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Commit b2934b16 authored by Andreas Huber's avatar Andreas Huber
Browse files

Change timestamp handling in RTSP, remove unused, experimental, gtalk support 

related-to-bug: 3216447

NTP timestamp handling is now done at a higher layer than before.

Change-Id: I9fb23f1335110ec59e534f9aa0fe6f6a6406dd52
parent c77d7b42

media/libstagefright/AwesomePlayer.cpp

+6 −125
Original line number Diff line number Diff line
@@ -27,11 +27,6 @@ 
#include "include/ThrottledSource.h"

#include "include/MPEG2TSExtractor.h"



#include "ARTPSession.h"

#include "APacketSource.h"

#include "ASessionDescription.h"

#include "UDPPusher.h"



#include <binder/IPCThreadState.h>

#include <media/stagefright/foundation/hexdump.h>

#include <media/stagefright/foundation/ADebug.h>
@@ -58,6 +53,7 @@ namespace android { 


static int64_t kLowWaterMarkUs = 2000000ll;  // 2secs

static int64_t kHighWaterMarkUs = 10000000ll;  // 10secs

static int64_t kHighWaterMarkRTSPUs = 4000000ll;  // 4secs

static const size_t kLowWaterMarkBytes = 40000;

static const size_t kHighWaterMarkBytes = 200000;
@@ -463,10 +459,6 @@ void AwesomePlayer::reset_l() { 
        mLiveSession.clear();

    }



    mRTPPusher.clear();

    mRTCPPusher.clear();

    mRTPSession.clear();



    if (mVideoSource != NULL) {

        mVideoSource->stop();
@@ -644,6 +636,9 @@ void AwesomePlayer::onBufferingUpdate() { 
        LOGV("cachedDurationUs = %.2f secs, eos=%d",

             cachedDurationUs / 1E6, eos);



        int64_t highWaterMarkUs =

            (mRTSPController != NULL) ? kHighWaterMarkRTSPUs : kHighWaterMarkUs;



        if ((mFlags & PLAYING) && !eos

                && (cachedDurationUs < kLowWaterMarkUs)) {

            LOGI("cache is running low (%.2f secs) , pausing.",
@@ -652,7 +647,7 @@ void AwesomePlayer::onBufferingUpdate() { 
            pause_l();

            ensureCacheIsFetching_l();

            notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);

        } else if (eos || cachedDurationUs > kHighWaterMarkUs) {

        } else if (eos || cachedDurationUs > highWaterMarkUs) {

            if (mFlags & CACHE_UNDERRUN) {

                LOGI("cache has filled up (%.2f secs), resuming.",

                     cachedDurationUs / 1E6);
@@ -1331,10 +1326,8 @@ void AwesomePlayer::onVideoEvent() { 
        mTimeSourceDeltaUs = realTimeUs - mediaTimeUs;

    }



    if (!wasSeeking && mRTPSession == NULL) {

    if (!wasSeeking) {

        // Let's display the first frame after seeking right away.

        // We'll completely ignore timestamps for gtalk videochat

        // and we'll play incoming video as fast as we get it.



        int64_t nowUs = ts->getRealTimeUs() - mTimeSourceDeltaUs;
@@ -1594,118 +1587,6 @@ status_t AwesomePlayer::finishSetDataSource_l() { 
            ->setLiveSession(mLiveSession);



        return setDataSource_l(extractor);

    } else if (!strncmp("rtsp://gtalk/", mUri.string(), 13)) {

        if (mLooper == NULL) {

            mLooper = new ALooper;

            mLooper->setName("gtalk rtp");

            mLooper->start(

                    false /* runOnCallingThread */,

                    false /* canCallJava */,

                    PRIORITY_HIGHEST);

        }



        const char *startOfCodecString = &mUri.string()[13];

        const char *startOfSlash1 = strchr(startOfCodecString, '/');

        if (startOfSlash1 == NULL) {

            return BAD_VALUE;

        }

        const char *startOfWidthString = &startOfSlash1[1];

        const char *startOfSlash2 = strchr(startOfWidthString, '/');

        if (startOfSlash2 == NULL) {

            return BAD_VALUE;

