PlaylistFetcher.cpp revision f5b7c3b3c9a6da29f3bbd02e4031ad19bc7ad0f7
1/*
2 * Copyright (C) 2012 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 *      http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17//#define LOG_NDEBUG 0
18#define LOG_TAG "PlaylistFetcher"
19#include <utils/Log.h>
20
21#include "PlaylistFetcher.h"
22
23#include "LiveDataSource.h"
24#include "LiveSession.h"
25#include "M3UParser.h"
26
27#include "include/avc_utils.h"
28#include "include/HTTPBase.h"
29#include "include/ID3.h"
30#include "mpeg2ts/AnotherPacketSource.h"
31
32#include <media/IStreamSource.h>
33#include <media/stagefright/foundation/ABitReader.h>
34#include <media/stagefright/foundation/ABuffer.h>
35#include <media/stagefright/foundation/ADebug.h>
36#include <media/stagefright/foundation/hexdump.h>
37#include <media/stagefright/FileSource.h>
38#include <media/stagefright/MediaDefs.h>
39#include <media/stagefright/MetaData.h>
40#include <media/stagefright/Utils.h>
41
42#include <ctype.h>
43#include <inttypes.h>
44#include <openssl/aes.h>
45#include <openssl/md5.h>
46
47namespace android {
48
49// static
50const int64_t PlaylistFetcher::kMinBufferedDurationUs = 10000000ll;
51const int64_t PlaylistFetcher::kMaxMonitorDelayUs = 3000000ll;
52const int32_t PlaylistFetcher::kDownloadBlockSize = 2048;
53const int32_t PlaylistFetcher::kNumSkipFrames = 10;
54
55PlaylistFetcher::PlaylistFetcher(
56        const sp<AMessage> &notify,
57        const sp<LiveSession> &session,
58        const char *uri,
59        int32_t subtitleGeneration)
60    : mNotify(notify),
61      mStartTimeUsNotify(notify->dup()),
62      mSession(session),
63      mURI(uri),
64      mStreamTypeMask(0),
65      mStartTimeUs(-1ll),
66      mSegmentStartTimeUs(-1ll),
67      mDiscontinuitySeq(-1ll),
68      mStartTimeUsRelative(false),
69      mLastPlaylistFetchTimeUs(-1ll),
70      mSeqNumber(-1),
71      mNumRetries(0),
72      mStartup(true),
73      mAdaptive(false),
74      mPrepared(false),
75      mNextPTSTimeUs(-1ll),
76      mMonitorQueueGeneration(0),
77      mSubtitleGeneration(subtitleGeneration),
78      mRefreshState(INITIAL_MINIMUM_RELOAD_DELAY),
79      mFirstPTSValid(false),
80      mAbsoluteTimeAnchorUs(0ll),
81      mVideoBuffer(new AnotherPacketSource(NULL)) {
82    memset(mPlaylistHash, 0, sizeof(mPlaylistHash));
83    mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
84    mStartTimeUsNotify->setInt32("streamMask", 0);
85}
86
87PlaylistFetcher::~PlaylistFetcher() {
88}
89
90int64_t PlaylistFetcher::getSegmentStartTimeUs(int32_t seqNumber) const {
91    CHECK(mPlaylist != NULL);
92
93    int32_t firstSeqNumberInPlaylist;
94    if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
95                "media-sequence", &firstSeqNumberInPlaylist)) {
96        firstSeqNumberInPlaylist = 0;
97    }
98
99    int32_t lastSeqNumberInPlaylist =
100        firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
101
102    CHECK_GE(seqNumber, firstSeqNumberInPlaylist);
103    CHECK_LE(seqNumber, lastSeqNumberInPlaylist);
104
105    int64_t segmentStartUs = 0ll;
106    for (int32_t index = 0;
107            index < seqNumber - firstSeqNumberInPlaylist; ++index) {
108        sp<AMessage> itemMeta;
109        CHECK(mPlaylist->itemAt(
110                    index, NULL /* uri */, &itemMeta));
111
112        int64_t itemDurationUs;
113        CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
114
115        segmentStartUs += itemDurationUs;
116    }
117
118    return segmentStartUs;
119}
120
121int64_t PlaylistFetcher::delayUsToRefreshPlaylist() const {
122    int64_t nowUs = ALooper::GetNowUs();
123
124    if (mPlaylist == NULL || mLastPlaylistFetchTimeUs < 0ll) {
125        CHECK_EQ((int)mRefreshState, (int)INITIAL_MINIMUM_RELOAD_DELAY);
126        return 0ll;
127    }
128
129    if (mPlaylist->isComplete()) {
130        return (~0llu >> 1);
131    }
132
133    int32_t targetDurationSecs;
134    CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
135
136    int64_t targetDurationUs = targetDurationSecs * 1000000ll;
137
138    int64_t minPlaylistAgeUs;
139
140    switch (mRefreshState) {
141        case INITIAL_MINIMUM_RELOAD_DELAY:
142        {
143            size_t n = mPlaylist->size();
144            if (n > 0) {
145                sp<AMessage> itemMeta;
146                CHECK(mPlaylist->itemAt(n - 1, NULL /* uri */, &itemMeta));
147
148                int64_t itemDurationUs;
149                CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
150
151                minPlaylistAgeUs = itemDurationUs;
152                break;
153            }
154
155            // fall through
156        }
157
158        case FIRST_UNCHANGED_RELOAD_ATTEMPT:
159        {
160            minPlaylistAgeUs = targetDurationUs / 2;
161            break;
162        }
163
164        case SECOND_UNCHANGED_RELOAD_ATTEMPT:
165        {
166            minPlaylistAgeUs = (targetDurationUs * 3) / 2;
167            break;
168        }
169
170        case THIRD_UNCHANGED_RELOAD_ATTEMPT:
171        {
172            minPlaylistAgeUs = targetDurationUs * 3;
173            break;
174        }
175
176        default:
177            TRESPASS();
178            break;
179    }
180
181    int64_t delayUs = mLastPlaylistFetchTimeUs + minPlaylistAgeUs - nowUs;
182    return delayUs > 0ll ? delayUs : 0ll;
183}
184
185status_t PlaylistFetcher::decryptBuffer(
186        size_t playlistIndex, const sp<ABuffer> &buffer,
187        bool first) {
188    sp<AMessage> itemMeta;
189    bool found = false;
190    AString method;
191
192    for (ssize_t i = playlistIndex; i >= 0; --i) {
193        AString uri;
194        CHECK(mPlaylist->itemAt(i, &uri, &itemMeta));
195
196        if (itemMeta->findString("cipher-method", &method)) {
197            found = true;
198            break;
199        }
200    }
201
202    if (!found) {
203        method = "NONE";
204    }
205    buffer->meta()->setString("cipher-method", method.c_str());
206
207    if (method == "NONE") {
208        return OK;
209    } else if (!(method == "AES-128")) {
210        ALOGE("Unsupported cipher method '%s'", method.c_str());
211        return ERROR_UNSUPPORTED;
212    }
213
214    AString keyURI;
215    if (!itemMeta->findString("cipher-uri", &keyURI)) {
216        ALOGE("Missing key uri");
217        return ERROR_MALFORMED;
218    }
219
220    ssize_t index = mAESKeyForURI.indexOfKey(keyURI);
221
222    sp<ABuffer> key;
223    if (index >= 0) {
224        key = mAESKeyForURI.valueAt(index);
225    } else {
226        ssize_t err = mSession->fetchFile(keyURI.c_str(), &key);
227
228        if (err < 0) {
229            ALOGE("failed to fetch cipher key from '%s'.", keyURI.c_str());
230            return ERROR_IO;
231        } else if (key->size() != 16) {
232            ALOGE("key file '%s' wasn't 16 bytes in size.", keyURI.c_str());
233            return ERROR_MALFORMED;
234        }
235
236        mAESKeyForURI.add(keyURI, key);
237    }
238
239    AES_KEY aes_key;
240    if (AES_set_decrypt_key(key->data(), 128, &aes_key) != 0) {
241        ALOGE("failed to set AES decryption key.");
242        return UNKNOWN_ERROR;
243    }
244
245    size_t n = buffer->size();
246    if (!n) {
247        return OK;
248    }
249    CHECK(n % 16 == 0);
250
251    if (first) {
252        // If decrypting the first block in a file, read the iv from the manifest
253        // or derive the iv from the file's sequence number.
