| /* |
| * Copyright 2012, The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * limitations under the License. |
| */ |
| |
| //#define LOG_NDEBUG 0 |
| #define LOG_TAG "Sender" |
| #include <utils/Log.h> |
| |
| #include "Sender.h" |
| |
| #include "ANetworkSession.h" |
| |
| #include <media/stagefright/foundation/ABuffer.h> |
| #include <media/stagefright/foundation/ADebug.h> |
| #include <media/stagefright/foundation/AMessage.h> |
| #include <media/stagefright/foundation/hexdump.h> |
| #include <media/stagefright/MediaErrors.h> |
| #include <media/stagefright/Utils.h> |
| |
| #include <math.h> |
| |
| #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; |
| |
| Sender::Sender( |
| const sp<ANetworkSession> &netSession, |
| const sp<AMessage> ¬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 |
| mRTPRetransmissionSessionID(0), |
| mRTCPRetransmissionSessionID(0), |
| #endif |
| mClientRTPPort(0), |
| mClientRTCPPort(0), |
| mRTPConnected(false), |
| mRTCPConnected(false), |
| mFirstOutputBufferReadyTimeUs(-1ll), |
| mFirstOutputBufferSentTimeUs(-1ll), |
| mRTPSeqNo(0), |
| #if ENABLE_RETRANSMISSION |
| mRTPRetransmissionSeqNo(0), |
| #endif |
| mLastNTPTime(0), |
| mLastRTPTime(0), |
| mNumRTPSent(0), |
| mNumRTPOctetsSent(0), |
| mNumSRsSent(0), |
| mSendSRPending(false) |
| #if ENABLE_RETRANSMISSION |
| ,mHistoryLength(0) |
| #endif |
| #if TRACK_BANDWIDTH |
| ,mFirstPacketTimeUs(-1ll) |
| ,mTotalBytesSent(0ll) |
| #endif |
| #if LOG_TRANSPORT_STREAM |
| ,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 (mRTCPRetransmissionSessionID != 0) { |
| mNetSession->destroySession(mRTCPRetransmissionSessionID); |
| } |
| |
| if (mRTPRetransmissionSessionID != 0) { |
| mNetSession->destroySession(mRTPRetransmissionSessionID); |
| } |
| #endif |
| |
| if (mRTCPSessionID != 0) { |
| mNetSession->destroySession(mRTCPSessionID); |
| } |
| |
| if (mRTPSessionID != 0) { |
| mNetSession->destroySession(mRTPSessionID); |
| } |
| |
| #if LOG_TRANSPORT_STREAM |
| if (mLogFile != NULL) { |
| fclose(mLogFile); |
| mLogFile = NULL; |
| } |
| #endif |
| } |
| |
| status_t Sender::init( |
| const char *clientIP, int32_t clientRtp, int32_t clientRtcp, |
| TransportMode transportMode) { |
| mClientIP = clientIP; |
| mTransportMode = transportMode; |
| |
| if (transportMode == TRANSPORT_TCP_INTERLEAVED) { |
| mRTPChannel = clientRtp; |
| mRTCPChannel = clientRtcp; |
| mRTPPort = 0; |
| mRTPSessionID = 0; |
| mRTCPSessionID = 0; |
| return OK; |
| } |
| |
| mRTPChannel = 0; |
| mRTCPChannel = 0; |
| |
| if (mTransportMode == TRANSPORT_TCP) { |
| // XXX This is wrong, we need to allocate sockets here, we only |
| // need to do this because the dongles are not establishing their |
| // end until after PLAY instead of before SETUP. |
| mRTPPort = 20000; |
| mRTPSessionID = 0; |
| mRTCPSessionID = 0; |
| mClientRTPPort = clientRtp; |
| mClientRTCPPort = clientRtcp; |
| return OK; |
| } |
| |
| int serverRtp; |
| |
| sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, id()); |
