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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/modules/rtp_rtcp/source/rtp_utility.cc
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
Diffstat (limited to 'webrtc/modules/rtp_rtcp/source/rtp_utility.cc')
-rw-r--r--webrtc/modules/rtp_rtcp/source/rtp_utility.cc506
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diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
new file mode 100644
index 0000000000..bf0b30a064
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
@@ -0,0 +1,506 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+#include <assert.h>
+#include <math.h> // ceil
+#include <string.h> // memcpy
+
+#if defined(_WIN32)
+// Order for these headers are important
+#include <winsock2.h> // timeval
+#include <windows.h> // FILETIME
+#include <MMSystem.h> // timeGetTime
+#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
+#include <sys/time.h> // gettimeofday
+#include <time.h>
+#endif
+#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
+#include <stdio.h>
+#endif
+
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+
+#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
+#define DEBUG_PRINT(...) \
+ { \
+ char msg[256]; \
+ sprintf(msg, __VA_ARGS__); \
+ OutputDebugString(msg); \
+ }
+#else
+// special fix for visual 2003
+#define DEBUG_PRINT(exp) ((void)0)
+#endif // defined(_DEBUG) && defined(_WIN32)
+
+namespace webrtc {
+
+RtpData* NullObjectRtpData() {
+ static NullRtpData null_rtp_data;
+ return &null_rtp_data;
+}
+
+RtpFeedback* NullObjectRtpFeedback() {
+ static NullRtpFeedback null_rtp_feedback;
+ return &null_rtp_feedback;
+}
+
+RtpAudioFeedback* NullObjectRtpAudioFeedback() {
+ static NullRtpAudioFeedback null_rtp_audio_feedback;
+ return &null_rtp_audio_feedback;
+}
+
+ReceiveStatistics* NullObjectReceiveStatistics() {
+ static NullReceiveStatistics null_receive_statistics;
+ return &null_receive_statistics;
+}
+
+namespace RtpUtility {
+
+enum {
+ kRtcpExpectedVersion = 2,
+ kRtcpMinHeaderLength = 4,
+ kRtcpMinParseLength = 8,
+
+ kRtpExpectedVersion = 2,
+ kRtpMinParseLength = 12
+};
+
+/*
+ * Time routines.
+ */
+
+uint32_t GetCurrentRTP(Clock* clock, uint32_t freq) {
+ const bool use_global_clock = (clock == NULL);
+ Clock* local_clock = clock;
+ if (use_global_clock) {
+ local_clock = Clock::GetRealTimeClock();
+ }
+ uint32_t secs = 0, frac = 0;
+ local_clock->CurrentNtp(secs, frac);
+ if (use_global_clock) {
+ delete local_clock;
+ }
+ return ConvertNTPTimeToRTP(secs, frac, freq);
+}
+
+uint32_t ConvertNTPTimeToRTP(uint32_t NTPsec, uint32_t NTPfrac, uint32_t freq) {
+ float ftemp = (float)NTPfrac / (float)NTP_FRAC;
+ uint32_t tmp = (uint32_t)(ftemp * freq);
+ return NTPsec * freq + tmp;
+}
+
+uint32_t ConvertNTPTimeToMS(uint32_t NTPsec, uint32_t NTPfrac) {
+ int freq = 1000;
+ float ftemp = (float)NTPfrac / (float)NTP_FRAC;
+ uint32_t tmp = (uint32_t)(ftemp * freq);
+ uint32_t MStime = NTPsec * freq + tmp;
+ return MStime;
+}
+
+/*
+ * Misc utility routines
+ */
+
+#if defined(_WIN32)
+bool StringCompare(const char* str1, const char* str2,
+ const uint32_t length) {
+ return (_strnicmp(str1, str2, length) == 0) ? true : false;
+}
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
+bool StringCompare(const char* str1, const char* str2,
+ const uint32_t length) {
