audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
diff --git a/include/media/AudioParameter.h b/include/media/AudioParameter.h
index dc0bd2e..79d5d82 100644
--- a/include/media/AudioParameter.h
+++ b/include/media/AudioParameter.h
@@ -33,12 +33,12 @@
// 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
+ // keyRouting: to change audio routing, value is an int in audio_devices_t
// 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
+ // keyFormat: to change audio format, value is an int in audio_format_t
+ // keyChannels: to change audio channel configuration, value is an int in audio_channels_t
// keyFrameCount: to change audio output frame count, value is an int
- // keyInputSource: to change audio input source, value is an int in audio_source
+ // keyInputSource: to change audio input source, value is an int in audio_source_t
// (defined in media/mediarecorder.h)
static const char *keyRouting;
static const char *keySamplingRate;
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 293764d..def3612 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -30,6 +30,7 @@
#include <binder/IMemory.h>
#include <utils/threads.h>
+#include <hardware/audio.h>
namespace android {
@@ -127,9 +128,9 @@
*
* inputSource: Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT).
* sampleRate: Track sampling rate in Hz.
- * format: Audio format (e.g AudioSystem::PCM_16_BIT for signed
+ * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
- * channels: Channel mask: see AudioSystem::audio_channels.
+ * channels: Channel mask: see audio_channels_t.
* 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
@@ -142,15 +143,15 @@
*/
enum record_flags {
- RECORD_AGC_ENABLE = AudioSystem::AGC_ENABLE,
- RECORD_NS_ENABLE = AudioSystem::NS_ENABLE,
- RECORD_IIR_ENABLE = AudioSystem::TX_IIR_ENABLE
+ RECORD_AGC_ENABLE = AUDIO_IN_ACOUSTICS_AGC_ENABLE,
+ RECORD_NS_ENABLE = AUDIO_IN_ACOUSTICS_NS_ENABLE,
+ RECORD_IIR_ENABLE = AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE,
};
AudioRecord(int inputSource,
uint32_t sampleRate = 0,
int format = 0,
- uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
+ uint32_t channels = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
@@ -176,7 +177,7 @@
status_t set(int inputSource = 0,
uint32_t sampleRate = 0,
int format = 0,
- uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
+ uint32_t channels = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
callback_t cbf = 0,
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index b0c82d8..eb61a87 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -21,13 +21,15 @@
#include <utils/threads.h>
#include <media/IAudioFlinger.h>
+#include <hardware/audio.h>
+#include <hardware/audio_policy.h>
+
/* XXX: Should be include by all the users instead */
#include <media/AudioParameter.h>
namespace android {
typedef void (*audio_error_callback)(status_t err);
-typedef int audio_io_handle_t;
class IAudioPolicyService;
class String8;
@@ -36,155 +38,6 @@
{
public:
- // must match android/media/AudioSystem.java STREAM_* constants
- 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
- };
-
- // FIXME These sub_format enums are currently unused
-
- // 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 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)
- };
-
- // must match android/media/AudioSystem.java MODE_* values
- enum audio_mode {
- MODE_INVALID = -2,
- MODE_CURRENT = -1,
- MODE_NORMAL = 0,
- MODE_RINGTONE,
- MODE_IN_CALL,
- MODE_IN_COMMUNICATION,
- NUM_MODES // not a valid entry, denotes end-of-list
- };
-
- enum audio_in_acoustics {
- AGC_ENABLE = 0x0001,
- AGC_DISABLE = 0,
- NS_ENABLE = 0x0002,
- NS_DISABLE = 0,
- TX_IIR_ENABLE = 0x0004,
- TX_DISABLE = 0
- };
-
- // special audio session values
- enum audio_sessions {
- SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream
- // (value must be less than 0)
- SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can
- // be moved by audio policy manager to another output stream
- // (value must be 0)
- };
-
/* These are static methods to control the system-wide AudioFlinger
* only privileged processes can have access to them
*/
@@ -209,7 +62,7 @@
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)
+ // set audio mode in audio hardware (see audio_mode_t)
static status_t setMode(int mode);
// returns true in *state if tracks are active on the specified stream or has been active
@@ -230,9 +83,9 @@
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 status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT);
+ static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT);
+ static status_t getOutputLatency(uint32_t* latency, int stream = AUDIO_STREAM_DEFAULT);
static bool routedToA2dpOutput(int streamType);
