Merge "Make sure we only reallocate buffers on a genuine port definition change."
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 42a22b7..12f8f32 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -263,6 +263,10 @@
}
}
+ if (maxSocket == -1) {
+ return;
+ }
+
int res = select(maxSocket + 1, &rs, NULL, NULL, &tv);
CHECK_GE(res, 0);
@@ -292,6 +296,10 @@
it != mStreams.end(); ++it) {
StreamInfo *s = &*it;
+ if (s->mIsInjected) {
+ continue;
+ }
+
if (s->mNumRTCPPacketsReceived == 0) {
// We have never received any RTCP packets on this stream,
// we don't even know where to send a report.
diff --git a/media/libstagefright/rtsp/ARTPSession.cpp b/media/libstagefright/rtsp/ARTPSession.cpp
index d2c56f7..f60c1a9 100644
--- a/media/libstagefright/rtsp/ARTPSession.cpp
+++ b/media/libstagefright/rtsp/ARTPSession.cpp
@@ -167,6 +167,7 @@
#endif
accessUnit->meta()->setInt64("ntp-time", ntpTime);
+ accessUnit->meta()->setInt64("timeUs", 0);
#if 0
int32_t damaged;
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 7c06db3..3d8d86b 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -31,8 +31,6 @@
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MediaErrors.h>
-#define USE_TCP_INTERLEAVED 0
-
// If no access units are received within 3 secs, assume that the rtp
// stream has ended and signal end of stream.
static int64_t kAccessUnitTimeoutUs = 3000000ll;
@@ -83,7 +81,8 @@
mFirstAccessUnit(true),
mFirstAccessUnitNTP(0),
mNumAccessUnitsReceived(0),
- mCheckPending(false) {
+ mCheckPending(false),
+ mTryTCPInterleaving(false) {
mNetLooper->setName("rtsp net");
mNetLooper->start(false /* runOnCallingThread */,
false /* canCallJava */,
@@ -158,7 +157,13 @@
case 'disc':
{
- (new AMessage('quit', id()))->post();
+ int32_t reconnect;
+ if (msg->findInt32("reconnect", &reconnect) && reconnect) {
+ sp<AMessage> reply = new AMessage('conn', id());
+ mConn->connect(mSessionURL.c_str(), reply);
+ } else {
+ (new AMessage('quit', id()))->post();
+ }
break;
}
@@ -325,10 +330,6 @@
parsePlayResponse(response);
- mDoneMsg->setInt32("result", OK);
- mDoneMsg->post();
- mDoneMsg = NULL;
-
sp<AMessage> timeout = new AMessage('tiou', id());
timeout->post(kStartupTimeoutUs);
} else {
@@ -342,12 +343,26 @@
case 'abor':
{
for (size_t i = 0; i < mTracks.size(); ++i) {
- mTracks.editItemAt(i).mPacketSource->signalEOS(
- ERROR_END_OF_STREAM);
+ TrackInfo *info = &mTracks.editItemAt(i);
+
+ info->mPacketSource->signalEOS(ERROR_END_OF_STREAM);
+
+ if (!info->mUsingInterleavedTCP) {
+ mRTPConn->removeStream(info->mRTPSocket, info->mRTCPSocket);
+
+ close(info->mRTPSocket);
+ close(info->mRTCPSocket);
+ }
}
+ mTracks.clear();
sp<AMessage> reply = new AMessage('tear', id());
+ int32_t reconnect;
+ if (msg->findInt32("reconnect", &reconnect) && reconnect) {
+ reply->setInt32("reconnect", true);
+ }
+
AString request;
request = "TEARDOWN ";
@@ -374,6 +389,12 @@
<< result << " (" << strerror(-result) << ")";
sp<AMessage> reply = new AMessage('disc', id());
+
+ int32_t reconnect;
+ if (msg->findInt32("reconnect", &reconnect) && reconnect) {
+ reply->setInt32("reconnect", true);
+ }
+
mConn->disconnect(reply);
break;
}
@@ -414,6 +435,11 @@
size_t trackIndex;
CHECK(msg->findSize("track-index", &trackIndex));
+ if (trackIndex >= mTracks.size()) {
+ LOG(ERROR) << "late packets ignored.";
+ break;
+ }
+
TrackInfo *track = &mTracks.editItemAt(trackIndex);
int32_t eos;
@@ -457,6 +483,10 @@
}
if (mFirstAccessUnit) {
+ mDoneMsg->setInt32("result", OK);
+ mDoneMsg->post();
+ mDoneMsg = NULL;
+
mFirstAccessUnit = false;
mFirstAccessUnitNTP = ntpTime;
}
@@ -583,8 +613,19 @@
case 'tiou':
{
if (mFirstAccessUnit) {
- LOG(WARNING) << "Never received any data, disconnecting.";
- (new AMessage('abor', id()))->post();
+ if (mTryTCPInterleaving) {
+ LOG(WARNING) << "Never received any data, disconnecting.";
+ (new AMessage('abor', id()))->post();
+ } else {
+ LOG(WARNING)
+ << "Never received any data, switching transports.";
+
+ mTryTCPInterleaving = true;
+
+ sp<AMessage> msg = new AMessage('abor', id());
+ msg->setInt32("reconnect", true);
+ msg->post();
+ }
}
break;
}
@@ -705,6 +746,7 @@
uint64_t mFirstAccessUnitNTP;
int64_t mNumAccessUnitsReceived;
bool mCheckPending;
+ bool mTryTCPInterleaving;
struct TrackInfo {
AString mURL;
@@ -723,6 +765,7 @@
void setupTrack(size_t index) {
sp<APacketSource> source =
new APacketSource(mSessionDesc, index);
+
if (source->initCheck() != OK) {
LOG(WARNING) << "Unsupported format. Ignoring track #"
<< index << ".";
@@ -754,26 +797,26 @@
request.append(trackURL);
request.append(" RTSP/1.0\r\n");
-#if USE_TCP_INTERLEAVED
- size_t interleaveIndex = 2 * (mTracks.size() - 1);
- info->mUsingInterleavedTCP = true;
- info->mRTPSocket = interleaveIndex;
- info->mRTCPSocket = interleaveIndex + 1;
+ if (mTryTCPInterleaving) {
+ size_t interleaveIndex = 2 * (mTracks.size() - 1);
+ info->mUsingInterleavedTCP = true;
+ info->mRTPSocket = interleaveIndex;
+ info->mRTCPSocket = interleaveIndex + 1;
- request.append("Transport: RTP/AVP/TCP;interleaved=");
- request.append(interleaveIndex);
- request.append("-");
- request.append(interleaveIndex + 1);
-#else
- unsigned rtpPort;
- ARTPConnection::MakePortPair(
- &info->mRTPSocket, &info->mRTCPSocket, &rtpPort);
+ request.append("Transport: RTP/AVP/TCP;interleaved=");
+ request.append(interleaveIndex);
+ request.append("-");
+ request.append(interleaveIndex + 1);
+ } else {
+ unsigned rtpPort;
+ ARTPConnection::MakePortPair(
+ &info->mRTPSocket, &info->mRTCPSocket, &rtpPort);
- request.append("Transport: RTP/AVP/UDP;unicast;client_port=");
- request.append(rtpPort);
- request.append("-");
- request.append(rtpPort + 1);
-#endif
+ request.append("Transport: RTP/AVP/UDP;unicast;client_port=");
+ request.append(rtpPort);
+ request.append("-");
+ request.append(rtpPort + 1);
+ }
request.append("\r\n");