| /* | 
 |  *  Copyright (c) 2014 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 <stdio.h> | 
 |  | 
 | #include <memory> | 
 |  | 
 | #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" | 
 | #include "modules/remote_bitrate_estimator/tools/bwe_rtp.h" | 
 | #include "modules/rtp_rtcp/include/rtp_header_parser.h" | 
 | #include "modules/rtp_rtcp/include/rtp_payload_registry.h" | 
 | #include "rtc_base/format_macros.h" | 
 | #include "test/rtp_file_reader.h" | 
 |  | 
 | class Observer : public webrtc::RemoteBitrateObserver { | 
 |  public: | 
 |   explicit Observer(webrtc::Clock* clock) : clock_(clock) {} | 
 |  | 
 |   // Called when a receive channel group has a new bitrate estimate for the | 
 |   // incoming streams. | 
 |   virtual void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs, | 
 |                                        uint32_t bitrate) { | 
 |     printf("[%u] Num SSRCs: %d, bitrate: %u\n", | 
 |            static_cast<uint32_t>(clock_->TimeInMilliseconds()), | 
 |            static_cast<int>(ssrcs.size()), bitrate); | 
 |   } | 
 |  | 
 |   virtual ~Observer() {} | 
 |  | 
 |  private: | 
 |   webrtc::Clock* clock_; | 
 | }; | 
 |  | 
 | int main(int argc, char* argv[]) { | 
 |   webrtc::test::RtpFileReader* reader; | 
 |   webrtc::RemoteBitrateEstimator* estimator; | 
 |   webrtc::RtpHeaderParser* parser; | 
 |   std::string estimator_used; | 
 |   webrtc::SimulatedClock clock(0); | 
 |   Observer observer(&clock); | 
 |   if (!ParseArgsAndSetupEstimator(argc, argv, &clock, &observer, &reader, | 
 |                                   &parser, &estimator, &estimator_used)) { | 
 |     return -1; | 
 |   } | 
 |   std::unique_ptr<webrtc::test::RtpFileReader> rtp_reader(reader); | 
 |   std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(parser); | 
 |   std::unique_ptr<webrtc::RemoteBitrateEstimator> rbe(estimator); | 
 |  | 
 |   // Process the file. | 
 |   int packet_counter = 0; | 
 |   int64_t next_rtp_time_ms = 0; | 
 |   int64_t first_rtp_time_ms = -1; | 
 |   int abs_send_time_count = 0; | 
 |   int ts_offset_count = 0; | 
 |   webrtc::test::RtpPacket packet; | 
 |   if (!rtp_reader->NextPacket(&packet)) { | 
 |     printf("No RTP packet found\n"); | 
 |     return 0; | 
 |   } | 
 |   first_rtp_time_ms = packet.time_ms; | 
 |   packet.time_ms = packet.time_ms - first_rtp_time_ms; | 
 |   while (true) { | 
 |     if (next_rtp_time_ms <= clock.TimeInMilliseconds()) { | 
 |       if (!parser->IsRtcp(packet.data, packet.length)) { | 
 |         webrtc::RTPHeader header; | 
 |         parser->Parse(packet.data, packet.length, &header); | 
 |         if (header.extension.hasAbsoluteSendTime) | 
 |           ++abs_send_time_count; | 
 |         if (header.extension.hasTransmissionTimeOffset) | 
 |           ++ts_offset_count; | 
 |         size_t packet_length = packet.length; | 
 |         // Some RTP dumps only include the header, in which case packet.length | 
 |         // is equal to the header length. In those cases packet.original_length | 
 |         // usually contains the original packet length. | 
 |         if (packet.original_length > 0) { | 
 |           packet_length = packet.original_length; | 
 |         } | 
 |         rbe->IncomingPacket(clock.TimeInMilliseconds(), | 
 |                             packet_length - header.headerLength, header); | 
 |         ++packet_counter; | 
 |       } | 
 |       if (!rtp_reader->NextPacket(&packet)) { | 
 |         break; | 
 |       } | 
 |       packet.time_ms = packet.time_ms - first_rtp_time_ms; | 
 |       next_rtp_time_ms = packet.time_ms; | 
 |     } | 
 |     int64_t time_until_process_ms = rbe->TimeUntilNextProcess(); | 
 |     if (time_until_process_ms <= 0) { | 
 |       rbe->Process(); | 
 |     } | 
 |     int64_t time_until_next_event = | 
 |         std::min(rbe->TimeUntilNextProcess(), | 
 |                  next_rtp_time_ms - clock.TimeInMilliseconds()); | 
 |     clock.AdvanceTimeMilliseconds(std::max<int64_t>(time_until_next_event, 0)); | 
 |   } | 
 |   printf("Parsed %d packets\nTime passed: %" PRId64 " ms\n", packet_counter, | 
 |          clock.TimeInMilliseconds()); | 
 |   printf("Estimator used: %s\n", estimator_used.c_str()); | 
 |   printf("Packets with absolute send time: %d\n", | 
 |          abs_send_time_count); | 
 |   printf("Packets with timestamp offset: %d\n", | 
 |          ts_offset_count); | 
 |   printf("Packets with no extension: %d\n", | 
 |          packet_counter - ts_offset_count - abs_send_time_count); | 
 |   return 0; | 
 | } |