diff options
Diffstat (limited to 'media/libstagefright/wifi-display/rtp')
-rw-r--r-- | media/libstagefright/wifi-display/rtp/RTPBase.h | 51 | ||||
-rw-r--r-- | media/libstagefright/wifi-display/rtp/RTPSender.cpp | 805 | ||||
-rw-r--r-- | media/libstagefright/wifi-display/rtp/RTPSender.h | 121 |
3 files changed, 977 insertions, 0 deletions
diff --git a/media/libstagefright/wifi-display/rtp/RTPBase.h b/media/libstagefright/wifi-display/rtp/RTPBase.h new file mode 100644 index 0000000..6178f00 --- /dev/null +++ b/media/libstagefright/wifi-display/rtp/RTPBase.h @@ -0,0 +1,51 @@ +/* + * Copyright 2013, 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. + */ + +#ifndef RTP_BASE_H_ + +#define RTP_BASE_H_ + +namespace android { + +struct RTPBase { + enum PacketizationMode { + PACKETIZATION_TRANSPORT_STREAM, + PACKETIZATION_H264, + PACKETIZATION_AAC, + PACKETIZATION_NONE, + }; + + enum TransportMode { + TRANSPORT_UNDEFINED, + TRANSPORT_NONE, + TRANSPORT_UDP, + TRANSPORT_TCP, + TRANSPORT_TCP_INTERLEAVED, + }; + + enum { + // Really UDP _payload_ size + kMaxUDPPacketSize = 1472, // 1472 good, 1473 bad on Android@Home + }; + + static int32_t PickRandomRTPPort(); +}; + +} // namespace android + +#endif // RTP_BASE_H_ + + diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.cpp b/media/libstagefright/wifi-display/rtp/RTPSender.cpp new file mode 100644 index 0000000..1887b8b --- /dev/null +++ b/media/libstagefright/wifi-display/rtp/RTPSender.cpp @@ -0,0 +1,805 @@ +/* + * Copyright 2013, 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 "RTPSender" +#include <utils/Log.h> + +#include "RTPSender.h" + +#include <media/stagefright/foundation/ABuffer.h> +#include <media/stagefright/foundation/ADebug.h> +#include <media/stagefright/foundation/AMessage.h> +#include <media/stagefright/foundation/ANetworkSession.h> +#include <media/stagefright/foundation/hexdump.h> +#include <media/stagefright/MediaErrors.h> +#include <media/stagefright/Utils.h> + +#include "include/avc_utils.h" + +namespace android { + +RTPSender::RTPSender( + const sp<ANetworkSession> &netSession, + const sp<AMessage> ¬ify) + : mNetSession(netSession), + mNotify(notify), + mRTPMode(TRANSPORT_UNDEFINED), + mRTCPMode(TRANSPORT_UNDEFINED), + mRTPSessionID(0), + mRTCPSessionID(0), + mRTPConnected(false), + mRTCPConnected(false), + mLastNTPTime(0), + mLastRTPTime(0), + mNumRTPSent(0), + mNumRTPOctetsSent(0), + mNumSRsSent(0), + mRTPSeqNo(0), + mHistorySize(0) { +} + +RTPSender::~RTPSender() { + if (mRTCPSessionID != 0) { + mNetSession->destroySession(mRTCPSessionID); + mRTCPSessionID = 0; + } + + if (mRTPSessionID != 0) { + mNetSession->destroySession(mRTPSessionID); + mRTPSessionID = 0; + } +} + +// static +int32_t RTPBase::PickRandomRTPPort() { + // Pick an even integer in range [1024, 65534) + + static const size_t kRange = (65534 - 1024) / 2; + + return (int32_t)(((float)(kRange + 1) * rand()) / RAND_MAX) * 2 + 1024; +} + +status_t RTPSender::initAsync( + const char *remoteHost, + int32_t remoteRTPPort, + TransportMode rtpMode, + int32_t remoteRTCPPort, + TransportMode rtcpMode, + int32_t *outLocalRTPPort) { + if (mRTPMode != TRANSPORT_UNDEFINED + || rtpMode == TRANSPORT_UNDEFINED + || rtpMode == TRANSPORT_NONE + || rtcpMode == TRANSPORT_UNDEFINED) { + return INVALID_OPERATION; + } + + CHECK_NE(rtpMode, TRANSPORT_TCP_INTERLEAVED); + CHECK_NE(rtcpMode, TRANSPORT_TCP_INTERLEAVED); + + if ((rtcpMode == TRANSPORT_NONE && remoteRTCPPort >= 0) + || (rtcpMode != TRANSPORT_NONE && remoteRTCPPort < 0)) { + return INVALID_OPERATION; + } + + sp<AMessage> rtpNotify = new AMessage(kWhatRTPNotify, id()); + + sp<AMessage> rtcpNotify; + if (remoteRTCPPort >= 0) { + rtcpNotify = new AMessage(kWhatRTCPNotify, id()); + } + + CHECK_EQ(mRTPSessionID, 0); + CHECK_EQ(mRTCPSessionID, 0); + + int32_t localRTPPort; + + for (;;) { + localRTPPort = PickRandomRTPPort(); + + status_t err; + if (rtpMode == TRANSPORT_UDP) { + err = mNetSession->createUDPSession( + localRTPPort, + remoteHost, + remoteRTPPort, + rtpNotify, + &mRTPSessionID); + } else { + CHECK_EQ(rtpMode, TRANSPORT_TCP); + err = mNetSession->createTCPDatagramSession( + localRTPPort, + remoteHost, + remoteRTPPort, + rtpNotify, + &mRTPSessionID); + } + + if (err != OK) { + continue; + } + + if (remoteRTCPPort < 0) { + break; + } + + if (rtcpMode == TRANSPORT_UDP) { + err = mNetSession->createUDPSession( + localRTPPort + 1, + remoteHost, + remoteRTCPPort, + rtcpNotify, + &mRTCPSessionID); + } else { + CHECK_EQ(rtcpMode, TRANSPORT_TCP); + err = mNetSession->createTCPDatagramSession( + localRTPPort + 1, + remoteHost, + remoteRTCPPort, + rtcpNotify, + &mRTCPSessionID); + } + + if (err == OK) { + break; + } + + mNetSession->destroySession(mRTPSessionID); + mRTPSessionID = 0; + } + + if (rtpMode == TRANSPORT_UDP) { + mRTPConnected = true; + } + + if (rtcpMode == TRANSPORT_UDP) { + mRTCPConnected = true; + } + + mRTPMode = rtpMode; + mRTCPMode = rtcpMode; + *outLocalRTPPort = localRTPPort; + + if (mRTPMode == TRANSPORT_UDP + && (mRTCPMode == TRANSPORT_UDP || mRTCPMode == TRANSPORT_NONE)) { + notifyInitDone(OK); + } + + return OK; +} + +status_t RTPSender::queueBuffer( + const sp<ABuffer> &buffer, uint8_t packetType, PacketizationMode mode) { + status_t err; + + switch (mode) { + case PACKETIZATION_NONE: + err = queueRawPacket(buffer, packetType); + break; + + case PACKETIZATION_TRANSPORT_STREAM: + err = queueTSPackets(buffer, packetType); + break; + + case PACKETIZATION_H264: + err = queueAVCBuffer(buffer, packetType); + break; + + default: + TRESPASS(); + } + + return err; +} + +status_t RTPSender::queueRawPacket( + const sp<ABuffer> &packet, uint8_t packetType) { + CHECK_LE(packet->size(), kMaxUDPPacketSize - 12); + + int64_t timeUs; + CHECK(packet->meta()->findInt64("timeUs", &timeUs)); + + sp<ABuffer> udpPacket = new ABuffer(12 + packet->size()); + + udpPacket->setInt32Data(mRTPSeqNo); + + uint8_t *rtp = udpPacket->data(); + rtp[0] = 0x80; + rtp[1] = packetType; + + rtp[2] = (mRTPSeqNo >> 8) & 0xff; + rtp[3] = mRTPSeqNo & 0xff; + ++mRTPSeqNo; + + uint32_t rtpTime = (timeUs * 9) / 100ll; + + 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; + + memcpy(&rtp[12], packet->data(), packet->size()); + + return sendRTPPacket( + udpPacket, + true /* storeInHistory */, + true /* timeValid */, + ALooper::GetNowUs()); +} + +status_t RTPSender::queueTSPackets( + const sp<ABuffer> &tsPackets, uint8_t packetType) { + CHECK_EQ(0, tsPackets->size() % 188); + + int64_t timeUs; + CHECK(tsPackets->meta()->findInt64("timeUs", &timeUs)); + + const size_t numTSPackets = tsPackets->size() / 188; + + size_t srcOffset = 0; + while (srcOffset < tsPackets->size()) { + sp<ABuffer> udpPacket = + new ABuffer(12 + kMaxNumTSPacketsPerRTPPacket * 188); + + udpPacket->setInt32Data(mRTPSeqNo); + + uint8_t *rtp = udpPacket->data(); + rtp[0] = 0x80; + rtp[1] = packetType; + + rtp[2] = (mRTPSeqNo >> 8) & 0xff; + rtp[3] = mRTPSeqNo & 0xff; + ++mRTPSeqNo; + + int64_t nowUs = ALooper::GetNowUs(); + uint32_t rtpTime = (nowUs * 9) / 100ll; + + 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; + + size_t numTSPackets = (tsPackets->size() - srcOffset) / 188; + if (numTSPackets > kMaxNumTSPacketsPerRTPPacket) { + numTSPackets = kMaxNumTSPacketsPerRTPPacket; + } + + memcpy(&rtp[12], tsPackets->data() + srcOffset, numTSPackets * 188); + + udpPacket->setRange(0, 12 + numTSPackets * 188); + + srcOffset += numTSPackets * 188; + bool isLastPacket = (srcOffset == tsPackets->size()); + + status_t err = sendRTPPacket( + udpPacket, + true /* storeInHistory */, + isLastPacket /* timeValid */, + timeUs); + + if (err != OK) { + return err; + } + } + + return OK; +} + +status_t RTPSender::queueAVCBuffer( + const sp<ABuffer> &accessUnit, uint8_t packetType) { + int64_t timeUs; + CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs)); + + uint32_t rtpTime = (timeUs * 9 / 100ll); + + List<sp<ABuffer> > packets; + + sp<ABuffer> out = new ABuffer(kMaxUDPPacketSize); + size_t outBytesUsed = 12; // Placeholder for RTP header. + + const uint8_t *data = accessUnit->data(); + size_t size = accessUnit->size(); + const uint8_t *nalStart; + size_t nalSize; + while (getNextNALUnit( + &data, &size, &nalStart, &nalSize, + true /* startCodeFollows */) == OK) { + size_t bytesNeeded = nalSize + 2; + if (outBytesUsed == 12) { + ++bytesNeeded; + } + + if (outBytesUsed + bytesNeeded > out->capacity()) { + bool emitSingleNALPacket = false; + + if (outBytesUsed == 12 + && outBytesUsed + nalSize <= out->capacity()) { + // We haven't emitted anything into the current packet yet and + // this NAL unit fits into a single-NAL-unit-packet while + // it wouldn't have fit as part of a STAP-A packet. + + memcpy(out->data() + outBytesUsed, nalStart, nalSize); + outBytesUsed += nalSize; + + emitSingleNALPacket = true; + } + + if (outBytesUsed > 12) { + out->setRange(0, outBytesUsed); + packets.push_back(out); + out = new ABuffer(kMaxUDPPacketSize); + outBytesUsed = 12; // Placeholder for RTP header + } + + if (emitSingleNALPacket) { + continue; + } + } + + if (outBytesUsed + bytesNeeded <= out->capacity()) { + uint8_t *dst = out->data() + outBytesUsed; + + if (outBytesUsed == 12) { + *dst++ = 24; // STAP-A header + } + + *dst++ = (nalSize >> 8) & 0xff; + *dst++ = nalSize & 0xff; + memcpy(dst, nalStart, nalSize); + + outBytesUsed += bytesNeeded; + continue; + } + + // This single