391 lines
14 KiB
C
391 lines
14 KiB
C
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/*
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* Copyright (C) 2008 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef AUDIORECORD_H_
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#define AUDIORECORD_H_
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#include <stdint.h>
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#include <sys/types.h>
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#include <media/IAudioFlinger.h>
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#include <media/IAudioRecord.h>
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#include <media/AudioTrack.h>
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#include <utils/RefBase.h>
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#include <utils/Errors.h>
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#include <binder/IInterface.h>
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#include <binder/IMemory.h>
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#include <utils/threads.h>
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namespace android {
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// ----------------------------------------------------------------------------
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class AudioRecord
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{
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public:
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static const int DEFAULT_SAMPLE_RATE = 8000;
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/* Events used by AudioRecord callback function (callback_t).
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*
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* to keep in sync with frameworks/base/media/java/android/media/AudioRecord.java
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*/
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enum event_type {
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EVENT_MORE_DATA = 0, // Request to reqd more data from PCM buffer.
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EVENT_OVERRUN = 1, // PCM buffer overrun occured.
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EVENT_MARKER = 2, // Record head is at the specified marker position
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// (See setMarkerPosition()).
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EVENT_NEW_POS = 3, // Record head is at a new position
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// (See setPositionUpdatePeriod()).
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};
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/* Create Buffer on the stack and pass it to obtainBuffer()
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* and releaseBuffer().
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*/
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class Buffer
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{
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public:
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enum {
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MUTE = 0x00000001
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};
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uint32_t flags;
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int channelCount;
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int format;
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size_t frameCount;
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size_t size;
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union {
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void* raw;
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short* i16;
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int8_t* i8;
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};
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};
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/* These are static methods to control the system-wide AudioFlinger
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* only privileged processes can have access to them
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*/
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// static status_t setMasterMute(bool mute);
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/* As a convenience, if a callback is supplied, a handler thread
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* is automatically created with the appropriate priority. This thread
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* invokes the callback when a new buffer becomes ready or an overrun condition occurs.
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* Parameters:
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*
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* event: type of event notified (see enum AudioRecord::event_type).
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* user: Pointer to context for use by the callback receiver.
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* info: Pointer to optional parameter according to event type:
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* - EVENT_MORE_DATA: pointer to AudioRecord::Buffer struct. The callback must not read
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* more bytes than indicated by 'size' field and update 'size' if less bytes are
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* read.
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* - EVENT_OVERRUN: unused.
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* - EVENT_MARKER: pointer to an uin32_t containing the marker position in frames.
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* - EVENT_NEW_POS: pointer to an uin32_t containing the new position in frames.
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*/
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typedef void (*callback_t)(int event, void* user, void *info);
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/* Returns the minimum frame count required for the successful creation of
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* an AudioRecord object.
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* Returned status (from utils/Errors.h) can be:
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* - NO_ERROR: successful operation
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* - NO_INIT: audio server or audio hardware not initialized
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* - BAD_VALUE: unsupported configuration
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*/
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static status_t getMinFrameCount(int* frameCount,
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uint32_t sampleRate,
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int format,
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int channelCount);
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/* Constructs an uninitialized AudioRecord. No connection with
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* AudioFlinger takes place.
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*/
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AudioRecord();
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/* Creates an AudioRecord track and registers it with AudioFlinger.
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* Once created, the track needs to be started before it can be used.
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* Unspecified values are set to the audio hardware's current
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* values.
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*
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* Parameters:
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*
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* inputSource: Select the audio input to record to (e.g. AUDIO_SOURCE_DEFAULT).
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* sampleRate: Track sampling rate in Hz.
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* format: Audio format (e.g AudioSystem::PCM_16_BIT for signed
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* 16 bits per sample).
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* channels: Channel mask: see AudioSystem::audio_channels.
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* frameCount: Total size of track PCM buffer in frames. This defines the
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* latency of the track.
