We're now going to ignore timestamps completely in gtalk video conferencing,... (f88f8441) · Commits · e / os / android_frameworks_base

media/libstagefright/AwesomePlayer.cpp

+6 −0

Original line number	Diff line number	Diff line
		@@ -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);

+8 −0

Original line number	Diff line number	Diff line
		@@ -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);

+22 −17

Original line number	Diff line number	Diff line
		@@ -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();

+12 −12

Original line number	Diff line number	Diff line
		@@ -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) {
		}

		@@ -280,8 +279,11 @@ void ARTPConnection::onPollStreams() {
		sp<ARTPSource> source = s->mSources.valueAt(i);

		source->addReceiverReport(buffer);

		if (mFlags & kRegularlyRequestFIR) {
		source->addFIR(buffer);
		}
		}

		if (buffer->size() > 0) {
		LOG(VERBOSE) << "Sending RR...";
		@@ -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
		if ((mFlags & kFakeTimestamps) == 0) {
		source->timeUpdate(rtpTime, ntpTime);
		#endif
		}

		return 0;
		}

+8 −1

Original line number	Diff line number	Diff line
		@@ -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;