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authorGlenn Kasten <gkasten@google.com>2013-11-08 19:19:00 +0000
committerAndroid (Google) Code Review <android-gerrit@google.com>2013-11-08 19:19:00 +0000
commitd8a1de6204829b0379bbd13042f1450bdb203f2b (patch)
treeb7b513a563f76060ec82a73aa9ee9543a391b8db /services/audioflinger
parent2e70b42e2af66f4abb22a95c5d642175c007e256 (diff)
parentf551e9961a6f66d63bb57f288c4be01b00725289 (diff)
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Merge "Channel counts are uint32_t"
Diffstat (limited to 'services/audioflinger')
-rw-r--r--services/audioflinger/AudioMixer.cpp2
-rw-r--r--services/audioflinger/Threads.cpp4
2 files changed, 3 insertions, 3 deletions
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 1b0b64f..8bea752 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -543,7 +543,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
resampler = AudioResampler::create(
format,
// the resampler sees the number of channels after the downmixer, if any
- downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount,
+ (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),
devSampleRate, quality);
resampler->setLocalTimeFreq(sLocalTimeFreq);
}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 16c35a2..7474f1a 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5255,7 +5255,7 @@ void AudioFlinger::RecordThread::readInputParameters()
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) {
- int channelCount;
+ uint32_t channelCount;
// optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
// stereo to mono post process as the resampler always outputs stereo.
if (mChannelCount == 1 && mReqChannelCount == 2) {
@@ -5263,7 +5263,7 @@ void AudioFlinger::RecordThread::readInputParameters()
} else {
channelCount = 2;
}
- mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+ mResampler = AudioResampler::create(16, (int) channelCount, mReqSampleRate);
mResampler->setSampleRate(mSampleRate);
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2];