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android_frameworks_base/voip/jni/rtp/AudioGroup.cpp
Eric Laurent d7a724e6d8 Issue 4157048: mic gain for VoIP/SIP calls. 
Herring board exhibits a strong echo even in non speakerphone modes.
To compensate the lack of AEC or AES when not in speakerphone, the mic gain
had been reduced in the ADC. But this has an adverse effect on other VoIP applications
that have their own AEC and are penalized by the weak mic gain.

This workaround enables an acceptable mic gain for other VoIP apps while offering a
SIP call experience which is not worse than it was with the residual echo that was
present even with mic gain reduction.

Change-Id: I33fd37858758e94e42ef5b545d3f0dc233220bf1
2011-03-29 18:44:03 -07:00

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/*
* Copyright (C) 2010 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#include <stdio.h>
#include <stdint.h>
#include <string.h>
#include <errno.h>
#include <fcntl.h>
#include <sys/epoll.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/stat.h>
#include <sys/time.h>
#include <time.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#define LOG_TAG "AudioGroup"
#include <cutils/atomic.h>
#include <cutils/properties.h>
#include <utils/Log.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
#include <utils/threads.h>
#include <utils/SystemClock.h>
#include <media/AudioSystem.h>
#include <media/AudioRecord.h>
#include <media/AudioTrack.h>
#include <media/mediarecorder.h>
#include "jni.h"
#include "JNIHelp.h"
#include "AudioCodec.h"
#include "EchoSuppressor.h"
extern int parse(JNIEnv *env, jstring jAddress, int port, sockaddr_storage *ss);
namespace {
using namespace android;
int gRandom = -1;
// We use a circular array to implement jitter buffer. The simplest way is doing
// 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
// 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
// nothing to mix. However, this causes some problems with certain firewalls and
// proxies. A firewall might remove a port mapping when there is no outgoing
// packet for a preiod of time, and a proxy might wait for incoming packets from
// both sides before start forwarding. To solve these problems, we send out a
// silence packet on the stream for every second. It should be good enough to
// keep the stream alive with relatively low resources.
// Other notes:
// + We use elapsedRealtime() to get the time. Since we use 32bit variables
// instead of 64bit ones, comparison must be done by subtraction.
// + Sampling rate must be multiple of 1000Hz, and packet length must be in
// milliseconds. No floating points.
// + If we cannot get enough CPU, we drop samples and simulate packet loss.
// + 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
class AudioStream
{
public:
AudioStream();
~AudioStream();
bool set(int mode, int socket, sockaddr_storage *remote,
AudioCodec *codec, int sampleRate, int sampleCount,
int codecType, int dtmfType);
void sendDtmf(int event);
bool mix(int32_t *output, int head, int tail, int sampleRate);
void encode(int tick, AudioStream *chain);
void decode(int tick);
enum {
NORMAL = 0,
SEND_ONLY = 1,
RECEIVE_ONLY = 2,
LAST_MODE = 2,
};
private:
int mMode;
int mSocket;
sockaddr_storage mRemote;
AudioCodec *mCodec;
uint32_t mCodecMagic;
uint32_t mDtmfMagic;
bool mFixRemote;
int mTick;
int mSampleRate;
int mSampleCount;
int mInterval;
int mKeepAlive;
int16_t *mBuffer;
int mBufferMask;
int mBufferHead;
int mBufferTail;
int mLatencyTimer;
int mLatencyScore;
uint16_t mSequence;
uint32_t mTimestamp;
uint32_t mSsrc;
int mDtmfEvent;
int mDtmfStart;
AudioStream *mNext;
friend class AudioGroup;
};
AudioStream::AudioStream()
{
mSocket = -1;
mCodec = NULL;
mBuffer = NULL;
mNext = NULL;
}
AudioStream::~AudioStream()
{
close(mSocket);
delete mCodec;
delete [] mBuffer;
LOGD("stream[%d] is dead", mSocket);
}
bool AudioStream::set(int mode, int socket, sockaddr_storage *remote,
AudioCodec *codec, int sampleRate, int sampleCount,
int codecType, int dtmfType)
{
if (mode < 0 || mode > LAST_MODE) {
return false;
}
mMode = mode;
mCodecMagic = (0x8000 | codecType) << 16;
mDtmfMagic = (dtmfType == -1) ? 0 : (0x8000 | dtmfType) << 16;
mTick = elapsedRealtime();
mSampleRate = sampleRate / 1000;
mSampleCount = sampleCount;
mInterval = mSampleCount / mSampleRate;
// Allocate jitter buffer.