254
255        AString iv;
256        if (itemMeta->findString("cipher-iv", &iv)) {
257            if ((!iv.startsWith("0x") && !iv.startsWith("0X"))
258                    || iv.size() != 16 * 2 + 2) {
259                ALOGE("malformed cipher IV '%s'.", iv.c_str());
260                return ERROR_MALFORMED;
261            }
262
263            memset(mAESInitVec, 0, sizeof(mAESInitVec));
264            for (size_t i = 0; i < 16; ++i) {
265                char c1 = tolower(iv.c_str()[2 + 2 * i]);
266                char c2 = tolower(iv.c_str()[3 + 2 * i]);
267                if (!isxdigit(c1) || !isxdigit(c2)) {
268                    ALOGE("malformed cipher IV '%s'.", iv.c_str());
269                    return ERROR_MALFORMED;
270                }
271                uint8_t nibble1 = isdigit(c1) ? c1 - '0' : c1 - 'a' + 10;
272                uint8_t nibble2 = isdigit(c2) ? c2 - '0' : c2 - 'a' + 10;
273
274                mAESInitVec[i] = nibble1 << 4 | nibble2;
275            }
276        } else {
277            memset(mAESInitVec, 0, sizeof(mAESInitVec));
278            mAESInitVec[15] = mSeqNumber & 0xff;
279            mAESInitVec[14] = (mSeqNumber >> 8) & 0xff;
280            mAESInitVec[13] = (mSeqNumber >> 16) & 0xff;
281            mAESInitVec[12] = (mSeqNumber >> 24) & 0xff;
282        }
283    }
284
285    AES_cbc_encrypt(
286            buffer->data(), buffer->data(), buffer->size(),
287            &aes_key, mAESInitVec, AES_DECRYPT);
288
289    return OK;
290}
291
292status_t PlaylistFetcher::checkDecryptPadding(const sp<ABuffer> &buffer) {
293    status_t err;
294    AString method;
295    CHECK(buffer->meta()->findString("cipher-method", &method));
296    if (method == "NONE") {
297        return OK;
298    }
299
300    uint8_t padding = 0;
301    if (buffer->size() > 0) {
302        padding = buffer->data()[buffer->size() - 1];
303    }
304
305    if (padding > 16) {
306        return ERROR_MALFORMED;
307    }
308
309    for (size_t i = buffer->size() - padding; i < padding; i++) {
310        if (buffer->data()[i] != padding) {
311            return ERROR_MALFORMED;
312        }
313    }
314
315    buffer->setRange(buffer->offset(), buffer->size() - padding);
316    return OK;
317}
318
319void PlaylistFetcher::postMonitorQueue(int64_t delayUs, int64_t minDelayUs) {
320    int64_t maxDelayUs = delayUsToRefreshPlaylist();
321    if (maxDelayUs < minDelayUs) {
322        maxDelayUs = minDelayUs;
323    }
324    if (delayUs > maxDelayUs) {
325        ALOGV("Need to refresh playlist in %" PRId64 , maxDelayUs);
326        delayUs = maxDelayUs;
327    }
328    sp<AMessage> msg = new AMessage(kWhatMonitorQueue, id());
329    msg->setInt32("generation", mMonitorQueueGeneration);
330    msg->post(delayUs);
331}
332
333void PlaylistFetcher::cancelMonitorQueue() {
334    ++mMonitorQueueGeneration;
335}
336
337void PlaylistFetcher::startAsync(
338        const sp<AnotherPacketSource> &audioSource,
339        const sp<AnotherPacketSource> &videoSource,
340        const sp<AnotherPacketSource> &subtitleSource,
341        int64_t startTimeUs,
342        int64_t segmentStartTimeUs,
343        int32_t startDiscontinuitySeq,
344        bool adaptive) {
345    sp<AMessage> msg = new AMessage(kWhatStart, id());
346
347    uint32_t streamTypeMask = 0ul;
348
349    if (audioSource != NULL) {
350        msg->setPointer("audioSource", audioSource.get());
351        streamTypeMask |= LiveSession::STREAMTYPE_AUDIO;
352    }
353
354    if (videoSource != NULL) {
355        msg->setPointer("videoSource", videoSource.get());
356        streamTypeMask |= LiveSession::STREAMTYPE_VIDEO;
357    }
358
359    if (subtitleSource != NULL) {
360        msg->setPointer("subtitleSource", subtitleSource.get());
361        streamTypeMask |= LiveSession::STREAMTYPE_SUBTITLES;
362    }
363
364    msg->setInt32("streamTypeMask", streamTypeMask);
365    msg->setInt64("startTimeUs", startTimeUs);
366    msg->setInt64("segmentStartTimeUs", segmentStartTimeUs);
367    msg->setInt32("startDiscontinuitySeq", startDiscontinuitySeq);
368    msg->setInt32("adaptive", adaptive);
369    msg->post();
370}
371
372void PlaylistFetcher::pauseAsync() {
373    (new AMessage(kWhatPause, id()))->post();
374}
375
376void PlaylistFetcher::stopAsync(bool clear) {
377    sp<AMessage> msg = new AMessage(kWhatStop, id());
378    msg->setInt32("clear", clear);
379    msg->post();
380}
381
382void PlaylistFetcher::resumeUntilAsync(const sp<AMessage> &params) {
383    AMessage* msg = new AMessage(kWhatResumeUntil, id());
384    msg->setMessage("params", params);
385    msg->post();
386}
387
388void PlaylistFetcher::onMessageReceived(const sp<AMessage> &msg) {
389    switch (msg->what()) {
390        case kWhatStart:
391        {
392            status_t err = onStart(msg);
393
394            sp<AMessage> notify = mNotify->dup();
395            notify->setInt32("what", kWhatStarted);
396            notify->setInt32("err", err);
397            notify->post();
398            break;
399        }
400
401        case kWhatPause:
402        {
403            onPause();
404
405            sp<AMessage> notify = mNotify->dup();
406            notify->setInt32("what", kWhatPaused);
407            notify->post();
408            break;
409        }
410
411        case kWhatStop:
412        {
413            onStop(msg);
414
415            sp<AMessage> notify = mNotify->dup();
416            notify->setInt32("what", kWhatStopped);
417            notify->post();
418            break;
419        }
420
421        case kWhatMonitorQueue:
422        case kWhatDownloadNext:
423        {
424            int32_t generation;
425            CHECK(msg->findInt32("generation", &generation));
426
427            if (generation != mMonitorQueueGeneration) {
428                // Stale event
429                break;
430            }
431
432            if (msg->what() == kWhatMonitorQueue) {
433                onMonitorQueue();
434            } else {
435                onDownloadNext();
436            }
437            break;
438        }
439
440        case kWhatResumeUntil:
441        {
442            onResumeUntil(msg);
443            break;
444        }
445
446        default:
447            TRESPASS();
448    }
449}
450
451status_t PlaylistFetcher::onStart(const sp<AMessage> &msg) {
452    mPacketSources.clear();
453
454    uint32_t streamTypeMask;
455    CHECK(msg->findInt32("streamTypeMask", (int32_t *)&streamTypeMask));
456
457    int64_t startTimeUs;
458    int64_t segmentStartTimeUs;
459    int32_t startDiscontinuitySeq;
460    int32_t adaptive;
461    CHECK(msg->findInt64("startTimeUs", &startTimeUs));
462    CHECK(msg->findInt64("segmentStartTimeUs", &segmentStartTimeUs));
463    CHECK(msg->findInt32("startDiscontinuitySeq", &startDiscontinuitySeq));
464    CHECK(msg->findInt32("adaptive", &adaptive));
465
466    if (streamTypeMask & LiveSession::STREAMTYPE_AUDIO) {
467        void *ptr;
468        CHECK(msg->findPointer("audioSource", &ptr));
469
470        mPacketSources.add(
471                LiveSession::STREAMTYPE_AUDIO,
472                static_cast<AnotherPacketSource *>(ptr));
473    }
474
475    if (streamTypeMask & LiveSession::STREAMTYPE_VIDEO) {
476        void *ptr;
477        CHECK(msg->findPointer("videoSource", &ptr));
478
479        mPacketSources.add(
480                LiveSession::STREAMTYPE_VIDEO,
481                static_cast<AnotherPacketSource *>(ptr));
482    }
483
484    if (streamTypeMask & LiveSession::STREAMTYPE_SUBTITLES) {
485        void *ptr;
486        CHECK(msg->findPointer("subtitleSource", &ptr));
487
488        mPacketSources.add(
489                LiveSession::STREAMTYPE_SUBTITLES,
490                static_cast<AnotherPacketSource *>(ptr));
491    }
492
493    mStreamTypeMask = streamTypeMask;
494
495    mSegmentStartTimeUs = segmentStartTimeUs;
496    mDiscontinuitySeq = startDiscontinuitySeq;
497
498    if (startTimeUs >= 0) {
499        mStartTimeUs = startTimeUs;
500        mSeqNumber = -1;
501        mStartup = true;
502        mPrepared = false;
503        mAdaptive = adaptive;
504    }
505
506    postMonitorQueue();
507
508    return OK;
509}
510
511void PlaylistFetcher::onPause() {
512    cancelMonitorQueue();
513}
514
515void PlaylistFetcher::onStop(const sp<AMessage> &msg) {
516    cancelMonitorQueue();
517
518    int32_t clear;
519    CHECK(msg->findInt32("clear", &clear));
520    if (clear) {
521        for (size_t i = 0; i < mPacketSources.size(); i++) {
522            sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
523            packetSource->clear();
524        }
525    }
526
527    mPacketSources.clear();
528    mStreamTypeMask = 0;
529}
530
531// Resume until we have reached the boundary timestamps listed in `msg`; when
532// the remaining time is too short (within a resume threshold) stop immediately
533// instead.