| sp<AMessage> rtcpNotify = new AMessage(kWhatRTCPNotify, id()); |
| |
| #if ENABLE_RETRANSMISSION |
| sp<AMessage> rtpRetransmissionNotify = |
| new AMessage(kWhatRTPRetransmissionNotify, id()); |
| |
| sp<AMessage> rtcpRetransmissionNotify = |
| new AMessage(kWhatRTCPRetransmissionNotify, id()); |
| #endif |
| |
| status_t err; |
| for (serverRtp = 15550;; serverRtp += 2) { |
| int32_t rtpSession; |
| if (mTransportMode == TRANSPORT_UDP) { |
| err = mNetSession->createUDPSession( |
| serverRtp, clientIP, clientRtp, |
| rtpNotify, &rtpSession); |
| } else { |
| err = mNetSession->createTCPDatagramSession( |
| serverRtp, clientIP, clientRtp, |
| rtpNotify, &rtpSession); |
| } |
| |
| if (err != OK) { |
| ALOGI("failed to create RTP socket on port %d", serverRtp); |
| continue; |
| } |
| |
| int32_t rtcpSession = 0; |
| |
| if (clientRtcp >= 0) { |
| if (mTransportMode == TRANSPORT_UDP) { |
| err = mNetSession->createUDPSession( |
| serverRtp + 1, clientIP, clientRtcp, |
| rtcpNotify, &rtcpSession); |
| } else { |
| err = mNetSession->createTCPDatagramSession( |
| serverRtp + 1, clientIP, clientRtcp, |
| rtcpNotify, &rtcpSession); |
| } |
| |
| if (err != OK) { |
| ALOGI("failed to create RTCP socket on port %d", serverRtp + 1); |
| |
| mNetSession->destroySession(rtpSession); |
| continue; |
| } |
| } |
| |
| #if ENABLE_RETRANSMISSION |
| if (mTransportMode == TRANSPORT_UDP) { |
| int32_t rtpRetransmissionSession; |
| |
| err = mNetSession->createUDPSession( |
| serverRtp + kRetransmissionPortOffset, |
| clientIP, |
| clientRtp + kRetransmissionPortOffset, |
| rtpRetransmissionNotify, |
| &rtpRetransmissionSession); |
| |
| if (err != OK) { |
| mNetSession->destroySession(rtcpSession); |
| mNetSession->destroySession(rtpSession); |
| continue; |
| } |
| |
| CHECK_GE(clientRtcp, 0); |
| |
| int32_t rtcpRetransmissionSession; |
| err = mNetSession->createUDPSession( |
| serverRtp + 1 + kRetransmissionPortOffset, |
| clientIP, |
| clientRtp + 1 + kRetransmissionPortOffset, |
| rtcpRetransmissionNotify, |
| &rtcpRetransmissionSession); |
| |
| if (err != OK) { |
| mNetSession->destroySession(rtpRetransmissionSession); |
| mNetSession->destroySession(rtcpSession); |
| mNetSession->destroySession(rtpSession); |
| continue; |
| } |
| |
| mRTPRetransmissionSessionID = rtpRetransmissionSession; |
| mRTCPRetransmissionSessionID = rtcpRetransmissionSession; |
| |
| ALOGI("rtpRetransmissionSessionID = %d, " |
| "rtcpRetransmissionSessionID = %d", |
| rtpRetransmissionSession, rtcpRetransmissionSession); |
| } |
| #endif |
| |
| mRTPPort = serverRtp; |
| mRTPSessionID = rtpSession; |
| mRTCPSessionID = rtcpSession; |
| |
| ALOGI("rtpSessionID = %d, rtcpSessionID = %d", rtpSession, rtcpSession); |
| break; |
| } |
| |
| if (mRTPPort == 0) { |
| return UNKNOWN_ERROR; |
| } |
| |
| return OK; |
| } |
| |
| status_t Sender::finishInit() { |
| if (mTransportMode != TRANSPORT_TCP) { |
| notifyInitDone(); |
| return OK; |
| } |