        }

        const char *startOfHeightString = &startOfSlash2[1];



        String8 codecString(startOfCodecString, startOfSlash1 - startOfCodecString);

        String8 widthString(startOfWidthString, startOfSlash2 - startOfWidthString);

        String8 heightString(startOfHeightString);



#if 0

        mRTPPusher = new UDPPusher("/data/misc/rtpout.bin", 5434);

        mLooper->registerHandler(mRTPPusher);



        mRTCPPusher = new UDPPusher("/data/misc/rtcpout.bin", 5435);

        mLooper->registerHandler(mRTCPPusher);

#endif



        mRTPSession = new ARTPSession;

        mLooper->registerHandler(mRTPSession);



#if 0

        // My AMR SDP

        static const char *raw =

            "v=0\r\n"

            "o=- 64 233572944 IN IP4 127.0.0.0\r\n"

            "s=QuickTime\r\n"

            "t=0 0\r\n"

            "a=range:npt=0-315\r\n"

            "a=isma-compliance:2,2.0,2\r\n"

            "m=audio 5434 RTP/AVP 97\r\n"

            "c=IN IP4 127.0.0.1\r\n"

            "b=AS:30\r\n"

            "a=rtpmap:97 AMR/8000/1\r\n"

            "a=fmtp:97 octet-align\r\n";

#elif 1

        String8 sdp;

        sdp.appendFormat(

            "v=0\r\n"

            "o=- 64 233572944 IN IP4 127.0.0.0\r\n"

            "s=QuickTime\r\n"

            "t=0 0\r\n"

            "a=range:npt=0-315\r\n"

            "a=isma-compliance:2,2.0,2\r\n"

            "m=video 5434 RTP/AVP 97\r\n"

            "c=IN IP4 127.0.0.1\r\n"

            "b=AS:30\r\n"

            "a=rtpmap:97 %s/90000\r\n"

            "a=cliprect:0,0,%s,%s\r\n"

            "a=framesize:97 %s-%s\r\n",



            codecString.string(),

            heightString.string(), widthString.string(),

            widthString.string(), heightString.string()

            );

        const char *raw = sdp.string();



#endif



        sp<ASessionDescription> desc = new ASessionDescription;

        CHECK(desc->setTo(raw, strlen(raw)));



        CHECK_EQ(mRTPSession->setup(desc), (status_t)OK);



        if (mRTPPusher != NULL) {

            mRTPPusher->start();

        }



        if (mRTCPPusher != NULL) {

            mRTCPPusher->start();

        }



        CHECK_EQ(mRTPSession->countTracks(), 1u);

        sp<MediaSource> source = mRTPSession->trackAt(0);



#if 0

        bool eos;

        while (((APacketSource *)source.get())

                ->getQueuedDuration(&eos) < 5000000ll && !eos) {

            usleep(100000ll);

        }

#endif



        const char *mime;

        CHECK(source->getFormat()->findCString(kKeyMIMEType, &mime));



        if (!strncasecmp("video/", mime, 6)) {

            setVideoSource(source);

        } else {

            CHECK(!strncasecmp("audio/", mime, 6));

            setAudioSource(source);

        }



        mExtractorFlags = MediaExtractor::CAN_PAUSE;



        return OK;

    } else if (!strncasecmp("rtsp://", mUri.string(), 7)) {

        if (mLooper == NULL) {

            mLooper = new ALooper;

media/libstagefright/include/AwesomePlayer.h

+0 −4
Original line number Diff line number Diff line
@@ -39,8 +39,6 @@ struct NuCachedSource2; 


struct ALooper;

struct ARTSPController;

struct ARTPSession;

struct UDPPusher;



class DrmManagerClinet;

class DecryptHandle;
@@ -204,8 +202,6 @@ private: 


    sp<ALooper> mLooper;

    sp<ARTSPController> mRTSPController;

    sp<ARTPSession> mRTPSession;

    sp<UDPPusher> mRTPPusher, mRTCPPusher;



    sp<LiveSession> mLiveSession;

media/libstagefright/rtsp/ARTPAssembler.cpp

+0 −4
Original line number Diff line number Diff line
@@ -65,13 +65,9 @@ void ARTPAssembler::onPacketReceived(const sp<ARTPSource> &source) { 


