Change StatsCollector to use DCHECK instead of ASSERT.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46579004
Cr-Commit-Position: refs/heads/master@{#8729}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8729 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/statscollector.cc b/talk/app/webrtc/statscollector.cc
index 0567c3f..7dfc3ea 100644
--- a/talk/app/webrtc/statscollector.cc
+++ b/talk/app/webrtc/statscollector.cc
@@ -32,6 +32,7 @@
#include "talk/session/media/channel.h"
#include "webrtc/base/base64.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/timing.h"
@@ -248,7 +249,7 @@
double stats_gathering_started,
PeerConnectionInterface::StatsOutputLevel level,
StatsReport* report) {
- ASSERT(report->type() == StatsReport::kStatsReportTypeBwe);
+ DCHECK(report->type() == StatsReport::kStatsReportTypeBwe);
report->set_timestamp(stats_gathering_started);
const IntForAdd ints[] = {
@@ -319,7 +320,7 @@
if (candidate_type == cricket::RELAY_PORT_TYPE) {
return STATSREPORT_RELAY_PORT_TYPE;
}
- ASSERT(false);
+ DCHECK(false);
return "unknown";
}
@@ -338,7 +339,7 @@
case rtc::ADAPTER_TYPE_LOOPBACK:
return STATSREPORT_ADAPTER_TYPE_LOOPBACK;
default:
- ASSERT(false);
+ DCHECK(false);
return "";
}
}
@@ -346,11 +347,11 @@
StatsCollector::StatsCollector(WebRtcSession* session)
: session_(session),
stats_gathering_started_(0) {
- ASSERT(session_);
+ DCHECK(session_);
}
StatsCollector::~StatsCollector() {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
}
double StatsCollector::GetTimeNow() {
@@ -360,8 +361,8 @@
// Adds a MediaStream with tracks that can be used as a |selector| in a call
// to GetStats.
void StatsCollector::AddStream(MediaStreamInterface* stream) {
- ASSERT(session_->signaling_thread()->IsCurrent());
- ASSERT(stream != NULL);
+ DCHECK(session_->signaling_thread()->IsCurrent());
+ DCHECK(stream != NULL);
CreateTrackReports<AudioTrackVector>(stream->GetAudioTracks(),
&reports_);
@@ -371,11 +372,11 @@
void StatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track,
uint32 ssrc) {
- ASSERT(session_->signaling_thread()->IsCurrent());
- ASSERT(audio_track != NULL);
-#if ENABLE_DEBUG
+ DCHECK(session_->signaling_thread()->IsCurrent());
+ DCHECK(audio_track != NULL);
+#if (!defined(NDEBUG) || defined(DCHECK_ALWAYS_ON))
for (const auto& track : local_audio_tracks_)
- ASSERT(track.first != audio_track || track.second != ssrc);
+ DCHECK(track.first != audio_track || track.second != ssrc);
#endif
local_audio_tracks_.push_back(std::make_pair(audio_track, ssrc));
@@ -393,7 +394,7 @@
void StatsCollector::RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
uint32 ssrc) {
- ASSERT(audio_track != NULL);
+ DCHECK(audio_track != NULL);
local_audio_tracks_.erase(std::remove_if(local_audio_tracks_.begin(),
local_audio_tracks_.end(),
[audio_track, ssrc](const LocalAudioTrackVector::value_type& track) {
@@ -403,9 +404,9 @@
void StatsCollector::GetStats(MediaStreamTrackInterface* track,
StatsReports* reports) {
- ASSERT(session_->signaling_thread()->IsCurrent());
- ASSERT(reports != NULL);
- ASSERT(reports->empty());
+ DCHECK(session_->signaling_thread()->IsCurrent());
+ DCHECK(reports != NULL);
+ DCHECK(reports->empty());
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
@@ -443,7 +444,7 @@
void
StatsCollector::UpdateStats(PeerConnectionInterface::StatsOutputLevel level) {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
double time_now = GetTimeNow();
// Calls to UpdateStats() that occur less than kMinGatherStatsPeriod number of
// ms apart will be ignored.