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* flags: A bitmask of acoustic values from enum record_flags. It enables
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* AGC, NS, and IIR.
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* cbf: Callback function. If not null, this function is called periodically
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* to provide new PCM data.
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* notificationFrames: The callback function is called each time notificationFrames PCM
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* frames are ready in record track output buffer.
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* user Context for use by the callback receiver.
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*/
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enum record_flags {
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RECORD_AGC_ENABLE = 1,
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RECORD_NS_ENABLE = 2,
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RECORD_IIR_ENABLE = 4
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};
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AudioRecord(int inputSource,
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uint32_t sampleRate = 0,
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int format = 0,
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uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
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int frameCount = 0,
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uint32_t flags = 0,
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callback_t cbf = 0,
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void* user = 0,
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int notificationFrames = 0,
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int sessionId = 0);
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/* Terminates the AudioRecord and unregisters it from AudioFlinger.
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* Also destroys all resources assotiated with the AudioRecord.
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*/
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~AudioRecord();
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/* Initialize an uninitialized AudioRecord.
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* Returned status (from utils/Errors.h) can be:
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* - NO_ERROR: successful intialization
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* - INVALID_OPERATION: AudioRecord is already intitialized or record device is already in use
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* - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
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* - NO_INIT: audio server or audio hardware not initialized
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* - PERMISSION_DENIED: recording is not allowed for the requesting process
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* */
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status_t set(int inputSource = 0,
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uint32_t sampleRate = 0,
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int format = 0,
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uint32_t channels = AudioSystem::CHANNEL_IN_MONO,
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int frameCount = 0,
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uint32_t flags = 0,
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callback_t cbf = 0,
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void* user = 0,
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int notificationFrames = 0,
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bool threadCanCallJava = false,
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int sessionId = 0);
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/* Result of constructing the AudioRecord. This must be checked
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* before using any AudioRecord API (except for set()), using
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* an uninitialized AudioRecord produces undefined results.
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* See set() method above for possible return codes.
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*/
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status_t initCheck() const;
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/* Returns this track's latency in milliseconds.
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* This includes the latency due to AudioRecord buffer size
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* and audio hardware driver.
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*/
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uint32_t latency() const;
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/* getters, see constructor */
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int format() const;
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int channelCount() const;
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int channels() const;
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uint32_t frameCount() const;
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int frameSize() const;
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int inputSource() const;
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/* After it's created the track is not active. Call start() to
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* make it active. If set, the callback will start being called.
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*/
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status_t start();
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/* Stop a track. If set, the callback will cease being called and
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* obtainBuffer returns STOPPED. Note that obtainBuffer() still works
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* and will fill up buffers until the pool is exhausted.
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*/
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status_t stop();
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bool stopped() const;
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/* get sample rate for this record track
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*/
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uint32_t getSampleRate();
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/* Sets marker position. When record reaches the number of frames specified,
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* a callback with event type EVENT_MARKER is called. Calling setMarkerPosition
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* with marker == 0 cancels marker notification callback.
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* If the AudioRecord has been opened with no callback function associated,
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* the operation will fail.
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*
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* Parameters:
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*
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* marker: marker position expressed in frames.
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*
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* Returned status (from utils/Errors.h) can be:
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* - NO_ERROR: successful operation
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* - INVALID_OPERATION: the AudioRecord has no callback installed.
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*/
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status_t setMarkerPosition(uint32_t marker);
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status_t getMarkerPosition(uint32_t *marker);
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/* Sets position update period. Every time the number of frames specified has been recorded,
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* a callback with event type EVENT_NEW_POS is called.
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* Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
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* callback.
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* If the AudioRecord has been opened with no callback function associated,
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* the operation will fail.
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*
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* Parameters:
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*
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* updatePeriod: position update notification period expressed in frames.
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*
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* Returned status (from utils/Errors.h) can be:
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* - NO_ERROR: successful operation
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* - INVALID_OPERATION: the AudioRecord has no callback installed.
