void MPEGProgramStreamParser::parsePackHeader() { #ifdef DEBUG fprintf(stderr, "parsing pack header\n"); fflush(stderr); #endif unsigned first4Bytes; while (1) { first4Bytes = test4Bytes(); // We're supposed to have a pack header here, but check also for // a system header or a PES packet, just in case: if (first4Bytes == PACK_START_CODE) { skipBytes(4); break; } else if (first4Bytes == SYSTEM_HEADER_START_CODE) { #ifdef DEBUG fprintf(stderr, "found system header instead of pack header\n"); #endif setParseState(PARSING_SYSTEM_HEADER); return; } else if (isPacketStartCode(first4Bytes)) { #ifdef DEBUG fprintf(stderr, "found packet start code 0x%02x instead of pack header\n", first4Bytes); #endif setParseState(PARSING_PES_PACKET); return; } setParseState(PARSING_PACK_HEADER); // ensures we progress over bad data if ((first4Bytes & 0xFF) > 1) // a system code definitely doesn't start here { skipBytes(4); } else { skipBytes(1); } } // The size of the pack header differs depending on whether it's // MPEG-1 or MPEG-2. The next byte tells us this: unsigned char nextByte = get1Byte(); MPEG1or2Demux::SCR &scr = fUsingSource->fLastSeenSCR; // alias if ((nextByte & 0xF0) == 0x20) // MPEG-1 { fUsingSource->fMPEGversion = 1; scr.highBit = (nextByte & 0x08) >> 3; scr.remainingBits = (nextByte & 0x06) << 29; unsigned next4Bytes = get4Bytes(); scr.remainingBits |= (next4Bytes & 0xFFFE0000) >> 2; scr.remainingBits |= (next4Bytes & 0x0000FFFE) >> 1; scr.extension = 0; scr.isValid = True; skipBits(24); #if defined(DEBUG_TIMESTAMPS) || defined(DEBUG_SCR_TIMESTAMPS) fprintf(stderr, "pack hdr system_clock_reference_base: 0x%x", scr.highBit); fprintf(stderr, "%08x\n", scr.remainingBits); #endif }
unsigned MPEG1or2VideoStreamParser::parseGOPHeader(Boolean haveSeenStartCode) { // First check whether we should insert a previously-saved // 'video_sequence_header' here: if (needToUseSavedVSH()) return useSavedVSH(); #ifdef DEBUG fprintf(stderr, "parsing GOP header\n"); #endif unsigned first4Bytes; if (!haveSeenStartCode) { while ((first4Bytes = test4Bytes()) != GROUP_START_CODE) { #ifdef DEBUG fprintf(stderr, "ignoring non GOP start code: 0x%08x\n", first4Bytes); #endif get1Byte(); setParseState(PARSING_GOP_HEADER); // ensures we progress over bad data } first4Bytes = get4Bytes(); } else { // We've already seen the GROUP_START_CODE first4Bytes = GROUP_START_CODE; } save4Bytes(first4Bytes); // Next, extract the (25-bit) time code from the next 4 bytes: unsigned next4Bytes = get4Bytes(); unsigned time_code = (next4Bytes & 0xFFFFFF80) >> (32 - 25); #if defined(DEBUG) || defined(DEBUG_TIMESTAMPS) Boolean drop_frame_flag = (time_code & 0x01000000) != 0; #endif unsigned time_code_hours = (time_code & 0x00F80000) >> 19; unsigned time_code_minutes = (time_code & 0x0007E000) >> 13; unsigned time_code_seconds = (time_code & 0x00000FC0) >> 6; unsigned time_code_pictures = (time_code & 0x0000003F); #if defined(DEBUG) || defined(DEBUG_TIMESTAMPS) fprintf(stderr, "time_code: 0x%07x, drop_frame %d, hours %d, minutes %d, seconds %d, pictures %d\n", time_code, drop_frame_flag, time_code_hours, time_code_minutes, time_code_seconds, time_code_pictures); #endif #ifdef DEBUG Boolean closed_gop = (next4Bytes & 0x00000040) != 0; Boolean broken_link = (next4Bytes & 0x00000020) != 0; fprintf(stderr, "closed_gop: %d, broken_link: %d\n", closed_gop, broken_link); #endif // Now, copy all bytes that we see, up until we reach a PICTURE_START_CODE: do { saveToNextCode(next4Bytes); } while (next4Bytes != PICTURE_START_CODE); // Record the time code: usingSource()->setTimeCode(time_code_hours, time_code_minutes, time_code_seconds, time_code_pictures, fPicturesSinceLastGOP); fPicturesSinceLastGOP = 0; // Compute this frame's timestamp: usingSource()->computePresentationTime(0); setParseState(PARSING_PICTURE_HEADER); return curFrameSize(); }
