VT: RTPSource: Added a component as an one of NuPlayer::Source
RTPSource added to be used for RTP scenario.
This can be created by NuPlayer::setDataSource(String& rtpParams).
Code implementation includes specifications of RFC 3550, RFC 4566
Following parameters should be provided to setup RTPSource.
- video-param-decoder-profile : decoder profile to be set to codec.
- video-param-decoder-level : decoder level to be set to codec.
- video-param-width : video width to be set to codec.
- video-param-height : video height to be set to codec.
- rtp-param-local-ip : local-ip that the RTP/RTCP sockets be bound.
- rtp-param-local-port : local-port assigned that the RTP/RTCP sockets be bound.
- rtp-param-remote-ip : remote-ip that the RTCP socket where to sent.
- rtp-param-remote-port : remote-port assigned that the RTCP socket where to sent.
- rtp-param-payload-type : Section 5.1 of RFC 3550, Payload type value assigned for the RTP session.
- rtp-param-as : Section 5.8 of RFC 4566, Maximum bandwidth belong to this session.
Bug: 121230209
Change-Id: I9fdc71a854d441703c8dd2fc8115de36a5e9958e
Signed-off-by: Byeongjo Park <bjo.park@samsung.com>
Signed-off-by: Kim Sungyeon <sy85.kim@samsung.com>
diff --git a/media/libmediaplayerservice/nuplayer/RTPSource.cpp b/media/libmediaplayerservice/nuplayer/RTPSource.cpp
new file mode 100644
index 0000000..de1f8a1
--- /dev/null
+++ b/media/libmediaplayerservice/nuplayer/RTPSource.cpp
@@ -0,0 +1,708 @@
+/*
+ * Copyright (C) 2010 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 "RTPSource"
+#include <utils/Log.h>
+
+#include "RTPSource.h"
+
+
+
+
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <string.h>
+
+namespace android {
+
+const int64_t kNearEOSTimeoutUs = 2000000ll; // 2 secs
+static int32_t kMaxAllowedStaleAccessUnits = 20;
+
+NuPlayer::RTPSource::RTPSource(
+ const sp<AMessage> ¬ify,
+ const String8& rtpParams)
+ : Source(notify),
+ mRTPParams(rtpParams),
+ mFlags(0),
+ mState(DISCONNECTED),
+ mFinalResult(OK),
+ mBuffering(false),
+ mInPreparationPhase(true),
+ mRTPConn(new ARTPConnection),
+ mEOSTimeoutAudio(0),
+ mEOSTimeoutVideo(0) {
+ ALOGD("RTPSource initialized with rtpParams=%s", rtpParams.string());
+}
+
+NuPlayer::RTPSource::~RTPSource() {
+ if (mLooper != NULL) {
+ mLooper->unregisterHandler(id());
+ mLooper->unregisterHandler(mRTPConn->id());
+ mLooper->stop();
+ }
+}
+
+status_t NuPlayer::RTPSource::getBufferingSettings(
+ BufferingSettings* buffering /* nonnull */) {
+ Mutex::Autolock _l(mBufferingSettingsLock);
+ *buffering = mBufferingSettings;
+ return OK;
+}
+
+status_t NuPlayer::RTPSource::setBufferingSettings(const BufferingSettings& buffering) {
+ Mutex::Autolock _l(mBufferingSettingsLock);
+ mBufferingSettings = buffering;
+ return OK;
+}
+
+void NuPlayer::RTPSource::prepareAsync() {
+ if (mLooper == NULL) {
+ mLooper = new ALooper;
+ mLooper->setName("rtp");