+ return (strncasecmp(str1, str2, length) == 0) ? true : false;
+}
+#endif
+
+size_t Word32Align(size_t size) {
+ uint32_t remainder = size % 4;
+ if (remainder != 0)
+ return size + 4 - remainder;
+ return size;
+}
+
+uint32_t pow2(uint8_t exp) {
+ return 1 << exp;
+}
+
+RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData,
+ const size_t rtpDataLength)
+ : _ptrRTPDataBegin(rtpData),
+ _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) {
+}
+
+RtpHeaderParser::~RtpHeaderParser() {
+}
+
+bool RtpHeaderParser::RTCP() const {
+ // 72 to 76 is reserved for RTP
+ // 77 to 79 is not reserver but they are not assigned we will block them
+ // for RTCP 200 SR == marker bit + 72
+ // for RTCP 204 APP == marker bit + 76
+ /*
+ * RTCP
+ *
+ * FIR full INTRA-frame request 192 [RFC2032] supported
+ * NACK negative acknowledgement 193 [RFC2032]
+ * IJ Extended inter-arrival jitter report 195 [RFC-ietf-avt-rtp-toff
+ * set-07.txt] http://tools.ietf.org/html/draft-ietf-avt-rtp-toffset-07
+ * SR sender report 200 [RFC3551] supported
+ * RR receiver report 201 [RFC3551] supported
+ * SDES source description 202 [RFC3551] supported
+ * BYE goodbye 203 [RFC3551] supported
+ * APP application-defined 204 [RFC3551] ignored
+ * RTPFB Transport layer FB message 205 [RFC4585] supported
+ * PSFB Payload-specific FB message 206 [RFC4585] supported
+ * XR extended report 207 [RFC3611] supported
+ */
+
+ /* 205 RFC 5104
+ * FMT 1 NACK supported
+ * FMT 2 reserved
+ * FMT 3 TMMBR supported
+ * FMT 4 TMMBN supported
+ */
+
+ /* 206 RFC 5104
+ * FMT 1: Picture Loss Indication (PLI) supported
+ * FMT 2: Slice Lost Indication (SLI)
+ * FMT 3: Reference Picture Selection Indication (RPSI)
+ * FMT 4: Full Intra Request (FIR) Command supported
+ * FMT 5: Temporal-Spatial Trade-off Request (TSTR)
+ * FMT 6: Temporal-Spatial Trade-off Notification (TSTN)
+ * FMT 7: Video Back Channel Message (VBCM)
+ * FMT 15: Application layer FB message
+ */
+
+ const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
+ if (length < kRtcpMinHeaderLength) {
+ return false;
+ }
+
+ const uint8_t V = _ptrRTPDataBegin[0] >> 6;
+ if (V != kRtcpExpectedVersion) {
+ return false;
+ }
+
+ const uint8_t payloadType = _ptrRTPDataBegin[1];
+ switch (payloadType) {
+ case 192:
+ return true;
+ case 193:
+ // not supported
+ // pass through and check for a potential RTP packet
+ return false;
+ case 195:
+ case 200:
+ case 201:
+ case 202:
+ case 203:
+ case 204:
+ case 205:
+ case 206:
+ case 207:
+ return true;
+ default:
+ return false;
+ }
+}
+
+bool RtpHeaderParser::ParseRtcp(RTPHeader* header) const {
+ assert(header != NULL);
+
+ const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
+ if (length < kRtcpMinParseLength) {
+ return false;
+ }
+
+ const uint8_t V = _ptrRTPDataBegin[0] >> 6;
+ if (V != kRtcpExpectedVersion) {
+ return false;
+ }
+
+ const uint8_t PT = _ptrRTPDataBegin[1];
+ const size_t len = (_ptrRTPDataBegin[2] << 8) + _ptrRTPDataBegin[3];
+ const uint8_t* ptr = &_ptrRTPDataBegin[4];
+
+ uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
+ ptr += 4;
+
+ header->payloadType = PT;
+ header->ssrc = SSRC;
+ header->headerLength = 4 + (len << 2);
+
+ return true;
+}
+
+bool RtpHeaderParser::Parse(RTPHeader& header,
+ RtpHeaderExtensionMap* ptrExtensionMap) const {
+ const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
+ if (length < kRtpMinParseLength) {