WAVAudioFileSource::WAVAudioFileSource(UsageEnvironment& env, FILE* fid) : AudioInputDevice(env, 0, 0, 0, 0)/* set the real parameters later */, fFid(fid), fLastPlayTime(0), fWAVHeaderSize(0), fFileSize(0), fScaleFactor(1), fLimitNumBytesToStream(False), fNumBytesToStream(0), fAudioFormat(WA_UNKNOWN) { // Check the WAV file header for validity. // Note: The following web pages contain info about the WAV format: // http://www.ringthis.com/dev/wave_format.htm // http://www.lightlink.com/tjweber/StripWav/Canon.html // http://www.wotsit.org/list.asp?al=W Boolean success = False; // until we learn otherwise do { // RIFF Chunk: if (nextc != 'R' || nextc != 'I' || nextc != 'F' || nextc != 'F') break; if (!skipBytes(fid, 4)) break; if (nextc != 'W' || nextc != 'A' || nextc != 'V' || nextc != 'E') break; // FORMAT Chunk: if (nextc != 'f' || nextc != 'm' || nextc != 't' || nextc != ' ') break; unsigned formatLength; if (!get4Bytes(fid, formatLength)) break; unsigned short audioFormat; if (!get2Bytes(fid, audioFormat)) break; fAudioFormat = (unsigned char)audioFormat; if (fAudioFormat != WA_PCM && fAudioFormat != WA_PCMA && fAudioFormat != WA_PCMU && fAudioFormat != WA_IMA_ADPCM) { // It's a format that we don't (yet) understand env.setResultMsg("Audio format is not one that we handle (PCM/PCMU/PCMA or IMA ADPCM)"); break; } unsigned short numChannels; if (!get2Bytes(fid, numChannels)) break; fNumChannels = (unsigned char)numChannels; if (fNumChannels < 1 || fNumChannels > 2) { // invalid # channels char errMsg[100]; sprintf(errMsg, "Bad # channels: %d", fNumChannels); env.setResultMsg(errMsg); break; } if (!get4Bytes(fid, fSamplingFrequency)) break; if (fSamplingFrequency == 0) { env.setResultMsg("Bad sampling frequency: 0"); break; } if (!skipBytes(fid, 6)) break; // "nAvgBytesPerSec" (4 bytes) + "nBlockAlign" (2 bytes) unsigned short bitsPerSample; if (!get2Bytes(fid, bitsPerSample)) break; fBitsPerSample = (unsigned char)bitsPerSample; if (fBitsPerSample == 0) { env.setResultMsg("Bad bits-per-sample: 0"); break; } if (!skipBytes(fid, formatLength - 16)) break; // FACT chunk (optional): int c = nextc; if (c == 'f') { if (nextc != 'a' || nextc != 'c' || nextc != 't') break; unsigned factLength; if (!get4Bytes(fid, factLength)) break; if (!skipBytes(fid, factLength)) break; c = nextc; } // DATA Chunk: if (c != 'd' || nextc != 'a' || nextc != 't' || nextc != 'a') break; if (!skipBytes(fid, 4)) break; // The header is good; the remaining data are the sample bytes. fWAVHeaderSize = ftell(fid); success = True; } while (0); if (!success) { env.setResultMsg("Bad WAV file format"); // Set "fBitsPerSample" to zero, to indicate failure: fBitsPerSample = 0; return; } fPlayTimePerSample = 1e6/(double)fSamplingFrequency; // Although PCM is a sample-based format, we group samples into // 'frames' for efficient delivery