| Commit message (Collapse) | Author | Age | Files | Lines |
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The cause is very likely that the WaveGenerator *lpWaveGen returned by lpToneGen->mWaveGens.valueFor(lFrequency) just before calling lpWaveGen->getSamples(lpOut, lGenSmp, lWaveCmd) is invalid. The frequency lFrequency is not part of the frequencies in mWaveGens.
This can happen if a different tone is started while the callback function is active: The state is changed to TONE_RESTARTING and the call to prepareWave() at line 1226 will change the tone descriptor pointed to by mpToneDesc as well as the content of mWaveGens. However, mpToneDesc was cached in a local variable lpToneDesc when entering the callback and is not reloaded when exiting prepareWave(). This causes a mismatch between the tone frequencies listed in lpToneDesc and the frequencies present in mWaveGens.
This regression was introduced in change 973 when mpToneDesc was cached in a local variable.
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to twice of the ouputSR
Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
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DO NOT MERGE
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This change is the first part of a fix for issue 1846343, :
- Added new enum values for input sources in AudioRecord and MediaRecorder for voice uplink, downlink and uplink+downlink sources.
- renamed streamType to inputSource in all native functions handling audio record.
A second change is required in opencore author driver and android audio input to completely fix the issue.
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Bug 1836596
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Added new tone types for CDMA IS-95 specific tones.
Automatic selection between IS-95, CEPT and JAPAN version base on operator
country code for call supervisory tones.
Also improved tone generator capabilities:
- Each tone segment can now generate its own set of frequencies
- A tone does not have to be a succession of alternating ON/OFF segments
- The sequence repetition does not have to start from first segment
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AudioTrack, AudioRecord:
- remove useless mAudioFlinger member of AudioTrack and AudioRecord.
- signal cblk.cv condition in stop() method to speed up stop completion.
- extend wait condition timeout in obtainBuffer() when waitCount is -1 to avoid waking up callback thread unnecessarily
AudioFlinger:
- remove some warnings in AudioFlinger.cpp.
- remove function AudioFlinger::MixerThread::removetrack_l() as its content is never executed.
- remove useless call to setMasterVolume in AudioFlinger::handleForcedSpeakerRoute().
- Offset VOICE_CALL stream volume to reflect actual volume that is never 0 in hardware (this fix has been made in the open source): 0.01 + v * 0.99.
AudioSystem.java:
- correct typo in comment
IAudioflinger, IAudioFlingerClient:
- make AudioFlinger binder interfaces used for callbacks ONEWAY.
AudioHardwareInterface:
- correct routeStrings[] table in AudioHardwareInteface.cpp
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Invalid buffer size: minFrameCount 10240, frameCount 4800
The problem comes from the fact that AudioSystem::getOutputFrameCount() calls getOutput() to retrieve the active output (A2DP or Hardware) before calling get_audio_flinger(). If it is the first time AudioSystem::getOutputFrameCount() is called in a given process, getOutput() will return a wrong value because gA2dpEnabled has not yet been updated by get_audio_flinger().
The fix consists in calling get_audio_flinger() in getOutput() to be sure that gA2dpEnabled is valid when getOutput() reads it.
Original author: elaurent
Merged from: //branches/cupcake/...
Automated import of CL 144054
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