for (mBufferMask = 8; mBufferMask < mSampleRate; mBufferMask <<= 1);
mBufferMask *= BUFFER_SIZE;
mBuffer = new int16_t[mBufferMask];
--mBufferMask;
mBufferHead = 0;
mBufferTail = 0;
mLatencyTimer = 0;
mLatencyScore = 0;
// Initialize random bits.
read(gRandom, &mSequence, sizeof(mSequence));
read(gRandom, &mTimestamp, sizeof(mTimestamp));
read(gRandom, &mSsrc, sizeof(mSsrc));
mDtmfEvent = -1;
mDtmfStart = 0;
// Only take over these things when succeeded.
mSocket = socket;
if (codec) {
mRemote = *remote;
mCodec = codec;
// Here we should never get an private address, but some buggy proxy
// servers do give us one. To solve this, we replace the address when
// the first time we successfully decode an incoming packet.
mFixRemote = false;
if (remote->ss_family == AF_INET) {
unsigned char *address =
(unsigned char *)&((sockaddr_in *)remote)->sin_addr;
if (address[0] == 10 ||
(address[0] == 172 && (address[1] >> 4) == 1) ||
(address[0] == 192 && address[1] == 168)) {
mFixRemote = true;
}
}
}
LOGD("stream[%d] is configured as %s %dkHz %dms mode %d", mSocket,
(codec ? codec->name : "RAW"), mSampleRate, mInterval, mMode);
return true;
}
void AudioStream::sendDtmf(int event)
{
if (mDtmfMagic != 0) {
mDtmfEvent = event << 24;
mDtmfStart = mTimestamp + mSampleCount;
}
}
bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate)
{
if (mMode == SEND_ONLY) {
return false;
}
if (head - mBufferHead < 0) {
head = mBufferHead;
}
if (tail - mBufferTail > 0) {
tail = mBufferTail;
}
if (tail - head <= 0) {
return false;
}
head *= mSampleRate;
tail *= mSampleRate;
if (sampleRate == mSampleRate) {
for (int i = head; i - tail < 0; ++i) {
output[i - head] += mBuffer[i & mBufferMask];
}
} else {
// TODO: implement resampling.
return false;
}
return true;
}
void AudioStream::encode(int tick, AudioStream *chain)
{
if (tick - mTick >= mInterval) {
// We just missed the train. Pretend that packets in between are lost.
int skipped = (tick - mTick) / mInterval;
mTick += skipped * mInterval;
mSequence += skipped;
mTimestamp += skipped * mSampleCount;
LOGV("stream[%d] skips %d packets", mSocket, skipped);
}
tick = mTick;
mTick += mInterval;
++mSequence;
mTimestamp += mSampleCount;
// If there is an ongoing DTMF event, send it now.
if (mMode != RECEIVE_ONLY && mDtmfEvent != -1) {
int duration = mTimestamp - mDtmfStart;
// Make sure duration is reasonable.
if (duration >= 0 && duration < mSampleRate * 100) {
duration += mSampleCount;
int32_t buffer[4] = {
htonl(mDtmfMagic | mSequence),
htonl(mDtmfStart),
mSsrc,
htonl(mDtmfEvent | duration),
};
if (duration >= mSampleRate * 100) {
buffer[3] |= htonl(1 << 23);
mDtmfEvent = -1;
}
sendto(mSocket, buffer, sizeof(buffer), MSG_DONTWAIT,
(sockaddr *)&mRemote, sizeof(mRemote));
return;
}
mDtmfEvent = -1;
}
int32_t buffer[mSampleCount + 3];
int16_t samples[mSampleCount];
if (mMode == RECEIVE_ONLY) {
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));
LOGV("stream[%d] no data", mSocket);
}
}
if (!mCodec) {
// Special case for device stream.