NAL unit does not fit into a single RTP packet, + // we need to emit an FU-A. + + CHECK_EQ(outBytesUsed, 12u); + + uint8_t nalType = nalStart[0] & 0x1f; + uint8_t nri = (nalStart[0] >> 5) & 3; + + size_t srcOffset = 1; + while (srcOffset < nalSize) { + size_t copy = out->capacity() - outBytesUsed - 2; + if (copy > nalSize - srcOffset) { + copy = nalSize - srcOffset; + } + + uint8_t *dst = out->data() + outBytesUsed; + dst[0] = (nri << 5) | 28; + + dst[1] = nalType; + + if (srcOffset == 1) { + dst[1] |= 0x80; + } + + if (srcOffset + copy == nalSize) { + dst[1] |= 0x40; + } + + memcpy(&dst[2], nalStart + srcOffset, copy); + srcOffset += copy; + + out->setRange(0, outBytesUsed + copy + 2); + + packets.push_back(out); + out = new ABuffer(kMaxUDPPacketSize); + outBytesUsed = 12; // Placeholder for RTP header + } + } + + if (outBytesUsed > 12) { + out->setRange(0, outBytesUsed); + packets.push_back(out); + } + + while (!packets.empty()) { + sp<ABuffer> out = *packets.begin(); + packets.erase(packets.begin()); + + out->setInt32Data(mRTPSeqNo); + + bool last = packets.empty(); + + uint8_t *dst = out->data(); + + dst[0] = 0x80; + + dst[1] = packetType; + if (last) { + dst[1] |= 1 << 7; // M-bit + } + + dst[2] = (mRTPSeqNo >> 8) & 0xff; + dst[3] = mRTPSeqNo & 0xff; + ++mRTPSeqNo; + + dst[4] = rtpTime >> 24; + dst[5] = (rtpTime >> 16) & 0xff; + dst[6] = (rtpTime >> 8) & 0xff; + dst[7] = rtpTime & 0xff; + dst[8] = kSourceID >> 24; + dst[9] = (kSourceID >> 16) & 0xff; + dst[10] = (kSourceID >> 8) & 0xff; + dst[11] = kSourceID & 0xff; + + status_t err = sendRTPPacket(out, true /* storeInHistory */); + + if (err != OK) { + return err; + } + } + + return OK; +} + +status_t RTPSender::sendRTPPacket( + const sp<ABuffer> &buffer, bool storeInHistory, + bool timeValid, int64_t timeUs) { + CHECK(mRTPConnected); + + status_t err = mNetSession->sendRequest( + mRTPSessionID, buffer->data(), buffer->size(), + timeValid, timeUs); + + if (err != OK) { + return err; + } + + mLastNTPTime = GetNowNTP(); + mLastRTPTime = U32_AT(buffer->data() + 4); + + ++mNumRTPSent; + mNumRTPOctetsSent += buffer->size() - 12; + + if (storeInHistory) { + if (mHistorySize == kMaxHistorySize) { + mHistory.erase(mHistory.begin()); + } else { + ++mHistorySize; + } + mHistory.push_back(buffer); + } + + return OK; +} + +// static +uint64_t RTPSender::GetNowNTP() { + struct timeval tv; + gettimeofday(&tv, NULL /* timezone */); + + uint64_t nowUs = tv.tv_sec * 1000000ll + tv.tv_usec; + + 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 RTPSender::onMessageReceived(const sp<AMessage> &msg) { + switch (msg->what()) { + case kWhatRTPNotify: + case kWhatRTCPNotify: + onNetNotify(msg->what() == kWhatRTPNotify, msg); + break; + + default: + TRESPASS(); + } +} + +void RTPSender::onNetNotify(bool isRTP, const sp<AMessage> &msg) { + 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)); + + 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 (!mRTPConnected + || (mRTPMode != TRANSPORT_NONE && !mRTCPConnected)) { + // We haven't completed initialization, attach the error + // to the notification