+ mLooper->start();
+
+ mLooper->registerHandler(this);
+ mLooper->registerHandler(mRTPConn);
+ }
+
+ setParameters(mRTPParams);
+
+ TrackInfo *info = NULL;
+ unsigned i;
+ for (i = 0; i < mTracks.size(); i++) {
+ info = &mTracks.editItemAt(i);
+
+ if (info == NULL)
+ break;
+
+ AString sdp;
+ ASessionDescription::SDPStringFactory(sdp, info->mLocalIp,
+ info->mIsAudio, info->mLocalPort, info->mPayloadType, info->mAS, info->mCodecName,
+ NULL, info->mWidth, info->mHeight);
+ ALOGD("RTPSource SDP =>\n%s", sdp.c_str());
+
+ sp<ASessionDescription> desc = new ASessionDescription;
+ bool isValidSdp = desc->setTo(sdp.c_str(), sdp.size());
+ ALOGV("RTPSource isValidSdp => %d", isValidSdp);
+
+ int sockRtp, sockRtcp;
+ ARTPConnection::MakeRTPSocketPair(&sockRtp, &sockRtcp, info->mLocalIp, info->mRemoteIp,
+ info->mLocalPort, info->mRemotePort);
+
+ sp<AMessage> notify = new AMessage('accu', this);
+
+ ALOGV("RTPSource addStream. track-index=%d", i);
+ notify->setSize("trackIndex", i);
+ // index(i) should be started from 1. 0 is reserved for [root]
+ mRTPConn->addStream(sockRtp, sockRtcp, desc, i + 1, notify, false);
+
+ info->mRTPSocket = sockRtp;
+ info->mRTCPSocket = sockRtcp;
+ info->mFirstSeqNumInSegment = 0;
+ info->mNewSegment = true;
+ info->mAllowedStaleAccessUnits = kMaxAllowedStaleAccessUnits;
+ info->mRTPAnchor = 0;
+ info->mNTPAnchorUs = -1;
+ info->mNormalPlayTimeRTP = 0;
+ info->mNormalPlayTimeUs = 0ll;
+
+ // index(i) should be started from 1. 0 is reserved for [root]
+ info->mPacketSource = new APacketSource(desc, i + 1);
+
+ int32_t timeScale;
+ sp<MetaData> format = getTrackFormat(i, &timeScale);
+ sp<AnotherPacketSource> source = new AnotherPacketSource(format);
+
+ if (info->mIsAudio) {
+ mAudioTrack = source;
+ } else {
+ mVideoTrack = source;
+ }
+
+ info->mSource = source;
+ }
+
+ CHECK_EQ(mState, (int)DISCONNECTED);
+ mState = CONNECTING;
+
+ if (mInPreparationPhase) {
+ mInPreparationPhase = false;
+ notifyPrepared();
+ }
+}
+
+void NuPlayer::RTPSource::start() {
+}
+
+void NuPlayer::RTPSource::pause() {
+ mState = PAUSED;
+}
+
+void NuPlayer::RTPSource::resume() {
+ mState = CONNECTING;
+}
+
+void NuPlayer::RTPSource::stop() {
+ if (mLooper == NULL) {
+ return;
+ }
+ sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
+
+ sp<AMessage> dummy;
+ msg->postAndAwaitResponse(&dummy);
+}
+
+status_t NuPlayer::RTPSource::feedMoreTSData() {
+ Mutex::Autolock _l(mBufferingLock);
+ return mFinalResult;
+}
+
+sp<MetaData> NuPlayer::RTPSource::getFormatMeta(bool audio) {
+ sp<AnotherPacketSource> source = getSource(audio);
+
+ if (source == NULL) {
+ return NULL;
+ }
+
+ return source->getFormat();
+}
+
+bool NuPlayer::RTPSource::haveSufficientDataOnAllTracks() {
+ // We're going to buffer at least 2 secs worth data on all tracks before
+ // starting playback (both at startup and after a seek).