+ return false;
+ }
+
+ // Version
+ const uint8_t V = _ptrRTPDataBegin[0] >> 6;
+ // Padding
+ const bool P = ((_ptrRTPDataBegin[0] & 0x20) == 0) ? false : true;
+ // eXtension
+ const bool X = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
+ const uint8_t CC = _ptrRTPDataBegin[0] & 0x0f;
+ const bool M = ((_ptrRTPDataBegin[1] & 0x80) == 0) ? false : true;
+
+ const uint8_t PT = _ptrRTPDataBegin[1] & 0x7f;
+
+ const uint16_t sequenceNumber = (_ptrRTPDataBegin[2] << 8) +
+ _ptrRTPDataBegin[3];
+
+ const uint8_t* ptr = &_ptrRTPDataBegin[4];
+
+ uint32_t RTPTimestamp = ByteReader<uint32_t>::ReadBigEndian(ptr);
+ ptr += 4;
+
+ uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
+ ptr += 4;
+
+ if (V != kRtpExpectedVersion) {
+ return false;
+ }
+
+ const size_t CSRCocts = CC * 4;
+
+ if ((ptr + CSRCocts) > _ptrRTPDataEnd) {
+ return false;
+ }
+
+ header.markerBit = M;
+ header.payloadType = PT;
+ header.sequenceNumber = sequenceNumber;
+ header.timestamp = RTPTimestamp;
+ header.ssrc = SSRC;
+ header.numCSRCs = CC;
+ header.paddingLength = P ? *(_ptrRTPDataEnd - 1) : 0;
+
+ for (uint8_t i = 0; i < CC; ++i) {
+ uint32_t CSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
+ ptr += 4;
+ header.arrOfCSRCs[i] = CSRC;
+ }
+
+ header.headerLength = 12 + CSRCocts;
+
+ // If in effect, MAY be omitted for those packets for which the offset
+ // is zero.
+ header.extension.hasTransmissionTimeOffset = false;
+ header.extension.transmissionTimeOffset = 0;
+
+ // May not be present in packet.
+ header.extension.hasAbsoluteSendTime = false;
+ header.extension.absoluteSendTime = 0;
+
+ // May not be present in packet.
+ header.extension.hasAudioLevel = false;
+ header.extension.voiceActivity = false;
+ header.extension.audioLevel = 0;
+
+ // May not be present in packet.
+ header.extension.hasVideoRotation = false;
+ header.extension.videoRotation = 0;
+
+ if (X) {
+ /* RTP header extension, RFC 3550.
+ 0 1 2 3
+ 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | defined by profile | length |
+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ | header extension |
+ | .... |
+ */
+ const ptrdiff_t remain = _ptrRTPDataEnd - ptr;
+ if (remain < 4) {
+ return false;
+ }
+
+ header.headerLength += 4;
+
+ uint16_t definedByProfile = ByteReader<uint16_t>::ReadBigEndian(ptr);
+ ptr += 2;
+
+ // in 32 bit words
+ size_t XLen = ByteReader<uint16_t>::ReadBigEndian(ptr);
+ ptr += 2;
+ XLen *= 4; // in bytes
+
+ if (static_cast<size_t>(remain) < (4 + XLen)) {
+ return false;
+ }
+ if (definedByProfile == kRtpOneByteHeaderExtensionId) {
+ const uint8_t* ptrRTPDataExtensionEnd = ptr + XLen;
+ ParseOneByteExtensionHeader(header,
+ ptrExtensionMap,
+ ptrRTPDataExtensionEnd,
+ ptr);
+ }
+ header.headerLength += XLen;
+ }
+ if (header.headerLength + header.paddingLength > static_cast<size_t>(length))
+ return false;
+ return true;
+}
+
+void RtpHeaderParser::ParseOneByteExtensionHeader(
+ RTPHeader& header,
+ const RtpHeaderExtensionMap* ptrExtensionMap,
+ const uint8_t* ptrRTPDataExtensionEnd,
+ const uint8_t* ptr) const {
+ if (!ptrExtensionMap) {
+ return;
+ }
+
+ while (ptrRTPDataExtensionEnd - ptr > 0) {
+ // 0
+ // 0 1 2 3 4 5 6 7
+ // +-+-+-+-+-+-+-+-+
+ // | ID | len |
+ // +-+-+-+-+-+-+-+-+
+
+ // Note that 'len' is the header extension element length, which is the
+ // number of bytes - 1.
