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android_frameworks_base/voip/jni/rtp/AudioGroup.cpp
Chia-chi Yeh 557b04de23 RTP: prevent buffer overflow in AudioRecord. 
This change simply reduces the receive timeout of DeviceSocket. It works
because AudioRecord will block us till there is enough data, which makes
AudioSocket overlap AudioRecord.

Change-Id: I4700224fb407e148ef359a9d99279e10240128d0
2010-09-08 09:56:02 +08:00

959 lines
26 KiB
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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 <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"
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 256ms long and assume packet interval is 32ms or less.
// The first 64ms is the place where samples get mixed. The rest 192ms is the
// real jitter buffer. For a stream at 8000Hz it takes 4096 bytes. These numbers
// are chosen by experiments and each of them can be adjusted as needed.
// 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 we might only support 8kHz and 16kHz.
class AudioStream
{
public:
AudioStream();
~AudioStream();
bool set(int mode, int socket, sockaddr_storage *remote,
const char *codecName, 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);
private:
enum {
NORMAL = 0,
SEND_ONLY = 1,
RECEIVE_ONLY = 2,
LAST_MODE = 2,
};
int mMode;
int mSocket;
sockaddr_storage mRemote;
AudioCodec *mCodec;
uint32_t mCodecMagic;
uint32_t mDtmfMagic;
int mTick;
int mSampleRate;
int mSampleCount;
int mInterval;
int16_t *mBuffer;
int mBufferMask;
int mBufferHead;
int mBufferTail;
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,
const char *codecName, int sampleRate, int sampleCount,
int codecType, int dtmfType)
{
if (mode < 0 || mode > LAST_MODE) {
return false;
}
mMode = mode;
if (codecName) {
mRemote = *remote;
mCodec = newAudioCodec(codecName);
if (!mCodec || !mCodec->set(sampleRate, sampleCount)) {
return false;
}
}
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 = 8192; mBufferMask < sampleRate; mBufferMask <<= 1);
mBufferMask >>= 2;
mBuffer = new int16_t[mBufferMask];
--mBufferMask;
mBufferHead = 0;
mBufferTail = 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 the socket when succeeded.
mSocket = socket;
LOGD("stream[%d] is configured as %s %dkHz %dms", mSocket,
(codecName ? codecName : "RAW"), mSampleRate, mInterval);
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;
LOGD("stream[%d] skips %d packets", mSocket, skipped);
}
tick = mTick;
mTick += mInterval;
++mSequence;
mTimestamp += mSampleCount;
if (mMode == RECEIVE_ONLY) {
return;
}
// If there is an ongoing DTMF event, send it now.
if (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;
}
// It is time to mix streams.
bool mixed = false;
int32_t buffer[mSampleCount + 3];
memset(buffer, 0, sizeof(buffer));
while (chain) {
if (chain != this &&
chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
mixed = true;
}
chain = chain->mNext;
}
if (!mixed) {
LOGD("stream[%d] no data", mSocket);
return;
}
// Cook the packet and send it out.
int16_t samples[mSampleCount];
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;
}
if (!mCodec) {
// Special case for device stream.
send(mSocket, samples, sizeof(samples), MSG_DONTWAIT);
return;
}
buffer[0] = htonl(mCodecMagic | mSequence);
buffer[1] = htonl(mTimestamp);
buffer[2] = mSsrc;
int length = mCodec->encode(&buffer[3], samples);
if (length <= 0) {
LOGD("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 + 256 - mBufferHead) > 1024) {
mBufferHead = tick - 64;
mBufferTail = mBufferHead;
}
if (tick - mBufferHead > 64) {
// Throw away outdated samples.
mBufferHead = tick - 64;
if (mBufferTail - mBufferHead < 0) {
mBufferTail = mBufferHead;
}
}
if (mBufferTail - tick <= 80) {
mLatencyScore = tick;
} else if (tick - mLatencyScore >= 5000) {
// Reset the jitter buffer to 40ms if the latency keeps larger than 80ms
// in the past 5s. This rarely happens, so let us just keep it simple.
