From 83c9bd167600b3ff77008a6c06c05bb1f189d4ca Mon Sep 17 00:00:00 2001 From: Andreas Huber Date: Mon, 12 Nov 2012 13:08:44 -0800 Subject: Various improvements of wifi display code - manually prepend SPS/PPS if encoder doesn't support it - latency improvements - support for "our" method of optional RTP retransmission - improvements to the wfd commandline tool for testing - make it easier to turn on/off suspension of the video pipeline on idle - fixes an issue where an error during encryption would cause a SEGV - add HDCP descriptor if necessary Squashed commit of the following: commit 1115be0ebb3b885b4f1b7dba56761ca013d0ec4a Author: Andreas Huber Date: Fri Nov 9 11:32:23 2012 -0800 Better shutdown of wfd -l sessions. Change-Id: Id898a14ae21efd3b065b00a729830063d39195a7 commit 0e7d106dfe4eb6e2640b0b66c65deaba265f7ff0 Author: Andreas Huber Date: Thu Nov 8 16:38:55 2012 -0800 No more sending delay, create rtp packets upfront. Change-Id: I809a225f664fdb485c7d9a49a27886601a6a26b2 commit d399e8571b77353d59afb57508dfd2a82c1ef93a Author: Andreas Huber Date: Thu Nov 8 14:19:43 2012 -0800 Restore AudioSource buffer size, factor out TimeSeries, make suspending video optional. Change-Id: Ifdfe4d447b901e714abf52856b4641d1d55a5d41 commit f8b649f0b8f917d59f4b8a2e8e6d7db61a684a78 Author: Andreas Huber Date: Thu Nov 8 09:34:06 2012 -0800 Pull 480 frames at a time from AudioSource/AudioRecord Change-Id: I1e215abd329faec3da026631122c0f4c800c0ac4 commit 1bc13452eb35eebbba00f5da93fa86535be5db59 Author: Andreas Huber Date: Thu Nov 8 08:50:30 2012 -0800 fixed bitrate traffic simulation Change-Id: Ic5efb7cbb0b5d3b4917bc77b8ba73d447249e695 commit 016cdff18e74bdc631a5679e97192645ed095aa2 Author: Andreas Huber Date: Wed Nov 7 14:00:03 2012 -0800 resurrected "our" style of retransmission. Change-Id: I34d757aba67428437cb39b8293a9651750ad20d9 commit 384cf1a3c8fb4ec410bdf8fa5722c298e6028f3e Author: Andreas Huber Date: Tue Nov 6 09:38:55 2012 -0800 Changes to make wfd work on manta. Change-Id: I7a4e00cf16581fe2146edd1b359af195774090e4 commit 9628f24b22b35f28630d99dda3614babf51bc07e Author: Andreas Huber Date: Wed Nov 7 09:15:44 2012 -0800 Patch up rtp timestamps to more accurately measure network jitter. Change-Id: I9502a4615575f97f98a215a13131a89a6ae93c6d commit 7c891a1a24f08bbd50f55be13f7d05f43e807eb8 Author: Andreas Huber Date: Tue Nov 6 09:37:24 2012 -0800 Additions to the "wfd" tool to create a local wfd source. Change-Id: I99558653a70fdc703f9d13990b3ce1c4d3ae227a Change-Id: Ia94c63fc390f597014531073485f0cfc53b3418a --- .../libstagefright/wifi-display/source/Sender.cpp | 415 ++++++++------------- 1 file changed, 153 insertions(+), 262 deletions(-) (limited to 'media/libstagefright/wifi-display/source/Sender.cpp') diff --git a/media/libstagefright/wifi-display/source/Sender.cpp b/media/libstagefright/wifi-display/source/Sender.cpp index ea12424..9048691 100644 --- a/media/libstagefright/wifi-display/source/Sender.cpp +++ b/media/libstagefright/wifi-display/source/Sender.cpp @@ -21,6 +21,7 @@ #include "Sender.h" #include "ANetworkSession.h" +#include "TimeSeries.h" #include #include @@ -29,79 +30,8 @@ #include #include -#include - -#define DEBUG_JITTER 0 - namespace android { -//////////////////////////////////////////////////////////////////////////////// - -#if DEBUG_JITTER -struct TimeSeries { - TimeSeries(); - - void add(double val); - - double mean() const; - double sdev() const; - -private: - enum { - kHistorySize = 20 - }; - double mValues[kHistorySize]; - - size_t mCount; - double mSum; -}; - -TimeSeries::TimeSeries() - : mCount(0), - mSum(0.0) { -} - -void TimeSeries::add(double val) { - if (mCount < kHistorySize) { - mValues[mCount++] = val; - mSum += val; - } else { - mSum -= mValues[0]; - memmove(&mValues[0], &mValues[1], (kHistorySize - 1) * sizeof(double)); - mValues[kHistorySize - 1] = val; - mSum += val; - } -} - -double TimeSeries::mean() const { - if (mCount < 1) { - return 0.0; - } - - return mSum / mCount; -} - -double TimeSeries::sdev() const { - if (mCount < 1) { - return 0.0; - } - - double m = mean(); - - double sum = 0.0; - for (size_t i = 0; i < mCount; ++i) { - double tmp = mValues[i] - m; - tmp *= tmp; - - sum += tmp; - } - - return sqrt(sum / mCount); -} -#endif // DEBUG_JITTER - -//////////////////////////////////////////////////////////////////////////////// - static size_t kMaxRTPPacketSize = 1500; static size_t kMaxNumTSPacketsPerRTPPacket = (kMaxRTPPacketSize - 12) / 188; @@ -110,14 +40,13 @@ Sender::Sender( const sp ¬ify) : mNetSession(netSession), mNotify(notify), - mTSQueue(new ABuffer(12 + kMaxNumTSPacketsPerRTPPacket * 188)), mTransportMode(TRANSPORT_UDP), mRTPChannel(0), mRTCPChannel(0), mRTPPort(0), mRTPSessionID(0), mRTCPSessionID(0), -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX mRTPRetransmissionSessionID(0), mRTCPRetransmissionSessionID(0), #endif @@ -128,7 +57,7 @@ Sender::Sender( mFirstOutputBufferReadyTimeUs(-1ll), mFirstOutputBufferSentTimeUs(-1ll), mRTPSeqNo(0), -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX mRTPRetransmissionSeqNo(0), #endif mLastNTPTime(0), @@ -148,15 +77,13 @@ Sender::Sender( ,mLogFile(NULL) #endif { - mTSQueue->setRange(0, 12); - #if LOG_TRANSPORT_STREAM mLogFile = fopen("/system/etc/log.ts", "wb"); #endif } Sender::~Sender() { -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX if (mRTCPRetransmissionSessionID != 0) { mNetSession->destroySession(mRTCPRetransmissionSessionID); } @@ -217,7 +144,7 @@ status_t Sender::init( sp rtpNotify = new AMessage(kWhatRTPNotify, id()); sp rtcpNotify = new AMessage(kWhatRTCPNotify, id()); -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX sp rtpRetransmissionNotify = new AMessage(kWhatRTPRetransmissionNotify, id()); @@ -264,7 +191,7 @@ status_t Sender::init( } } -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX if (mTransportMode == TRANSPORT_UDP) { int32_t rtpRetransmissionSession; @@ -358,44 +285,67 @@ int32_t Sender::getRTPPort() const { } void Sender::queuePackets( - int64_t timeUs, const sp &packets) { - bool isVideo = false; + int64_t timeUs, const sp &tsPackets) { + const size_t numTSPackets = tsPackets->size() / 188; + + const size_t numRTPPackets = + (numTSPackets + kMaxNumTSPacketsPerRTPPacket - 1) + / kMaxNumTSPacketsPerRTPPacket; + + sp udpPackets = new ABuffer( + numRTPPackets * (12 + kMaxNumTSPacketsPerRTPPacket * 188)); + + udpPackets->meta()->setInt64("timeUs", timeUs); + + size_t dstOffset = 