blob: 9ea3f08eab55f163e3fce2c7a3240545626c7249 [file] [log] [blame]
/*
* 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/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/modules/video_coding/main/test/rtp_file_reader.h"
#include "webrtc/modules/video_coding/main/test/rtp_player.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
using webrtc::rtpplayer::RtpPacketSourceInterface;
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) {
if (argc < 4) {
printf("Usage: bwe_rtp_play <extension type> <extension id> "
"<input_file.rtp>\n");
printf("<extension type> can either be:\n"
" abs for absolute send time or\n"
" tsoffset for timestamp offset.\n"
"<extension id> is the id associated with the extension.\n");
return -1;
}
RtpPacketSourceInterface* 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;
}
webrtc::scoped_ptr<RtpPacketSourceInterface> rtp_reader(reader);
webrtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
webrtc::scoped_ptr<webrtc::RemoteBitrateEstimator> rbe(estimator);
// Process the file.
int packet_counter = 0;
int64_t next_process_time_ms = 0;
int64_t next_rtp_time_ms = 0;
int64_t first_rtp_time_ms = -1;
const uint32_t kMaxPacketSize = 1500;
uint8_t packet_buffer[kMaxPacketSize];
uint8_t* packet = packet_buffer;
int non_zero_abs_send_time = 0;
int non_zero_ts_offsets = 0;
while (true) {
uint32_t next_rtp_time;
if (next_rtp_time_ms <= clock.TimeInMilliseconds()) {
uint32_t packet_length = kMaxPacketSize;
if (rtp_reader->NextPacket(packet, &packet_length,
&next_rtp_time) == -1) {
break;
}
if (first_rtp_time_ms == -1)
first_rtp_time_ms = next_rtp_time;
next_rtp_time_ms = next_rtp_time - first_rtp_time_ms;
webrtc::RTPHeader header;
parser->Parse(packet, packet_length, &header);
if (header.extension.absoluteSendTime != 0)
++non_zero_abs_send_time;
if (header.extension.transmissionTimeOffset != 0)
++non_zero_ts_offsets;
rbe->IncomingPacket(clock.TimeInMilliseconds(),
packet_length - header.headerLength,
header);
++packet_counter;
}
next_process_time_ms = rbe->TimeUntilNextProcess() +
clock.TimeInMilliseconds();
if (next_process_time_ms <= clock.TimeInMilliseconds()) {
rbe->Process();
}
int time_until_next_event =
std::min(next_process_time_ms, next_rtp_time_ms) -
clock.TimeInMilliseconds();
clock.AdvanceTimeMilliseconds(std::max(time_until_next_event, 0));
}
printf("Parsed %d packets\nTime passed: %u ms\n", packet_counter,
static_cast<uint32_t>(clock.TimeInMilliseconds()));
printf("Estimator used: %s\n", estimator_used.c_str());
printf("Packets with non-zero absolute send time: %d\n",
non_zero_abs_send_time);
printf("Packets with non-zero timestamp offset: %d\n",
non_zero_ts_offsets);
return 0;
}