+
+ static const int64_t kMinDurationUs = 2000000ll;
+
+ int64_t mediaDurationUs = 0;
+ getDuration(&mediaDurationUs);
+ if ((mAudioTrack != NULL && mAudioTrack->isFinished(mediaDurationUs))
+ || (mVideoTrack != NULL && mVideoTrack->isFinished(mediaDurationUs))) {
+ return true;
+ }
+
+ status_t err;
+ int64_t durationUs;
+ if (mAudioTrack != NULL
+ && (durationUs = mAudioTrack->getBufferedDurationUs(&err))
+ < kMinDurationUs
+ && err == OK) {
+ ALOGV("audio track doesn't have enough data yet. (%.2f secs buffered)",
+ durationUs / 1E6);
+ return false;
+ }
+
+ if (mVideoTrack != NULL
+ && (durationUs = mVideoTrack->getBufferedDurationUs(&err))
+ < kMinDurationUs
+ && err == OK) {
+ ALOGV("video track doesn't have enough data yet. (%.2f secs buffered)",
+ durationUs / 1E6);
+ return false;
+ }
+
+ return true;
+}
+
+status_t NuPlayer::RTPSource::dequeueAccessUnit(
+ bool audio, sp<ABuffer> *accessUnit) {
+
+ sp<AnotherPacketSource> source = getSource(audio);
+
+ if (mState == PAUSED) {
+ ALOGV("-EWOULDBLOCK");
+ return -EWOULDBLOCK;
+ }
+
+ status_t finalResult;
+ if (!source->hasBufferAvailable(&finalResult)) {
+ if (finalResult == OK) {
+ int64_t mediaDurationUs = 0;
+ getDuration(&mediaDurationUs);
+ sp<AnotherPacketSource> otherSource = getSource(!audio);
+ status_t otherFinalResult;
+
+ // If other source already signaled EOS, this source should also signal EOS
+ if (otherSource != NULL &&
+ !otherSource->hasBufferAvailable(&otherFinalResult) &&
+ otherFinalResult == ERROR_END_OF_STREAM) {
+ source->signalEOS(ERROR_END_OF_STREAM);
+ return ERROR_END_OF_STREAM;
+ }
+
+ // If this source has detected near end, give it some time to retrieve more
+ // data before signaling EOS
+ if (source->isFinished(mediaDurationUs)) {
+ int64_t eosTimeout = audio ? mEOSTimeoutAudio : mEOSTimeoutVideo;
+ if (eosTimeout == 0) {
+ setEOSTimeout(audio, ALooper::GetNowUs());
+ } else if ((ALooper::GetNowUs() - eosTimeout) > kNearEOSTimeoutUs) {
+ setEOSTimeout(audio, 0);
+ source->signalEOS(ERROR_END_OF_STREAM);
+ return ERROR_END_OF_STREAM;
+ }
+ return -EWOULDBLOCK;
+ }
+
+ if (!(otherSource != NULL && otherSource->isFinished(mediaDurationUs))) {
+ // We should not enter buffering mode
+ // if any of the sources already have detected EOS.
+ // TODO: needs to be checked whether below line is needed or not.
+ // startBufferingIfNecessary();
+ }
+
+ return -EWOULDBLOCK;
+ }
+ return finalResult;
+ }
+
+ setEOSTimeout(audio, 0);
+
+ return source->dequeueAccessUnit(accessUnit);
+}
+
+sp<AnotherPacketSource> NuPlayer::RTPSource::getSource(bool audio) {
+ return audio ? mAudioTrack : mVideoTrack;
+}
+
+void NuPlayer::RTPSource::setEOSTimeout(bool audio, int64_t timeout) {
+ if (audio) {
+ mEOSTimeoutAudio = timeout;
+ } else {
+ mEOSTimeoutVideo = timeout;
+ }
+}
+
+status_t NuPlayer::RTPSource::getDuration(int64_t *durationUs) {
+ *durationUs = 0ll;
+
+ int64_t audioDurationUs;
+ if (mAudioTrack != NULL
+ && mAudioTrack->getFormat()->findInt64(
+ kKeyDuration, &audioDurationUs)
+ && audioDurationUs > *durationUs) {
+ *durationUs = audioDurationUs;
+ }
+
+ int64_t videoDurationUs;
+ if (mVideoTrack != NULL
+ && mVideoTrack->getFormat()->findInt64(
+ kKeyDuration, &videoDurationUs)
+ && videoDurationUs > *durationUs) {
+ *durationUs = videoDurationUs;
+ }
+
+ return OK;
+}
+
+status_t NuPlayer::RTPSource::seekTo(int64_t seekTimeUs, MediaPlayerSeekMode mode) {
+ ALOGV("RTPSource::seekTo=%d, mode=%d", (int)seekTimeUs, mode);
+ return OK;
+}
+
+void NuPlayer::RTPSource::schedulePollBuffering() {
+ sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
+ msg->post(1000000ll); // 1 second intervals
+}
+
+void NuPlayer::RTPSource::onPollBuffering() {
+ schedulePollBuffering();
+}
+
+void NuPlayer::RTPSource::onMessageReceived(const sp<AMessage> &msg) {
+ ALOGV("onMessageReceived =%d", msg->what());
+
+ switch (msg->what()) {
+ case kWhatAccessUnitComplete:
+ {
+ if (mState == CONNECTING) {
+ mState = CONNECTED;
+ }
+
+ int32_t timeUpdate;
+ //"time-update" raised from ARTPConnection::parseSR()
+ if (msg->findInt32("time-update", &timeUpdate) && timeUpdate) {
+ size_t trackIndex;
+ CHECK(msg->findSize("trackIndex", &trackIndex));
+
+ uint32_t rtpTime;
+ uint64_t ntpTime;
+ CHECK(msg->findInt32("rtp-time", (int32_t *)&rtpTime));
+ CHECK(msg->findInt64("ntp-time", (int64_t *)&ntpTime));
+
+ onTimeUpdate(trackIndex, rtpTime, ntpTime);
+ break;
+ }
+
+ int32_t firstRTCP;
+ if (msg->findInt32("first-rtcp", &firstRTCP)) {
+ // There won't be an access unit here, it's just a notification
+ // that the data communication worked since we got the first
+ // rtcp packet.
