/* ** ** Copyright 2012, 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 INCLUDING_FROM_AUDIOFLINGER_H #error This header file should only be included from AudioFlinger.h #endif // record track class RecordTrack : public TrackBase { public: RecordTrack(RecordThread *thread, const sp& client, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount, int sessionId, int uid, IAudioFlinger::track_flags_t flags); virtual ~RecordTrack(); virtual status_t start(AudioSystem::sync_event_t event, int triggerSession); virtual void stop(); void destroy(); void invalidate(); // clear the buffer overflow flag void clearOverflow() { mOverflow = false; } // set the buffer overflow flag and return previous value bool setOverflow() { bool tmp = mOverflow; mOverflow = true; return tmp; } static void appendDumpHeader(String8& result); void dump(char* buffer, size_t size, bool active); void handleSyncStartEvent(const sp& event); void clearSyncStartEvent(); private: friend class AudioFlinger; // for mState RecordTrack(const RecordTrack&); RecordTrack& operator = (const RecordTrack&); // AudioBufferProvider interface virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts = kInvalidPTS); // releaseBuffer() not overridden bool mOverflow; // overflow on most recent attempt to fill client buffer // updated by RecordThread::readInputParameters_l() AudioResampler *mResampler; // interleaved stereo pairs of fixed-point Q4.27 int32_t *mRsmpOutBuffer; // current allocated frame count for the above, which may be larger than needed size_t mRsmpOutFrameCount; size_t mRsmpInUnrel; // unreleased frames remaining from // most recent getNextBuffer // for debug only // rolling counter that is never cleared int32_t mRsmpInFront; // next available frame AudioBufferProvider::Buffer mSink; // references client's buffer sink in shared memory // sync event triggering actual audio capture. Frames read before this event will // be dropped and therefore not read by the application. sp mSyncStartEvent; // number of captured frames to drop after the start sync event has been received. // when < 0, maximum frames to drop before starting capture even if sync event is // not received ssize_t mFramesToDrop; // used by resampler to find source frames ResamplerBufferProvider *mResamplerBufferProvider; };