Parameters: @@ -238,10 +228,10 @@ public: status_t getMarkerPosition(uint32_t *marker); - /* Sets position update period. Every time the number of frames specified has been recorded, - * a callback with event type EVENT_NEW_POS is called. - * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification - * callback. + /* Sets position update period. Every time the number of frames specified has been recorded, + * a callback with event type EVENT_NEW_POS is called. + * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification + * callback. * If the AudioRecord has been opened with no callback function associated, * the operation will fail. * @@ -257,8 +247,8 @@ public: status_t getPositionUpdatePeriod(uint32_t *updatePeriod); - /* Gets record head position. The position is the total number of frames - * recorded since record start. + /* Gets record head position. The position is the total number of frames + * recorded since record start. * * Parameters: * @@ -270,8 +260,16 @@ public: */ status_t getPosition(uint32_t *position); - - + /* returns a handle on the audio input used by this AudioRecord. + * + * Parameters: + * none. + * + * Returned value: + * handle on audio hardware input + */ + audio_io_handle_t getInput() { return mInput; } + /* 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, @@ -342,6 +340,7 @@ private: bool mMarkerReached; uint32_t mNewPosition; uint32_t mUpdatePeriod; + audio_io_handle_t mInput; }; }; // namespace android diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h index 3a3a714..0ea04a4 100644 --- a/include/media/AudioSystem.h +++ b/include/media/AudioSystem.h @@ -24,36 +24,130 @@ namespace android { typedef void (*audio_error_callback)(status_t err); +typedef void * 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, + 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 }; - enum audio_output_type { - AUDIO_OUTPUT_DEFAULT =-1, - AUDIO_OUTPUT_HARDWARE = 0, - AUDIO_OUTPUT_A2DP = 1, - NUM_AUDIO_OUTPUT_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 { - FORMAT_DEFAULT = 0, - PCM_16_BIT, - PCM_8_BIT, - INVALID_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 = 0x1, + CHANNEL_OUT_FRONT_RIGHT = 0x2, + CHANNEL_OUT_FRONT_CENTER = 0x4, + CHANNEL_OUT_LOW_FREQUENCY = 0x8, + CHANNEL_OUT_BACK_LEFT = 0x10, + CHANNEL_OUT_BACK_RIGHT = 0x20, + CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x40, + CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80, + CHANNEL_OUT_BACK_CENTER = 0x100, + 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 = 0x10000, + CHANNEL_IN_RIGHT = 0x20000, + CHANNEL_IN_FRONT = 0x40000, + CHANNEL_IN_BACK = 0x80000, + CHANNEL_IN_LEFT_PROCESSED = 0x100000, + CHANNEL_IN_RIGHT_PROCESSED = 0x200000, + CHANNEL_IN_FRONT_PROCESSED = 0x400000, + CHANNEL_IN_BACK_PROCESSED = 0x800000, + CHANNEL_IN_PRESSURE = 0x1000000, + CHANNEL_IN_X_AXIS = 0x2000000, + CHANNEL_IN_Y_AXIS = 0x4000000, + CHANNEL_IN_Z_AXIS = 0x8000000, + CHANNEL_IN_VOICE_UPLINK = 0x10000000, + CHANNEL_IN_VOICE_DNLINK = 0x20000000, + 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 { @@ -65,15 +159,6 @@ public: NUM_MODES // not a valid entry, denotes end-of-list }; - enum audio_routes { - ROUTE_EARPIECE = (1 << 0), - ROUTE_SPEAKER = (1 << 1), - ROUTE_BLUETOOTH_SCO = (1 << 2), - ROUTE_HEADSET = (1 << 3), - ROUTE_BLUETOOTH_A2DP = (1 << 4), - ROUTE_ALL = -1UL, - }; - enum audio_in_acoustics { AGC_ENABLE = 0x0001, AGC_DISABLE = 0, @@ -87,36 +172,37 @@ public: * only privileged processes can have access to them */ - // routing helper functions - static status_t speakerphone(bool state); - static status_t isSpeakerphoneOn(bool* state); - static status_t bluetoothSco(bool state); - static status_t isBluetoothScoOn(bool* state); + // 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 setMasterMute(bool mute); static status_t getMasterVolume(float* volume); + // mute/unmute audio outputs + static status_t setMasterMute(bool mute); static status_t getMasterMute(bool* mute); - static status_t setStreamVolume(int stream, float value); + // set/get stream volume on specified output + static