blob: 24e558c2d71eed65418c3b5e3dfe010053639925 [file] [log] [blame]
* Copyright (c) 2016 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 <stddef.h>
#include <stdint.h>
#include <memory>
#include <string>
#include <vector>
#include "api/transport/field_trial_based_config.h"
#include "api/transport/network_types.h"
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "rtc_base/constructor_magic.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
class TestBitrateObserver {
TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
~TestBitrateObserver() {}
void OnReceiveBitrateChanged(uint32_t bitrate);
void Reset() { updated_ = false; }
bool updated() const { return updated_; }
uint32_t latest_bitrate() const { return latest_bitrate_; }
bool updated_;
uint32_t latest_bitrate_;
class RtpStream {
enum { kSendSideOffsetUs = 1000000 };
RtpStream(int fps, int bitrate_bps);
// Generates a new frame for this stream. If called too soon after the
// previous frame, no frame will be generated. The frame is split into
// packets.
int64_t GenerateFrame(int64_t time_now_us,
std::vector<PacketResult>* packets);
// The send-side time when the next frame can be generated.
int64_t next_rtp_time() const;
void set_bitrate_bps(int bitrate_bps);
int bitrate_bps() const;
static bool Compare(const std::unique_ptr<RtpStream>& lhs,
const std::unique_ptr<RtpStream>& rhs);
int fps_;
int bitrate_bps_;
int64_t next_rtp_time_;
class StreamGenerator {
StreamGenerator(int capacity, int64_t time_now);
// Add a new stream.
void AddStream(RtpStream* stream);
// Set the link capacity.
void set_capacity_bps(int capacity_bps);
// Divides |bitrate_bps| among all streams. The allocated bitrate per stream
// is decided by the initial allocation ratios.
void SetBitrateBps(int bitrate_bps);
// Set the RTP timestamp offset for the stream identified by |ssrc|.
void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
// TODO(holmer): Break out the channel simulation part from this class to make
// it possible to simulate different types of channels.
int64_t GenerateFrame(std::vector<PacketResult>* packets,
int64_t time_now_us);
// Capacity of the simulated channel in bits per second.
int capacity_;
// The time when the last packet arrived.
int64_t prev_arrival_time_us_;
// All streams being transmitted on this simulated channel.
std::vector<std::unique_ptr<RtpStream>> streams_;
} // namespace test
class DelayBasedBweTest : public ::testing::TestWithParam<std::string> {
~DelayBasedBweTest() override;
void AddDefaultStream();
// Helpers to insert a single packet into the delay-based BWE.
void IncomingFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
size_t payload_size);
void IncomingFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
size_t payload_size,
const PacedPacketInfo& pacing_info);
// Generates a frame of packets belonging to a stream at a given bitrate and
// with a given ssrc. The stream is pushed through a very simple simulated
// network, and is then given to the receive-side bandwidth estimator.
// Returns true if an over-use was seen, false otherwise.
// The StreamGenerator::updated() should be used to check for any changes in
// target bitrate after the call to this function.
bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);
// Run the bandwidth estimator with a stream of |number_of_frames| frames, or
// until it reaches |target_bitrate|.
// Can for instance be used to run the estimator for some time to get it
// into a steady state.
uint32_t SteadyStateRun(uint32_t ssrc,
int number_of_frames,
uint32_t start_bitrate,
uint32_t min_bitrate,
uint32_t max_bitrate,
uint32_t target_bitrate);
void TestTimestampGroupingTestHelper();
void TestWrappingHelper(int silence_time_s);
void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
void CapacityDropTestHelper(int number_of_streams,
bool wrap_time_stamp,
uint32_t expected_bitrate_drop_delta,
int64_t receiver_clock_offset_change_ms);
static const uint32_t kDefaultSsrc;
FieldTrialBasedConfig field_trial_config_;
field_trial; // Must be initialized first.
SimulatedClock clock_; // Time at the receiver.
test::TestBitrateObserver bitrate_observer_;
const std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
std::unique_ptr<test::StreamGenerator> stream_generator_;
int64_t arrival_time_offset_ms_;
bool first_update_;
} // namespace webrtc