diff options
author | Dima Zavin <dima@android.com> | 2011-04-19 22:30:36 -0700 |
---|---|---|
committer | Dima Zavin <dima@android.com> | 2011-04-27 13:10:10 -0700 |
commit | 24fc2fb1c541e954b83fd31ea9f786a5e9b45501 (patch) | |
tree | e85b3ff86b73d56e16ab640919d251c046371c65 /include | |
parent | 4dc22e77cfd2a1c3671e5646ee87c5e4c15596a0 (diff) | |
download | frameworks_base-24fc2fb1c541e954b83fd31ea9f786a5e9b45501.zip frameworks_base-24fc2fb1c541e954b83fd31ea9f786a5e9b45501.tar.gz frameworks_base-24fc2fb1c541e954b83fd31ea9f786a5e9b45501.tar.bz2 |
audio/media: convert to using the audio HAL and new audio defs
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
Diffstat (limited to 'include')
-rw-r--r-- | include/media/AudioParameter.h | 8 | ||||
-rw-r--r-- | include/media/AudioRecord.h | 15 | ||||
-rw-r--r-- | include/media/AudioSystem.h | 292 | ||||
-rw-r--r-- | include/media/AudioTrack.h | 7 | ||||
-rw-r--r-- | include/media/EffectApi.h | 6 | ||||
-rw-r--r-- | include/media/IAudioPolicyService.h | 35 | ||||
-rw-r--r-- | include/media/MediaPlayerInterface.h | 2 | ||||
-rw-r--r-- | include/media/MediaRecorderBase.h | 4 | ||||
-rw-r--r-- | include/media/mediarecorder.h | 17 | ||||
-rw-r--r-- | include/media/stagefright/AudioSource.h | 6 |
10 files changed, 70 insertions, 322 deletions
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 @@ public: // 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 @@ public: * * 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 @@ public: */ 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 @@ public: 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 @@ class AudioSystem { 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 @@ public: 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 @@ public: 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 @@ public: // - 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 @@ public: // // 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 @@ public: // ---------------------------------------------------------------------------- - 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 @@ public: * 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 @@ enum audio_device_e { // 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 @@ public: // // 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 @@ public: // 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 @@ struct MediaRecorderBase { 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 @@ class ICamera; 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; |