| 1 | /********** |
| 2 | This library is free software; you can redistribute it and/or modify it under |
| 3 | the terms of the GNU Lesser General Public License as published by the |
| 4 | Free Software Foundation; either version 3 of the License, or (at your |
| 5 | option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.) |
| 6 | |
| 7 | This library is distributed in the hope that it will be useful, but WITHOUT |
| 8 | ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS |
| 9 | FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for |
| 10 | more details. |
| 11 | |
| 12 | You should have received a copy of the GNU Lesser General Public License |
| 13 | along with this library; if not, write to the Free Software Foundation, Inc., |
| 14 | 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| 15 | **********/ |
| 16 | // "liveMedia" |
| 17 | // Copyright (c) 1996-2020 Live Networks, Inc. All rights reserved. |
| 18 | // A filter that breaks up an AC3 audio elementary stream into frames |
| 19 | // Implementation |
| 20 | |
| 21 | #include "AC3AudioStreamFramer.hh" |
| 22 | #include "StreamParser.hh" |
| 23 | #include <GroupsockHelper.hh> |
| 24 | |
| 25 | ////////// AC3AudioStreamParser definition ////////// |
| 26 | |
| 27 | class AC3FrameParams { |
| 28 | public: |
| 29 | AC3FrameParams() : samplingFreq(0) {} |
| 30 | // 8-byte header at the start of each frame: |
| 31 | // u_int32_t hdr0, hdr1; |
| 32 | unsigned hdr0, hdr1; |
| 33 | |
| 34 | // parameters derived from the headers |
| 35 | unsigned kbps, samplingFreq, frameSize; |
| 36 | |
| 37 | void setParamsFromHeader(); |
| 38 | }; |
| 39 | |
| 40 | class AC3AudioStreamParser: public StreamParser { |
| 41 | public: |
| 42 | AC3AudioStreamParser(AC3AudioStreamFramer* usingSource, |
| 43 | FramedSource* inputSource); |
| 44 | virtual ~AC3AudioStreamParser(); |
| 45 | |
| 46 | public: |
| 47 | void testStreamCode(unsigned char ourStreamCode, |
| 48 | unsigned char* ptr, unsigned size); |
| 49 | unsigned parseFrame(unsigned& numTruncatedBytes); |
| 50 | // returns the size of the frame that was acquired, or 0 if none was |
| 51 | |
| 52 | void registerReadInterest(unsigned char* to, unsigned maxSize); |
| 53 | |
| 54 | AC3FrameParams const& currentFrame() const { return fCurrentFrame; } |
| 55 | |
| 56 | Boolean haveParsedAFrame() const { return fHaveParsedAFrame; } |
| 57 | void readAndSaveAFrame(); |
| 58 | |
| 59 | private: |
| 60 | static void afterGettingSavedFrame(void* clientData, unsigned frameSize, |
| 61 | unsigned numTruncatedBytes, |
| 62 | struct timeval presentationTime, |
| 63 | unsigned durationInMicroseconds); |
| 64 | void afterGettingSavedFrame1(unsigned frameSize); |
| 65 | static void onSavedFrameClosure(void* clientData); |
| 66 | void onSavedFrameClosure1(); |
| 67 | |
| 68 | private: |
| 69 | AC3AudioStreamFramer* fUsingSource; |
| 70 | unsigned char* fTo; |
| 71 | unsigned fMaxSize; |
| 72 | |
| 73 | Boolean fHaveParsedAFrame; |
| 74 | unsigned char* fSavedFrame; |
| 75 | unsigned fSavedFrameSize; |
| 76 | char fSavedFrameFlag; |
| 77 | |
| 78 | // Parameters of the most recently read frame: |
| 79 | AC3FrameParams fCurrentFrame; |
| 80 | }; |
| 81 | |
| 82 | |
| 83 | ////////// AC3AudioStreamFramer implementation ////////// |