instead. + notifyInitDone(err); + break; + } + + notifyError(err); + break; + } + + case ANetworkSession::kWhatDatagram: + { + sp<ABuffer> data; + CHECK(msg->findBuffer("data", &data)); + + if (isRTP) { + ALOGW("Huh? Received data on RTP connection..."); + } else { + onRTCPData(data); + } + break; + } + + case ANetworkSession::kWhatConnected: + { + int32_t sessionID; + CHECK(msg->findInt32("sessionID", &sessionID)); + + if (isRTP) { + CHECK_EQ(mRTPMode, TRANSPORT_TCP); + CHECK_EQ(sessionID, mRTPSessionID); + mRTPConnected = true; + } else { + CHECK_EQ(mRTCPMode, TRANSPORT_TCP); + CHECK_EQ(sessionID, mRTCPSessionID); + mRTCPConnected = true; + } + + if (mRTPConnected + && (mRTCPMode == TRANSPORT_NONE || mRTCPConnected)) { + notifyInitDone(OK); + } + break; + } + + case ANetworkSession::kWhatNetworkStall: + { + size_t numBytesQueued; + CHECK(msg->findSize("numBytesQueued", &numBytesQueued)); + + notifyNetworkStall(numBytesQueued); + break; + } + + default: + TRESPASS(); + } +} + +status_t RTPSender::onRTCPData(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 + parseReceiverReport(data, headerLength); + break; + + case 202: // SDES + case 203: + break; + + case 204: // APP + parseAPP(data, headerLength); + break; + + case 205: // TSFB (transport layer specific feedback) + parseTSFB(data, headerLength); + break; + + 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 RTPSender::parseReceiverReport(const uint8_t *data, size_t size) { + // hexdump(data, size); + + float fractionLost = data[12] / 256.0f; + + ALOGI("lost %.2f %% of packets during report interval.", + 100.0f * fractionLost); + + return OK; +} + +status_t RTPSender::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) { + ALOGV("retransmitting seqNo %d", bufferSeqNo); + + CHECK_EQ((status_t)OK, + sendRTPPacket(buffer, false /* storeInHistory */)); + + 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 (seqNo = %d, foundSeqNo = %d, blp = 0x%04x)", + seqNo, foundSeqNo, blp); + + if (!mHistory.empty()) { + int32_t earliest = (*mHistory.begin())->int32Data() & 0xffff; + int32_t latest = (*--mHistory.end())->int32Data() & 0xffff; + + ALOGI("have seq numbers from %d - %d", earliest, latest); + } + } + } + + return OK; +} + +status_t RTPSender::parseAPP(const uint8_t *data, size_t size) { + if (!memcmp("late", &data[8], 4)) { + int64_t avgLatencyUs = (int64_t)U64_AT(&data[12]); + int64_t maxLatencyUs = (int64_t)U64_AT(&data[20]); + + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatInformSender); + notify->setInt64("avgLatencyUs", avgLatencyUs); + notify->setInt64("maxLatencyUs", maxLatencyUs); + notify->post(); + } + + return OK; +} + +void RTPSender::notifyInitDone(status_t err) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatInitDone); + notify->setInt32("err", err); + notify->post(); +} + +void RTPSender::notifyError(status_t err) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatError); + notify->setInt32("err", err); + notify->post(); +} + +void RTPSender::notifyNetworkStall(size_t numBytesQueued) { + sp<AMessage> notify = mNotify->dup(); + notify->setInt32("what", kWhatNetworkStall); + notify->setSize("numBytesQueued", numBytesQueued); + notify->post(); +} + +} // namespace