| |
| sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, id()); |
| |
| status_t err = mNetSession->createTCPDatagramSession( |
| mRTPPort, mClientIP.c_str(), mClientRTPPort, |
| rtpNotify, &mRTPSessionID); |
| |
| if (err != OK) { |
| return err; |
| } |
| |
| if (mClientRTCPPort >= 0) { |
| sp<AMessage> rtcpNotify = new AMessage(kWhatRTCPNotify, id()); |
| |
| err = mNetSession->createTCPDatagramSession( |
| mRTPPort + 1, mClientIP.c_str(), mClientRTCPPort, |
| rtcpNotify, &mRTCPSessionID); |
| |
| if (err != OK) { |
| return err; |
| } |
| } |
| |
| return OK; |
| } |
| |
| int32_t Sender::getRTPPort() const { |
| return mRTPPort; |
| } |
| |
| void Sender::queuePackets( |
| int64_t timeUs, const sp<ABuffer> &packets) { |
| bool isVideo = false; |
| |
| int32_t dummy; |
| if (packets->meta()->findInt32("isVideo", &dummy)) { |
| isVideo = true; |
| } |
| |
| int64_t delayUs; |
| int64_t whenUs; |
| |
| if (mFirstOutputBufferReadyTimeUs < 0ll) { |
| mFirstOutputBufferReadyTimeUs = timeUs; |
| mFirstOutputBufferSentTimeUs = whenUs = ALooper::GetNowUs(); |
| delayUs = 0ll; |
| } else { |
| int64_t nowUs = ALooper::GetNowUs(); |
| |
| whenUs = (timeUs - mFirstOutputBufferReadyTimeUs) |
| + mFirstOutputBufferSentTimeUs; |
| |
| delayUs = whenUs - nowUs; |
| } |
| |
| sp<AMessage> 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); |
| } |
| |
| void Sender::onMessageReceived(const sp<AMessage> &msg) { |
| switch (msg->what()) { |
| case kWhatRTPNotify: |
| case kWhatRTCPNotify: |
| #if ENABLE_RETRANSMISSION |
| case kWhatRTPRetransmissionNotify: |
| case kWhatRTCPRetransmissionNotify: |
| #endif |
| { |
| int32_t reason; |
| CHECK(msg->findInt32("reason", &reason)); |
| |
| switch (reason) { |
| case ANetworkSession::kWhatError: |
| { |
| int32_t sessionID; |
| CHECK(msg->findInt32("sessionID", &sessionID)); |
| |
| int32_t err; |
| CHECK(msg->findInt32("err", &err)); |
| |
| int32_t errorOccuredDuringSend; |
| CHECK(msg->findInt32("send", &errorOccuredDuringSend)); |
| |
| AString detail; |
| CHECK(msg->findString("detail", &detail)); |
| |
| if ((msg->what() == kWhatRTPNotify |
| #if ENABLE_RETRANSMISSION |
| || msg->what() == kWhatRTPRetransmissionNotify |
| #endif |
| ) && !errorOccuredDuringSend) { |
| // This is ok, we don't expect to receive anything on |
| // the RTP socket. |
| break; |
| } |
| |
| ALOGE("An error occurred during %s in session %d " |
| "(%d, '%s' (%s)).", |
| errorOccuredDuringSend ? "send" : "receive", |
| sessionID, |
| err, |
| detail.c_str(), |
| strerror(-err)); |
| |
| mNetSession->destroySession(sessionID); |
| |
| if (sessionID == mRTPSessionID) { |
| mRTPSessionID = 0; |
| } else if (sessionID == mRTCPSessionID) { |
| mRTCPSessionID = 0; |
| } |
| #if ENABLE_RETRANSMISSION |
| else if (sessionID == mRTPRetransmissionSessionID) { |
| mRTPRetransmissionSessionID = 0; |
| } else if (sessionID == mRTCPRetransmissionSessionID) { |
| mRTCPRetransmissionSessionID = 0; |
| } |
| #endif |
| |
| notifySessionDead(); |