| 84 | |
| 85 | AC3AudioStreamFramer::AC3AudioStreamFramer(UsageEnvironment& env, |
| 86 | FramedSource* inputSource, |
| 87 | unsigned char streamCode) |
| 88 | : FramedFilter(env, inputSource), fOurStreamCode(streamCode) { |
| 89 | // Use the current wallclock time as the initial 'presentation time': |
| 90 | gettimeofday(&fNextFramePresentationTime, NULL); |
| 91 | |
| 92 | fParser = new AC3AudioStreamParser(this, inputSource); |
| 93 | } |
| 94 | |
| 95 | AC3AudioStreamFramer::~AC3AudioStreamFramer() { |
| 96 | delete fParser; |
| 97 | } |
| 98 | |
| 99 | AC3AudioStreamFramer* |
| 100 | AC3AudioStreamFramer::createNew(UsageEnvironment& env, |
| 101 | FramedSource* inputSource, |
| 102 | unsigned char streamCode) { |
| 103 | // Need to add source type checking here??? ##### |
| 104 | return new AC3AudioStreamFramer(env, inputSource, streamCode); |
| 105 | } |
| 106 | |
| 107 | unsigned AC3AudioStreamFramer::samplingRate() { |
| 108 | if (!fParser->haveParsedAFrame()) { |
| 109 | // Because we haven't yet parsed a frame, we don't yet know the input |
| 110 | // stream's sampling rate. So, we first need to read a frame |
| 111 | // (into a special buffer that we keep around for later use). |
| 112 | fParser->readAndSaveAFrame(); |
| 113 | } |
| 114 | |
| 115 | return fParser->currentFrame().samplingFreq; |
| 116 | } |
| 117 | |
| 118 | void AC3AudioStreamFramer::flushInput() { |
| 119 | fParser->flushInput(); |
| 120 | } |
| 121 | |
| 122 | void AC3AudioStreamFramer::doGetNextFrame() { |
| 123 | fParser->registerReadInterest(fTo, fMaxSize); |
| 124 | parseNextFrame(); |
| 125 | } |
| 126 | |
| 127 | #define MILLION 1000000 |
| 128 | |
| 129 | struct timeval AC3AudioStreamFramer::currentFramePlayTime() const { |
| 130 | AC3FrameParams const& fr = fParser->currentFrame(); |
| 131 | unsigned const numSamples = 1536; |
| 132 | unsigned const freq = fr.samplingFreq; |
| 133 | |
| 134 | // result is numSamples/freq |
| 135 | unsigned const uSeconds = (freq == 0) ? 0 |
| 136 | : ((numSamples*2*MILLION)/freq + 1)/2; // rounds to nearest integer |
| 137 | |
| 138 | struct timeval result; |
| 139 | result.tv_sec = uSeconds/MILLION; |
| 140 | result.tv_usec = uSeconds%MILLION; |
| 141 | return result; |
| 142 | } |
| 143 | |
| 144 | void AC3AudioStreamFramer |
| 145 | ::handleNewData(void* clientData, unsigned char* ptr, unsigned size, |
| 146 | struct timeval /*presentationTime*/) { |
| 147 | AC3AudioStreamFramer* framer = (AC3AudioStreamFramer*)clientData; |
| 148 | framer->handleNewData(ptr, size); |
| 149 | } |
| 150 | |
| 151 | void AC3AudioStreamFramer |
| 152 | ::handleNewData(unsigned char* ptr, unsigned size) { |
| 153 | fParser->testStreamCode(fOurStreamCode, ptr, size); |
| 154 | |
| 155 | parseNextFrame(); |
| 156 | } |
| 157 | |
| 158 | void AC3AudioStreamFramer::parseNextFrame() { |
| 159 | unsigned acquiredFrameSize = fParser->parseFrame(fNumTruncatedBytes); |
| 160 | if (acquiredFrameSize > 0) { |
| 161 | // We were able to acquire a frame from the input. |
| 162 | // It has already been copied to the reader's space. |
| 163 | fFrameSize = acquiredFrameSize; |
| 164 | |
| 165 | // Also set the presentation time, and increment it for next time, |
| 166 | // based on the length of this frame: |