android + diff --git a/media/libstagefright/wifi-display/rtp/RTPSender.h b/media/libstagefright/wifi-display/rtp/RTPSender.h new file mode 100644 index 0000000..fefcab7 --- /dev/null +++ b/media/libstagefright/wifi-display/rtp/RTPSender.h @@ -0,0 +1,121 @@ +/* + * Copyright 2013, 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. + */ + +#ifndef RTP_SENDER_H_ + +#define RTP_SENDER_H_ + +#include "RTPBase.h" + +#include <media/stagefright/foundation/AHandler.h> + +namespace android { + +struct ABuffer; +struct ANetworkSession; + +// An object of this class facilitates sending of media data over an RTP +// channel. The channel is established over a UDP or TCP connection depending +// on which "TransportMode" was chosen. In addition different RTP packetization +// schemes are supported such as "Transport Stream Packets over RTP", +// or "AVC/H.264 encapsulation as specified in RFC 3984 (non-interleaved mode)" +struct RTPSender : public RTPBase, public AHandler { + enum { + kWhatInitDone, + kWhatError, + kWhatNetworkStall, + kWhatInformSender, + }; + RTPSender( + const sp<ANetworkSession> &netSession, + const sp<AMessage> ¬ify); + + status_t initAsync( + const char *remoteHost, + int32_t remoteRTPPort, + TransportMode rtpMode, + int32_t remoteRTCPPort, + TransportMode rtcpMode, + int32_t *outLocalRTPPort); + + status_t queueBuffer( + const sp<ABuffer> &buffer, + uint8_t packetType, + PacketizationMode mode); + +protected: + virtual ~RTPSender(); + virtual void onMessageReceived(const sp<AMessage> &msg); + +private: + enum { + kWhatRTPNotify, + kWhatRTCPNotify, + }; + + enum { + kMaxNumTSPacketsPerRTPPacket = (kMaxUDPPacketSize - 12) / 188, + kMaxHistorySize = 1024, + kSourceID = 0xdeadbeef, + }; + + sp<ANetworkSession> mNetSession; + sp<AMessage> mNotify; + TransportMode mRTPMode; + TransportMode mRTCPMode; + int32_t mRTPSessionID; + int32_t mRTCPSessionID; + bool mRTPConnected; + bool mRTCPConnected; + + uint64_t mLastNTPTime; + uint32_t mLastRTPTime; + uint32_t mNumRTPSent; + uint32_t mNumRTPOctetsSent; + uint32_t mNumSRsSent; + + uint32_t mRTPSeqNo; + + List<sp<ABuffer> > mHistory; + size_t mHistorySize; + + static uint64_t GetNowNTP(); + + status_t queueRawPacket(const sp<ABuffer> &tsPackets, uint8_t packetType); + status_t queueTSPackets(const sp<ABuffer> &tsPackets, uint8_t packetType); + status_t queueAVCBuffer(const sp<ABuffer> &accessUnit, uint8_t packetType); + + status_t sendRTPPacket( + const sp<ABuffer> &packet, bool storeInHistory, + bool timeValid = false, int64_t timeUs = -1ll); + + void onNetNotify(bool isRTP, const sp<AMessage> &msg); + + status_t onRTCPData(const sp<ABuffer> &data); + status_t parseReceiverReport(const uint8_t *data, size_t size); + status_t parseTSFB(const uint8_t *data, size_t size); + status_t parseAPP(const uint8_t *data, size_t size); + + void notifyInitDone(status_t err); + void notifyError(status_t err); + void notifyNetworkStall(size_t numBytesQueued); + + DISALLOW_EVIL_CONSTRUCTORS(RTPSender); +}; + +} // namespace android + +#endif // RTP_SENDER_H_ |