534status_t PlaylistFetcher::onResumeUntil(const sp<AMessage> &msg) {
535    sp<AMessage> params;
536    CHECK(msg->findMessage("params", &params));
537
538    bool stop = false;
539    for (size_t i = 0; i < mPacketSources.size(); i++) {
540        sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
541
542        const char *stopKey;
543        int streamType = mPacketSources.keyAt(i);
544        switch (streamType) {
545        case LiveSession::STREAMTYPE_VIDEO:
546            stopKey = "timeUsVideo";
547            break;
548
549        case LiveSession::STREAMTYPE_AUDIO:
550            stopKey = "timeUsAudio";
551            break;
552
553        case LiveSession::STREAMTYPE_SUBTITLES:
554            stopKey = "timeUsSubtitle";
555            break;
556
557        default:
558            TRESPASS();
559        }
560
561        // Don't resume if we would stop within a resume threshold.
562        int32_t discontinuitySeq;
563        int64_t latestTimeUs = 0, stopTimeUs = 0;
564        sp<AMessage> latestMeta = packetSource->getLatestDequeuedMeta();
565        if (latestMeta != NULL
566                && latestMeta->findInt32("discontinuitySeq", &discontinuitySeq)
567                && discontinuitySeq == mDiscontinuitySeq
568                && latestMeta->findInt64("timeUs", &latestTimeUs)
569                && params->findInt64(stopKey, &stopTimeUs)
570                && stopTimeUs - latestTimeUs < resumeThreshold(latestMeta)) {
571            stop = true;
572        }
573    }
574
575    if (stop) {
576        for (size_t i = 0; i < mPacketSources.size(); i++) {
577            mPacketSources.valueAt(i)->queueAccessUnit(mSession->createFormatChangeBuffer());
578        }
579        stopAsync(/* clear = */ false);
580        return OK;
581    }
582
583    mStopParams = params;
584    postMonitorQueue();
585
586    return OK;
587}
588
589void PlaylistFetcher::notifyError(status_t err) {
590    sp<AMessage> notify = mNotify->dup();
591    notify->setInt32("what", kWhatError);
592    notify->setInt32("err", err);
593    notify->post();
594}
595
596void PlaylistFetcher::queueDiscontinuity(
597        ATSParser::DiscontinuityType type, const sp<AMessage> &extra) {
598    for (size_t i = 0; i < mPacketSources.size(); ++i) {
599        // do not discard buffer upon #EXT-X-DISCONTINUITY tag
600        // (seek will discard buffer by abandoning old fetchers)
601        mPacketSources.valueAt(i)->queueDiscontinuity(
602                type, extra, false /* discard */);
603    }
604}
605
606void PlaylistFetcher::onMonitorQueue() {
607    bool downloadMore = false;
608    refreshPlaylist();
609
610    int32_t targetDurationSecs;
611    int64_t targetDurationUs = kMinBufferedDurationUs;
612    if (mPlaylist != NULL) {
613        CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
614        targetDurationUs = targetDurationSecs * 1000000ll;
615    }
616
617    // buffer at least 3 times the target duration, or up to 10 seconds
618    int64_t durationToBufferUs = targetDurationUs * 3;
619    if (durationToBufferUs > kMinBufferedDurationUs)  {
620        durationToBufferUs = kMinBufferedDurationUs;
621    }
622
623    int64_t bufferedDurationUs = 0ll;
624    status_t finalResult = NOT_ENOUGH_DATA;
625    if (mStreamTypeMask == LiveSession::STREAMTYPE_SUBTITLES) {
626        sp<AnotherPacketSource> packetSource =
627            mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
628
629        bufferedDurationUs =
630                packetSource->getBufferedDurationUs(&finalResult);
631        finalResult = OK;
632    } else {
633        // Use max stream duration to prevent us from waiting on a non-existent stream;
634        // when we cannot make out from the manifest what streams are included in a playlist
635        // we might assume extra streams.
636        for (size_t i = 0; i < mPacketSources.size(); ++i) {
637            if ((mStreamTypeMask & mPacketSources.keyAt(i)) == 0) {
638                continue;
639            }
640
641            int64_t bufferedStreamDurationUs =
642                mPacketSources.valueAt(i)->getBufferedDurationUs(&finalResult);
643            ALOGV("buffered %" PRId64 " for stream %d",
644                    bufferedStreamDurationUs, mPacketSources.keyAt(i));
645            if (bufferedStreamDurationUs > bufferedDurationUs) {
646                bufferedDurationUs = bufferedStreamDurationUs;
647            }
648        }
649    }
650    downloadMore = (bufferedDurationUs < durationToBufferUs);
651
652    // signal start if buffered up at least the target size
653    if (!mPrepared && bufferedDurationUs > targetDurationUs && downloadMore) {
654        mPrepared = true;
655
656        ALOGV("prepared, buffered=%" PRId64 " > %" PRId64 "",
657                bufferedDurationUs, targetDurationUs);
658        sp<AMessage> msg = mNotify->dup();
659        msg->setInt32("what", kWhatTemporarilyDoneFetching);
660        msg->post();
661    }
662
663    if (finalResult == OK && downloadMore) {
664        ALOGV("monitoring, buffered=%" PRId64 " < %" PRId64 "",
665                bufferedDurationUs, durationToBufferUs);
666        // delay the next download slightly; hopefully this gives other concurrent fetchers
667        // a better chance to run.
668        // onDownloadNext();
669        sp<AMessage> msg = new AMessage(kWhatDownloadNext, id());
670        msg->setInt32("generation", mMonitorQueueGeneration);
671        msg->post(1000l);
672    } else {
673        // Nothing to do yet, try again in a second.