@@ -250,93 +103,11 @@
// - 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 status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = AUDIO_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_ANLG_DOCK_HEADSET = 0x800,
- DEVICE_OUT_DGTL_DOCK_HEADSET = 0x1000,
- 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_ANLG_DOCK_HEADSET | DEVICE_OUT_DGTL_DOCK_HEADSET |
- DEVICE_OUT_DEFAULT),
- DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
- DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
-
- // input devices
- DEVICE_IN_COMMUNICATION = 0x10000,
- DEVICE_IN_AMBIENT = 0x20000,
- DEVICE_IN_BUILTIN_MIC = 0x40000,
- DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,
- DEVICE_IN_WIRED_HEADSET = 0x100000,
- DEVICE_IN_AUX_DIGITAL = 0x200000,
- DEVICE_IN_VOICE_CALL = 0x400000,
- DEVICE_IN_BACK_MIC = 0x800000,
- DEVICE_IN_DEFAULT = 0x80000000,
-
- DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC |
- DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL |
- DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT)
- };
-
- // device connection states used for setDeviceConnectionState()
- enum device_connection_state {
- DEVICE_STATE_UNAVAILABLE,
- DEVICE_STATE_AVAILABLE,
- NUM_DEVICE_STATES
- };
-
- // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks)
- enum output_flags {
- OUTPUT_FLAG_INDIRECT = 0x0,
- OUTPUT_FLAG_DIRECT = 0x1
- };
-
- // device categories used for setForceUse()
- enum forced_config {
- FORCE_NONE,
- FORCE_SPEAKER,
- FORCE_HEADPHONES,
- FORCE_BT_SCO,
- FORCE_BT_A2DP,
- FORCE_WIRED_ACCESSORY,
- FORCE_BT_CAR_DOCK,
- FORCE_BT_DESK_DOCK,
- FORCE_ANALOG_DOCK,
- FORCE_DIGITAL_DOCK,
- NUM_FORCE_CONFIG,
- FORCE_DEFAULT = FORCE_NONE
- };
-
- // usages used for setForceUse(), must match AudioSystem.java
- 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 {
@@ -367,40 +138,40 @@
//
// 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 setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address);
+ static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t 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,
+ static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
+ static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+ static audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = FORMAT_DEFAULT,
- uint32_t channels = CHANNEL_OUT_STEREO,
- output_flags flags = OUTPUT_FLAG_INDIRECT);
+ uint32_t format = AUDIO_FORMAT_DEFAULT,
+ uint32_t channels = AUDIO_CHANNEL_OUT_STEREO,
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);
static status_t startOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
int session = 0);
static status_t stopOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t 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);
+ uint32_t format = AUDIO_FORMAT_DEFAULT,
+ uint32_t channels = AUDIO_CHANNEL_IN_MONO,
+ audio_in_acoustics_t acoustics = (audio_in_acoustics_t)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,
+ static status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax);
- static status_t setStreamVolumeIndex(stream_type stream, int index);
- static status_t getStreamVolumeIndex(stream_type stream, int *index);
+ static status_t setStreamVolumeIndex(audio_stream_type_t stream, int index);
+ static status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index);
- static uint32_t getStrategyForStream(stream_type stream);
- static uint32_t getDevicesForStream(stream_type stream);
+ static uint32_t getStrategyForStream(audio_stream_type_t stream);
+ static uint32_t getDevicesForStream(audio_stream_type_t stream);
static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
static status_t registerEffect(effect_descriptor_t *desc,
@@ -414,17 +185,6 @@
// ----------------------------------------------------------------------------
- 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
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 3e346db..de928da 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -30,7 +30,6 @@
#include <binder/IMemory.h>
#include <utils/threads.h>
-
namespace android {
// ----------------------------------------------------------------------------
@@ -126,11 +125,11 @@
* Parameters:
*
* streamType: Select the type of audio stream this track is attached to
- * (e.g. AudioSystem::MUSIC).
+ * (e.g. AUDIO_STREAM_MUSIC).
* sampleRate: Track sampling rate in Hz.
- * format: Audio format (e.g AudioSystem::PCM_16_BIT for signed
+ * format: Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
* 16 bits per sample).
- * channels: Channel mask: see AudioSystem::audio_channels.
+ * channels: Channel mask: see audio_channels_t.
* frameCount: Total size of track PCM buffer in frames. This defines the
* latency of the track.