| break; |
| } |
| |
| case ANetworkSession::kWhatDatagram: |
| { |
| int32_t sessionID; |
| CHECK(msg->findInt32("sessionID", &sessionID)); |
| |
| sp<ABuffer> data; |
| CHECK(msg->findBuffer("data", &data)); |
| |
| status_t err; |
| if (msg->what() == kWhatRTCPNotify |
| #if ENABLE_RETRANSMISSION |
| || msg->what() == kWhatRTCPRetransmissionNotify |
| #endif |
| ) |
| { |
| err = parseRTCP(data); |
| } |
| break; |
| } |
| |
| case ANetworkSession::kWhatConnected: |
| { |
| CHECK_EQ(mTransportMode, TRANSPORT_TCP); |
| |
| int32_t sessionID; |
| CHECK(msg->findInt32("sessionID", &sessionID)); |
| |
| if (sessionID == mRTPSessionID) { |
| CHECK(!mRTPConnected); |
| mRTPConnected = true; |
| ALOGI("RTP Session now connected."); |
| } else if (sessionID == mRTCPSessionID) { |
| CHECK(!mRTCPConnected); |
| mRTCPConnected = true; |
| ALOGI("RTCP Session now connected."); |
| } else { |
| TRESPASS(); |
| } |
| |
| if (mRTPConnected |
| && (mClientRTCPPort < 0 || mRTCPConnected)) { |
| notifyInitDone(); |
| } |
| break; |
| } |
| |
| default: |
| TRESPASS(); |
| } |
| break; |
| } |
| |
| case kWhatQueuePackets: |
| { |
| sp<ABuffer> packets; |
| CHECK(msg->findBuffer("packets", &packets)); |
| |
| onQueuePackets(packets); |
| break; |
| } |
| |
| case kWhatSendSR: |
| { |
| mSendSRPending = false; |
| |
| if (mRTCPSessionID == 0) { |
| break; |
| } |
| |
| onSendSR(); |
| |
| scheduleSendSR(); |
| break; |
| } |
| } |
| } |
| |
| void Sender::onQueuePackets(const sp<ABuffer> &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<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatBinaryData); |
| |
| sp<ABuffer> 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; |
| } |
| |
| mSendSRPending = true; |
| (new AMessage(kWhatSendSR, id()))->post(kSendSRIntervalUs); |
| } |
| |
| void Sender::addSR(const sp<ABuffer> &buffer) { |
| uint8_t *data = buffer->data() + buffer->size(); |
| |
| // TODO: Use macros/utility functions to clean up all the bitshifts below. |
| |
| data[0] = 0x80 | 0; |
| data[1] = 200; // SR |
| data[2] = 0; |
| data[3] = 6; |
| data[4] = kSourceID >> 24; |
| data[5] = (kSourceID >> 16) & 0xff; |
| data[6] = (kSourceID >> 8) & 0xff; |
| data[7] = kSourceID & 0xff; |
| |
| data[8] = mLastNTPTime >> (64 - 8); |
| data[9] = (mLastNTPTime >> (64 - 16)) & 0xff; |
| data[10] = (mLastNTPTime >> (64 - 24)) & 0xff; |
| data[11] = (mLastNTPTime >> 32) & 0xff; |
| data[12] = (mLastNTPTime >> 24) & 0xff; |
| data[13] = (mLastNTPTime >> 16) & 0xff; |
| data[14] = (mLastNTPTime >> 8) & 0xff; |
| data[15] = mLastNTPTime & 0xff; |
| |
| data[16] = (mLastRTPTime >> 24) & 0xff; |
| data[17] = (mLastRTPTime >> 16) & 0xff; |
| data[18] = (mLastRTPTime >> 8) & 0xff; |
| data[19] = mLastRTPTime & 0xff; |
| |
| data[20] = mNumRTPSent >> 24; |
| data[21] = (mNumRTPSent >> 16) & 0xff; |
| data[22] = (mNumRTPSent >> 8) & 0xff; |
| data[23] = mNumRTPSent & 0xff; |
| |
| data[24] = mNumRTPOctetsSent >> 24; |
| data[25] = (mNumRTPOctetsSent >> 16) & 0xff; |