674
675        sp<AMessage> msg = mNotify->dup();
676        msg->setInt32("what", kWhatTemporarilyDoneFetching);
677        msg->post();
678
679        int64_t delayUs = mPrepared ? kMaxMonitorDelayUs : targetDurationUs / 2;
680        ALOGV("pausing for %" PRId64 ", buffered=%" PRId64 " > %" PRId64 "",
681                delayUs, bufferedDurationUs, durationToBufferUs);
682        // :TRICKY: need to enforce minimum delay because the delay to
683        // refresh the playlist will become 0
684        postMonitorQueue(delayUs, mPrepared ? targetDurationUs * 2 : 0);
685    }
686}
687
688status_t PlaylistFetcher::refreshPlaylist() {
689    if (delayUsToRefreshPlaylist() <= 0) {
690        bool unchanged;
691        sp<M3UParser> playlist = mSession->fetchPlaylist(
692                mURI.c_str(), mPlaylistHash, &unchanged);
693
694        if (playlist == NULL) {
695            if (unchanged) {
696                // We succeeded in fetching the playlist, but it was
697                // unchanged from the last time we tried.
698
699                if (mRefreshState != THIRD_UNCHANGED_RELOAD_ATTEMPT) {
700                    mRefreshState = (RefreshState)(mRefreshState + 1);
701                }
702            } else {
703                ALOGE("failed to load playlist at url '%s'", mURI.c_str());
704                notifyError(ERROR_IO);
705                return ERROR_IO;
706            }
707        } else {
708            mRefreshState = INITIAL_MINIMUM_RELOAD_DELAY;
709            mPlaylist = playlist;
710
711            if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
712                updateDuration();
713            }
714        }
715
716        mLastPlaylistFetchTimeUs = ALooper::GetNowUs();
717    }
718    return OK;
719}
720
721// static
722bool PlaylistFetcher::bufferStartsWithTsSyncByte(const sp<ABuffer>& buffer) {
723    return buffer->size() > 0 && buffer->data()[0] == 0x47;
724}
725
726void PlaylistFetcher::onDownloadNext() {
727    if (refreshPlaylist() != OK) {
728        return;
729    }
730
731    int32_t firstSeqNumberInPlaylist;
732    if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
733                "media-sequence", &firstSeqNumberInPlaylist)) {
734        firstSeqNumberInPlaylist = 0;
735    }
736
737    bool discontinuity = false;
738
739    const int32_t lastSeqNumberInPlaylist =
740        firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1;
741
742    if (mDiscontinuitySeq < 0) {
743        mDiscontinuitySeq = mPlaylist->getDiscontinuitySeq();
744    }
745
746    if (mSeqNumber < 0) {
747        CHECK_GE(mStartTimeUs, 0ll);
748
749        if (mSegmentStartTimeUs < 0) {
750            if (!mPlaylist->isComplete() && !mPlaylist->isEvent()) {
751                // If this is a live session, start 3 segments from the end on connect
752                mSeqNumber = lastSeqNumberInPlaylist - 3;
753                if (mSeqNumber < firstSeqNumberInPlaylist) {
754                    mSeqNumber = firstSeqNumberInPlaylist;
755                }
756            } else {
757                mSeqNumber = getSeqNumberForTime(mStartTimeUs);
758                mStartTimeUs -= getSegmentStartTimeUs(mSeqNumber);
759            }
760            mStartTimeUsRelative = true;
761            ALOGV("Initial sequence number for time %" PRId64 " is %d from (%d .. %d)",
762                    mStartTimeUs, mSeqNumber, firstSeqNumberInPlaylist,
763                    lastSeqNumberInPlaylist);
764        } else {
765            mSeqNumber = getSeqNumberForTime(mSegmentStartTimeUs);
766            if (mAdaptive) {
767                // avoid double fetch/decode
768                mSeqNumber += 1;
769            }
770            ssize_t minSeq = getSeqNumberForDiscontinuity(mDiscontinuitySeq);
771            if (mSeqNumber < minSeq) {
772                mSeqNumber = minSeq;
773            }
774
775            if (mSeqNumber < firstSeqNumberInPlaylist) {
776                mSeqNumber = firstSeqNumberInPlaylist;
777            }
778
779            if (mSeqNumber > lastSeqNumberInPlaylist) {
780                mSeqNumber = lastSeqNumberInPlaylist;
781            }
782            ALOGV("Initial sequence number for live event %d from (%d .. %d)",
783                    mSeqNumber, firstSeqNumberInPlaylist,
784                    lastSeqNumberInPlaylist);
785        }
786    }
787
788    if (mSeqNumber < firstSeqNumberInPlaylist
789            || mSeqNumber > lastSeqNumberInPlaylist) {
790        if (!mPlaylist->isComplete() && mNumRetries < kMaxNumRetries) {
791            ++mNumRetries;
792
793            if (mSeqNumber > lastSeqNumberInPlaylist) {
794                // refresh in increasing fraction (1/2, 1/3, ...) of the
795                // playlist's target duration or 3 seconds, whichever is less
796                int32_t targetDurationSecs;
797                CHECK(mPlaylist->meta()->findInt32(
798                        "target-duration", &targetDurationSecs));
799                int64_t delayUs = mPlaylist->size() * targetDurationSecs *
800                        1000000ll / (1 + mNumRetries);
801                if (delayUs > kMaxMonitorDelayUs) {
802                    delayUs = kMaxMonitorDelayUs;
803                }
804                ALOGV("sequence number high: %d from (%d .. %d), "
805                      "monitor in %" PRId64 " (retry=%d)",
806                        mSeqNumber, firstSeqNumberInPlaylist,
807                        lastSeqNumberInPlaylist, delayUs, mNumRetries);
808                postMonitorQueue(delayUs);
809                return;
810            }
811
812            // we've missed the boat, let's start from the lowest sequence
813            // number available and signal a discontinuity.
814
815            ALOGI("We've missed the boat, restarting playback."
816                  "  mStartup=%d, was  looking for %d in %d-%d",
817                    mStartup, mSeqNumber, firstSeqNumberInPlaylist,
818                    lastSeqNumberInPlaylist);
819            mSeqNumber = lastSeqNumberInPlaylist - 3;
820            if (mSeqNumber < firstSeqNumberInPlaylist) {
821                mSeqNumber = firstSeqNumberInPlaylist;
822            }
823            discontinuity = true;
824
825            // fall through
826        } else {
827            ALOGE("Cannot find sequence number %d in playlist "
828                 "(contains %d - %d)",
829                 mSeqNumber, firstSeqNumberInPlaylist,
830                  firstSeqNumberInPlaylist + (int32_t)mPlaylist->size() - 1);
831
832            notifyError(ERROR_END_OF_STREAM);
833            return;
834        }
835    }
836
837    mNumRetries = 0;
838
839    AString uri;
840    sp<AMessage> itemMeta;
841    CHECK(mPlaylist->itemAt(
842                mSeqNumber - firstSeqNumberInPlaylist,
843                &uri,
844                &itemMeta));
845
846    int32_t val;
847    if (itemMeta->findInt32("discontinuity", &val) && val != 0) {
848        mDiscontinuitySeq++;
849        discontinuity = true;
850    }
851
852    int64_t range_offset, range_length;
853    if (!itemMeta->findInt64("range-offset", &range_offset)
854            || !itemMeta->findInt64("range-length", &range_length)) {
855        range_offset = 0;
856        range_length = -1;
857    }
858
859    ALOGV("fetching segment %d from (%d .. %d)",
860          mSeqNumber, firstSeqNumberInPlaylist, lastSeqNumberInPlaylist);
861
862    ALOGV("fetching '%s'", uri.c_str());
863
864    sp<DataSource> source;
865    sp<ABuffer> buffer, tsBuffer;
866    // decrypt a junk buffer to prefetch key; since a session uses only one http connection,
867    // this avoids interleaved connections to the key and segment file.