to clients. Set up our preferred // frame size to be close to 20 ms, if possible, but always no greater // than 1400 bytes (to ensure that it will fit in a single RTP packet) unsigned maxSamplesPerFrame = (1400*8)/(fNumChannels*fBitsPerSample); unsigned desiredSamplesPerFrame = (unsigned)(0.02*fSamplingFrequency); unsigned samplesPerFrame = desiredSamplesPerFrame < maxSamplesPerFrame ? desiredSamplesPerFrame : maxSamplesPerFrame; fPreferredFrameSize = (samplesPerFrame*fNumChannels*fBitsPerSample)/8; }
unsigned MPEG1or2VideoStreamParser ::parseVideoSequenceHeader(Boolean haveSeenStartCode) { #ifdef DEBUG fprintf(stderr, "parsing video sequence header\n"); #endif unsigned first4Bytes; if (!haveSeenStartCode) { while ((first4Bytes = test4Bytes()) != VIDEO_SEQUENCE_HEADER_START_CODE) { #ifdef DEBUG fprintf(stderr, "ignoring non video sequence header: 0x%08x\n", first4Bytes); #endif get1Byte(); setParseState(PARSING_VIDEO_SEQUENCE_HEADER); // ensures we progress over bad data } first4Bytes = get4Bytes(); } else { // We've already seen the start code first4Bytes = VIDEO_SEQUENCE_HEADER_START_CODE; } save4Bytes(first4Bytes); // Next, extract the size and rate parameters from the next 8 bytes unsigned paramWord1 = get4Bytes(); save4Bytes(paramWord1); unsigned next4Bytes = get4Bytes(); #ifdef DEBUG unsigned short horizontal_size_value = (paramWord1 & 0xFFF00000) >> (32 - 12); unsigned short vertical_size_value = (paramWord1 & 0x000FFF00) >> 8; unsigned char aspect_ratio_information = (paramWord1 & 0x000000F0) >> 4; #endif unsigned char frame_rate_code = (paramWord1 & 0x0000000F); usingSource()->fFrameRate = frameRateFromCode[frame_rate_code]; #ifdef DEBUG unsigned bit_rate_value = (next4Bytes & 0xFFFFC000) >> (32 - 18); unsigned vbv_buffer_size_value = (next4Bytes & 0x00001FF8) >> 3; fprintf(stderr, "horizontal_size_value: %d, vertical_size_value: %d, aspect_ratio_information: %d, frame_rate_code: %d (=>%f fps), bit_rate_value: %d (=>%d bps), vbv_buffer_size_value: %d\n", horizontal_size_value, vertical_size_value, aspect_ratio_information, frame_rate_code, usingSource()->fFrameRate, bit_rate_value, bit_rate_value * 400, vbv_buffer_size_value); #endif // Now, copy all bytes that we see, up until we reach a GROUP_START_CODE // or a PICTURE_START_CODE: do { saveToNextCode(next4Bytes); } while (next4Bytes != GROUP_START_CODE && next4Bytes != PICTURE_START_CODE); setParseState((next4Bytes == GROUP_START_CODE) ? PARSING_GOP_HEADER_SEEN_CODE : PARSING_PICTURE_HEADER); // Compute this frame's timestamp by noting how many pictures we've seen // since the last GOP header: usingSource()->computePresentationTime(fPicturesSinceLastGOP); // Save this video_sequence_header, in case we need to insert a copy // into the stream later: saveCurrentVSH(); return curFrameSize(); }
WAVAudioFileSource::WAVAudioFileSource(UsageEnvironment& env, FILE* fid) : AudioInputDevice(env, 0, 0, 0, 0)/* set the real parameters later */, fFid(fid), fFidIsSeekable(False), fLastPlayTime(0), fHaveStartedReading(False), fWAVHeaderSize(0), fFileSize(0), fScaleFactor(1), fLimitNumBytesToStream(False), fNumBytesToStream(0), fAudioFormat(WA_UNKNOWN) { // Check the WAV file header for validity. // Note: The following web pages contain info about the WAV format: // http://www.ringthis.com/dev/wave_format.htm // http://www.lightlink.com/tjweber/StripWav/Canon.html // http://www.onicos.com/staff/iz/formats/wav.html Boolean success = False; // until we learn otherwise do { // RIFF Chunk: if (nextc != 'R' || nextc != 'I' || nextc != 'F' || nextc != 'F') break; if (!skipBytes(fid, 4)) break; if (nextc != 'W' || nextc != 'A' || nextc != 'V' || nextc != 'E') break; // Skip over any chunk that's not a FORMAT ('fmt ') chunk: u_int32_t tmp; if (!get4Bytes(fid, tmp)) break; if (tmp != 0x20746d66/*'fmt ', little-endian*/) { // Skip this chunk: if (!get4Bytes(fid, tmp)) break; if (!skipBytes(fid, tmp)) break; } // FORMAT Chunk (the 4-byte header code has already been parsed): unsigned formatLength; if (!get4Bytes(fid, formatLength)) break; unsigned short audioFormat; if (!get2Bytes(fid, audioFormat)) break; fAudioFormat = (unsigned char)audioFormat; if (fAudioFormat != WA_PCM && fAudioFormat != WA_PCMA && fAudioFormat != WA_PCMU && fAudioFormat != WA_IMA_ADPCM) { // It's a format that we don't (yet) understand env.setResultMsg("Audio format is not one that we handle (PCM/PCMU/PCMA or IMA ADPCM)"); break; } unsigned short numChannels; if (!get2Bytes(fid, numChannels)) break; fNumChannels = (unsigned char)numChannels; if (fNumChannels < 1 || fNumChannels > 2) { // invalid # channels char errMsg[100]; sprintf(errMsg, "Bad # channels: %d", fNumChannels); env.setResultMsg(errMsg); break; } if (!get4Bytes(fid, fSamplingFrequency)) break; if (fSamplingFrequency == 0) { env.setResultMsg("Bad sampling frequency: 0"); break; } if (!skipBytes(fid, 6)) break; // "nAvgBytesPerSec" (4 bytes) + "nBlockAlign" (2 bytes) unsigned short bitsPerSample; if (!get2Bytes(fid, bitsPerSample)) break; fBitsPerSample = (unsigned char)bitsPerSample; if (fBitsPerSample == 0) { env.setResultMsg("Bad bits-per-sample: 0"); break; } if (!skipBytes(fid, formatLength - 16)) break; // FACT chunk (optional): int c = nextc; if (c == 'f') { if (nextc != 'a' || nextc != 'c' || nextc != 't') break; unsigned factLength; if (!get4Bytes(fid, factLength)) break; if (!skipBytes(fid, factLength)) break; c = nextc; } // DATA Chunk: if (c != 'd' || nextc != 'a' || nextc != 't' || nextc != 'a') break; if (!skipBytes(fid, 4)) break; // The header is good; the remaining data are the sample bytes. fWAVHeaderSize = (unsigned)TellFile64(fid); success = True; } while (0); if (!success) { env.setResultMsg("Bad WAV file format"); // Set "fBitsPerSample" to zero, to indicate failure: fBitsPerSample = 0; return; } fPlayTimePerSample = 1e6/(double)fSamplingFrequency; // Although PCM is a sample-based format, we group samples into // 'frames' for efficient delivery to clients. Set up our preferred // frame size to be close to 20 ms, if possible, but always no greater // than 1400 bytes (to ensure that it will fit in a single RTP packet) unsigned maxSamplesPerFrame = (1400*8)/(fNumChannels*fBitsPerSample); unsigned desiredSamplesPerFrame = (unsigned)(0.02*fSamplingFrequency); unsigned samplesPerFrame = desiredSamplesPerFrame < maxSamplesPerFrame ? desiredSamplesPerFrame : maxSamplesPerFrame; fPreferredFrameSize = (samplesPerFrame*fNumChannels*fBitsPerSample)/8; fFidIsSeekable = FileIsSeekable(fFid); #ifndef READ_FROM_FILES_SYNCHRONOUSLY // Now that we've finished reading the WAV header, all future reads (of audio samples) from the file will be asynchronous: makeSocketNonBlocking(fileno(fFid)); #endif }