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/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
#include "MediaConduitInterface.h"
#include "nsTArray.h"
#include "mozilla/Assertions.h"
#include "MainThreadUtils.h"
#include "SystemTime.h"
#include "system_wrappers/include/clock.h"
namespace mozilla {
void MediaSessionConduit::GetRtpSources(
nsTArray<dom::RTCRtpSourceEntry>& outSources) const {
MOZ_ASSERT(NS_IsMainThread());
if (mSourcesUpdateNeeded) {
UpdateRtpSources(GetUpstreamRtpSources());
OnSourcesUpdated();
}
outSources.Clear();
for (auto& [key, entry] : mSourcesCache) {
(void)key;
outSources.AppendElement(entry);
}
struct TimestampComparator {
bool LessThan(const dom::RTCRtpSourceEntry& aLhs,
const dom::RTCRtpSourceEntry& aRhs) const {
// Sort descending!
return aLhs.mTimestamp > aRhs.mTimestamp;
}
bool Equals(const dom::RTCRtpSourceEntry& aLhs,
const dom::RTCRtpSourceEntry& aRhs) const {
return aLhs.mTimestamp == aRhs.mTimestamp;
}
};
// *sigh* We have to re-sort this by JS timestamp; we can run into cases
// where the libwebrtc timestamps are not in exactly the same order as JS
// timestamps due to clock differences (wibbly-wobbly, timey-wimey stuff)
outSources.Sort(TimestampComparator());
}
static double rtpToDomAudioLevel(uint8_t aAudioLevel) {
if (aAudioLevel == 127) {
// Spec indicates that a value of 127 should be set to 0
return 0;
}
// All other values are calculated as 10^(-rfc_level/20)
return std::pow(10, -aAudioLevel / 20.0);
}
void MediaSessionConduit::UpdateRtpSources(
const std::vector<webrtc::RtpSource>& aSources) const {
MOZ_ASSERT(NS_IsMainThread());
// Empty out the cache; we'll copy things back as needed
auto cache = std::move(mSourcesCache);
for (const auto& source : aSources) {
SourceKey key(source);
auto it = cache.find(key);
if (it != cache.end()) {
// This source entry was already in the cache, and should continue to be
// present in exactly the same form as before. This means we do _not_
// want to perform the timestamp adjustment again, since it might yield a
// slightly different result. This is why we copy this entry from the old
// cache instead of simply rebuilding it, and is also why we key the
// cache based on timestamp (keying the cache based on timestamp also
// gets us the ordering we want, conveniently).
mSourcesCache[key] = it->second;
continue;
}
// This is something we did not already have in the cache.
dom::RTCRtpSourceEntry domEntry;
domEntry.mSource = source.source_id();
switch (source.source_type()) {
case webrtc::RtpSourceType::SSRC:
domEntry.mSourceType = dom::RTCRtpSourceEntryType::Synchronization;
break;
case webrtc::RtpSourceType::CSRC:
domEntry.mSourceType = dom::RTCRtpSourceEntryType::Contributing;
break;
default:
MOZ_CRASH("Unexpected RTCRtpSourceEntryType");
}
if (source.audio_level()) {
domEntry.mAudioLevel.Construct(rtpToDomAudioLevel(*source.audio_level()));
}
// These timestamps are always **rounded** to milliseconds. That means they
// can jump up to half a millisecond into the future. We compensate for that
// here so that things seem consistent to js.
domEntry.mTimestamp =
dom::RTCStatsTimestamp::FromRealtime(
GetTimestampMaker(),
webrtc::Timestamp::Millis(source.timestamp().ms()) -
webrtc::TimeDelta::Micros(500))
.ToDom();
domEntry.mRtpTimestamp = source.rtp_timestamp();
mSourcesCache[key] = domEntry;
}
}
void MediaSessionConduit::OnSourcesUpdated() const {
MOZ_ASSERT(NS_IsMainThread());
MOZ_ASSERT(mSourcesUpdateNeeded);
mSourcesUpdateNeeded = false;
// Reset the updateNeeded flag and clear the cache in a direct task, i.e.,
// as soon as the current task has finished.
AbstractThread::GetCurrent()->TailDispatcher().AddDirectTask(
NS_NewRunnableFunction(
__func__, [this, self = RefPtr<const MediaSessionConduit>(this)] {
mSourcesUpdateNeeded = true;
mSourcesCache.clear();
}));
}
void MediaSessionConduit::InsertAudioLevelForContributingSource(
const uint32_t aCsrcSource, const int64_t aTimestamp,
const uint32_t aRtpTimestamp, const bool aHasAudioLevel,
const uint8_t aAudioLevel) {
MOZ_ASSERT(NS_IsMainThread());
if (mSourcesUpdateNeeded) {
OnSourcesUpdated();
}
dom::RTCRtpSourceEntry domEntry;
domEntry.mSource = aCsrcSource;
domEntry.mSourceType = dom::RTCRtpSourceEntryType::Contributing;
domEntry.mTimestamp = aTimestamp;
domEntry.mRtpTimestamp = aRtpTimestamp;
if (aHasAudioLevel) {
domEntry.mAudioLevel.Construct(rtpToDomAudioLevel(aAudioLevel));
}
auto now = GetTimestampMaker().GetNow();
webrtc::Timestamp convertedTimestamp =
now.ToRealtime() - webrtc::TimeDelta::Millis(now.ToDom() - aTimestamp);
SourceKey key(convertedTimestamp.ms<uint32_t>(), aCsrcSource);
mSourcesCache[key] = domEntry;
}
} // namespace mozilla