868    {
869        sp<ABuffer> junk = new ABuffer(16);
870        junk->setRange(0, 16);
871        status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, junk,
872                true /* first */);
873        if (err != OK) {
874            notifyError(err);
875            return;
876        }
877    }
878
879    // block-wise download
880    bool startup = mStartup;
881    ssize_t bytesRead;
882    do {
883        bytesRead = mSession->fetchFile(
884                uri.c_str(), &buffer, range_offset, range_length, kDownloadBlockSize, &source);
885
886        if (bytesRead < 0) {
887            status_t err = bytesRead;
888            ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());
889            notifyError(err);
890            return;
891        }
892
893        CHECK(buffer != NULL);
894
895        size_t size = buffer->size();
896        // Set decryption range.
897        buffer->setRange(size - bytesRead, bytesRead);
898        status_t err = decryptBuffer(mSeqNumber - firstSeqNumberInPlaylist, buffer,
899                buffer->offset() == 0 /* first */);
900        // Unset decryption range.
901        buffer->setRange(0, size);
902
903        if (err != OK) {
904            ALOGE("decryptBuffer failed w/ error %d", err);
905
906            notifyError(err);
907            return;
908        }
909
910        if (startup || discontinuity) {
911            // Signal discontinuity.
912
913            if (mPlaylist->isComplete() || mPlaylist->isEvent()) {
914                // If this was a live event this made no sense since
915                // we don't have access to all the segment before the current
916                // one.
917                mNextPTSTimeUs = getSegmentStartTimeUs(mSeqNumber);
918            }
919
920            if (discontinuity) {
921                ALOGI("queueing discontinuity (explicit=%d)", discontinuity);
922
923                queueDiscontinuity(
924                        ATSParser::DISCONTINUITY_FORMATCHANGE,
925                        NULL /* extra */);
926
927                discontinuity = false;
928            }
929
930            startup = false;
931        }
932
933        err = OK;
934        if (bufferStartsWithTsSyncByte(buffer)) {
935            // Incremental extraction is only supported for MPEG2 transport streams.
936            if (tsBuffer == NULL) {
937                tsBuffer = new ABuffer(buffer->data(), buffer->capacity());
938                tsBuffer->setRange(0, 0);
939            } else if (tsBuffer->capacity() != buffer->capacity()) {
940                size_t tsOff = tsBuffer->offset(), tsSize = tsBuffer->size();
941                tsBuffer = new ABuffer(buffer->data(), buffer->capacity());
942                tsBuffer->setRange(tsOff, tsSize);
943            }
944            tsBuffer->setRange(tsBuffer->offset(), tsBuffer->size() + bytesRead);
945
946            err = extractAndQueueAccessUnitsFromTs(tsBuffer);
947        }
948
949        if (err == -EAGAIN) {
950            // starting sequence number too low/high
951            mTSParser.clear();
952            postMonitorQueue();
953            return;
954        } else if (err == ERROR_OUT_OF_RANGE) {
955            // reached stopping point
956            stopAsync(/* clear = */ false);
957            return;
958        } else if (err != OK) {
959            notifyError(err);
960            return;
961        }
962
963    } while (bytesRead != 0);
964
965    if (bufferStartsWithTsSyncByte(buffer)) {
966        // If we still don't see a stream after fetching a full ts segment mark it as
967        // nonexistent.
968        const size_t kNumTypes = ATSParser::NUM_SOURCE_TYPES;
969        ATSParser::SourceType srcTypes[kNumTypes] =
970                { ATSParser::VIDEO, ATSParser::AUDIO };
971        LiveSession::StreamType streamTypes[kNumTypes] =
972                { LiveSession::STREAMTYPE_VIDEO, LiveSession::STREAMTYPE_AUDIO };
973
974        for (size_t i = 0; i < kNumTypes; i++) {
975            ATSParser::SourceType srcType = srcTypes[i];
976            LiveSession::StreamType streamType = streamTypes[i];
977
978            sp<AnotherPacketSource> source =
979                static_cast<AnotherPacketSource *>(
980                    mTSParser->getSource(srcType).get());
981
982            if (source == NULL) {
983                ALOGW("MPEG2 Transport stream does not contain %s data.",
984                      srcType == ATSParser::VIDEO ? "video" : "audio");
985
986                mStreamTypeMask &= ~streamType;
987                mPacketSources.removeItem(streamType);
988            }
989        }
990
991    }
992
993    if (checkDecryptPadding(buffer) != OK) {
994        ALOGE("Incorrect padding bytes after decryption.");
995        notifyError(ERROR_MALFORMED);
996        return;
997    }
998
999    status_t err = OK;
1000    if (tsBuffer != NULL) {
1001        AString method;
1002        CHECK(buffer->meta()->findString("cipher-method", &method));
1003        if ((tsBuffer->size() > 0 && method == "NONE")
1004                || tsBuffer->size() > 16) {
1005            ALOGE("MPEG2 transport stream is not an even multiple of 188 "
1006                    "bytes in length.");
1007            notifyError(ERROR_MALFORMED);
1008            return;
1009        }
1010    }
1011
1012    // bulk extract non-ts files
1013    if (tsBuffer == NULL) {
1014        err = extractAndQueueAccessUnits(buffer, itemMeta);
1015        if (err == -EAGAIN) {
1016            // starting sequence number too low/high
1017            postMonitorQueue();
1018            return;
1019        } else if (err == ERROR_OUT_OF_RANGE) {
1020            // reached stopping point
1021            stopAsync(/* clear = */false);
1022            return;
1023        }
1024    }
1025
1026    if (err != OK) {
1027        notifyError(err);
1028        return;
1029    }
1030
1031    ++mSeqNumber;
1032
1033    postMonitorQueue();
1034}
1035
1036int32_t PlaylistFetcher::getSeqNumberWithAnchorTime(int64_t anchorTimeUs) const {
1037    int32_t firstSeqNumberInPlaylist, lastSeqNumberInPlaylist;
1038    if (mPlaylist->meta() == NULL
1039            || !mPlaylist->meta()->findInt32("media-sequence", &firstSeqNumberInPlaylist)) {
1040        firstSeqNumberInPlaylist = 0;
1041    }
1042    lastSeqNumberInPlaylist = firstSeqNumberInPlaylist + mPlaylist->size() - 1;
1043
1044    int32_t index = mSeqNumber - firstSeqNumberInPlaylist - 1;
1045    while (index >= 0 && anchorTimeUs > mStartTimeUs) {
1046        sp<AMessage> itemMeta;
1047        CHECK(mPlaylist->itemAt(index, NULL /* uri */, &itemMeta));
1048
1049        int64_t itemDurationUs;
1050        CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
1051
1052        anchorTimeUs -= itemDurationUs;
1053        --index;
1054    }
1055
1056    int32_t newSeqNumber = firstSeqNumberInPlaylist + index + 1;
1057    if (newSeqNumber <= lastSeqNumberInPlaylist) {
1058        return newSeqNumber;
1059    } else {
1060        return lastSeqNumberInPlaylist;
1061    }
1062}
1063
1064int32_t PlaylistFetcher::getSeqNumberForDiscontinuity(size_t discontinuitySeq) const {
1065    int32_t firstSeqNumberInPlaylist;
1066    if (mPlaylist->meta() == NULL
1067            || !mPlaylist->meta()->findInt32("media-sequence", &firstSeqNumberInPlaylist)) {
1068        firstSeqNumberInPlaylist = 0;
1069    }
1070
1071    size_t curDiscontinuitySeq = mPlaylist->getDiscontinuitySeq();
1072    if (discontinuitySeq < curDiscontinuitySeq) {
1073        return firstSeqNumberInPlaylist <= 0 ? 0 : (firstSeqNumberInPlaylist - 1);
1074    }
1075
1076    size_t index = 0;
1077    while (index < mPlaylist->size()) {
1078        sp<AMessage> itemMeta;
1079        CHECK(mPlaylist->itemAt( index, NULL /* uri */, &itemMeta));
1080
1081        int64_t discontinuity;
1082        if (itemMeta->findInt64("discontinuity", &discontinuity)) {
1083            curDiscontinuitySeq++;
1084        }
1085
1086        if (curDiscontinuitySeq == discontinuitySeq) {
1087            return firstSeqNumberInPlaylist + index;
1088        }
1089