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*/
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status_t setPositionUpdatePeriod(uint32_t updatePeriod);
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status_t getPositionUpdatePeriod(uint32_t *updatePeriod);
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/* Gets record head position. The position is the total number of frames
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* recorded since record start.
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*
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* Parameters:
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*
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* position: Address where to return record head position within AudioRecord buffer.
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*
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* Returned status (from utils/Errors.h) can be:
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* - NO_ERROR: successful operation
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* - BAD_VALUE: position is NULL
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*/
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status_t getPosition(uint32_t *position);
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/* returns a handle on the audio input used by this AudioRecord.
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*
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* Parameters:
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* none.
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*
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* Returned value:
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* handle on audio hardware input
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*/
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audio_io_handle_t getInput();
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/* returns the audio session ID associated to this AudioRecord.
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*
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* Parameters:
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* none.
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*
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* Returned value:
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* AudioRecord session ID.
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*/
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int getSessionId();
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/* obtains a buffer of "frameCount" frames. The buffer must be
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* filled entirely. If the track is stopped, obtainBuffer() returns
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* STOPPED instead of NO_ERROR as long as there are buffers availlable,
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* at which point NO_MORE_BUFFERS is returned.
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* Buffers will be returned until the pool (buffercount())
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* is exhausted, at which point obtainBuffer() will either block
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* or return WOULD_BLOCK depending on the value of the "blocking"
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* parameter.
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*/
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enum {
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NO_MORE_BUFFERS = 0x80000001,
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STOPPED = 1
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};
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status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
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void releaseBuffer(Buffer* audioBuffer);
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/* As a convenience we provide a read() interface to the audio buffer.
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* This is implemented on top of lockBuffer/unlockBuffer.
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*/
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ssize_t read(void* buffer, size_t size);
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/* Return the amount of input frames lost in the audio driver since the last call of this function.
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* Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call.
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* Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers.
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* Unit: the number of input audio frames
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*/
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unsigned int getInputFramesLost();
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private:
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/* copying audio tracks is not allowed */
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AudioRecord(const AudioRecord& other);
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AudioRecord& operator = (const AudioRecord& other);
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/* a small internal class to handle the callback */
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class ClientRecordThread : public Thread
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{
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public:
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ClientRecordThread(AudioRecord& receiver, bool bCanCallJava = false);
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private:
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friend class AudioRecord;
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virtual bool threadLoop();
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virtual status_t readyToRun() { return NO_ERROR; }
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virtual void onFirstRef() {}
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AudioRecord& mReceiver;
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Mutex mLock;
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};
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bool processAudioBuffer(const sp<ClientRecordThread>& thread);
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status_t openRecord(uint32_t sampleRate,
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int format,
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int channelCount,
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int frameCount,
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uint32_t flags,
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audio_io_handle_t input);
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sp<IAudioRecord> mAudioRecord;
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sp<IMemory> mCblkMemory;
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sp<ClientRecordThread> mClientRecordThread;
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Mutex mRecordThreadLock;
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uint32_t mFrameCount;
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audio_track_cblk_t* mCblk;
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int32_t mFormat;
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uint8_t mChannelCount;
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uint8_t mInputSource;
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uint8_t mReserved;
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status_t mStatus;
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uint32_t mLatency;
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volatile int32_t mActive;
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callback_t mCbf;
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void* mUserData;
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uint32_t mNotificationFrames;
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uint32_t mRemainingFrames;
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uint32_t mMarkerPosition;
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bool mMarkerReached;
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uint32_t mNewPosition;
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uint32_t mUpdatePeriod;
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uint32_t mFlags;
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uint32_t mChannels;
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audio_io_handle_t mInput;
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bool mFirstread;
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int mSessionId;
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};
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}; // namespace android
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#endif /*AUDIORECORD_H_*/
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