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RTP: Update parameters for larger packet intervals.

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Also remove some duplicated code.

Change-Id: I64576e5442a962eb4b0dfa83b52a8127567ba597
This commit is contained in:
Chia-chi Yeh
2011-09-06 23:45:17 -07:00
parent 35d05dcba1
commit be57bfe853
1 changed files with 42 additions and 46 deletions
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88
voip/jni/rtp/AudioGroup.cpp
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@@ -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 @@ int gRandom = -1;
// 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 @@ int gRandom = -1;
// + 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 @@ void AudioStream::encode(int tick, AudioStream *chain)
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 @@ void AudioStream::encode(int tick, AudioStream *chain)
mSsrc,
htonl(mDtmfEvent | duration),
};
if (duration >= mSampleRate * 100) {
if (duration >= mSampleRate * DTMF_PERIOD) {
buffer[3] |= htonl(1 << 23);
mDtmfEvent = -1;
}
@@ -298,43 +301,39 @@ void AudioStream::encode(int tick, AudioStream *chain)
}
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 @@ void AudioStream::decode(int tick)
}
}
// 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;