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()
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 @@
{
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 @@
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 @@
*
* 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 @@
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 @@
* 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 @@
int format() const;
int channelCount() const;
+ int channels() const;
uint32_t frameCount() const;
int frameSize() const;
int inputSource() const;
@@ -222,8 +212,8 @@
/* 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 @@
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 @@
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 @@
*/
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 @@
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 @@
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 @@
* 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 @@
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<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() {
+ AudioFlingerClient() {
}
-
+
// DeathRecipient
virtual void binderDied(const wp<IBinder>& 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);
+
+ 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 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<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);
+
+ size_t size() { return mParameters.size(); }
+
+private:
+ String8 mKeyValuePairs;
+ KeyedVector <String8, String8> 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 @@
* 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 @@
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 @@
AudioTrack( int streamType,
uint32_t sampleRate = 0,
int format = 0,
- int channelCount = 0,
+ int channels = 0,
const sp<IMemory>& sharedBuffer = 0,
uint32_t flags = 0,
callback_t cbf = 0,
@@ -169,13 +169,13 @@
* 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 @@
*/
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 @@
sp<AudioTrackThread> mAudioTrackThread;
float mVolume[2];
- uint32_t mSampleRate;
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
@@ -395,6 +404,7 @@
uint8_t mFormat;
uint8_t mChannelCount;
uint8_t mMuted;
+ uint32_t mChannels;
status_t mStatus;
uint32_t mLatency;
@@ -410,6 +420,7 @@
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 <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/IAudioFlingerClient.h>
-
+#include <utils/String8.h>
namespace android {
@@ -50,11 +50,12 @@
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
+ void *output,
status_t *status) = 0;
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int inputSource,
+ void *input,
uint32_t sampleRate,
int format,
int channelCount,
@@ -65,11 +66,11 @@
/* 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 @@
/* 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 @@
// 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<IAudioFlingerClient>& 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 <utils/RefBase.h>
#include <binder/IInterface.h>
-
+#include <utils/KeyedVector.h>
namespace android {
@@ -31,8 +31,8 @@
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 <stdint.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <binder/IInterface.h>
+#include <media/AudioSystem.h>
+
+
+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<IAudioPolicyService>
+{
+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 @@
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);
diff --git a/include/tts/TtsEngine.h b/include/tts/TtsEngine.h
index ed084ca..28b0d2f 100644
--- a/include/tts/TtsEngine.h
+++ b/include/tts/TtsEngine.h
@@ -43,7 +43,7 @@
// @param [inout] void *& - The userdata pointer set in the original
// synth call
// @param [in] uint32_t - Track sampling rate in Hz
-// @param [in] audio_format - The AudioSystem::audio_format enum
+// @param [in] uint32_t - The audio format
// @param [in] int - The number of channels
// @param [inout] int8_t *& - A buffer of audio data only valid during the
// execution of the callback
@@ -54,7 +54,7 @@
// TTS_CALLBACK_CONTINUE to indicate the synthesis must continue if
// there is more data to produce.
typedef tts_callback_status (synthDoneCB_t)(void *&, uint32_t,
- AudioSystem::audio_format, int, int8_t *&, size_t&, tts_synth_status);
+ uint32_t, int, int8_t *&, size_t&, tts_synth_status);
class TtsEngine;
extern "C" TtsEngine* getTtsEngine();