1090        ++index;
1091    }
1092
1093    return firstSeqNumberInPlaylist + mPlaylist->size();
1094}
1095
1096int32_t PlaylistFetcher::getSeqNumberForTime(int64_t timeUs) const {
1097    int32_t firstSeqNumberInPlaylist;
1098    if (mPlaylist->meta() == NULL || !mPlaylist->meta()->findInt32(
1099                "media-sequence", &firstSeqNumberInPlaylist)) {
1100        firstSeqNumberInPlaylist = 0;
1101    }
1102
1103    size_t index = 0;
1104    int64_t segmentStartUs = 0;
1105    while (index < mPlaylist->size()) {
1106        sp<AMessage> itemMeta;
1107        CHECK(mPlaylist->itemAt(
1108                    index, NULL /* uri */, &itemMeta));
1109
1110        int64_t itemDurationUs;
1111        CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
1112
1113        if (timeUs < segmentStartUs + itemDurationUs) {
1114            break;
1115        }
1116
1117        segmentStartUs += itemDurationUs;
1118        ++index;
1119    }
1120
1121    if (index >= mPlaylist->size()) {
1122        index = mPlaylist->size() - 1;
1123    }
1124
1125    return firstSeqNumberInPlaylist + index;
1126}
1127
1128const sp<ABuffer> &PlaylistFetcher::setAccessUnitProperties(
1129        const sp<ABuffer> &accessUnit, const sp<AnotherPacketSource> &source, bool discard) {
1130    sp<MetaData> format = source->getFormat();
1131    if (format != NULL) {
1132        // for simplicity, store a reference to the format in each unit
1133        accessUnit->meta()->setObject("format", format);
1134    }
1135
1136    if (discard) {
1137        accessUnit->meta()->setInt32("discard", discard);
1138    }
1139
1140    accessUnit->meta()->setInt32("discontinuitySeq", mDiscontinuitySeq);
1141    accessUnit->meta()->setInt64("segmentStartTimeUs", getSegmentStartTimeUs(mSeqNumber));
1142    return accessUnit;
1143}
1144
1145status_t PlaylistFetcher::extractAndQueueAccessUnitsFromTs(const sp<ABuffer> &buffer) {
1146    if (mTSParser == NULL) {
1147        // Use TS_TIMESTAMPS_ARE_ABSOLUTE so pts carry over between fetchers.
1148        mTSParser = new ATSParser(ATSParser::TS_TIMESTAMPS_ARE_ABSOLUTE);
1149    }
1150
1151    if (mNextPTSTimeUs >= 0ll) {
1152        sp<AMessage> extra = new AMessage;
1153        // Since we are using absolute timestamps, signal an offset of 0 to prevent
1154        // ATSParser from skewing the timestamps of access units.
1155        extra->setInt64(IStreamListener::kKeyMediaTimeUs, 0);
1156
1157        mTSParser->signalDiscontinuity(
1158                ATSParser::DISCONTINUITY_SEEK, extra);
1159
1160        mAbsoluteTimeAnchorUs = mNextPTSTimeUs;
1161        mNextPTSTimeUs = -1ll;
1162        mFirstPTSValid = false;
1163    }
1164
1165    size_t offset = 0;
1166    while (offset + 188 <= buffer->size()) {
1167        status_t err = mTSParser->feedTSPacket(buffer->data() + offset, 188);
1168
1169        if (err != OK) {
1170            return err;
1171        }
1172
1173        offset += 188;
1174    }
1175    // setRange to indicate consumed bytes.
1176    buffer->setRange(buffer->offset() + offset, buffer->size() - offset);
1177
1178    status_t err = OK;
1179    for (size_t i = mPacketSources.size(); i-- > 0;) {
1180        sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
1181
1182        const char *key;
1183        ATSParser::SourceType type;
1184        const LiveSession::StreamType stream = mPacketSources.keyAt(i);
1185        switch (stream) {
1186            case LiveSession::STREAMTYPE_VIDEO:
1187                type = ATSParser::VIDEO;
1188                key = "timeUsVideo";
1189                break;
1190
1191            case LiveSession::STREAMTYPE_AUDIO:
1192                type = ATSParser::AUDIO;
1193                key = "timeUsAudio";
1194                break;
1195
1196            case LiveSession::STREAMTYPE_SUBTITLES:
1197            {
1198                ALOGE("MPEG2 Transport streams do not contain subtitles.");
1199                return ERROR_MALFORMED;
1200                break;
1201            }
1202
1203            default:
1204                TRESPASS();
1205        }
1206
1207        sp<AnotherPacketSource> source =
1208            static_cast<AnotherPacketSource *>(
1209                    mTSParser->getSource(type).get());
1210
1211        if (source == NULL) {
1212            continue;
1213        }
1214
1215        int64_t timeUs;
1216        sp<ABuffer> accessUnit;
1217        status_t finalResult;
1218        while (source->hasBufferAvailable(&finalResult)
1219                && source->dequeueAccessUnit(&accessUnit) == OK) {
1220
1221            CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
1222
1223            if (mStartup) {
1224                if (!mFirstPTSValid) {
1225                    mFirstTimeUs = timeUs;
1226                    mFirstPTSValid = true;
1227                }
1228                if (mStartTimeUsRelative) {
1229                    timeUs -= mFirstTimeUs;
1230                    if (timeUs < 0) {
1231                        timeUs = 0;
1232                    }
1233                }
1234
1235                if (timeUs < mStartTimeUs) {
1236                    // buffer up to the closest preceding IDR frame
1237                    ALOGV("timeUs %" PRId64 " us < mStartTimeUs %" PRId64 " us",
1238                            timeUs, mStartTimeUs);
1239                    const char *mime;
1240                    sp<MetaData> format  = source->getFormat();
1241                    bool isAvc = false;
1242                    if (format != NULL && format->findCString(kKeyMIMEType, &mime)
1243                            && !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC)) {
1244                        isAvc = true;
1245                    }
1246                    if (isAvc && IsIDR(accessUnit)) {
1247                        mVideoBuffer->clear();
1248                    }
1249                    if (isAvc) {
1250                        mVideoBuffer->queueAccessUnit(accessUnit);
1251                    }
1252
1253                    continue;
1254                }
1255            }
1256
1257            CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
1258            if (mStartTimeUsNotify != NULL && timeUs > mStartTimeUs) {
1259
1260                int32_t targetDurationSecs;
1261                CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
1262                int64_t targetDurationUs = targetDurationSecs * 1000000ll;
1263                // mStartup
1264                //   mStartup is true until we have queued a packet for all the streams
1265                //   we are fetching. We queue packets whose timestamps are greater than
1266                //   mStartTimeUs.
1267                // mSegmentStartTimeUs >= 0
1268                //   mSegmentStartTimeUs is non-negative when adapting or switching tracks
1269                // timeUs - mStartTimeUs > targetDurationUs:
1270                //   This and the 2 above conditions should only happen when adapting in a live
1271                //   stream; the old fetcher has already fetched to mStartTimeUs; the new fetcher
1272                //   would start fetching after timeUs, which should be greater than mStartTimeUs;
1273                //   the old fetcher would then continue fetching data until timeUs. We don't want
1274                //   timeUs to be too far ahead of mStartTimeUs because we want the old fetcher to
1275                //   stop as early as possible. The definition of being "too far ahead" is
1276                //   arbitrary; here we use targetDurationUs as threshold.
1277                if (mStartup && mSegmentStartTimeUs >= 0
1278                        && timeUs - mStartTimeUs > targetDurationUs) {
1279                    // we just guessed a starting timestamp that is too high when adapting in a
1280                    // live stream; re-adjust based on the actual timestamp extracted from the
1281                    // media segment; if we didn't move backward after the re-adjustment
1282                    // (newSeqNumber), start at least 1 segment prior.