@@ -473,7 +474,7 @@
uint32 ssrc,
const StatsReport::Id& transport_id,
StatsReport::Direction direction) {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
StatsReport::Id id(StatsReport::NewIdWithDirection(
local ? StatsReport::kStatsReportTypeSsrc :
StatsReport::kStatsReportTypeRemoteSsrc,
@@ -512,7 +513,7 @@
StatsReport* StatsCollector::AddOneCertificateReport(
const rtc::SSLCertificate* cert, const StatsReport* issuer) {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
// TODO(bemasc): Move this computation to a helper class that caches these
// values to reduce CPU use in GetStats. This will require adding a fast
@@ -555,13 +556,13 @@
StatsReport* StatsCollector::AddCertificateReports(
const rtc::SSLCertificate* cert) {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
// Produces a chain of StatsReports representing this certificate and the rest
// of its chain, and adds those reports to |reports_|. The return value is
// the id of the leaf report. The provided cert must be non-null, so at least
// one report will always be provided and the returned string will never be
// empty.
- ASSERT(cert != NULL);
+ DCHECK(cert != NULL);
StatsReport* issuer = nullptr;
rtc::scoped_ptr<rtc::SSLCertChain> chain;
@@ -655,7 +656,7 @@
}
void StatsCollector::ExtractSessionInfo() {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
// Extract information from the base session.
StatsReport::Id id(StatsReport::NewTypedId(
@@ -747,7 +748,7 @@
}
void StatsCollector::ExtractVoiceInfo() {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
if (!session_->voice_channel()) {
return;
@@ -780,7 +781,7 @@
void StatsCollector::ExtractVideoInfo(
PeerConnectionInterface::StatsOutputLevel level) {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
if (!session_->video_channel())
return;
@@ -817,7 +818,7 @@
}
void StatsCollector::ExtractDataInfo() {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
@@ -838,14 +839,14 @@
StatsReport* StatsCollector::GetReport(const StatsReport::StatsType& type,
const std::string& id,
StatsReport::Direction direction) {
- ASSERT(session_->signaling_thread()->IsCurrent());
- ASSERT(type == StatsReport::kStatsReportTypeSsrc ||
+ DCHECK(session_->signaling_thread()->IsCurrent());
+ DCHECK(type == StatsReport::kStatsReportTypeSsrc ||
type == StatsReport::kStatsReportTypeRemoteSsrc);
return reports_.Find(StatsReport::NewIdWithDirection(type, id, direction));
}
void StatsCollector::UpdateStatsFromExistingLocalAudioTracks() {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
// Loop through the existing local audio tracks.
for (const auto& it : local_audio_tracks_) {
AudioTrackInterface* track = it.first;
@@ -872,8 +873,8 @@
void StatsCollector::UpdateReportFromAudioTrack(AudioTrackInterface* track,
StatsReport* report) {
- ASSERT(session_->signaling_thread()->IsCurrent());
- ASSERT(track != NULL);
+ DCHECK(session_->signaling_thread()->IsCurrent());
+ DCHECK(track != NULL);
int signal_level = 0;
if (!track->GetSignalLevel(&signal_level))
@@ -894,7 +895,7 @@
bool StatsCollector::GetTrackIdBySsrc(uint32 ssrc, std::string* track_id,
StatsReport::Direction direction) {
- ASSERT(session_->signaling_thread()->IsCurrent());
+ DCHECK(session_->signaling_thread()->IsCurrent());
if (direction == StatsReport::kSend) {
if (!session_->GetLocalTrackIdBySsrc(ssrc, track_id)) {
LOG(LS_WARNING) << "The SSRC " << ssrc
@@ -902,7 +903,7 @@
return false;
}
} else {
- ASSERT(direction == StatsReport::kReceive);
+ DCHECK(direction == StatsReport::kReceive);
if (!session_->GetRemoteTrackIdBySsrc(ssrc, track_id)) {
LOG(LS_WARNING) << "The SSRC " << ssrc
<< " is not associated with a receiving track";
diff --git a/talk/app/webrtc/statstypes.cc b/talk/app/webrtc/statstypes.cc
index cc6eaa0..bca7280 100644
--- a/talk/app/webrtc/statstypes.cc
+++ b/talk/app/webrtc/statstypes.cc
@@ -37,7 +37,7 @@
// generate strongly typed inline C++ code that forces the correct type to be
// used for a given name at compile time.
- using rtc::RefCountedObject;
+using rtc::RefCountedObject;
namespace webrtc {
namespace {
