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authorSteve Block <steveblock@google.com>2011-10-20 11:56:00 +0100
committerSteve Block <steveblock@google.com>2011-10-26 09:57:54 +0100
commit71f2cf116aab893e224056c38ab146bd1538dd3e (patch)
tree75a9162a0ea00830184b12a9ca51d3a1a040a5bb /voip
parent1da79501066a74b630c8aa138db0f86ab6c690bb (diff)
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Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE
See https://android-git.corp.google.com/g/#/c/143865 Bug: 5449033 Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
Diffstat (limited to 'voip')
-rw-r--r--voip/jni/rtp/AudioGroup.cpp18
1 files changed, 9 insertions, 9 deletions
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp
index 459756d..6f8a232 100644
--- a/voip/jni/rtp/AudioGroup.cpp
+++ b/voip/jni/rtp/AudioGroup.cpp
@@ -269,7 +269,7 @@ void AudioStream::encode(int tick, AudioStream *chain)
mTick += skipped * mInterval;
mSequence += skipped;
mTimestamp += skipped * mSampleCount;
- LOGV("stream[%d] skips %d packets", mSocket, skipped);
+ ALOGV("stream[%d] skips %d packets", mSocket, skipped);
}
tick = mTick;
@@ -334,7 +334,7 @@ void AudioStream::encode(int tick, AudioStream *chain)
memset(samples, 0, sizeof(samples));
if (mMode != RECEIVE_ONLY) {
- LOGV("stream[%d] no data", mSocket);
+ ALOGV("stream[%d] no data", mSocket);
}
}
@@ -350,7 +350,7 @@ void AudioStream::encode(int tick, AudioStream *chain)
buffer[2] = mSsrc;
int length = mCodec->encode(&buffer[3], samples);
if (length <= 0) {
- LOGV("stream[%d] encoder error", mSocket);
+ ALOGV("stream[%d] encoder error", mSocket);
return;
}
sendto(mSocket, buffer, length + 12, MSG_DONTWAIT, (sockaddr *)&mRemote,
@@ -386,7 +386,7 @@ void AudioStream::decode(int tick)
mLatencyScore = score;
mLatencyTimer = tick;
} else if (tick - mLatencyTimer >= MEASURE_PERIOD) {
- LOGV("stream[%d] reduces latency of %dms", mSocket, mLatencyScore);
+ ALOGV("stream[%d] reduces latency of %dms", mSocket, mLatencyScore);
mBufferTail -= mLatencyScore;
mLatencyScore = -1;
}
@@ -394,7 +394,7 @@ void AudioStream::decode(int tick)
int count = (BUFFER_SIZE - (mBufferTail - mBufferHead)) * mSampleRate;
if (count < mSampleCount) {
// Buffer overflow. Drop the packet.
- LOGV("stream[%d] buffer overflow", mSocket);
+ ALOGV("stream[%d] buffer overflow", mSocket);
recv(mSocket, &c, 1, MSG_DONTWAIT);
return;
}
@@ -417,7 +417,7 @@ void AudioStream::decode(int tick)
// reliable but at least they can be used to identify duplicates?
if (length < 12 || length > (int)sizeof(buffer) ||
(ntohl(*(uint32_t *)buffer) & 0xC07F0000) != mCodecMagic) {
- LOGV("stream[%d] malformed packet", mSocket);
+ ALOGV("stream[%d] malformed packet", mSocket);
return;
}
int offset = 12 + ((buffer[0] & 0x0F) << 2);
@@ -438,13 +438,13 @@ void AudioStream::decode(int tick)
count = length;
}
if (count <= 0) {
- LOGV("stream[%d] decoder error", mSocket);
+ ALOGV("stream[%d] decoder error", mSocket);
return;
}
if (tick - mBufferTail > 0) {
// Buffer underrun. Reset the jitter buffer.
- LOGV("stream[%d] buffer underrun", mSocket);
+ ALOGV("stream[%d] buffer underrun", mSocket);
if (mBufferTail - mBufferHead <= 0) {
mBufferHead = tick + mInterval;
mBufferTail = mBufferHead;
@@ -913,7 +913,7 @@ bool AudioGroup::DeviceThread::threadLoop()
if (mode != MUTED) {
if (echo != NULL) {
- LOGV("echo->run()");
+ ALOGV("echo->run()");
echo->run(output, input);
}
send(deviceSocket, input, sizeof(input), MSG_DONTWAIT);