send(mSocket, samples, sizeof(samples), MSG_DONTWAIT);
return;
}
// Cook the packet and send it out.
buffer[0] = htonl(mCodecMagic | mSequence);
buffer[1] = htonl(mTimestamp);
buffer[2] = mSsrc;
int length = mCodec->encode(&buffer[3], samples);
if (length <= 0) {
LOGV("stream[%d] encoder error", mSocket);
return;
}
sendto(mSocket, buffer, length + 12, MSG_DONTWAIT, (sockaddr *)&mRemote,
sizeof(mRemote));
}
void AudioStream::decode(int tick)
{
char c;
if (mMode == SEND_ONLY) {
recv(mSocket, &c, 1, MSG_DONTWAIT);
return;
}
// Make sure mBufferHead and mBufferTail are reasonable.
if ((unsigned int)(tick + BUFFER_SIZE - mBufferHead) > BUFFER_SIZE * 2) {
mBufferHead = tick - HISTORY_SIZE;
mBufferTail = mBufferHead;
}
if (tick - mBufferHead > HISTORY_SIZE) {
// Throw away outdated samples.
mBufferHead = tick - HISTORY_SIZE;
if (mBufferTail - mBufferHead < 0) {
mBufferTail = mBufferHead;
}
}
// 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) {
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;
}
if (mBufferTail - mBufferHead > BUFFER_SIZE - mInterval) {
// Buffer overflow. Drop the packet.
LOGV("stream[%d] buffer overflow", mSocket);
recv(mSocket, &c, 1, MSG_DONTWAIT);
return;
}
// Receive the packet and decode it.
int16_t samples[mSampleCount];
int length = 0;
if (!mCodec) {
// Special case for device stream.
length = recv(mSocket, samples, sizeof(samples),
MSG_TRUNC | MSG_DONTWAIT) >> 1;
} else {
__attribute__((aligned(4))) uint8_t buffer[2048];
sockaddr_storage remote;
socklen_t len = sizeof(remote);
length = recvfrom(mSocket, buffer, sizeof(buffer),
MSG_TRUNC | MSG_DONTWAIT, (sockaddr *)&remote, &len);
// Do we need to check SSRC, sequence, and timestamp? They are not
// 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);
return;
}
int offset = 12 + ((buffer[0] & 0x0F) << 2);
if ((buffer[0] & 0x10) != 0) {
offset += 4 + (ntohs(*(uint16_t *)&buffer[offset + 2]) << 2);
}
if ((buffer[0] & 0x20) != 0) {
length -= buffer[length - 1];
}
length -= offset;
if (length >= 0) {
length = mCodec->decode(samples, &buffer[offset], length);
}
if (length > 0 && mFixRemote) {
mRemote = remote;
mFixRemote = false;
}
}
if (length <= 0) {
LOGV("stream[%d] decoder error", mSocket);
return;
}
if (tick - mBufferTail > 0) {
// Buffer underrun. Reset the jitter buffer.
LOGV("stream[%d] buffer underrun", mSocket);
if (mBufferTail - mBufferHead <= 0) {
mBufferHead = tick + mInterval;
mBufferTail = mBufferHead;
} else {
int tail = (tick + mInterval) * mSampleRate;
for (int i = mBufferTail * mSampleRate; i - tail < 0; ++i) {
mBuffer[i & mBufferMask] = 0;
}
mBufferTail = tick + mInterval;
}
}
// Append to the jitter buffer.