0; + for (size_t i = 0; i < numTSPackets; ++i) { + if ((i % kMaxNumTSPacketsPerRTPPacket) == 0) { + static const bool kMarkerBit = false; + + uint8_t *rtp = udpPackets->data() + dstOffset; + rtp[0] = 0x80; + rtp[1] = 33 | (kMarkerBit ? (1 << 7) : 0); // M-bit + rtp[2] = (mRTPSeqNo >> 8) & 0xff; + rtp[3] = mRTPSeqNo & 0xff; + rtp[4] = 0x00; // rtp time to be filled in later. + rtp[5] = 0x00; + rtp[6] = 0x00; + rtp[7] = 0x00; + rtp[8] = kSourceID >> 24; + rtp[9] = (kSourceID >> 16) & 0xff; + rtp[10] = (kSourceID >> 8) & 0xff; + rtp[11] = kSourceID & 0xff; + + ++mRTPSeqNo; + + dstOffset += 12; + } + + memcpy(udpPackets->data() + dstOffset, + tsPackets->data() + 188 * i, + 188); - int32_t dummy; - if (packets->meta()->findInt32("isVideo", &dummy)) { - isVideo = true; + dstOffset += 188; } - int64_t delayUs; - int64_t whenUs; + udpPackets->setRange(0, dstOffset); - if (mFirstOutputBufferReadyTimeUs < 0ll) { - mFirstOutputBufferReadyTimeUs = timeUs; - mFirstOutputBufferSentTimeUs = whenUs = ALooper::GetNowUs(); - delayUs = 0ll; - } else { - int64_t nowUs = ALooper::GetNowUs(); - - whenUs = (timeUs - mFirstOutputBufferReadyTimeUs) - + mFirstOutputBufferSentTimeUs; + sp msg = new AMessage(kWhatDrainQueue, id()); + msg->setBuffer("udpPackets", udpPackets); + msg->post(); - delayUs = whenUs - nowUs; +#if LOG_TRANSPORT_STREAM + if (mLogFile != NULL) { + fwrite(tsPackets->data(), 1, tsPackets->size(), mLogFile); } - - sp msg = new AMessage(kWhatQueuePackets, id()); - msg->setBuffer("packets", packets); - - packets->meta()->setInt64("timeUs", timeUs); - packets->meta()->setInt64("whenUs", whenUs); - packets->meta()->setInt64("delayUs", delayUs); - msg->post(delayUs > 0 ? delayUs : 0); +#endif } void Sender::onMessageReceived(const sp &msg) { switch (msg->what()) { case kWhatRTPNotify: case kWhatRTCPNotify: -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX case kWhatRTPRetransmissionNotify: case kWhatRTCPRetransmissionNotify: #endif @@ -419,7 +369,7 @@ void Sender::onMessageReceived(const sp &msg) { CHECK(msg->findString("detail", &detail)); if ((msg->what() == kWhatRTPNotify -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX || msg->what() == kWhatRTPRetransmissionNotify #endif ) && !errorOccuredDuringSend) { @@ -443,7 +393,7 @@ void Sender::onMessageReceived(const sp &msg) { } else if (sessionID == mRTCPSessionID) { mRTCPSessionID = 0; } -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX else if (sessionID == mRTPRetransmissionSessionID) { mRTPRetransmissionSessionID = 0; } else if (sessionID == mRTCPRetransmissionSessionID) { @@ -465,7 +415,7 @@ void Sender::onMessageReceived(const sp &msg) { status_t err; if (msg->what() == kWhatRTCPNotify -#if ENABLE_RETRANSMISSION +#if ENABLE_RETRANSMISSION && RETRANSMISSION_ACCORDING_TO_RFC_XXXX || msg->what() == kWhatRTCPRetransmissionNotify #endif ) @@ -507,12 +457,12 @@ void Sender::onMessageReceived(const sp &msg) { break; } - case kWhatQueuePackets: + case kWhatDrainQueue: { - sp packets; - CHECK(msg->findBuffer("packets", &packets)); + sp udpPackets; + CHECK(msg->findBuffer("udpPackets", &udpPackets)); - onQueuePackets(packets); + onDrainQueue(udpPackets); break; } @@ -532,156 +482,6 @@ void Sender::onMessageReceived(const