include/media/IAudioFlinger.h

/*
 * Copyright (C) 2007 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_IAUDIOFLINGER_H
#define ANDROID_IAUDIOFLINGER_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/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/IAudioFlingerClient.h>
#include <system/audio.h>
#include <system/audio_policy.h>
#include <hardware/audio_policy.h>
#include <hardware/audio_effect.h>
#include <media/IEffect.h>
#include <media/IEffectClient.h>
#include <utils/String8.h>

namespace android {

// ----------------------------------------------------------------------------

class IAudioFlinger : public IInterface
{
public:
    DECLARE_META_INTERFACE(AudioFlinger);

    // or-able bits shared by createTrack and openRecord, but not all combinations make sense
    enum {
        TRACK_DEFAULT = 0,  // client requests a default AudioTrack
        TRACK_TIMED   = 1,  // client requests a TimedAudioTrack
        TRACK_FAST    = 2,  // client requests a fast AudioTrack or AudioRecord
        TRACK_OFFLOAD = 4,  // client requests offload to hw codec
        TRACK_DIRECT = 8,   // client requests a direct output
    };
    typedef uint32_t track_flags_t;

    // invariant on exit for all APIs that return an sp<>:
    //   (return value != 0) == (*status == NO_ERROR)

    /* create an audio track and registers it with AudioFlinger.
     * return null if the track cannot be created.
     */
    virtual sp<IAudioTrack> createTrack(
                                audio_stream_type_t streamType,
                                uint32_t sampleRate,
                                audio_format_t format,
                                audio_channel_mask_t channelMask,
                                size_t *pFrameCount,
                                track_flags_t *flags,
                                const sp<IMemory>& sharedBuffer,
                                // On successful return, AudioFlinger takes over the handle
                                // reference and will release it when the track is destroyed.
                                // However on failure, the client is responsible for release.
                                audio_io_handle_t output,
                                pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                int *sessionId,
                                int clientUid,
                                status_t *status) = 0;

    virtual sp<IAudioRecord> openRecord(
                                // On successful return, AudioFlinger takes over the handle
                                // reference and will release it when the track is destroyed.
                                // However on failure, the client is responsible for release.
                                audio_io_handle_t input,
                                uint32_t sampleRate,
                                audio_format_t format,
                                audio_channel_mask_t channelMask,
                                size_t *pFrameCount,
                                track_flags_t *flags,
                                pid_t tid,  // -1 means unused, otherwise must be valid non-0
                                int *sessionId,
                                size_t *notificationFrames,
                                sp<IMemory>& cblk,
                                sp<IMemory>& buffers,   // return value 0 means it follows cblk
                                status_t *status) = 0;

    /* query the audio hardware state. This state never changes,
     * and therefore can be cached.
     */
    virtual     uint32_t    sampleRate(audio_io_handle_t output) const = 0;
    virtual     audio_format_t format(audio_io_handle_t output) const = 0;
    virtual     size_t      frameCount(audio_io_handle_t output) const = 0;

    // return estimated latency in milliseconds
    virtual     uint32_t    latency(audio_io_handle_t output) const = 0;

    /* set/get the audio hardware state. This will probably be used by
     * the preference panel, mostly.
     */
    virtual     status_t    setMasterVolume(float value) = 0;
    virtual     status_t    setMasterMute(bool muted) = 0;

    virtual     float       masterVolume() const = 0;
    virtual     bool        masterMute() const = 0;

    /* set/get stream type state. This will probably be used by
     * the preference panel, mostly.
     */
    virtual     status_t    setStreamVolume(audio_stream_type_t stream, float value,
                                    audio_io_handle_t output) = 0;
    virtual     status_t    setStreamMute(audio_stream_type_t stream, bool muted) = 0;

    virtual     float       streamVolume(audio_stream_type_t stream,
                                    audio_io_handle_t output) const = 0;
    virtual     bool        streamMute(audio_stream_type_t stream) const = 0;

    // set audio mode
    virtual     status_t    setMode(audio_mode_t mode) = 0;

    // mic mute/state
    virtual     status_t    setMicMute(bool state) = 0;
    virtual     bool        getMicMute() const = 0;

    virtual     status_t    setParameters(audio_io_handle_t ioHandle,
                                    const String8& keyValuePairs) = 0;
    virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys)
                                    const = 0;