int tail = mBufferTail * mSampleRate;
for (int i = 0; i < mSampleCount; ++i) {
mBuffer[tail & mBufferMask] = samples[i];
++tail;
}
mBufferTail += mInterval;
}
//------------------------------------------------------------------------------
class AudioGroup
{
public:
AudioGroup();
~AudioGroup();
bool set(int sampleRate, int sampleCount);
bool setMode(int mode);
bool sendDtmf(int event);
bool add(AudioStream *stream);
bool remove(int socket);
enum {
ON_HOLD = 0,
MUTED = 1,
NORMAL = 2,
ECHO_SUPPRESSION = 3,
LAST_MODE = 3,
};
private:
AudioStream *mChain;
int mEventQueue;
volatile int mDtmfEvent;
int mMode;
int mSampleRate;
int mSampleCount;
int mDeviceSocket;
class NetworkThread : public Thread
{
public:
NetworkThread(AudioGroup *group) : Thread(false), mGroup(group) {}
bool start()
{
if (run("Network", ANDROID_PRIORITY_AUDIO) != NO_ERROR) {
LOGE("cannot start network thread");
return false;
}
return true;
}
private:
AudioGroup *mGroup;
bool threadLoop();
};
sp<NetworkThread> mNetworkThread;
class DeviceThread : public Thread
{
public:
DeviceThread(AudioGroup *group) : Thread(false), mGroup(group) {}
bool start()
{
if (run("Device", ANDROID_PRIORITY_AUDIO) != NO_ERROR) {
LOGE("cannot start device thread");
return false;
}
return true;
}
private:
AudioGroup *mGroup;
bool threadLoop();
};
sp<DeviceThread> mDeviceThread;
};
AudioGroup::AudioGroup()
{
mMode = ON_HOLD;
mChain = NULL;
mEventQueue = -1;
mDtmfEvent = -1;
mDeviceSocket = -1;
mNetworkThread = new NetworkThread(this);
mDeviceThread = new DeviceThread(this);
}
AudioGroup::~AudioGroup()
{
mNetworkThread->requestExitAndWait();
mDeviceThread->requestExitAndWait();
close(mEventQueue);
close(mDeviceSocket);
while (mChain) {
AudioStream *next = mChain->mNext;
delete mChain;
mChain = next;
}
LOGD("group[%d] is dead", mDeviceSocket);
}
bool AudioGroup::set(int sampleRate, int sampleCount)
{
mEventQueue = epoll_create(2);
if (mEventQueue == -1) {
LOGE("epoll_create: %s", strerror(errno));
return false;
}
mSampleRate = sampleRate;
mSampleCount = sampleCount;
// Create device socket.
int pair[2];
if (socketpair(AF_UNIX, SOCK_DGRAM, 0, pair)) {
LOGE("socketpair: %s", strerror(errno));
return false;
}
mDeviceSocket = pair[0];
// Create device stream.
mChain = new AudioStream;
if (!mChain->set(AudioStream::NORMAL, pair[1], NULL, NULL,
sampleRate, sampleCount, -1, -1)) {
close(pair[1]);
LOGE("cannot initialize device stream");
return false;
}
// Give device socket a reasonable timeout.
timeval tv;
tv.tv_sec = 0;
tv.tv_usec = 1000 * sampleCount / sampleRate * 500;
if (setsockopt(pair[0], SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv))) {
LOGE("setsockopt: %s", strerror(errno));
return false;
}
// Add device stream into event queue.
epoll_event event;
event.events = EPOLLIN;
event.data.ptr = mChain;
if (epoll_ctl(mEventQueue, EPOLL_CTL_ADD, pair[1], &event)) {
LOGE("epoll_ctl: %s", strerror(errno));
return false;
}
// Anything else?
LOGD("stream[%d] joins group[%d]", pair[1], pair[0]);
return true;
}
bool AudioGroup::setMode(int mode)
{
if (mode < 0 || mode > LAST_MODE) {
return false;
}
//FIXME: temporary code to overcome echo and mic gain issues on herring board.