| data[26] = (mNumRTPOctetsSent >> 8) & 0xff; |
| data[27] = mNumRTPOctetsSent & 0xff; |
| |
| buffer->setRange(buffer->offset(), buffer->size() + 28); |
| } |
| |
| void Sender::addSDES(const sp<ABuffer> &buffer) { |
| uint8_t *data = buffer->data() + buffer->size(); |
| data[0] = 0x80 | 1; |
| data[1] = 202; // SDES |
| data[4] = kSourceID >> 24; |
| data[5] = (kSourceID >> 16) & 0xff; |
| data[6] = (kSourceID >> 8) & 0xff; |
| data[7] = kSourceID & 0xff; |
| |
| size_t offset = 8; |
| |
| data[offset++] = 1; // CNAME |
| |
| static const char *kCNAME = "someone@somewhere"; |
| data[offset++] = strlen(kCNAME); |
| |
| memcpy(&data[offset], kCNAME, strlen(kCNAME)); |
| offset += strlen(kCNAME); |
| |
| data[offset++] = 7; // NOTE |
| |
| static const char *kNOTE = "Hell's frozen over."; |
| data[offset++] = strlen(kNOTE); |
| |
| memcpy(&data[offset], kNOTE, strlen(kNOTE)); |
| offset += strlen(kNOTE); |
| |
| data[offset++] = 0; |
| |
| if ((offset % 4) > 0) { |
| size_t count = 4 - (offset % 4); |
| switch (count) { |
| case 3: |
| data[offset++] = 0; |
| case 2: |
| data[offset++] = 0; |
| case 1: |
| data[offset++] = 0; |
| } |
| } |
| |
| size_t numWords = (offset / 4) - 1; |
| data[2] = numWords >> 8; |
| data[3] = numWords & 0xff; |
| |
| buffer->setRange(buffer->offset(), buffer->size() + offset); |
| } |
| |
| // static |
| uint64_t Sender::GetNowNTP() { |
| uint64_t nowUs = ALooper::GetNowUs(); |
| |
| nowUs += ((70ll * 365 + 17) * 24) * 60 * 60 * 1000000ll; |
| |
| uint64_t hi = nowUs / 1000000ll; |
| uint64_t lo = ((1ll << 32) * (nowUs % 1000000ll)) / 1000000ll; |
| |
| return (hi << 32) | lo; |
| } |
| |
| void Sender::onSendSR() { |
| sp<ABuffer> buffer = new ABuffer(1500); |
| buffer->setRange(0, 0); |
| |
| addSR(buffer); |
| addSDES(buffer); |
| |
| if (mTransportMode == TRANSPORT_TCP_INTERLEAVED) { |
| sp<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatBinaryData); |
| notify->setInt32("channel", mRTCPChannel); |
| notify->setBuffer("data", buffer); |
| notify->post(); |
| } else { |
| sendPacket(mRTCPSessionID, buffer->data(), buffer->size()); |
| } |
| |
| ++mNumSRsSent; |
| } |
| |
| #if ENABLE_RETRANSMISSION |
| status_t Sender::parseTSFB( |
| const uint8_t *data, size_t size) { |
| if ((data[0] & 0x1f) != 1) { |
| return ERROR_UNSUPPORTED; // We only support NACK for now. |
| } |
| |
| uint32_t srcId = U32_AT(&data[8]); |
| if (srcId != kSourceID) { |
| return ERROR_MALFORMED; |
| } |
| |
| for (size_t i = 12; i < size; i += 4) { |
| uint16_t seqNo = U16_AT(&data[i]); |
| uint16_t blp = U16_AT(&data[i + 2]); |
| |
| List<sp<ABuffer> >::iterator it = mHistory.begin(); |
| bool foundSeqNo = false; |
| while (it != mHistory.end()) { |
| const sp<ABuffer> &buffer = *it; |
| |
| uint16_t bufferSeqNo = buffer->int32Data() & 0xffff; |
| |
| bool retransmit = false; |
| if (bufferSeqNo == seqNo) { |
| retransmit = true; |
| } else if (blp != 0) { |
| for (size_t i = 0; i < 16; ++i) { |
| if ((blp & (1 << i)) |
