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author | Steve Block <steveblock@google.com> | 2011-10-20 11:56:00 +0100 |
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committer | Steve Block <steveblock@google.com> | 2011-10-26 09:57:54 +0100 |
commit | 71f2cf116aab893e224056c38ab146bd1538dd3e (patch) | |
tree | 75a9162a0ea00830184b12a9ca51d3a1a040a5bb /voip | |
parent | 1da79501066a74b630c8aa138db0f86ab6c690bb (diff) | |
download | frameworks_base-71f2cf116aab893e224056c38ab146bd1538dd3e.zip frameworks_base-71f2cf116aab893e224056c38ab146bd1538dd3e.tar.gz frameworks_base-71f2cf116aab893e224056c38ab146bd1538dd3e.tar.bz2 |
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.cpp | 18 |
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); |