// Must be modified/removed when proper support for voice processing query and control
// is included in audio framework
char value[PROPERTY_VALUE_MAX];
property_get("ro.product.board", value, "");
if (mode == NORMAL && !strcmp(value, "herring")) {
mode = ECHO_SUPPRESSION;
}
if (mode == ECHO_SUPPRESSION && AudioSystem::getParameters(
0, String8("ec_supported")) == "ec_supported=yes") {
mode = NORMAL;
}
if (mMode == mode) {
return true;
}
mDeviceThread->requestExitAndWait();
LOGD("group[%d] switches from mode %d to %d", mDeviceSocket, mMode, mode);
mMode = mode;
return (mode == ON_HOLD) || mDeviceThread->start();
}
bool AudioGroup::sendDtmf(int event)
{
if (event < 0 || event > 15) {
return false;
}
// DTMF is rarely used, so we try to make it as lightweight as possible.
// Using volatile might be dodgy, but using a pipe or pthread primitives
// or stop-set-restart threads seems too heavy. Will investigate later.
timespec ts;
ts.tv_sec = 0;
ts.tv_nsec = 100000000;
for (int i = 0; mDtmfEvent != -1 && i < 20; ++i) {
nanosleep(&ts, NULL);
}
if (mDtmfEvent != -1) {
return false;
}
mDtmfEvent = event;
nanosleep(&ts, NULL);
return true;
}
bool AudioGroup::add(AudioStream *stream)
{
mNetworkThread->requestExitAndWait();
epoll_event event;
event.events = EPOLLIN;
event.data.ptr = stream;
if (epoll_ctl(mEventQueue, EPOLL_CTL_ADD, stream->mSocket, &event)) {
LOGE("epoll_ctl: %s", strerror(errno));
return false;
}
stream->mNext = mChain->mNext;
mChain->mNext = stream;
if (!mNetworkThread->start()) {
// Only take over the stream when succeeded.
mChain->mNext = stream->mNext;
return false;
}
LOGD("stream[%d] joins group[%d]", stream->mSocket, mDeviceSocket);
return true;
}
bool AudioGroup::remove(int socket)
{
mNetworkThread->requestExitAndWait();
for (AudioStream *stream = mChain; stream->mNext; stream = stream->mNext) {
AudioStream *target = stream->mNext;
if (target->mSocket == socket) {
if (epoll_ctl(mEventQueue, EPOLL_CTL_DEL, socket, NULL)) {
LOGE("epoll_ctl: %s", strerror(errno));
return false;
}
stream->mNext = target->mNext;
LOGD("stream[%d] leaves group[%d]", socket, mDeviceSocket);
delete target;
break;
}
}
// Do not start network thread if there is only one stream.
if (!mChain->mNext || !mNetworkThread->start()) {
return false;
}
return true;
}
bool AudioGroup::NetworkThread::threadLoop()
{
AudioStream *chain = mGroup->mChain;
int tick = elapsedRealtime();
int deadline = tick + 10;
int count = 0;
for (AudioStream *stream = chain; stream; stream = stream->mNext) {
if (tick - stream->mTick >= 0) {
stream->encode(tick, chain);
}
if (deadline - stream->mTick > 0) {
deadline = stream->mTick;
}
++count;
}
int event = mGroup->mDtmfEvent;
if (event != -1) {
for (AudioStream *stream = chain; stream; stream = stream->mNext) {
stream->sendDtmf(event);
}
mGroup->mDtmfEvent = -1;
}
deadline -= tick;
if (deadline < 1) {
deadline = 1;
}
epoll_event events[count];
count = epoll_wait(mGroup->mEventQueue, events, count, deadline);
if (count == -1) {
LOGE("epoll_wait: %s", strerror(errno));
return false;
}
for (int i = 0; i < count; ++i) {
((AudioStream *)events[i].data.ptr)->decode(tick);
}
return true;
}
bool AudioGroup::DeviceThread::threadLoop()
{
int mode = mGroup->mMode;
int sampleRate = mGroup->mSampleRate;
int sampleCount = mGroup->mSampleCount;
int deviceSocket = mGroup->mDeviceSocket;
// Find out the frame count for AudioTrack and AudioRecord.
int output = 0;
int input = 0;
if (AudioTrack::getMinFrameCount(&output, AudioSystem::VOICE_CALL,
sampleRate) != NO_ERROR || output <= 0 ||
AudioRecord::getMinFrameCount(&input, sampleRate,
AudioSystem::PCM_16_BIT, 1) != NO_ERROR || input <= 0) {
LOGE("cannot compute frame count");
return false;
}
LOGD("reported frame count: output %d, input %d", output, input);
if (output < sampleCount * 2) {
output = sampleCount * 2;
}
if (input < sampleCount * 2) {
input = sampleCount * 2;
}
LOGD("adjusted frame count: output %d, input %d", output, input);
// Initialize AudioTrack and AudioRecord.