status_t setStreamVolume(int stream, float value, void *output); + static status_t getStreamVolume(int stream, float* volume, void *output); + + // mute/unmute stream static status_t setStreamMute(int stream, bool mute); - static status_t getStreamVolume(int stream, float* volume); static status_t getStreamMute(int stream, bool* mute); + // set audio mode in audio hardware (see AudioSystem::audio_mode) static status_t setMode(int mode); - static status_t getMode(int* mode); - - static status_t setRouting(int mode, uint32_t routes, uint32_t mask); - static status_t getRouting(int mode, uint32_t* routes); + // returns true if tracks are active on AudioSystem::MUSIC stream static status_t isMusicActive(bool *state); - // Temporary interface, do not use - // TODO: Replace with a more generic key:value get/set mechanism - static status_t setParameter(const char* key, const char* value); - + // 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 @@ -130,47 +216,247 @@ public: 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, + + static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, size_t* buffSize); + + // + // 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_FM_HEADPHONE = 0x800, + DEVICE_OUT_FM_SPEAKER = 0x1000, + DEVICE_OUT_TTY = 0x2000, + 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_FM_HEADPHONE | + DEVICE_OUT_FM_SPEAKER | DEVICE_OUT_TTY | DEVICE_OUT_DEFAULT), + + // 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_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_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, + NUM_FORCE_CONFIG, + FORCE_DEFAULT = FORCE_NONE + }; + + // usages used for setForceUse() + enum force_use { + FOR_COMMUNICATION, + FOR_MEDIA, + FOR_RECORD, + 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); + static status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream); + 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 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 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() { + AudioFlingerClient() { } - + // DeathRecipient virtual void binderDied(const wp& who); - + // IAudioFlingerClient - virtual void a2dpEnabledChanged(bool enabled); - + + // 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, void *param1, void *param2); }; - static int getOutput(int streamType); - static sp gAudioFlingerClient; + 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 int gOutSamplingRate[NUM_AUDIO_OUTPUT_TYPES]; - static int gOutFrameCount[NUM_AUDIO_OUTPUT_TYPES]; - static uint32_t gOutLatency[NUM_AUDIO_OUTPUT_TYPES]; - static bool gA2dpEnabled; - + 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 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); + + size_t size() { return mParameters.size(); } + +private: + String8 mKeyValuePairs; + KeyedVector mParameters; }; }; // namespace android diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h index 2e1fbda..981c2f6 100644 --- a/include/media/AudioTrack.h +++ b/include/media/AudioTrack.h @@ -117,9 +117,9 @@ public: * streamType: Select the type of audio stream this track is attached to * (e.g. AudioSystem::MUSIC). * sampleRate: Track sampling rate in Hz. - * format: PCM sample format (e.g AudioSystem::PCM_16_BIT for signed + * format: Audio format (e.g AudioSystem::PCM_16_BIT for signed * 16 bits per sample). - * channelCount: Number of PCM channels (e.g 2 for stereo). + * 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. @@ -133,7 +133,7 @@ public: AudioTrack( int streamType, uint32_t sampleRate = 0, int format = 0, - int channelCount = 0, + int channels = 0, int frameCount = 0, uint32_t flags = 0, callback_t cbf = 0, @@ -152,7 +152,7 @@ public: AudioTrack( int streamType, uint32_t sampleRate = 0, int format = 0, - int channelCount = 0, + int channels = 0, const sp& sharedBuffer = 0, uint32_t flags = 0, callback_t cbf = 0, @@ -169,13 +169,13 @@ public: * Returned status (from utils/Errors.h) can be: * - NO_ERROR: successful intialization * - INVALID_OPERATION: AudioTrack is already intitialized - * - BAD_VALUE: invalid parameter (channelCount, format, sampleRate...) + * - 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 channelCount = 0, + int channels = 0, int frameCount = 0, uint32_t flags = 0, callback_t cbf = 0, @@ -330,6 +330,16 @@ public: */ 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(); + /* 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, @@ -387,7 +397,6 @@ private: sp mAudioTrackThread; float mVolume[2]; - uint32_t mSampleRate; uint32_t mFrameCount; audio_track_cblk_t* mCblk; @@ -395,6 +404,7 @@ private: uint8_t mFormat; uint8_t mChannelCount; uint8_t mMuted; + uint32_t mChannels; status_t mStatus; uint32_t mLatency; @@ -410,6 +420,7 @@ private: bool mMarkerReached; uint32_t mNewPosition; uint32_t mUpdatePeriod; + uint32_t mFlags; }; diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h index bac3d29..26e6972 100644 --- a/include/media/IAudioFlinger.h +++ b/include/media/IAudioFlinger.h @@ -27,7 +27,7 @@ #include #include #include - +#include namespace android { @@ -50,11 +50,12 @@ public: int frameCount, uint32_t flags, const sp& sharedBuffer, + void *output, status_t *status) = 0; virtual sp openRecord( pid_t pid, - int inputSource, + void *input, uint32_t sampleRate, int format, int channelCount, @@ -65,11 +66,11 @@ public: /* 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; + virtual uint32_t sampleRate(void *output) const = 0; + virtual int channelCount(void *output) const = 0; + virtual int format(void *output) const = 0; + virtual size_t frameCount(void *output) const = 0; + virtual uint32_t latency(void *output) const = 0; /* set/get the audio hardware state. This will probably be used by * the preference panel, mostly. @@ -83,19 +84,14 @@ public: /* set/get stream type state. This will probably be used by * the preference panel, mostly. */ - virtual status_t setStreamVolume(int stream, float value) = 0; + virtual status_t setStreamVolume(int stream, float value, void *output) = 0; virtual status_t setStreamMute(int stream, bool muted) = 0; - virtual float streamVolume(int stream) const = 0; + virtual float streamVolume(int stream, void *output) const = 0; virtual bool streamMute(int stream) const = 0; - // set/get audio routing - virtual status_t setRouting(int mode, uint32_t routes, uint32_t mask) = 0; - virtual uint32_t getRouting(int mode) const = 0; - - // set/get audio mode + // set audio mode virtual status_t setMode(int mode) = 0; - virtual int getMode() const = 0; // mic mute/state virtual status_t setMicMute(bool state) = 0; @@ -104,22 +100,34 @@ public: // is a music stream active? virtual bool isMusicActive() const = 0; - // pass a generic configuration parameter to libaudio - // Temporary interface, do not use - // TODO: Replace with a more generic key:value get/set mechanism - virtual status_t setParameter(const char* key, const char* value) = 0; + virtual status_t setParameters(void *ioHandle, const String8& keyValuePairs) = 0; + virtual String8 getParameters(void *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; - - // force AudioFlinger thread out of standby - virtual void wakeUp() = 0; - // is A2DP output enabled - virtual bool isA2dpEnabled() const = 0; + virtual void *openOutput(uint32_t *pDevices, + uint32_t *pSamplingRate, + uint32_t *pFormat, + uint32_t *pChannels, + uint32_t *pLatencyMs, + uint32_t flags) = 0; + virtual void *openDuplicateOutput(void *output1, void *output2) = 0; + virtual status_t closeOutput(void *output) = 0; + virtual status_t suspendOutput(void *output) = 0; + virtual status_t restoreOutput(void *output) = 0; + + virtual void *openInput(uint32_t *pDevices, + uint32_t *pSamplingRate, + uint32_t *pFormat, + uint32_t *pChannels, + uint32_t acoustics) = 0; + virtual status_t closeInput(void *input) = 0; + + virtual status_t setStreamOutput(uint32_t stream, void *output) = 0; }; diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h index 383ec0c..78142ce 100644 --- a/include/media/IAudioFlingerClient.h +++ b/include/media/IAudioFlingerClient.h @@ -20,7 +20,7 @@ #include #include - +#include namespace android { @@ -31,8 +31,8 @@ class