1283                    int32_t newSeqNumber = getSeqNumberWithAnchorTime(timeUs);
1284                    if (newSeqNumber >= mSeqNumber) {
1285                        --mSeqNumber;
1286                    } else {
1287                        mSeqNumber = newSeqNumber;
1288                    }
1289                    mStartTimeUsNotify = mNotify->dup();
1290                    mStartTimeUsNotify->setInt32("what", kWhatStartedAt);
1291                    return -EAGAIN;
1292                }
1293
1294                int32_t seq;
1295                if (!mStartTimeUsNotify->findInt32("discontinuitySeq", &seq)) {
1296                    mStartTimeUsNotify->setInt32("discontinuitySeq", mDiscontinuitySeq);
1297                }
1298                int64_t startTimeUs;
1299                if (!mStartTimeUsNotify->findInt64(key, &startTimeUs)) {
1300                    mStartTimeUsNotify->setInt64(key, timeUs);
1301
1302                    uint32_t streamMask = 0;
1303                    mStartTimeUsNotify->findInt32("streamMask", (int32_t *) &streamMask);
1304                    streamMask |= mPacketSources.keyAt(i);
1305                    mStartTimeUsNotify->setInt32("streamMask", streamMask);
1306
1307                    if (streamMask == mStreamTypeMask) {
1308                        mStartup = false;
1309                        mStartTimeUsNotify->post();
1310                        mStartTimeUsNotify.clear();
1311                    }
1312                }
1313            }
1314
1315            if (mStopParams != NULL) {
1316                // Queue discontinuity in original stream.
1317                int32_t discontinuitySeq;
1318                int64_t stopTimeUs;
1319                if (!mStopParams->findInt32("discontinuitySeq", &discontinuitySeq)
1320                        || discontinuitySeq > mDiscontinuitySeq
1321                        || !mStopParams->findInt64(key, &stopTimeUs)
1322                        || (discontinuitySeq == mDiscontinuitySeq
1323                                && timeUs >= stopTimeUs)) {
1324                    packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
1325                    mStreamTypeMask &= ~stream;
1326                    mPacketSources.removeItemsAt(i);
1327                    break;
1328                }
1329            }
1330
1331            // Note that we do NOT dequeue any discontinuities except for format change.
1332            if (stream == LiveSession::STREAMTYPE_VIDEO) {
1333                const bool discard = true;
1334                status_t status;
1335                while (mVideoBuffer->hasBufferAvailable(&status)) {
1336                    sp<ABuffer> videoBuffer;
1337                    mVideoBuffer->dequeueAccessUnit(&videoBuffer);
1338                    setAccessUnitProperties(videoBuffer, source, discard);
1339                    packetSource->queueAccessUnit(videoBuffer);
1340                }
1341            }
1342
1343            setAccessUnitProperties(accessUnit, source);
1344            packetSource->queueAccessUnit(accessUnit);
1345        }
1346
1347        if (err != OK) {
1348            break;
1349        }
1350    }
1351
1352    if (err != OK) {
1353        for (size_t i = mPacketSources.size(); i-- > 0;) {
1354            sp<AnotherPacketSource> packetSource = mPacketSources.valueAt(i);
1355            packetSource->clear();
1356        }
1357        return err;
1358    }
1359
1360    if (!mStreamTypeMask) {
1361        // Signal gap is filled between original and new stream.
1362        ALOGV("ERROR OUT OF RANGE");
1363        return ERROR_OUT_OF_RANGE;
1364    }
1365
1366    return OK;
1367}
1368
1369/* static */
1370bool PlaylistFetcher::bufferStartsWithWebVTTMagicSequence(
1371        const sp<ABuffer> &buffer) {
1372    size_t pos = 0;
1373
1374    // skip possible BOM
1375    if (buffer->size() >= pos + 3 &&
1376            !memcmp("\xef\xbb\xbf", buffer->data() + pos, 3)) {
1377        pos += 3;
1378    }
1379
1380    // accept WEBVTT followed by SPACE, TAB or (CR) LF
1381    if (buffer->size() < pos + 6 ||
1382            memcmp("WEBVTT", buffer->data() + pos, 6)) {
1383        return false;
1384    }
1385    pos += 6;
1386
1387    if (buffer->size() == pos) {
1388        return true;
1389    }
1390
1391    uint8_t sep = buffer->data()[pos];
1392    return sep == ' ' || sep == '\t' || sep == '\n' || sep == '\r';
1393}
1394
1395status_t PlaylistFetcher::extractAndQueueAccessUnits(
1396        const sp<ABuffer> &buffer, const sp<AMessage> &itemMeta) {
1397    if (bufferStartsWithWebVTTMagicSequence(buffer)) {
1398        if (mStreamTypeMask != LiveSession::STREAMTYPE_SUBTITLES) {
1399            ALOGE("This stream only contains subtitles.");
1400            return ERROR_MALFORMED;
1401        }
1402
1403        const sp<AnotherPacketSource> packetSource =
1404            mPacketSources.valueFor(LiveSession::STREAMTYPE_SUBTITLES);
1405
1406        int64_t durationUs;
1407        CHECK(itemMeta->findInt64("durationUs", &durationUs));
1408        buffer->meta()->setInt64("timeUs", getSegmentStartTimeUs(mSeqNumber));
1409        buffer->meta()->setInt64("durationUs", durationUs);
1410        buffer->meta()->setInt64("segmentStartTimeUs", getSegmentStartTimeUs(mSeqNumber));
1411        buffer->meta()->setInt32("discontinuitySeq", mDiscontinuitySeq);
1412        buffer->meta()->setInt32("subtitleGeneration", mSubtitleGeneration);
1413
1414        packetSource->queueAccessUnit(buffer);
1415        return OK;
1416    }
1417
1418    if (mNextPTSTimeUs >= 0ll) {
1419        mFirstPTSValid = false;
1420        mAbsoluteTimeAnchorUs = mNextPTSTimeUs;
1421        mNextPTSTimeUs = -1ll;
1422    }
1423
1424    // This better be an ISO 13818-7 (AAC) or ISO 13818-1 (MPEG) audio
1425    // stream prefixed by an ID3 tag.
1426
1427    bool firstID3Tag = true;
1428    uint64_t PTS = 0;
1429
1430    for (;;) {
1431        // Make sure to skip all ID3 tags preceding the audio data.
1432        // At least one must be present to provide the PTS timestamp.