+ ALOGV("first-rtcp");
+ break;
+ }
+
+ size_t trackIndex;
+ CHECK(msg->findSize("trackIndex", &trackIndex));
+
+ sp<ABuffer> accessUnit;
+ if (msg->findBuffer("access-unit", &accessUnit) == false) {
+ break;
+ }
+
+ int32_t damaged;
+ if (accessUnit->meta()->findInt32("damaged", &damaged)
+ && damaged) {
+ ALOGD("dropping damaged access unit.");
+ break;
+ }
+
+ TrackInfo *info = &mTracks.editItemAt(trackIndex);
+
+ sp<AnotherPacketSource> source = info->mSource;
+ if (source != NULL) {
+ uint32_t rtpTime;
+ CHECK(accessUnit->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
+
+ /* AnotherPacketSource make an assertion if there is no ntp provided
+ RTPSource should provide ntpUs all the times.
+ if (!info->mNPTMappingValid) {
+ // This is a live stream, we didn't receive any normal
+ // playtime mapping. We won't map to npt time.
+ source->queueAccessUnit(accessUnit);
+ break;
+ }
+ */
+
+ int64_t nptUs =
+ ((double)rtpTime - (double)info->mRTPTime)
+ / info->mTimeScale
+ * 1000000ll
+ + info->mNormalPlaytimeUs;
+
+ accessUnit->meta()->setInt64("timeUs", nptUs);
+
+ source->queueAccessUnit(accessUnit);
+ }
+
+ break;
+ }
+ case kWhatDisconnect:
+ {
+ sp<AReplyToken> replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ TrackInfo *info = &mTracks.editItemAt(i);
+
+ if (info->mIsAudio) {
+ mAudioTrack->signalEOS(ERROR_END_OF_STREAM);
+ mAudioTrack = NULL;
+ ALOGV("mAudioTrack disconnected");
+ } else {
+ mVideoTrack->signalEOS(ERROR_END_OF_STREAM);
+ mVideoTrack = NULL;
+ ALOGV("mVideoTrack disconnected");
+ }
+
+ mRTPConn->removeStream(info->mRTPSocket, info->mRTCPSocket);
+ close(info->mRTPSocket);
+ close(info->mRTCPSocket);
+ }
+
+ mTracks.clear();
+ mFirstAccessUnit = true;
+ mAllTracksHaveTime = false;
+ mNTPAnchorUs = -1;
+ mMediaAnchorUs = -1;
+ mLastMediaTimeUs = -1;
+ mNumAccessUnitsReceived = 0;
+ mReceivedFirstRTCPPacket = false;
+ mReceivedFirstRTPPacket = false;
+ mPausing = false;
+ mPauseGeneration = 0;
+
+ (new AMessage)->postReply(replyID);
+
+ break;
+ }
+ case kWhatPollBuffering:
+ break;
+ default:
+ TRESPASS();
+ }
+}
+
+void NuPlayer::RTPSource::onTimeUpdate(int32_t trackIndex, uint32_t rtpTime, uint64_t ntpTime) {
+ ALOGV("onTimeUpdate track %d, rtpTime = 0x%08x, ntpTime = %#016llx",
+ trackIndex, rtpTime, (long long)ntpTime);
+
+ int64_t ntpTimeUs = (int64_t)(ntpTime * 1E6 / (1ll << 32));
+
+ TrackInfo *track = &mTracks.editItemAt(trackIndex);
+
+ track->mRTPAnchor = rtpTime;
+ track->mNTPAnchorUs = ntpTimeUs;
+
+ if (mNTPAnchorUs < 0) {
+ mNTPAnchorUs = ntpTimeUs;
+ mMediaAnchorUs = mLastMediaTimeUs;
+ }
+
+ if (!mAllTracksHaveTime) {
+ bool allTracksHaveTime = (mTracks.size() > 0);
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ TrackInfo *track = &mTracks.editItemAt(i);
+ if (track->mNTPAnchorUs < 0) {
+ allTracksHaveTime = false;
+ break;
+ }
+ }
+ if (allTracksHaveTime) {
+ mAllTracksHaveTime = true;
+ ALOGI("Time now established for all tracks.");