    // Register an object to receive audio input/output change and track notifications.
    // For a given calling pid, AudioFlinger disregards any registrations after the first.
    // Thus the IAudioFlingerClient must be a singleton per process.
    virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;

    // retrieve the audio recording buffer size
    // FIXME This API assumes a route, and so should be deprecated.
    virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
            audio_channel_mask_t channelMask) const = 0;

    virtual status_t openOutput(audio_module_handle_t module,
                                audio_io_handle_t *output,
                                audio_config_t *config,
                                audio_devices_t *devices,
                                const String8& address,
                                uint32_t *latencyMs,
                                audio_output_flags_t flags) = 0;
    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
                                    audio_io_handle_t output2) = 0;
    virtual status_t closeOutput(audio_io_handle_t output) = 0;
    virtual status_t suspendOutput(audio_io_handle_t output) = 0;
    virtual status_t restoreOutput(audio_io_handle_t output) = 0;

    virtual status_t openInput(audio_module_handle_t module,
                               audio_io_handle_t *input,
                               audio_config_t *config,
                               audio_devices_t *device,
                               const String8& address,
                               audio_source_t source,
                               audio_input_flags_t flags) = 0;
    virtual status_t closeInput(audio_io_handle_t input) = 0;

    virtual status_t invalidateStream(audio_stream_type_t stream) = 0;

    virtual status_t setVoiceVolume(float volume) = 0;

    virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
                                    audio_io_handle_t output) const = 0;

    virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const = 0;

    virtual audio_unique_id_t newAudioUniqueId() = 0;

    virtual void acquireAudioSessionId(int audioSession, pid_t pid) = 0;
    virtual void releaseAudioSessionId(int audioSession, pid_t pid) = 0;

    virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;

    virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const = 0;

    virtual status_t getEffectDescriptor(const effect_uuid_t *pEffectUUID,
                                        effect_descriptor_t *pDescriptor) const = 0;

    virtual sp<IEffect> createEffect(
                                    effect_descriptor_t *pDesc,
                                    const sp<IEffectClient>& client,
                                    int32_t priority,
                                    // AudioFlinger doesn't take over handle reference from client
                                    audio_io_handle_t output,
                                    int sessionId,
                                    status_t *status,
                                    int *id,
                                    int *enabled) = 0;

    virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
                                    audio_io_handle_t dstOutput) = 0;

    virtual audio_module_handle_t loadHwModule(const char *name) = 0;

    // helpers for android.media.AudioManager.getProperty(), see description there for meaning
    // FIXME move these APIs to AudioPolicy to permit a more accurate implementation
    // that looks on primary device for a stream with fast flag, primary flag, or first one.
    virtual uint32_t getPrimaryOutputSamplingRate() = 0;
    virtual size_t getPrimaryOutputFrameCount() = 0;

    // Intended for AudioService to inform AudioFlinger of device's low RAM attribute,
    // and should be called at most once.  For a definition of what "low RAM" means, see
    // android.app.ActivityManager.isLowRamDevice().
    virtual status_t setLowRamDevice(bool isLowRamDevice) = 0;

    /* List available audio ports and their attributes */
    virtual status_t listAudioPorts(unsigned int *num_ports,
                                    struct audio_port *ports) = 0;

    /* Get attributes for a given audio port */
    virtual status_t getAudioPort(struct audio_port *port) = 0;

    /* Create an audio patch between several source and sink ports */
    virtual status_t createAudioPatch(const struct audio_patch *patch,
                                       audio_patch_handle_t *handle) = 0;

    /* Release an audio patch */
    virtual status_t releaseAudioPatch(audio_patch_handle_t handle) = 0;

    /* List existing audio patches */
    virtual status_t listAudioPatches(unsigned int *num_patches,
                                      struct audio_patch *patches) = 0;
    /* Set audio port configuration */
    virtual status_t setAudioPortConfig(const struct audio_port_config *config) = 0;

    /* Get the HW synchronization source used for an audio session */
    virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) = 0;
};


// ----------------------------------------------------------------------------

class BnAudioFlinger : public BnInterface<IAudioFlinger>
{
public:
    virtual status_t    onTransact( uint32_t code,
                                    const Parcel& data,
                                    Parcel* reply,
                                    uint32_t flags = 0);
};

// ----------------------------------------------------------------------------

}; // namespace android

#endif // ANDROID_IAUDIOFLINGER_H