1433
1434        ID3 id3(buffer->data(), buffer->size(), true /* ignoreV1 */);
1435        if (!id3.isValid()) {
1436            if (firstID3Tag) {
1437                ALOGE("Unable to parse ID3 tag.");
1438                return ERROR_MALFORMED;
1439            } else {
1440                break;
1441            }
1442        }
1443
1444        if (firstID3Tag) {
1445            bool found = false;
1446
1447            ID3::Iterator it(id3, "PRIV");
1448            while (!it.done()) {
1449                size_t length;
1450                const uint8_t *data = it.getData(&length);
1451
1452                static const char *kMatchName =
1453                    "com.apple.streaming.transportStreamTimestamp";
1454                static const size_t kMatchNameLen = strlen(kMatchName);
1455
1456                if (length == kMatchNameLen + 1 + 8
1457                        && !strncmp((const char *)data, kMatchName, kMatchNameLen)) {
1458                    found = true;
1459                    PTS = U64_AT(&data[kMatchNameLen + 1]);
1460                }
1461
1462                it.next();
1463            }
1464
1465            if (!found) {
1466                ALOGE("Unable to extract transportStreamTimestamp from ID3 tag.");
1467                return ERROR_MALFORMED;
1468            }
1469        }
1470
1471        // skip the ID3 tag
1472        buffer->setRange(
1473                buffer->offset() + id3.rawSize(), buffer->size() - id3.rawSize());
1474
1475        firstID3Tag = false;
1476    }
1477
1478    if (mStreamTypeMask != LiveSession::STREAMTYPE_AUDIO) {
1479        ALOGW("This stream only contains audio data!");
1480
1481        mStreamTypeMask &= LiveSession::STREAMTYPE_AUDIO;
1482
1483        if (mStreamTypeMask == 0) {
1484            return OK;
1485        }
1486    }
1487
1488    sp<AnotherPacketSource> packetSource =
1489        mPacketSources.valueFor(LiveSession::STREAMTYPE_AUDIO);
1490
1491    if (packetSource->getFormat() == NULL && buffer->size() >= 7) {
1492        ABitReader bits(buffer->data(), buffer->size());
1493
1494        // adts_fixed_header
1495
1496        CHECK_EQ(bits.getBits(12), 0xfffu);
1497        bits.skipBits(3);  // ID, layer
1498        bool protection_absent = bits.getBits(1) != 0;
1499
1500        unsigned profile = bits.getBits(2);
1501        CHECK_NE(profile, 3u);
1502        unsigned sampling_freq_index = bits.getBits(4);
1503        bits.getBits(1);  // private_bit
1504        unsigned channel_configuration = bits.getBits(3);
1505        CHECK_NE(channel_configuration, 0u);
1506        bits.skipBits(2);  // original_copy, home
1507
1508        sp<MetaData> meta = MakeAACCodecSpecificData(
1509                profile, sampling_freq_index, channel_configuration);
1510
1511        meta->setInt32(kKeyIsADTS, true);
1512
1513        packetSource->setFormat(meta);
1514    }
1515
1516    int64_t numSamples = 0ll;
1517    int32_t sampleRate;
1518    CHECK(packetSource->getFormat()->findInt32(kKeySampleRate, &sampleRate));
1519
1520    int64_t timeUs = (PTS * 100ll) / 9ll;
1521    if (!mFirstPTSValid) {
1522        mFirstPTSValid = true;
1523        mFirstTimeUs = timeUs;
1524    }
1525
1526    size_t offset = 0;
1527    while (offset < buffer->size()) {
1528        const uint8_t *adtsHeader = buffer->data() + offset;
1529        CHECK_LT(offset + 5, buffer->size());
1530
1531        unsigned aac_frame_length =
1532            ((adtsHeader[3] & 3) << 11)
1533            | (adtsHeader[4] << 3)
1534            | (adtsHeader[5] >> 5);
1535
1536        if (aac_frame_length == 0) {
1537            const uint8_t *id3Header = adtsHeader;
1538            if (!memcmp(id3Header, "ID3", 3)) {
1539                ID3 id3(id3Header, buffer->size() - offset, true);
1540                if (id3.isValid()) {
1541                    offset += id3.rawSize();
1542                    continue;
1543                };
1544            }
1545            return ERROR_MALFORMED;
1546        }
1547
1548        CHECK_LE(offset + aac_frame_length, buffer->size());
1549
1550        int64_t unitTimeUs = timeUs + numSamples * 1000000ll / sampleRate;
1551        offset += aac_frame_length;
1552
1553        // Each AAC frame encodes 1024 samples.
1554        numSamples += 1024;
1555
1556        if (mStartup) {
1557            int64_t startTimeUs = unitTimeUs;
1558            if (mStartTimeUsRelative) {
1559                startTimeUs -= mFirstTimeUs;
1560                if (startTimeUs  < 0) {
1561                    startTimeUs = 0;
1562                }
1563            }
1564            if (startTimeUs < mStartTimeUs) {
1565                continue;
1566            }
1567
1568            if (mStartTimeUsNotify != NULL) {
1569                int32_t targetDurationSecs;
1570                CHECK(mPlaylist->meta()->findInt32("target-duration", &targetDurationSecs));
1571                int64_t targetDurationUs = targetDurationSecs * 1000000ll;
1572
1573                // Duplicated logic from how we handle .ts playlists.
1574                if (mStartup && mSegmentStartTimeUs >= 0
1575                        && timeUs - mStartTimeUs > targetDurationUs) {
1576                    int32_t newSeqNumber = getSeqNumberWithAnchorTime(timeUs);
1577                    if (newSeqNumber >= mSeqNumber) {
1578                        --mSeqNumber;
1579                    } else {
1580                        mSeqNumber = newSeqNumber;
1581                    }
1582                    return -EAGAIN;
1583                }
1584
1585                mStartTimeUsNotify->setInt64("timeUsAudio", timeUs);
1586                mStartTimeUsNotify->setInt32("discontinuitySeq", mDiscontinuitySeq);
1587                mStartTimeUsNotify->setInt32("streamMask", LiveSession::STREAMTYPE_AUDIO);
1588                mStartTimeUsNotify->post();
1589                mStartTimeUsNotify.clear();
1590                mStartup = false;
1591            }
1592        }
1593
1594        if (mStopParams != NULL) {
1595            // Queue discontinuity in original stream.
1596            int32_t discontinuitySeq;
1597            int64_t stopTimeUs;
1598            if (!mStopParams->findInt32("discontinuitySeq", &discontinuitySeq)
1599                    || discontinuitySeq > mDiscontinuitySeq
1600                    || !mStopParams->findInt64("timeUsAudio", &stopTimeUs)
1601                    || (discontinuitySeq == mDiscontinuitySeq && unitTimeUs >= stopTimeUs)) {
1602                packetSource->queueAccessUnit(mSession->createFormatChangeBuffer());
1603                mStreamTypeMask = 0;
1604                mPacketSources.clear();
1605                return ERROR_OUT_OF_RANGE;
1606            }
1607        }
1608
1609        sp<ABuffer> unit = new ABuffer(aac_frame_length);
1610        memcpy(unit->data(), adtsHeader, aac_frame_length);
1611
1612        unit->meta()->setInt64("timeUs", unitTimeUs);
1613        setAccessUnitProperties(unit, packetSource);
1614        packetSource->queueAccessUnit(unit);
1615    }
1616
1617    return OK;
1618}
1619
1620void PlaylistFetcher::updateDuration() {
1621    int64_t durationUs = 0ll;
1622    for (size_t index = 0; index < mPlaylist->size(); ++index) {
1623        sp<AMessage> itemMeta;
1624        CHECK(mPlaylist->itemAt(
1625                    index, NULL /* uri */, &itemMeta));
1626
1627        int64_t itemDurationUs;
1628        CHECK(itemMeta->findInt64("durationUs", &itemDurationUs));
1629
1630        durationUs += itemDurationUs;
1631    }
1632
1633    sp<AMessage> msg = mNotify->dup();
1634    msg->setInt32("what", kWhatDurationUpdate);
1635    msg->setInt64("durationUs", durationUs);
1636    msg->post();
1637}
1638
1639int64_t PlaylistFetcher::resumeThreshold(const sp<AMessage> &msg) {
1640    int64_t durationUs, threshold;
1641    if (msg->findInt64("durationUs", &durationUs)) {
1642        return kNumSkipFrames * durationUs;
1643    }
1644
1645    sp<RefBase> obj;
1646    msg->findObject("format", &obj);
1647    MetaData *format = static_cast<MetaData *>(obj.get());
1648
1649    const char *mime;
1650    CHECK(format->findCString(kKeyMIMEType, &mime));
1651    bool audio = !strncasecmp(mime, "audio/", 6);
1652    if (audio) {
1653        // Assumes 1000 samples per frame.
1654        int32_t sampleRate;
1655        CHECK(format->findInt32(kKeySampleRate, &sampleRate));
1656        return kNumSkipFrames  /* frames */ * 1000 /* samples */
1657                * (1000000 / sampleRate) /* sample duration (us) */;
1658    } else {
1659        int32_t frameRate;
1660        if (format->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) {
1661            return kNumSkipFrames * (1000000 / frameRate);
1662        }
1663    }
1664
1665    return 500000ll;
1666}
1667
1668}  // namespace android
1669