+ }
+ }
+ if (mAllTracksHaveTime && dataReceivedOnAllChannels()) {
+ // Time is now established, lets start timestamping immediately
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ TrackInfo *trackInfo = &mTracks.editItemAt(i);
+ while (!trackInfo->mPackets.empty()) {
+ sp<ABuffer> accessUnit = *trackInfo->mPackets.begin();
+ trackInfo->mPackets.erase(trackInfo->mPackets.begin());
+
+ if (addMediaTimestamp(i, trackInfo, accessUnit)) {
+ postQueueAccessUnit(i, accessUnit);
+ }
+ }
+ }
+ }
+}
+
+bool NuPlayer::RTPSource::addMediaTimestamp(
+ int32_t trackIndex, const TrackInfo *track,
+ const sp<ABuffer> &accessUnit) {
+
+ uint32_t rtpTime;
+ CHECK(accessUnit->meta()->findInt32(
+ "rtp-time", (int32_t *)&rtpTime));
+
+ int64_t relRtpTimeUs =
+ (((int64_t)rtpTime - (int64_t)track->mRTPAnchor) * 1000000ll)
+ / track->mTimeScale;
+
+ int64_t ntpTimeUs = track->mNTPAnchorUs + relRtpTimeUs;
+
+ int64_t mediaTimeUs = mMediaAnchorUs + ntpTimeUs - mNTPAnchorUs;
+
+ if (mediaTimeUs > mLastMediaTimeUs) {
+ mLastMediaTimeUs = mediaTimeUs;
+ }
+
+ if (mediaTimeUs < 0) {
+ ALOGV("dropping early accessUnit.");
+ return false;
+ }
+
+ ALOGV("track %d rtpTime=%u mediaTimeUs = %lld us (%.2f secs)",
+ trackIndex, rtpTime, (long long)mediaTimeUs, mediaTimeUs / 1E6);
+
+ accessUnit->meta()->setInt64("timeUs", mediaTimeUs);
+
+ return true;
+}
+
+bool NuPlayer::RTPSource::dataReceivedOnAllChannels() {
+ TrackInfo *track;
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ track = &mTracks.editItemAt(i);
+ if (track->mPackets.empty()) {
+ return false;
+ }
+ }
+ return true;
+}
+
+void NuPlayer::RTPSource::postQueueAccessUnit(
+ size_t trackIndex, const sp<ABuffer> &accessUnit) {
+ sp<AMessage> msg = new AMessage(kWhatAccessUnit, this);
+ msg->setInt32("what", kWhatAccessUnit);
+ msg->setSize("trackIndex", trackIndex);
+ msg->setBuffer("accessUnit", accessUnit);
+ msg->post();
+}
+
+void NuPlayer::RTPSource::postQueueEOS(size_t trackIndex, status_t finalResult) {
+ sp<AMessage> msg = new AMessage(kWhatEOS, this);
+ msg->setInt32("what", kWhatEOS);
+ msg->setSize("trackIndex", trackIndex);
+ msg->setInt32("finalResult", finalResult);
+ msg->post();
+}
+
+sp<MetaData> NuPlayer::RTPSource::getTrackFormat(size_t index, int32_t *timeScale) {
+ CHECK_GE(index, 0u);
+ CHECK_LT(index, mTracks.size());
+
+ const TrackInfo &info = mTracks.itemAt(index);
+
+ *timeScale = info.mTimeScale;
+
+ return info.mPacketSource->getFormat();
+}
+
+void NuPlayer::RTPSource::onConnected() {
+ ALOGV("onConnected");
+ mState = CONNECTED;
+}
+
+void NuPlayer::RTPSource::onDisconnected(const sp<AMessage> &msg) {
+ if (mState == DISCONNECTED) {
+ return;
+ }
+
+ status_t err;
+ CHECK(msg->findInt32("result", &err));
+ CHECK_NE(err, (status_t)OK);
+
+// mLooper->unregisterHandler(mHandler->id());
+// mHandler.clear();
+
+ if (mState == CONNECTING) {
+ // We're still in the preparation phase, signal that it
+ // failed.