// static

void ARTPAssembler::CopyTimes(const sp<ABuffer> &to, const sp<ABuffer> &from) {

    uint64_t ntpTime;

    CHECK(from->meta()->findInt64("ntp-time", (int64_t *)&ntpTime));



    uint32_t rtpTime;

    CHECK(from->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));



    to->meta()->setInt64("ntp-time", ntpTime);

    to->meta()->setInt32("rtp-time", rtpTime);



    // Copy the seq number.

media/libstagefright/rtsp/ARTPConnection.cpp

+1 −37
Original line number Diff line number Diff line
@@ -169,12 +169,6 @@ void ARTPConnection::onMessageReceived(const sp<AMessage> &msg) { 
            break;

        }



        case kWhatFakeTimestamps:

        {

            onFakeTimestamps();

            break;

        }



        default:

        {

            TRESPASS();
@@ -461,12 +455,6 @@ status_t ARTPConnection::parseRTP(StreamInfo *s, const sp<ABuffer> &buffer) { 
    buffer->setInt32Data(u16at(&data[2]));

    buffer->setRange(payloadOffset, size - payloadOffset);



    if ((mFlags & kFakeTimestamps) && !source->timeEstablished()) {

        source->timeUpdate(rtpTime, 0);

        source->timeUpdate(rtpTime + 90000, 0x100000000ll);

        CHECK(source->timeEstablished());

    }



    source->processRTPPacket(buffer);



    return OK;
@@ -592,9 +580,7 @@ status_t ARTPConnection::parseSR( 


    sp<ARTPSource> source = findSource(s, id);



    if ((mFlags & kFakeTimestamps) == 0) {

    source->timeUpdate(rtpTime, ntpTime);

    }



    return 0;

}
@@ -652,27 +638,5 @@ void ARTPConnection::onInjectPacket(const sp<AMessage> &msg) { 
    }

}



void ARTPConnection::fakeTimestamps() {

    (new AMessage(kWhatFakeTimestamps, id()))->post();

}



void ARTPConnection::onFakeTimestamps() {

    List<StreamInfo>::iterator it = mStreams.begin();

    while (it != mStreams.end()) {

        StreamInfo &info = *it++;



        for (size_t j = 0; j < info.mSources.size(); ++j) {

            sp<ARTPSource> source = info.mSources.valueAt(j);



            if (!source->timeEstablished()) {

                source->timeUpdate(0, 0);

                source->timeUpdate(0 + 90000, 0x100000000ll);



                mFlags |= kFakeTimestamps;

            }

        }

    }

}



}  // namespace android

media/libstagefright/rtsp/ARTPConnection.h

+0 −5
Original line number Diff line number Diff line
@@ -29,7 +29,6 @@ struct ASessionDescription; 


struct ARTPConnection : public AHandler {

    enum Flags {

        kFakeTimestamps      = 1,

        kRegularlyRequestFIR = 2,

    };
@@ -51,8 +50,6 @@ struct ARTPConnection : public AHandler { 
    static void MakePortPair(

            int *rtpSocket, int *rtcpSocket, unsigned *rtpPort);



    void fakeTimestamps();



protected:

    virtual ~ARTPConnection();

    virtual void onMessageReceived(const sp<AMessage> &msg);
@@ -63,7 +60,6 @@ private: 
        kWhatRemoveStream,

        kWhatPollStreams,

        kWhatInjectPacket,

        kWhatFakeTimestamps,

    };



    static const int64_t kSelectTimeoutUs;
@@ -81,7 +77,6 @@ private: 
    void onPollStreams();

    void onInjectPacket(const sp<AMessage> &msg);

    void onSendReceiverReports();

    void onFakeTimestamps();



    status_t receive(StreamInfo *info, bool receiveRTP);