* flags: Reserved for future use.
diff --git a/include/media/EffectApi.h b/include/media/EffectApi.h
index b97c22e..a5ad846 100644
--- a/include/media/EffectApi.h
+++ b/include/media/EffectApi.h
@@ -602,9 +602,9 @@
// Audio mode
enum audio_mode_e {
- AUDIO_MODE_NORMAL, // device idle
- AUDIO_MODE_RINGTONE, // device ringing
- AUDIO_MODE_IN_CALL // audio call connected (VoIP or telephony)
+ AUDIO_EFFECT_MODE_NORMAL, // device idle
+ AUDIO_EFFECT_MODE_RINGTONE, // device ringing
+ AUDIO_EFFECT_MODE_IN_CALL, // audio call connected (VoIP or telephony)
};
// Values for "accessMode" field of buffer_config_t:
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index 720a562..09b2bfe 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -26,6 +26,7 @@
#include <binder/IInterface.h>
#include <media/AudioSystem.h>
+#include <hardware/audio_policy.h>
namespace android {
@@ -39,42 +40,42 @@
//
// IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
//
- virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
- AudioSystem::device_connection_state state,
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
const char *device_address) = 0;
- virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+ virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t 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,
+ virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0;
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
+ virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
- AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT) = 0;
+ audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT) = 0;
virtual status_t startOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
int session = 0) = 0;
virtual status_t stopOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
+ audio_stream_type_t stream,
int session = 0) = 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 format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
- AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0) = 0;
+ audio_in_acoustics_t acoustics = (audio_in_acoustics_t)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,
+ virtual status_t initStreamVolume(audio_stream_type_t 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;
- virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream) = 0;
- virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream) = 0;
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index) = 0;
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index) = 0;
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream) = 0;
+ virtual uint32_t getDevicesForStream(audio_stream_type_t stream) = 0;
virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc) = 0;
virtual status_t registerEffect(effect_descriptor_t *desc,
audio_io_handle_t output,
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index e1b6dd6..bebecc0 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -85,7 +85,7 @@
// audio data.
virtual status_t open(
uint32_t sampleRate, int channelCount,
- int format=AudioSystem::PCM_16_BIT,
+ int format=AUDIO_FORMAT_PCM_16_BIT,
int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
AudioCallback cb = NULL,
void *cookie = NULL) = 0;
diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h
index c42346e..5fe7722 100644
--- a/include/media/MediaRecorderBase.h
+++ b/include/media/MediaRecorderBase.h
@@ -20,6 +20,8 @@
#include <media/mediarecorder.h>
+#include <hardware/audio.h>
+
namespace android {
class Surface;
@@ -29,7 +31,7 @@
virtual ~MediaRecorderBase() {}
virtual status_t init() = 0;
- virtual status_t setAudioSource(audio_source as) = 0;
+ virtual status_t setAudioSource(audio_source_t as) = 0;
virtual status_t setVideoSource(video_source vs) = 0;
virtual status_t setOutputFormat(output_format of) = 0;
virtual status_t setAudioEncoder(audio_encoder ae) = 0;
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index 67d940b..18a3c6a 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -33,23 +33,6 @@
typedef void (*media_completion_f)(status_t status, void *cookie);
-/* Do not change these values without updating their counterparts
- * in media/java/android/media/MediaRecorder.java!
- */
-enum audio_source {
- AUDIO_SOURCE_DEFAULT = 0,
- AUDIO_SOURCE_MIC = 1,
- AUDIO_SOURCE_VOICE_UPLINK = 2,
- AUDIO_SOURCE_VOICE_DOWNLINK = 3,
- AUDIO_SOURCE_VOICE_CALL = 4,
- AUDIO_SOURCE_CAMCORDER = 5,
- AUDIO_SOURCE_VOICE_RECOGNITION = 6,
- AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
- AUDIO_SOURCE_MAX = AUDIO_SOURCE_VOICE_COMMUNICATION,
-
- AUDIO_SOURCE_LIST_END // must be last - used to validate audio source type
-};
-
enum video_source {
VIDEO_SOURCE_DEFAULT = 0,
VIDEO_SOURCE_CAMERA = 1,
diff --git a/include/media/stagefright/AudioSource.h b/include/media/stagefright/AudioSource.h
index 9e6f0e2..20a9e16 100644
--- a/include/media/stagefright/AudioSource.h
+++ b/include/media/stagefright/AudioSource.h
@@ -24,16 +24,18 @@
#include <media/stagefright/MediaBuffer.h>
#include <utils/List.h>
+#include <hardware/audio.h>
+
namespace android {
class AudioRecord;
struct AudioSource : public MediaSource, public MediaBufferObserver {
// Note that the "channels" parameter is _not_ the number of channels,
- // but a bitmask of AudioSystem::audio_channels constants.
+ // but a bitmask of audio_channels_t constants.
AudioSource(
int inputSource, uint32_t sampleRate,
- uint32_t channels = AudioSystem::CHANNEL_IN_MONO);
+ uint32_t channels = AUDIO_CHANNEL_IN_MONO);
status_t initCheck() const;