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authorAndreas Huber <andih@google.com>2010-08-10 12:59:27 -0700
committerAndroid (Google) Code Review <android-gerrit@google.com>2010-08-10 12:59:27 -0700
commit18f0174ff4eef46f121e77a0dc1d7a8271621f48 (patch)
tree8b26b96602be5cf3aa59097317b9e63c1ab4d49a /media
parent605f402796afa9c88e2f91284e98d1f59a0726aa (diff)
parentf88f84414ae7baead03497f1d650ad8ea2f87688 (diff)
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Merge "We're now going to ignore timestamps completely in gtalk video conferencing, playing video as soon as it comes in. We also make up fake timestamps in the rtp code, ignoring rtcp SR information to enable early startup." into gingerbread
Diffstat (limited to 'media')
-rw-r--r--media/libstagefright/AwesomePlayer.cpp6
-rw-r--r--media/libstagefright/OMXCodec.cpp8
-rw-r--r--media/libstagefright/rtsp/APacketSource.cpp39
-rw-r--r--media/libstagefright/rtsp/ARTPConnection.cpp24
-rw-r--r--media/libstagefright/rtsp/ARTPConnection.h9
-rw-r--r--media/libstagefright/rtsp/ARTPSession.cpp5
-rw-r--r--media/libstagefright/rtsp/ARTPSource.cpp9
7 files changed, 61 insertions, 39 deletions
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index e426fca..b16a136 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -1019,6 +1019,12 @@ void AwesomePlayer::onVideoEvent() {
int64_t latenessUs = nowUs - timeUs;
+ if (mRTPSession != NULL) {
+ // We'll completely ignore timestamps for gtalk videochat
+ // and we'll play incoming video as fast as we get it.
+ latenessUs = 0;
+ }
+
if (latenessUs > 40000) {
// We're more than 40ms late.
LOGV("we're late by %lld us (%.2f secs)", latenessUs, latenessUs / 1E6);
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 11396ef..4741b1d 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -1685,6 +1685,14 @@ void OMXCodec::on_message(const omx_message &msg) {
MediaBuffer *buffer = info->mMediaBuffer;
+ if (msg.u.extended_buffer_data.range_offset
+ + msg.u.extended_buffer_data.range_length
+ > buffer->size()) {
+ CODEC_LOGE(
+ "Codec lied about its buffer size requirements, "
+ "sending a buffer larger than the originally "
+ "advertised size in FILL_BUFFER_DONE!");
+ }
buffer->set_range(
msg.u.extended_buffer_data.range_offset,
msg.u.extended_buffer_data.range_length);
diff --git a/media/libstagefright/rtsp/APacketSource.cpp b/media/libstagefright/rtsp/APacketSource.cpp
index a577704..395cd28 100644
--- a/media/libstagefright/rtsp/APacketSource.cpp
+++ b/media/libstagefright/rtsp/APacketSource.cpp
@@ -356,24 +356,10 @@ status_t APacketSource::read(
if (!mBuffers.empty()) {
const sp<ABuffer> buffer = *mBuffers.begin();
- uint64_t ntpTime;
- CHECK(buffer->meta()->findInt64(
- "ntp-time", (int64_t *)&ntpTime));
-
MediaBuffer *mediaBuffer = new MediaBuffer(buffer->size());
- mediaBuffer->meta_data()->setInt64(kKeyNTPTime, ntpTime);
-
- if (mFirstAccessUnit) {
- mFirstAccessUnit = false;
- mFirstAccessUnitNTP = ntpTime;
- }
- if (ntpTime > mFirstAccessUnitNTP) {
- ntpTime -= mFirstAccessUnitNTP;
- } else {
- ntpTime = 0;
- }
- int64_t timeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
+ int64_t timeUs;
+ CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
mediaBuffer->meta_data()->setInt64(kKeyTime, timeUs);
@@ -390,10 +376,29 @@ status_t APacketSource::read(
void APacketSource::queueAccessUnit(const sp<ABuffer> &buffer) {
int32_t damaged;
if (buffer->meta()->findInt32("damaged", &damaged) && damaged) {
- // LOG(VERBOSE) << "discarding damaged AU";
+ LOG(INFO) << "discarding damaged AU";
return;
}
+ uint64_t ntpTime;
+ CHECK(buffer->meta()->findInt64(
+ "ntp-time", (int64_t *)&ntpTime));
+
+ if (mFirstAccessUnit) {
+ mFirstAccessUnit = false;
+ mFirstAccessUnitNTP = ntpTime;
+ }
+
+ if (ntpTime > mFirstAccessUnitNTP) {
+ ntpTime -= mFirstAccessUnitNTP;
+ } else {
+ ntpTime = 0;
+ }
+