sp &msg) { } } -void Sender::onQueuePackets(const sp &packets) { -#if DEBUG_JITTER - int32_t dummy; - if (packets->meta()->findInt32("isVideo", &dummy)) { - static int64_t lastTimeUs = 0ll; - int64_t nowUs = ALooper::GetNowUs(); - - static TimeSeries series; - series.add((double)(nowUs - lastTimeUs)); - - ALOGI("deltaTimeUs = %lld us, mean %.2f, sdev %.2f", - nowUs - lastTimeUs, series.mean(), series.sdev()); - - lastTimeUs = nowUs; - } -#endif - - int64_t startTimeUs = ALooper::GetNowUs(); - - for (size_t offset = 0; - offset < packets->size(); offset += 188) { - bool lastTSPacket = (offset + 188 >= packets->size()); - - appendTSData( - packets->data() + offset, - 188, - true /* timeDiscontinuity */, - lastTSPacket /* flush */); - } - -#if 0 - int64_t netTimeUs = ALooper::GetNowUs() - startTimeUs; - - int64_t whenUs; - CHECK(packets->meta()->findInt64("whenUs", &whenUs)); - - int64_t delayUs; - CHECK(packets->meta()->findInt64("delayUs", &delayUs)); - - bool isVideo = false; - int32_t dummy; - if (packets->meta()->findInt32("isVideo", &dummy)) { - isVideo = true; - } - - int64_t nowUs = ALooper::GetNowUs(); - - if (nowUs - whenUs > 2000) { - ALOGI("[%s] delayUs = %lld us, delta = %lld us", - isVideo ? "video" : "audio", delayUs, nowUs - netTimeUs - whenUs); - } -#endif - -#if LOG_TRANSPORT_STREAM - if (mLogFile != NULL) { - fwrite(packets->data(), 1, packets->size(), mLogFile); - } -#endif -} - -ssize_t Sender::appendTSData( - const void *data, size_t size, bool timeDiscontinuity, bool flush) { - CHECK_EQ(size, 188); - - CHECK_LE(mTSQueue->size() + size, mTSQueue->capacity()); - - memcpy(mTSQueue->data() + mTSQueue->size(), data, size); - mTSQueue->setRange(0, mTSQueue->size() + size); - - if (flush || mTSQueue->size() == mTSQueue->capacity()) { - // flush - - int64_t nowUs = ALooper::GetNowUs(); - -#if TRACK_BANDWIDTH - if (mFirstPacketTimeUs < 0ll) { - mFirstPacketTimeUs = nowUs; - } -#endif - - // 90kHz time scale - uint32_t rtpTime = (nowUs * 9ll) / 100ll; - - uint8_t *rtp = mTSQueue->data(); - rtp[0] = 0x80; - rtp[1] = 33 | (timeDiscontinuity ? (1 << 7) : 0); // M-bit - rtp[2] = (mRTPSeqNo >> 8) & 0xff; - rtp[3] = mRTPSeqNo & 0xff; - rtp[4] = rtpTime >> 24; - rtp[5] = (rtpTime >> 16) & 0xff; - rtp[6] = (rtpTime >> 8) & 0xff; - rtp[7] = rtpTime & 0xff; - rtp[8] = kSourceID >> 24; - rtp[9] = (kSourceID >> 16) & 0xff; - rtp[10] = (kSourceID >> 8) & 0xff; - rtp[11] = kSourceID & 0xff; - - ++mRTPSeqNo; - ++mNumRTPSent; - mNumRTPOctetsSent += mTSQueue->size() - 12; - - mLastRTPTime = rtpTime; - mLastNTPTime = GetNowNTP(); - - if (mTransportMode == TRANSPORT_TCP_INTERLEAVED) { - sp notify = mNotify->dup(); - notify->setInt32("what", kWhatBinaryData); - - sp data = new ABuffer(mTSQueue->size()); - memcpy(data->data(), rtp, mTSQueue->size()); - - notify->setInt32("channel", mRTPChannel); - notify->setBuffer("data", data); - notify->post(); - } else { - sendPacket(mRTPSessionID, rtp, mTSQueue->size()); - -#if TRACK_BANDWIDTH - mTotalBytesSent += mTSQueue->size(); - int64_t delayUs = ALooper::GetNowUs() - mFirstPacketTimeUs; - - if (delayUs > 0ll) { - ALOGI("approx. net bandwidth used: %.2f Mbit/sec", - mTotalBytesSent * 8.0 / delayUs); - } -#endif - } - -#if ENABLE_RETRANSMISSION - mTSQueue->setInt32Data(mRTPSeqNo - 1); - - mHistory.push_back(mTSQueue); - ++mHistoryLength; - - if (mHistoryLength > kMaxHistoryLength) { - mTSQueue = *mHistory.begin(); - mHistory.erase(mHistory.begin()); - - --mHistoryLength; - } else { - mTSQueue = new ABuffer(12 + kMaxNumTSPacketsPerRTPPacket * 188); - } -#endif - - mTSQueue->setRange(0, 12); - } - - return size; -} - void Sender::scheduleSendSR() { if (mSendSRPending || mRTCPSessionID == 0) { return; @@ -851,6 +651,7 @@ status_t Sender::parseTSFB( if (retransmit) { ALOGI("retransmitting seqNo %d", bufferSeqNo); +#if RETRANSMISSION_ACCORDING_TO_RFC_XXXX sp retransRTP = new ABuffer(2 + buffer->size()); uint8_t *rtp = retransRTP->data(); memcpy(rtp, buffer->data(), 12); @@ -865,6 +666,10 @@ status_t Sender::parseTSFB( sendPacket( mRTPRetransmissionSessionID, retransRTP->data(), retransRTP->size()); +#else + sendPacket( + mRTPSessionID, buffer->data(), buffer->size()); +#endif if (bufferSeqNo == seqNo) { foundSeqNo = true; @@ -975,5 +780,91 @@ void Sender::notifySessionDead() { notify->post(); } +void Sender::onDrainQueue(const sp &udpPackets) { + static const size_t kFullRTPPacketSize = + 12 + 188 * kMaxNumTSPacketsPerRTPPacket; + + size_t srcOffset = 0; + while (srcOffset < udpPackets->size()) { + uint8_t *rtp = udpPackets->data() + srcOffset; + + size_t rtpPacketSize = udpPackets->size() - srcOffset; + if (rtpPacketSize > kFullRTPPacketSize) { + rtpPacketSize = kFullRTPPacketSize; + } + + int64_t nowUs = ALooper::GetNowUs(); + mLastNTPTime = GetNowNTP(); + + // 90kHz time scale + uint32_t rtpTime = (nowUs * 9ll) / 100ll; + + rtp[4] = rtpTime >> 24; + rtp[5] = (rtpTime >> 16) & 0xff; + rtp[6] = (rtpTime >> 8) & 0xff; + rtp[7] = rtpTime & 0xff; + + ++mNumRTPSent; + mNumRTPOctetsSent += rtpPacketSize - 12; + + mLastRTPTime = rtpTime; + + if (mTransportMode == TRANSPORT_TCP_INTERLEAVED) { + sp notify = mNotify->dup(); + notify->setInt32("what", kWhatBinaryData); + + sp data = new ABuffer(rtpPacketSize); + memcpy(data->data(), rtp, rtpPacketSize); + + notify->setInt32("channel", mRTPChannel); + notify->setBuffer("data", data); + notify->post(); + } else { + sendPacket(mRTPSessionID, rtp, rtpPacketSize); + +#if TRACK_BANDWIDTH + mTotalBytesSent += rtpPacketSize->size(); + int64_t delayUs = ALooper::GetNowUs() - mFirstPacketTimeUs; + + if (delayUs > 0ll) { + ALOGI("approx. net bandwidth used: %.2f Mbit/sec", + mTotalBytesSent * 8.0 / delayUs); + } +#endif + } + +#if ENABLE_RETRANSMISSION + addToHistory(rtp, rtpPacketSize); +#endif + + srcOffset += rtpPacketSize; + } + +#if 0 + int64_t timeUs; + CHECK(udpPackets->meta()->findInt64("timeUs", &timeUs)); + + ALOGI("dTimeUs = %lld us", ALooper::GetNowUs() - timeUs); +#endif +} + +#if ENABLE_RETRANSMISSION +void Sender::addToHistory(const uint8_t *rtp, size_t rtpPacketSize) { + sp packet = new ABuffer(rtpPacketSize); + memcpy(packet->data(), rtp, rtpPacketSize); + + unsigned rtpSeqNo = U16_AT(&rtp[2]); + packet->setInt32Data(rtpSeqNo); + + mHistory.push_back(packet); + ++mHistoryLength; + + if (mHistoryLength > kMaxHistoryLength) { + mHistory.erase(mHistory.begin()); + --mHistoryLength; + } +} +#endif + } // namespace android -- cgit v1.1