+ notifyPrepared(err);
+ }
+
+ mState = DISCONNECTED;
+// setError(err);
+
+}
+
+status_t NuPlayer::RTPSource::setParameter(const String8 &key, const String8 &value) {
+ ALOGV("setParameter: key (%s) => value (%s)", key.string(), value.string());
+
+ bool isAudioKey = key.contains("audio");
+ TrackInfo *info = NULL;
+ for (unsigned i = 0; i < mTracks.size(); ++i) {
+ info = &mTracks.editItemAt(i);
+ if (info != NULL && info->mIsAudio == isAudioKey) {
+ ALOGV("setParameter: %s track (%d) found", isAudioKey ? "audio" : "video" , i);
+ break;
+ }
+ }
+
+ if (info == NULL) {
+ TrackInfo newTrackInfo;
+ newTrackInfo.mIsAudio = isAudioKey;
+ mTracks.push(newTrackInfo);
+ info = &mTracks.editTop();
+ }
+
+ if (key == "rtp-param-mime-type") {
+ info->mMimeType = value;
+
+ const char *mime = value.string();
+ const char *delimiter = strchr(mime, '/');
+ info->mCodecName = (delimiter + 1);
+
+ ALOGV("rtp-param-mime-type: mMimeType (%s) => mCodecName (%s)",
+ info->mMimeType.string(), info->mCodecName.string());
+ } else if (key == "video-param-decoder-profile") {
+ info->mCodecProfile = atoi(value);
+ } else if (key == "video-param-decoder-level") {
+ info->mCodecLevel = atoi(value);
+ } else if (key == "video-param-width") {
+ info->mWidth = atoi(value);
+ } else if (key == "video-param-height") {
+ info->mHeight = atoi(value);
+ } else if (key == "rtp-param-local-ip") {
+ info->mLocalIp = value;
+ } else if (key == "rtp-param-local-port") {
+ info->mLocalPort = atoi(value);
+ } else if (key == "rtp-param-remote-ip") {
+ info->mRemoteIp = value;
+ } else if (key == "rtp-param-remote-port") {
+ info->mRemotePort = atoi(value);
+ } else if (key == "rtp-param-payload-type") {
+ info->mPayloadType = atoi(value);
+ } else if (key == "rtp-param-as") {
+ //AS means guaranteed bit rate that negotiated from sdp.
+ info->mAS = atoi(value);
+ } else if (key == "rtp-param-rtp-timeout") {
+ } else if (key == "rtp-param-rtcp-timeout") {
+ } else if (key == "rtp-param-time-scale") {
+ }
+
+ return OK;
+}
+
+status_t NuPlayer::RTPSource::setParameters(const String8 ¶ms) {
+ ALOGV("setParameters: %s", params.string());
+ const char *cparams = params.string();
+ const char *key_start = cparams;
+ for (;;) {
+ const char *equal_pos = strchr(key_start, '=');
+ if (equal_pos == NULL) {
+ ALOGE("Parameters %s miss a value", cparams);
+ return BAD_VALUE;
+ }
+ String8 key(key_start, equal_pos - key_start);
+ TrimString(&key);
+ if (key.length() == 0) {
+ ALOGE("Parameters %s contains an empty key", cparams);
+ return BAD_VALUE;
+ }
+ const char *value_start = equal_pos + 1;
+ const char *semicolon_pos = strchr(value_start, ';');
+ String8 value;
+ if (semicolon_pos == NULL) {
+ value.setTo(value_start);
+ } else {
+ value.setTo(value_start, semicolon_pos - value_start);
+ }
+ if (setParameter(key, value) != OK) {
+ return BAD_VALUE;
+ }
+ if (semicolon_pos == NULL) {
+ break; // Reaches the end
+ }
+ key_start = semicolon_pos + 1;
+ }
+ return OK;
+}
+
+// Trim both leading and trailing whitespace from the given string.
+//static
+void NuPlayer::RTPSource::TrimString(String8 *s) {
+ size_t num_bytes = s->bytes();
+ const char *data = s->string();
+
+ size_t leading_space = 0;
+ while (leading_space < num_bytes && isspace(data[leading_space])) {
+ ++leading_space;
+ }
+
+ size_t i = num_bytes;
+ while (i > leading_space && isspace(data[i - 1])) {
+ --i;
+ }
+
+ s->setTo(String8(&data[leading_space], i - leading_space));
+}
+
+} // namespace android