Files
platform-external-webrtc/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
Henrik Kjellander ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

131 lines
5.1 KiB
C++

/*
* Copyright (c) 2013 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 "webrtc/modules/remote_bitrate_estimator/inter_arrival.h"
#include <algorithm>
#include <cassert>
#include "webrtc/base/logging.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
static const int kBurstDeltaThresholdMs = 5;
InterArrival::InterArrival(uint32_t timestamp_group_length_ticks,
double timestamp_to_ms_coeff,
bool enable_burst_grouping)
: kTimestampGroupLengthTicks(timestamp_group_length_ticks),
current_timestamp_group_(),
prev_timestamp_group_(),
timestamp_to_ms_coeff_(timestamp_to_ms_coeff),
burst_grouping_(enable_burst_grouping) {}
bool InterArrival::ComputeDeltas(uint32_t timestamp,
int64_t arrival_time_ms,
size_t packet_size,
uint32_t* timestamp_delta,
int64_t* arrival_time_delta_ms,
int* packet_size_delta) {
assert(timestamp_delta != NULL);
assert(arrival_time_delta_ms != NULL);
assert(packet_size_delta != NULL);
bool calculated_deltas = false;
if (current_timestamp_group_.IsFirstPacket()) {
// We don't have enough data to update the filter, so we store it until we
// have two frames of data to process.
current_timestamp_group_.timestamp = timestamp;
current_timestamp_group_.first_timestamp = timestamp;
} else if (!PacketInOrder(timestamp)) {
return false;
} else if (NewTimestampGroup(arrival_time_ms, timestamp)) {
// First packet of a later frame, the previous frame sample is ready.
if (prev_timestamp_group_.complete_time_ms >= 0) {
*timestamp_delta = current_timestamp_group_.timestamp -
prev_timestamp_group_.timestamp;
*arrival_time_delta_ms = current_timestamp_group_.complete_time_ms -
prev_timestamp_group_.complete_time_ms;
if (*arrival_time_delta_ms < 0) {
// The group of packets has been reordered since receiving its local
// arrival timestamp.
LOG(LS_WARNING) << "Packets are being reordered on the path from the "
"socket to the bandwidth estimator. Ignoring this "
"packet for bandwidth estimation.";
return false;
}
assert(*arrival_time_delta_ms >= 0);
*packet_size_delta = static_cast<int>(current_timestamp_group_.size) -
static_cast<int>(prev_timestamp_group_.size);
calculated_deltas = true;
}
prev_timestamp_group_ = current_timestamp_group_;
// The new timestamp is now the current frame.
current_timestamp_group_.first_timestamp = timestamp;
current_timestamp_group_.timestamp = timestamp;
current_timestamp_group_.size = 0;
}
else {
current_timestamp_group_.timestamp = LatestTimestamp(
current_timestamp_group_.timestamp, timestamp);
}
// Accumulate the frame size.
current_timestamp_group_.size += packet_size;
current_timestamp_group_.complete_time_ms = arrival_time_ms;
return calculated_deltas;
}
bool InterArrival::PacketInOrder(uint32_t timestamp) {
if (current_timestamp_group_.IsFirstPacket()) {
return true;
} else {
// Assume that a diff which is bigger than half the timestamp interval
// (32 bits) must be due to reordering. This code is almost identical to
// that in IsNewerTimestamp() in module_common_types.h.
uint32_t timestamp_diff = timestamp -
current_timestamp_group_.first_timestamp;
return timestamp_diff < 0x80000000;
}
}
// Assumes that |timestamp| is not reordered compared to
// |current_timestamp_group_|.
bool InterArrival::NewTimestampGroup(int64_t arrival_time_ms,
uint32_t timestamp) const {
if (current_timestamp_group_.IsFirstPacket()) {
return false;
} else if (BelongsToBurst(arrival_time_ms, timestamp)) {
return false;
} else {
uint32_t timestamp_diff = timestamp -
current_timestamp_group_.first_timestamp;
return timestamp_diff > kTimestampGroupLengthTicks;
}
}
bool InterArrival::BelongsToBurst(int64_t arrival_time_ms,
uint32_t timestamp) const {
if (!burst_grouping_) {
return false;
}
assert(current_timestamp_group_.complete_time_ms >= 0);
int64_t arrival_time_delta_ms = arrival_time_ms -
current_timestamp_group_.complete_time_ms;
uint32_t timestamp_diff = timestamp - current_timestamp_group_.timestamp;
int64_t ts_delta_ms = timestamp_to_ms_coeff_ * timestamp_diff + 0.5;
if (ts_delta_ms == 0)
return true;
int propagation_delta_ms = arrival_time_delta_ms - ts_delta_ms;
return propagation_delta_ms < 0 &&
arrival_time_delta_ms <= kBurstDeltaThresholdMs;
}
} // namespace webrtc