| 167 | fPresentationTime = fNextFramePresentationTime; |
| 168 | |
| 169 | struct timeval framePlayTime = currentFramePlayTime(); |
| 170 | fDurationInMicroseconds = framePlayTime.tv_sec*MILLION + framePlayTime.tv_usec; |
| 171 | fNextFramePresentationTime.tv_usec += framePlayTime.tv_usec; |
| 172 | fNextFramePresentationTime.tv_sec |
| 173 | += framePlayTime.tv_sec + fNextFramePresentationTime.tv_usec/MILLION; |
| 174 | fNextFramePresentationTime.tv_usec %= MILLION; |
| 175 | |
| 176 | // Call our own 'after getting' function. Because we're not a 'leaf' |
| 177 | // source, we can call this directly, without risking infinite recursion. |
| 178 | afterGetting(this); |
| 179 | } else { |
| 180 | // We were unable to parse a complete frame from the input, because: |
| 181 | // - we had to read more data from the source stream, or |
| 182 | // - the source stream has ended. |
| 183 | } |
| 184 | } |
| 185 | |
| 186 | |
| 187 | ////////// AC3AudioStreamParser implementation ////////// |
| 188 | |
| 189 | static int const kbpsTable[] = {32, 40, 48, 56, 64, 80, 96, 112, |
| 190 | 128, 160, 192, 224, 256, 320, 384, 448, |
| 191 | 512, 576, 640}; |
| 192 | |
| 193 | void AC3FrameParams::() { |
| 194 | unsigned char byte4 = hdr1 >> 24; |
| 195 | |
| 196 | unsigned char kbpsIndex = (byte4&0x3E) >> 1; |
| 197 | if (kbpsIndex > 18) kbpsIndex = 18; |
| 198 | kbps = kbpsTable[kbpsIndex]; |
| 199 | |
| 200 | unsigned char samplingFreqIndex = (byte4&0xC0) >> 6; |
| 201 | switch (samplingFreqIndex) { |
| 202 | case 0: |
| 203 | samplingFreq = 48000; |
| 204 | frameSize = 4*kbps; |
| 205 | break; |
| 206 | case 1: |
| 207 | samplingFreq = 44100; |
| 208 | frameSize = 2*(320*kbps/147 + (byte4&1)); |
| 209 | break; |
| 210 | case 2: |
| 211 | case 3: // not legal? |
| 212 | samplingFreq = 32000; |
| 213 | frameSize = 6*kbps; |
| 214 | } |
| 215 | } |
| 216 | |
| 217 | AC3AudioStreamParser |
| 218 | ::AC3AudioStreamParser(AC3AudioStreamFramer* usingSource, |
| 219 | FramedSource* inputSource) |
| 220 | : StreamParser(inputSource, FramedSource::handleClosure, usingSource, |
| 221 | &AC3AudioStreamFramer::handleNewData, usingSource), |
| 222 | fUsingSource(usingSource), fHaveParsedAFrame(False), |
| 223 | fSavedFrame(NULL), fSavedFrameSize(0) { |
| 224 | } |
| 225 | |
| 226 | AC3AudioStreamParser::~AC3AudioStreamParser() { |
| 227 | } |
| 228 | |
| 229 | void AC3AudioStreamParser::registerReadInterest(unsigned char* to, |
| 230 | unsigned maxSize) { |
| 231 | fTo = to; |
| 232 | fMaxSize = maxSize; |
| 233 | } |
| 234 | |
| 235 | void AC3AudioStreamParser |
| 236 | ::testStreamCode(unsigned char ourStreamCode, |
| 237 | unsigned char* ptr, unsigned size) { |
| 238 | if (ourStreamCode == 0) return; // we assume that there's no stream code at the beginning of the data |
| 239 | |
| 240 | if (size < 4) return; |
| 241 | unsigned char streamCode = *ptr; |
| 242 | |
| 243 | if (streamCode == ourStreamCode) { |
| 244 | // Remove the first 4 bytes from the stream: |
| 245 | memmove(ptr, ptr + 4, size - 4); |
| 246 | totNumValidBytes() = totNumValidBytes() - 4; |
| 247 | } else { |
| 248 | // Discard all of the data that was just read: |
| 249 | totNumValidBytes() = totNumValidBytes() - size; |
| 250 | } |
| 251 | } |
| 252 | |