LOGD("stream[%d] latency control", mSocket);
mBufferTail = tick + 40;
}
if (mBufferTail - mBufferHead > 256 - mInterval) {
// Buffer overflow. Drop the packet.
LOGD("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];
length = recv(mSocket, buffer, sizeof(buffer),
MSG_TRUNC | MSG_DONTWAIT);
// 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) {
LOGD("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 != mSampleCount) {
LOGD("stream[%d] decoder error", mSocket);
return;
}
if (tick - mBufferTail > 0) {
// Buffer underrun. Reset the jitter buffer to 40ms.
LOGD("stream[%d] buffer underrun", mSocket);
if (mBufferTail - mBufferHead <= 0) {
mBufferHead = tick + 40;
mBufferTail = mBufferHead;
} else {
int tail = (tick + 40) * mSampleRate;
for (int i = mBufferTail * mSampleRate; i - tail < 0; ++i) {
mBuffer[i & mBufferMask] = 0;
}
mBufferTail = tick + 40;
}
}
// 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);
private:
enum {
ON_HOLD = 0,
MUTED = 1,
NORMAL = 2,
EC_ENABLED = 3,
LAST_MODE = 3,
};
int mMode;
AudioStream *mChain;
int mEventQueue;
volatile int mDtmfEvent;
int mSampleCount;
int mDeviceSocket;
AudioTrack mTrack;
AudioRecord mRecord;
bool networkLoop();
bool deviceLoop();
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()
{
return mGroup->networkLoop();
}
};
sp<NetworkThread> mNetworkThread;
class DeviceThread : public Thread
{
public:
DeviceThread(AudioGroup *group) : Thread(false), mGroup(group) {}
bool start()
{
char c;
while (recv(mGroup->mDeviceSocket, &c, 1, MSG_DONTWAIT) == 1);
if (run("Device", ANDROID_PRIORITY_AUDIO) != NO_ERROR) {
LOGE("cannot start device thread");
return false;
}
return true;
}
private:
AudioGroup *mGroup;
bool threadLoop()
{
return mGroup->deviceLoop();
}
};
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();
mTrack.stop();
mRecord.stop();
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;
}
mSampleCount = sampleCount;
// 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.
if (mTrack.set(AudioSystem::VOICE_CALL, sampleRate, AudioSystem::PCM_16_BIT,
AudioSystem::CHANNEL_OUT_MONO, output) != NO_ERROR ||
mRecord.set(AUDIO_SOURCE_MIC, 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", mTrack.latency(), mRecord.latency());
// TODO: initialize echo canceler here.
// 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 and buffer size.
timeval tv;
tv.tv_sec = 0;
tv.tv_usec = 1000 * sampleCount / sampleRate * 500;
if (setsockopt(pair[0], SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv)) ||
setsockopt(pair[0], SOL_SOCKET, SO_RCVBUF, &output, sizeof(output)) ||
setsockopt(pair[1], SOL_SOCKET, SO_SNDBUF, &output, sizeof(output))) {
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;
}
if (mMode == mode) {
return true;
}
LOGD("group[%d] switches from mode %d to %d", mDeviceSocket, mMode, mode);
mMode = mode;
mDeviceThread->requestExitAndWait();
if (mode == ON_HOLD) {
mTrack.stop();
mRecord.stop();
return true;
}
mTrack.start();
if (mode == MUTED) {
mRecord.stop();
} else {
mRecord.start();
}
if (!mDeviceThread->start()) {
mTrack.stop();
mRecord.stop();
return false;
}
return true;
}
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::networkLoop()
{
int tick = elapsedRealtime();
int deadline = tick + 10;
int count = 0;
for (AudioStream *stream = mChain; stream; stream = stream->mNext) {
if (!stream->mTick || tick - stream->mTick >= 0) {
stream->encode(tick, mChain);
}
if (deadline - stream->mTick > 0) {
