| /* |
| * 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 "webrtc/base/format_macros.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
| #include "webrtc/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; |
| } |