| 253 | unsigned AC3AudioStreamParser::parseFrame(unsigned& numTruncatedBytes) { |
| 254 | if (fSavedFrameSize > 0) { |
| 255 | // We've already read and parsed a frame. Use it instead: |
| 256 | memmove(fTo, fSavedFrame, fSavedFrameSize); |
| 257 | delete[] fSavedFrame; fSavedFrame = NULL; |
| 258 | unsigned frameSize = fSavedFrameSize; |
| 259 | fSavedFrameSize = 0; |
| 260 | return frameSize; |
| 261 | } |
| 262 | |
| 263 | try { |
| 264 | saveParserState(); |
| 265 | |
| 266 | // We expect an AC3 audio header (first 2 bytes == 0x0B77) at the start: |
| 267 | while (1) { |
| 268 | unsigned next4Bytes = test4Bytes(); |
| 269 | if (next4Bytes>>16 == 0x0B77) break; |
| 270 | skipBytes(1); |
| 271 | saveParserState(); |
| 272 | } |
| 273 | fCurrentFrame.hdr0 = get4Bytes(); |
| 274 | fCurrentFrame.hdr1 = test4Bytes(); |
| 275 | |
| 276 | fCurrentFrame.setParamsFromHeader(); |
| 277 | fHaveParsedAFrame = True; |
| 278 | |
| 279 | // Copy the frame to the requested destination: |
| 280 | unsigned frameSize = fCurrentFrame.frameSize; |
| 281 | if (frameSize > fMaxSize) { |
| 282 | numTruncatedBytes = frameSize - fMaxSize; |
| 283 | frameSize = fMaxSize; |
| 284 | } else { |
| 285 | numTruncatedBytes = 0; |
| 286 | } |
| 287 | |
| 288 | fTo[0] = fCurrentFrame.hdr0 >> 24; |
| 289 | fTo[1] = fCurrentFrame.hdr0 >> 16; |
| 290 | fTo[2] = fCurrentFrame.hdr0 >> 8; |
| 291 | fTo[3] = fCurrentFrame.hdr0; |
| 292 | getBytes(&fTo[4], frameSize-4); |
| 293 | skipBytes(numTruncatedBytes); |
| 294 | |
| 295 | return frameSize; |
| 296 | } catch (int /*e*/) { |
| 297 | #ifdef DEBUG |
| 298 | fUsingSource->envir() << "AC3AudioStreamParser::parseFrame() EXCEPTION (This is normal behavior - *not* an error)\n" ; |
| 299 | #endif |
| 300 | return 0; // the parsing got interrupted |
| 301 | } |
| 302 | } |
| 303 | |
| 304 | void AC3AudioStreamParser::readAndSaveAFrame() { |
| 305 | unsigned const maxAC3FrameSize = 4000; |
| 306 | fSavedFrame = new unsigned char[maxAC3FrameSize]; |
| 307 | fSavedFrameSize = 0; |
| 308 | |
| 309 | fSavedFrameFlag = 0; |
| 310 | fUsingSource->getNextFrame(fSavedFrame, maxAC3FrameSize, |
| 311 | afterGettingSavedFrame, this, |
| 312 | onSavedFrameClosure, this); |
| 313 | fUsingSource->envir().taskScheduler().doEventLoop(&fSavedFrameFlag); |
| 314 | } |
| 315 | |
| 316 | void AC3AudioStreamParser |
| 317 | ::afterGettingSavedFrame(void* clientData, unsigned frameSize, |
| 318 | unsigned /*numTruncatedBytes*/, |
| 319 | struct timeval /*presentationTime*/, |
| 320 | unsigned /*durationInMicroseconds*/) { |
| 321 | AC3AudioStreamParser* parser = (AC3AudioStreamParser*)clientData; |
| 322 | parser->afterGettingSavedFrame1(frameSize); |
| 323 | } |
| 324 | |
| 325 | void AC3AudioStreamParser |
| 326 | ::afterGettingSavedFrame1(unsigned frameSize) { |
| 327 | fSavedFrameSize = frameSize; |
| 328 | fSavedFrameFlag = ~0; |
| 329 | } |
| 330 | |
| 331 | void AC3AudioStreamParser::onSavedFrameClosure(void* clientData) { |
| 332 | AC3AudioStreamParser* parser = (AC3AudioStreamParser*)clientData; |
| 333 | parser->onSavedFrameClosure1(); |
| 334 | } |
| 335 | |
| 336 | void AC3AudioStreamParser::onSavedFrameClosure1() { |
| 337 | delete[] fSavedFrame; fSavedFrame = NULL; |
| 338 | fSavedFrameSize = 0; |
| 339 | fSavedFrameFlag = ~0; |
| 340 | } |
| 341 | |