Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these
Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
Unconditional delete for raw pointers.
Use "if (sp != 0)" not "if (sp.get() != 0)" or "if (sp != NULL)".
Use "if (raw != NULL)" not "if (raw)".
Change-Id: I531a8da7c37149261ed2f34b862ec4896a4b785b
Also remove defaults in startToneCommand(), they're not needed and the
default for tone type was nonsense.
Change-Id: I70fa8cee4f3dbb8c66ceb3719c8d3d2f447f05b9
At native level it was a mixture of audio_stream_type_t, int, uint32_t,
and uint8_t. Java is still int. Also fixed a couple of hard-coded -1
instead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0
instead of AUDIO_STREAM_VOICE_CALL.
Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
Create a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.
Do the same if start fails due to the same error after time out in obtainBuffer().
Do not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.
This avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.
Same modifications for AudioRecord.
Add a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.
This change is a complement to the main fix in kernel driver for the same issue (partner change #1250).
It removes clicks sometimes heard after the end of the tones while audio flinger is sending 0s to the audio output stream.
The problem was that the sleep time between two writes was more than the duration of one audio output stream buffer which could cause some underrun.
Also fixed a recent regression in ToneGenerator that made that the end of previous tone was repeated at the beginning of current one under certain timing circumstances when the maximum tone duration was specified.
When the AudioTrack callback notification size is relatively high (Which is the case on Sholes and over A2DP), it is likely that the end of tone is reached during the first callback. In this case, the AudioTrack is stopped before exiting the callback which causes 2 problems:
- 1: If the AudioFlinger thread is scheduled before we exit the ToneGenerator callback, the track can be stopped and reset before the data is actually marked as present in the buffer by the AudioTrack callback => no audio will be processed by AudioFlinger.
- 2: In this case, the data write index in the AudioTrack buffer is incremented after the track was reset by the AudioFlinger which leaves unplayed data in the buffer. This data will be played the next time the AudioTrack is started if not flushed in between.
The fix consists in adding an intermediate state to ToneGenerator state machine so that we exit the callback function when the stop condition is reached and stop the AudioTrack the next time we execute the callback.
Add a parameter to ToneGenerator.startTone() allowing the caller to specify the tone duration. This is used by the phone application to have a precise control on the DTMF tone duration which was not possible with the use of delayed messaged.
Also modified AudioFlinger output threads so that 0s are written to the audio output stream when no more tracks are ready to mix instead of just sleeping. This avoids an issue where the end of a previous DTMF tone could stay in audio hardware buffers and be played just before the beginning of the next DTMF tone.
There is a void statement at line 917 of ToneGenerator.cpp: mState == TONE_IDLE;
This problem is harmless as in current code this execution path is never taken; it can only happen if a "new" operator fails in prepareWave() which is a case we usually consider as unlikely in android audio framework.
Initial commit for review.
Integrated comments after patch set 1 review.
Fixed lockup in AudioFlinger::ThreadBase::exit()
Fixed lockup when playing tone with AudioPlocyService startTone()
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.
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
