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author | Jean-Baptiste Queru <jbq@google.com> | 2012-01-19 17:18:25 -0800 |
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committer | Jean-Baptiste Queru <jbq@google.com> | 2012-01-19 17:18:25 -0800 |
commit | 00326e74580e03334df5c51a9284ee2fcfb67023 (patch) | |
tree | 0d9a14fbfd313da6817a5de908fc7b8ba8227de7 | |
parent | b4074803f775e5a4616287804f3405fd17eecab3 (diff) | |
parent | 4c9d58091fb7847dbf92329f6118070f42866862 (diff) | |
download | voip-00326e74580e03334df5c51a9284ee2fcfb67023.tar.gz |
Merge ee4618bc
Change-Id: Ie1dc6ad38e7c30636d80f6caef11cf6673144940
-rw-r--r-- | jni/rtp/AudioGroup.cpp | 18 |
1 files changed, 9 insertions, 9 deletions
diff --git a/jni/rtp/AudioGroup.cpp b/jni/rtp/AudioGroup.cpp index 459756d..6f8a232 100644 --- a/jni/rtp/AudioGroup.cpp +++ b/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); |