AudioTrack track;
AudioRecord record;
if (track.set(AudioSystem::VOICE_CALL, sampleRate, AudioSystem::PCM_16_BIT,
AudioSystem::CHANNEL_OUT_MONO, output) != NO_ERROR || record.set(
AUDIO_SOURCE_VOICE_COMMUNICATION, sampleRate, AudioSystem::PCM_16_BIT,
AudioSystem::CHANNEL_IN_MONO, input) != NO_ERROR) {
LOGE("cannot initialize audio device");
return false;
}
LOGD("latency: output %d, input %d", track.latency(), record.latency());
// Initialize echo canceler.
EchoSuppressor echo(sampleCount,
(track.latency() + record.latency()) * sampleRate / 1000);
// Give device socket a reasonable buffer size.
setsockopt(deviceSocket, SOL_SOCKET, SO_RCVBUF, &output, sizeof(output));
setsockopt(deviceSocket, SOL_SOCKET, SO_SNDBUF, &output, sizeof(output));
// Drain device socket.
char c;
while (recv(deviceSocket, &c, 1, MSG_DONTWAIT) == 1);
// Start AudioRecord before AudioTrack. This prevents AudioTrack from being
// disabled due to buffer underrun while waiting for AudioRecord.
if (mode != MUTED) {
record.start();
int16_t one;
record.read(&one, sizeof(one));
}
track.start();
while (!exitPending()) {
int16_t output[sampleCount];
if (recv(deviceSocket, output, sizeof(output), 0) <= 0) {
memset(output, 0, sizeof(output));
}
int16_t input[sampleCount];
int toWrite = sampleCount;
int toRead = (mode == MUTED) ? 0 : sampleCount;
int chances = 100;
while (--chances > 0 && (toWrite > 0 || toRead > 0)) {
if (toWrite > 0) {
AudioTrack::Buffer buffer;
buffer.frameCount = toWrite;
status_t status = track.obtainBuffer(&buffer, 1);
if (status == NO_ERROR) {
int offset = sampleCount - toWrite;
memcpy(buffer.i8, &output[offset], buffer.size);
toWrite -= buffer.frameCount;
track.releaseBuffer(&buffer);
} else if (status != TIMED_OUT && status != WOULD_BLOCK) {
LOGE("cannot write to AudioTrack");
return true;
}
}
if (toRead > 0) {
AudioRecord::Buffer buffer;
buffer.frameCount = toRead;
status_t status = record.obtainBuffer(&buffer, 1);
if (status == NO_ERROR) {
int offset = sampleCount - toRead;
memcpy(&input[offset], buffer.i8, buffer.size);
toRead -= buffer.frameCount;
record.releaseBuffer(&buffer);
} else if (status != TIMED_OUT && status != WOULD_BLOCK) {
LOGE("cannot read from AudioRecord");
return true;
}
}
}
if (chances <= 0) {
LOGW("device loop timeout");
while (recv(deviceSocket, &c, 1, MSG_DONTWAIT) == 1);
}
if (mode != MUTED) {
if (mode == NORMAL) {
send(deviceSocket, input, sizeof(input), MSG_DONTWAIT);
} else {
echo.run(output, input);
send(deviceSocket, input, sizeof(input), MSG_DONTWAIT);
}
}
}
return false;
}
//------------------------------------------------------------------------------
static jfieldID gNative;
static jfieldID gMode;
void add(JNIEnv *env, jobject thiz, jint mode,
jint socket, jstring jRemoteAddress, jint remotePort,
jstring jCodecSpec, jint dtmfType)
{
AudioCodec *codec = NULL;
AudioStream *stream = NULL;
AudioGroup *group = NULL;
// Sanity check.