deadline = stream->mTick;
}
++count;
}
if (mDtmfEvent != -1) {
int event = mDtmfEvent;
for (AudioStream *stream = mChain; stream; stream = stream->mNext) {
stream->sendDtmf(event);
}
mDtmfEvent = -1;
}
deadline -= tick;
if (deadline < 1) {
deadline = 1;
}
epoll_event events[count];
count = epoll_wait(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::deviceLoop()
{
int16_t output[mSampleCount];
if (recv(mDeviceSocket, output, sizeof(output), 0) <= 0) {
memset(output, 0, sizeof(output));
}
int16_t input[mSampleCount];
int toWrite = mSampleCount;
int toRead = (mMode == MUTED) ? 0 : mSampleCount;
int chances = 100;
while (--chances > 0 && (toWrite > 0 || toRead > 0)) {
if (toWrite > 0) {
AudioTrack::Buffer buffer;
buffer.frameCount = toWrite;
status_t status = mTrack.obtainBuffer(&buffer, 1);
if (status == NO_ERROR) {
memcpy(buffer.i8, &output[mSampleCount - toWrite], buffer.size);
toWrite -= buffer.frameCount;
mTrack.releaseBuffer(&buffer);
} else if (status != TIMED_OUT && status != WOULD_BLOCK) {
LOGE("cannot write to AudioTrack");
return false;
}
}
if (toRead > 0) {
AudioRecord::Buffer buffer;
buffer.frameCount = mRecord.frameCount();
status_t status = mRecord.obtainBuffer(&buffer, 1);
if (status == NO_ERROR) {
int count = ((int)buffer.frameCount < toRead) ?
buffer.frameCount : toRead;
memcpy(&input[mSampleCount - toRead], buffer.i8, count * 2);
toRead -= count;
if (buffer.frameCount < mRecord.frameCount()) {
buffer.frameCount = count;
}
mRecord.releaseBuffer(&buffer);
} else if (status != TIMED_OUT && status != WOULD_BLOCK) {
LOGE("cannot read from AudioRecord");
return false;
}
}
}
if (!chances) {
LOGE("device loop timeout");
return false;
}
if (mMode != MUTED) {
if (mMode == NORMAL) {
send(mDeviceSocket, input, sizeof(input), MSG_DONTWAIT);
} else {
// TODO: Echo canceller runs here.
send(mDeviceSocket, input, sizeof(input), MSG_DONTWAIT);
}
}
return true;
}
//------------------------------------------------------------------------------
static jfieldID gNative;
static jfieldID gMode;
void add(JNIEnv *env, jobject thiz, jint mode,
jint socket, jstring jRemoteAddress, jint remotePort,
jstring jCodecName, jint sampleRate, jint sampleCount,
jint codecType, jint dtmfType)
{
const char *codecName = NULL;
AudioStream *stream = NULL;
AudioGroup *group = NULL;
// Sanity check.
sockaddr_storage remote;
if (parse(env, jRemoteAddress, remotePort, &remote) < 0) {
// Exception already thrown.
goto error;
}
if (sampleRate < 0 || sampleCount < 0 || codecType < 0 || codecType > 127) {
jniThrowException(env, "java/lang/IllegalArgumentException", NULL);
goto error;
}
if (!jCodecName) {
jniThrowNullPointerException(env, "codecName");
goto error;
}
codecName = env->GetStringUTFChars(jCodecName, NULL);
if (!codecName) {
// Exception already thrown.
goto error;
}
// Create audio stream.
stream = new AudioStream;
if (!stream->set(mode, socket, &remote, codecName, sampleRate, sampleCount,
codecType, dtmfType)) {
jniThrowException(env, "java/lang/IllegalStateException",
"cannot initialize audio stream");
env->ReleaseStringUTFChars(jCodecName, codecName);
goto error;
}
env->ReleaseStringUTFChars(jCodecName, codecName);
socket = -1;
// 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;
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)
{
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;IIII)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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