+ int64_t timeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
+
+ buffer->meta()->setInt64("timeUs", timeUs);
+
Mutex::Autolock autoLock(mLock);
mBuffers.push_back(buffer);
mCondition.signal();
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 5bd306b..469af3e 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -28,8 +28,6 @@
#include <arpa/inet.h>
#include <sys/socket.h>
-#define IGNORE_RTCP_TIME 0
-
namespace android {
static const size_t kMaxUDPSize = 1500;
@@ -61,8 +59,9 @@ struct ARTPConnection::StreamInfo {
struct sockaddr_in mRemoteRTCPAddr;
};
-ARTPConnection::ARTPConnection()
- : mPollEventPending(false),
+ARTPConnection::ARTPConnection(uint32_t flags)
+ : mFlags(flags),
+ mPollEventPending(false),
mLastReceiverReportTimeUs(-1) {
}
@@ -280,7 +279,10 @@ void ARTPConnection::onPollStreams() {
sp<ARTPSource> source = s->mSources.valueAt(i);
source->addReceiverReport(buffer);
- source->addFIR(buffer);
+
+ if (mFlags & kRegularlyRequestFIR) {
+ source->addFIR(buffer);
+ }
}
if (buffer->size() > 0) {
@@ -405,13 +407,11 @@ status_t ARTPConnection::parseRTP(StreamInfo *s, const sp<ABuffer> &buffer) {
buffer->setInt32Data(u16at(&data[2]));
buffer->setRange(payloadOffset, size - payloadOffset);
-#if IGNORE_RTCP_TIME
- if (!source->timeEstablished()) {
+ if ((mFlags & kFakeTimestamps) && !source->timeEstablished()) {
source->timeUpdate(rtpTime, 0);
- source->timeUpdate(rtpTime + 20, 0x100000000ll);
+ source->timeUpdate(rtpTime + 90000, 0x100000000ll);
CHECK(source->timeEstablished());
}
-#endif
source->processRTPPacket(buffer);
@@ -533,9 +533,9 @@ status_t ARTPConnection::parseSR(
sp<ARTPSource> source = findSource(s, id);
-#if !IGNORE_RTCP_TIME
- source->timeUpdate(rtpTime, ntpTime);
-#endif
+ if ((mFlags & kFakeTimestamps) == 0) {
+ source->timeUpdate(rtpTime, ntpTime);
+ }
return 0;
}
diff --git a/media/libstagefright/rtsp/ARTPConnection.h b/media/libstagefright/rtsp/ARTPConnection.h
index 49839ad..c535199 100644
--- a/media/libstagefright/rtsp/ARTPConnection.h
+++ b/media/libstagefright/rtsp/ARTPConnection.h
@@ -28,7 +28,12 @@ struct ARTPSource;
struct ASessionDescription;
struct ARTPConnection : public AHandler {
- ARTPConnection();
+ enum Flags {
+ kFakeTimestamps = 1,
+ kRegularlyRequestFIR = 2,
+ };
+
+ ARTPConnection(uint32_t flags = 0);
void addStream(
int rtpSocket, int rtcpSocket,
@@ -56,6 +61,8 @@ private:
static const int64_t kSelectTimeoutUs;
+ uint32_t mFlags;
+
struct StreamInfo;
List<StreamInfo> mStreams;
diff --git a/media/libstagefright/rtsp/ARTPSession.cpp b/media/libstagefright/rtsp/ARTPSession.cpp
index 0e0f45a..e082078 100644
--- a/media/libstagefright/rtsp/ARTPSession.cpp
+++ b/media/libstagefright/rtsp/ARTPSession.cpp
@@ -40,7 +40,10 @@ status_t ARTPSession::setup(const sp<ASessionDescription> &desc) {
mDesc = desc;
- mRTPConn = new ARTPConnection;
+ mRTPConn = new ARTPConnection(
+ ARTPConnection::kFakeTimestamps
+ | ARTPConnection::kRegularlyRequestFIR);
+
looper()->registerHandler(mRTPConn);
for (size_t i = 1; i < mDesc->countTracks(); ++i) {
diff --git a/media/libstagefright/rtsp/ARTPSource.cpp b/media/libstagefright/rtsp/ARTPSource.cpp
index e08183e..225f6e8 100644
--- a/media/libstagefright/rtsp/ARTPSource.cpp
+++ b/media/libstagefright/rtsp/ARTPSource.cpp
@@ -98,7 +98,7 @@ void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
mNTPTime[mNumTimes] = ntpTime;
mRTPTime[mNumTimes++] = rtpTime;
- if (mNumTimes == 2) {
+ if (timeEstablished()) {
for (List<sp<ABuffer> >::iterator it = mQueue.begin();
it != mQueue.end(); ++it) {
sp<AMessage> meta = (*it)->meta();
@@ -112,13 +112,6 @@ void ARTPSource::timeUpdate(uint32_t rtpTime, uint64_t ntpTime) {
}
bool ARTPSource::queuePacket(const sp<ABuffer> &buffer) {
-#if 1
- if (mNumTimes != 2) {
- // Drop incoming packets until we've established a time base.
- return false;
- }
-#endif
-
uint32_t seqNum = (uint32_t)buffer->int32Data();
if (mNumTimes == 2) {