+ const uint8_t id = (*ptr & 0xf0) >> 4;
+ const uint8_t len = (*ptr & 0x0f);
+ ptr++;
+
+ if (id == 15) {
+ LOG(LS_WARNING)
+ << "RTP extension header 15 encountered. Terminate parsing.";
+ return;
+ }
+
+ RTPExtensionType type;
+ if (ptrExtensionMap->GetType(id, &type) != 0) {
+ // If we encounter an unknown extension, just skip over it.
+ LOG(LS_WARNING) << "Failed to find extension id: "
+ << static_cast<int>(id);
+ } else {
+ switch (type) {
+ case kRtpExtensionTransmissionTimeOffset: {
+ if (len != 2) {
+ LOG(LS_WARNING) << "Incorrect transmission time offset len: "
+ << len;
+ return;
+ }
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | len=2 | transmission offset |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+ header.extension.transmissionTimeOffset =
+ ByteReader<int32_t, 3>::ReadBigEndian(ptr);
+ header.extension.hasTransmissionTimeOffset = true;
+ break;
+ }
+ case kRtpExtensionAudioLevel: {
+ if (len != 0) {
+ LOG(LS_WARNING) << "Incorrect audio level len: " << len;
+ return;
+ }
+ // 0 1
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | len=0 |V| level |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ //
+ header.extension.audioLevel = ptr[0] & 0x7f;
+ header.extension.voiceActivity = (ptr[0] & 0x80) != 0;
+ header.extension.hasAudioLevel = true;
+ break;
+ }
+ case kRtpExtensionAbsoluteSendTime: {
+ if (len != 2) {
+ LOG(LS_WARNING) << "Incorrect absolute send time len: " << len;
+ return;
+ }
+ // 0 1 2 3
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | len=2 | absolute send time |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+ header.extension.absoluteSendTime =
+ ByteReader<uint32_t, 3>::ReadBigEndian(ptr);
+ header.extension.hasAbsoluteSendTime = true;
+ break;
+ }
+ case kRtpExtensionVideoRotation: {
+ if (len != 0) {
+ LOG(LS_WARNING)
+ << "Incorrect coordination of video coordination len: " << len;
+ return;
+ }
+ // 0 1
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | len=0 |0 0 0 0 C F R R|
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ header.extension.hasVideoRotation = true;
+ header.extension.videoRotation = ptr[0];
+ break;
+ }
+ case kRtpExtensionTransportSequenceNumber: {
+ if (len != 1) {
+ LOG(LS_WARNING)
+ << "Incorrect peer connection sequence number len: " << len;
+ return;
+ }
+ // 0 1 2
+ // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ // | ID | L=1 |transport wide sequence number |
+ // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+ uint16_t sequence_number = ptr[0] << 8;
+ sequence_number += ptr[1];
+ header.extension.transportSequenceNumber = sequence_number;
+ header.extension.hasTransportSequenceNumber = true;
+ break;
+ }
+ default: {
+ LOG(LS_WARNING) << "Extension type not implemented: " << type;
+ return;
+ }
+ }
+ }
+ ptr += (len + 1);
+ uint8_t num_bytes = ParsePaddingBytes(ptrRTPDataExtensionEnd, ptr);
+ ptr += num_bytes;
+ }
+}
+
+uint8_t RtpHeaderParser::ParsePaddingBytes(
+ const uint8_t* ptrRTPDataExtensionEnd,
+ const uint8_t* ptr) const {
+ uint8_t num_zero_bytes = 0;
+ while (ptrRTPDataExtensionEnd - ptr > 0) {
+ if (*ptr != 0) {
+ return num_zero_bytes;
+ }
+ ptr++;
+ num_zero_bytes++;
+ }
+ return num_zero_bytes;
+}
+} // namespace RtpUtility
+
+} // namespace webrtc