RTP: Update parameters for larger packet intervals.
Also remove some duplicated code.
Change-Id: I64576e5442a962eb4b0dfa83b52a8127567ba597
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp
index 9b0455c..4955667 100644
--- a/voip/jni/rtp/AudioGroup.cpp
+++ b/voip/jni/rtp/AudioGroup.cpp
@@ -28,6 +28,7 @@
#include <arpa/inet.h>
#include <netinet/in.h>
+// #define LOG_NDEBUG 0
#define LOG_TAG "AudioGroup"
#include <cutils/atomic.h>
#include <cutils/properties.h>
@@ -62,9 +63,9 @@
// a modulo operation on the index while accessing the array. However modulo can
// be expensive on some platforms, such as ARM. Thus we round up the size of the
// array to the nearest power of 2 and then use bitwise-and instead of modulo.
-// Currently we make it 512ms long and assume packet interval is 40ms or less.
-// The first 80ms is the place where samples get mixed. The rest 432ms is the
-// real jitter buffer. For a stream at 8000Hz it takes 8192 bytes. These numbers
+// Currently we make it 2048ms long and assume packet interval is 50ms or less.
+// The first 100ms is the place where samples get mixed. The rest is the real
+// jitter buffer. For a stream at 8000Hz it takes 32 kilobytes. These numbers
// are chosen by experiments and each of them can be adjusted as needed.
// Originally a stream does not send packets when it is receive-only or there is
@@ -84,9 +85,11 @@
// + Resampling is not done yet, so streams in one group must use the same rate.
// For the first release only 8000Hz is supported.
-#define BUFFER_SIZE 512
-#define HISTORY_SIZE 80
-#define MEASURE_PERIOD 2000
+#define BUFFER_SIZE 2048
+#define HISTORY_SIZE 100
+#define MEASURE_BASE 100
+#define MEASURE_PERIOD 5000
+#define DTMF_PERIOD 200
class AudioStream
{
@@ -278,7 +281,7 @@
if (mMode != RECEIVE_ONLY && mDtmfEvent != -1) {
int duration = mTimestamp - mDtmfStart;
// Make sure duration is reasonable.
- if (duration >= 0 && duration < mSampleRate * 100) {
+ if (duration >= 0 && duration < mSampleRate * DTMF_PERIOD) {
duration += mSampleCount;
int32_t buffer[4] = {
htonl(mDtmfMagic | mSequence),
@@ -286,7 +289,7 @@
mSsrc,
htonl(mDtmfEvent | duration),
};
- if (duration >= mSampleRate * 100) {
+ if (duration >= mSampleRate * DTMF_PERIOD) {
buffer[3] |= htonl(1 << 23);
mDtmfEvent = -1;
}
@@ -298,43 +301,39 @@
}
int32_t buffer[mSampleCount + 3];
+ bool data = false;
+ if (mMode != RECEIVE_ONLY) {
+ // Mix all other streams.
+ memset(buffer, 0, sizeof(buffer));
+ while (chain) {
+ if (chain != this) {
+ data |= chain->mix(buffer, tick - mInterval, tick, mSampleRate);
+ }
+ chain = chain->mNext;
+ }
+ }
+
int16_t samples[mSampleCount];
- if (mMode == RECEIVE_ONLY) {
+ if (data) {
+ // Saturate into 16 bits.
+ for (int i = 0; i < mSampleCount; ++i) {
+ int32_t sample = buffer[i];
+ if (sample < -32768) {
+ sample = -32768;
+ }
+ if (sample > 32767) {
+ sample = 32767;
+ }
+ samples[i] = sample;
+ }
+ } else {
if ((mTick ^ mKeepAlive) >> 10 == 0) {
return;
}
mKeepAlive = mTick;
memset(samples, 0, sizeof(samples));
- } else {
- // Mix all other streams.
- bool mixed = false;
- memset(buffer, 0, sizeof(buffer));
- while (chain) {
- if (chain != this &&
- chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
- mixed = true;
- }
- chain = chain->mNext;
- }
- if (mixed) {
- // Saturate into 16 bits.
- for (int i = 0; i < mSampleCount; ++i) {
- int32_t sample = buffer[i];
- if (sample < -32768) {
- sample = -32768;
- }
- if (sample > 32767) {
- sample = 32767;
- }
- samples[i] = sample;
- }
- } else {
- if ((mTick ^ mKeepAlive) >> 10 == 0) {
- return;
- }
- mKeepAlive = mTick;
- memset(samples, 0, sizeof(samples));
+ if (mMode != RECEIVE_ONLY) {
LOGV("stream[%d] no data", mSocket);
}
}
@@ -380,19 +379,16 @@
}
}
- // Adjust the jitter buffer if the latency keeps larger than two times of the
- // packet interval in the past two seconds.
- int score = mBufferTail - tick - mInterval * 2;
- if (mLatencyScore > score) {
+ // Adjust the jitter buffer if the latency keeps larger than the threshold
+ // in the measurement period.
+ int score = mBufferTail - tick - MEASURE_BASE;
+ if (mLatencyScore > score || mLatencyScore <= 0) {
mLatencyScore = score;
- }
- if (mLatencyScore <= 0) {
mLatencyTimer = tick;
- mLatencyScore = score;
} else if (tick - mLatencyTimer >= MEASURE_PERIOD) {
LOGV("stream[%d] reduces latency of %dms", mSocket, mLatencyScore);
mBufferTail -= mLatencyScore;
- mLatencyTimer = tick;
+ mLatencyScore = -1;
}
int count = (BUFFER_SIZE - (mBufferTail - mBufferHead)) * mSampleRate;