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| author | Andy Hung <hunga@google.com> | 2015-05-06 08:46:52 -0700 |
|---|---|---|
| committer | Andy Hung <hunga@google.com> | 2015-05-12 09:30:51 -0700 |
| commit | db4c031f518ae5806af73756273ff32cd8d0e4f8 (patch) | |
| tree | 62f9e0541acccc3acacf808d2a3cdad130eb819b /services/audioflinger | |
| parent | 18aa27016a94d0fee243637a80fd0741f89e08f2 (diff) | |
| download | frameworks_av-db4c031f518ae5806af73756273ff32cd8d0e4f8.zip frameworks_av-db4c031f518ae5806af73756273ff32cd8d0e4f8.tar.gz frameworks_av-db4c031f518ae5806af73756273ff32cd8d0e4f8.tar.bz2 | |
Update sampling rate to 192kHz for devices
Change-Id: I0a83206be51d7ae18ccf85b94b2127356307be69
Diffstat (limited to 'services/audioflinger')
| -rw-r--r-- | services/audioflinger/AudioMixer.cpp | 7 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 7 |
2 files changed, 8 insertions, 6 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index 7040af4..1348d08 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -708,11 +708,10 @@ bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSam // FIXME this is flawed for dynamic sample rates, as we choose the resampler // quality level based on the initial ratio, but that could change later. // Should have a way to distinguish tracks with static ratios vs. dynamic ratios. - if (!((trackSampleRate == 44100 && devSampleRate == 48000) || - (trackSampleRate == 48000 && devSampleRate == 44100))) { - quality = AudioResampler::DYN_LOW_QUALITY; - } else { + if (isMusicRate(trackSampleRate)) { quality = AudioResampler::DEFAULT_QUALITY; + } else { + quality = AudioResampler::DYN_LOW_QUALITY; } // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 7feb63b..5ef017f 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -5328,11 +5328,11 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, } initFastCapture = // either capture sample rate is same as (a reasonable) primary output sample rate - (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && + ((isMusicRate(primaryOutputSampleRate) && (mSampleRate == primaryOutputSampleRate)) || // or primary output sample rate is unknown, and capture sample rate is reasonable ((primaryOutputSampleRate == 0) && - ((mSampleRate == 44100 || mSampleRate == 48000)))) && + isMusicRate(mSampleRate))) && // and the buffer size is < 12 ms (mFrameCount * 1000) / mSampleRate < 12; break; @@ -6435,6 +6435,9 @@ status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( return NO_ERROR; } + ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" + " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", + srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); const bool valid = audio_is_input_channel(srcChannelMask) && audio_is_input_channel(dstChannelMask) |