sockaddr_storage remote;
if (parse(env, jRemoteAddress, remotePort, &remote) < 0) {
// Exception already thrown.
return;
}
if (!jCodecSpec) {
jniThrowNullPointerException(env, "codecSpec");
return;
}
const char *codecSpec = env->GetStringUTFChars(jCodecSpec, NULL);
if (!codecSpec) {
// Exception already thrown.
return;
}
// Create audio codec.
int codecType = -1;
char codecName[16];
int sampleRate = -1;
sscanf(codecSpec, "%d %15[^/]%*c%d", &codecType, codecName, &sampleRate);
codec = newAudioCodec(codecName);
int sampleCount = (codec ? codec->set(sampleRate, codecSpec) : -1);
env->ReleaseStringUTFChars(jCodecSpec, codecSpec);
if (sampleCount <= 0) {
jniThrowException(env, "java/lang/IllegalStateException",
"cannot initialize audio codec");
goto error;
}
// Create audio stream.
stream = new AudioStream;
if (!stream->set(mode, socket, &remote, codec, sampleRate, sampleCount,
codecType, dtmfType)) {
jniThrowException(env, "java/lang/IllegalStateException",
"cannot initialize audio stream");
goto error;
}
socket = -1;
codec = NULL;
// Create audio group.
group = (AudioGroup *)env->GetIntField(thiz, gNative);
if (!group) {
int mode = env->GetIntField(thiz, gMode);
group = new AudioGroup;
if (!group->set(8000, 256) || !group->setMode(mode)) {
jniThrowException(env, "java/lang/IllegalStateException",
"cannot initialize audio group");
goto error;
}
}
// Add audio stream into audio group.
if (!group->add(stream)) {
jniThrowException(env, "java/lang/IllegalStateException",
"cannot add audio stream");
goto error;
}
// Succeed.
env->SetIntField(thiz, gNative, (int)group);
return;
error:
delete group;
delete stream;
delete codec;
close(socket);
env->SetIntField(thiz, gNative, NULL);
}
void remove(JNIEnv *env, jobject thiz, jint socket)
{
AudioGroup *group = (AudioGroup *)env->GetIntField(thiz, gNative);
if (group) {
if (socket == -1 || !group->remove(socket)) {
delete group;
env->SetIntField(thiz, gNative, NULL);
}
}
}
void setMode(JNIEnv *env, jobject thiz, jint mode)
{
if (mode < 0 || mode > AudioGroup::LAST_MODE) {
jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
return;
}
AudioGroup *group = (AudioGroup *)env->GetIntField(thiz, gNative);
if (group && !group->setMode(mode)) {
jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
return;
}
env->SetIntField(thiz, gMode, mode);
}
void sendDtmf(JNIEnv *env, jobject thiz, jint event)
{
AudioGroup *group = (AudioGroup *)env->GetIntField(thiz, gNative);
if (group && !group->sendDtmf(event)) {
jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
}
}
JNINativeMethod gMethods[] = {
{"add", "(IILjava/lang/String;ILjava/lang/String;I)V", (void *)add},
{"remove", "(I)V", (void *)remove},
{"setMode", "(I)V", (void *)setMode},
{"sendDtmf", "(I)V", (void *)sendDtmf},
};
} // namespace
int registerAudioGroup(JNIEnv *env)
{
gRandom = open("/dev/urandom", O_RDONLY);
if (gRandom == -1) {
LOGE("urandom: %s", strerror(errno));
return -1;
}
jclass clazz;
if ((clazz = env->FindClass("android/net/rtp/AudioGroup")) == NULL ||
(gNative = env->GetFieldID(clazz, "mNative", "I")) == NULL ||
(gMode = env->GetFieldID(clazz, "mMode", "I")) == NULL ||
env->RegisterNatives(clazz, gMethods, NELEM(gMethods)) < 0) {
LOGE("JNI registration failed");
return -1;
}
return 0;
}
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