| && (bufferSeqNo == ((seqNo + i + 1) & 0xffff))) { |
| blp &= ~(1 << i); |
| retransmit = true; |
| } |
| } |
| } |
| |
| if (retransmit) { |
| ALOGI("retransmitting seqNo %d", bufferSeqNo); |
| |
| sp<ABuffer> retransRTP = new ABuffer(2 + buffer->size()); |
| uint8_t *rtp = retransRTP->data(); |
| memcpy(rtp, buffer->data(), 12); |
| rtp[2] = (mRTPRetransmissionSeqNo >> 8) & 0xff; |
| rtp[3] = mRTPRetransmissionSeqNo & 0xff; |
| rtp[12] = (bufferSeqNo >> 8) & 0xff; |
| rtp[13] = bufferSeqNo & 0xff; |
| memcpy(&rtp[14], buffer->data() + 12, buffer->size() - 12); |
| |
| ++mRTPRetransmissionSeqNo; |
| |
| sendPacket( |
| mRTPRetransmissionSessionID, |
| retransRTP->data(), retransRTP->size()); |
| |
| if (bufferSeqNo == seqNo) { |
| foundSeqNo = true; |
| } |
| |
| if (foundSeqNo && blp == 0) { |
| break; |
| } |
| } |
| |
| ++it; |
| } |
| |
| if (!foundSeqNo || blp != 0) { |
| ALOGI("Some sequence numbers were no longer available for " |
| "retransmission"); |
| } |
| } |
| |
| return OK; |
| } |
| #endif |
| |
| status_t Sender::parseRTCP( |
| const sp<ABuffer> &buffer) { |
| const uint8_t *data = buffer->data(); |
| size_t size = buffer->size(); |
| |
| while (size > 0) { |
| if (size < 8) { |
| // Too short to be a valid RTCP header |
| return ERROR_MALFORMED; |
| } |
| |
| if ((data[0] >> 6) != 2) { |
| // Unsupported version. |
| return ERROR_UNSUPPORTED; |
| } |
| |
| if (data[0] & 0x20) { |
| // Padding present. |
| |
| size_t paddingLength = data[size - 1]; |
| |
| if (paddingLength + 12 > size) { |
| // If we removed this much padding we'd end up with something |
| // that's too short to be a valid RTP header. |
| return ERROR_MALFORMED; |
| } |
| |
| size -= paddingLength; |
| } |
| |
| size_t headerLength = 4 * (data[2] << 8 | data[3]) + 4; |
| |
| if (size < headerLength) { |
| // Only received a partial packet? |
| return ERROR_MALFORMED; |
| } |
| |
| switch (data[1]) { |
| case 200: |
| case 201: // RR |
| case 202: // SDES |
| case 203: |
| case 204: // APP |
| break; |
| |
| #if ENABLE_RETRANSMISSION |
| case 205: // TSFB (transport layer specific feedback) |
| parseTSFB(data, headerLength); |
| break; |
| #endif |
| |
| case 206: // PSFB (payload specific feedback) |
| hexdump(data, headerLength); |
| break; |
| |
| default: |
| { |
| ALOGW("Unknown RTCP packet type %u of size %d", |
| (unsigned)data[1], headerLength); |
| break; |
| } |
| } |
| |
| data += headerLength; |
| size -= headerLength; |
| } |
| |
| return OK; |
| } |
| |
| status_t Sender::sendPacket( |
| int32_t sessionID, const void *data, size_t size) { |
| return mNetSession->sendRequest(sessionID, data, size); |
| } |
| |
| void Sender::notifyInitDone() { |
| sp<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatInitDone); |
| notify->post(); |
| } |
| |
| void Sender::notifySessionDead() { |
| sp<AMessage> notify = mNotify->dup(); |
| notify->setInt32("what", kWhatSessionDead); |
| notify->post(); |
| } |
| |
| } // namespace android |
| |
