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/*
* 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 "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.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/interface/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/interface/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<unsigned int>& ssrcs,
unsigned int 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;
}
rtc::scoped_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
rtc::scoped_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, true);
++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;
}
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