IAudioFlingerClient : public IInterface public: DECLARE_META_INTERFACE(AudioFlingerClient); - // Notifies a change of audio output from/to hardware to/from A2DP. - virtual void a2dpEnabledChanged(bool enabled) = 0; + // Notifies a change of audio input/output configuration. + virtual void ioConfigChanged(int event, void *param1, void *param2) = 0; }; diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h new file mode 100644 index 0000000..4804bbd --- /dev/null +++ b/include/media/IAudioPolicyService.h @@ -0,0 +1,90 @@ +/* + * 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_IAUDIOPOLICYSERVICE_H +#define ANDROID_IAUDIOPOLICYSERVICE_H + +#include +#include +#include + +#include +#include +#include +#include + + +namespace android { + +// ---------------------------------------------------------------------------- + +class IAudioPolicyService : public IInterface +{ +public: + DECLARE_META_INTERFACE(AudioPolicyService); + + // + // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) + // + virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device, + AudioSystem::device_connection_state state, + const char *device_address) = 0; + virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device, + const char *device_address) = 0; + virtual status_t setPhoneState(int state) = 0; + virtual status_t setRingerMode(uint32_t mode, uint32_t mask) = 0; + virtual status_t setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0; + virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0; + virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream, + uint32_t samplingRate = 0, + uint32_t format = AudioSystem::FORMAT_DEFAULT, + uint32_t channels = 0, + AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT) = 0; + virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0; + virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0; + virtual void releaseOutput(audio_io_handle_t output) = 0; + virtual audio_io_handle_t getInput(int inputSource, + uint32_t samplingRate = 0, + uint32_t format = AudioSystem::FORMAT_DEFAULT, + uint32_t channels = 0, + AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0) = 0; + virtual status_t startInput(audio_io_handle_t input) = 0; + virtual status_t stopInput(audio_io_handle_t input) = 0; + virtual void releaseInput(audio_io_handle_t input) = 0; + virtual status_t initStreamVolume(AudioSystem::stream_type stream, + int indexMin, + int indexMax) = 0; + virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) = 0; + virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) = 0; +}; + + +// ---------------------------------------------------------------------------- + +class BnAudioPolicyService : public BnInterface +{ +public: + virtual status_t onTransact( uint32_t code, + const Parcel& data, + Parcel* reply, + uint32_t flags = 0); +}; + +// ---------------------------------------------------------------------------- + +}; // namespace android + +#endif // ANDROID_IAUDIOPOLICYSERVICE_H diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h index 496a739..8e2db20 100644 --- a/include/private/media/AudioTrackShared.h +++ b/include/private/media/AudioTrackShared.h @@ -55,17 +55,18 @@ struct audio_track_cblk_t uint32_t volumeLR; }; uint32_t sampleRate; + // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for + // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of + // 16 bit because data is converted to 16 bit before being stored in buffer + uint32_t frameSize; uint8_t channels; uint8_t flowControlFlag; // underrun (out) or overrrun (in) indication uint8_t out; // out equals 1 for AudioTrack and 0 for AudioRecord - uint8_t forceReady; + uint8_t forceReady; uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger uint16_t waitTimeMs; // Cumulated wait time - // Padding ensuring that data buffer starts on a cache line boundary (32 bytes). - // See AudioFlinger::TrackBase constructor - int32_t Padding[1]; - // Cache line boundary - + // Cache line boundary (32 bytes) + audio_track_cblk_t(); uint32_t stepUser(uint32_t frameCount); bool stepServer(uint32_t frameCount); -- cgit v1.1