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Add audio level to CSRC class

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This patch adds (optional) csrc to ContributingSources.
This will be used if using virtual audio ssrc, since
the audio level is otherwise unaccessible in that configuration.

BUG=webrtc:3333

Change-Id: Ied263b8f0850553cd637fd6bead373ed4252fd1e
Reviewed-on: https://webrtc-review.googlesource.com/c/109281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25516}
This commit is contained in:
Jonas Oreland
2018-11-06 07:35:06 +01:00
committed by Commit Bot
parent df351f4b54
commit 967f7d5497
5 changed files with 64 additions and 22 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
audio/channel_receive.cc
View File

@ -437,7 +437,9 @@ void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
if (has_audio_level)
last_received_rtp_audio_level_ = audio_level;
std::vector<uint32_t> csrcs = packet.Csrcs();
contributing_sources_.Update(now_ms, csrcs);
contributing_sources_.Update(
now_ms, csrcs,
has_audio_level ? absl::optional<uint8_t>(audio_level) : absl::nullopt);
}
// Store playout timestamp for the received RTP packet

30
modules/rtp_rtcp/source/contributing_sources.cc
View File

@ -25,9 +25,11 @@ ContributingSources::ContributingSources() = default;
ContributingSources::~ContributingSources() = default;
void ContributingSources::Update(int64_t now_ms,
rtc::ArrayView<const uint32_t> csrcs) {
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level) {
Entry entry = { now_ms, audio_level };
for (uint32_t csrc : csrcs) {
last_seen_ms_[csrc] = now_ms;
active_csrcs_[csrc] = entry;
}
if (!next_pruning_ms_) {
next_pruning_ms_ = now_ms + kPruningIntervalMs;
@ -43,9 +45,16 @@ void ContributingSources::Update(int64_t now_ms,
// non-const.
std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
std::vector<RtpSource> sources;
for (auto& record : last_seen_ms_) {
if (record.second >= now_ms - kHistoryMs) {
sources.emplace_back(record.second, record.first, RtpSourceType::CSRC);
for (auto& record : active_csrcs_) {
if (record.second.last_seen_ms >= now_ms - kHistoryMs) {
if (record.second.audio_level.has_value()) {
sources.emplace_back(record.second.last_seen_ms, record.first,
RtpSourceType::CSRC,
*record.second.audio_level);
} else {
sources.emplace_back(record.second.last_seen_ms, record.first,
RtpSourceType::CSRC);
}
}
}
@ -54,15 +63,20 @@ std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
// Delete stale entries.
void ContributingSources::DeleteOldEntries(int64_t now_ms) {
for (auto it = last_seen_ms_.begin(); it != last_seen_ms_.end();) {
if (it->second >= now_ms - kHistoryMs) {
for (auto it = active_csrcs_.begin(); it != active_csrcs_.end();) {
if (it->second.last_seen_ms >= now_ms - kHistoryMs) {
// Still relevant.
++it;
} else {
it = last_seen_ms_.erase(it);
it = active_csrcs_.erase(it);
}
}
next_pruning_ms_ = now_ms + kPruningIntervalMs;
}
ContributingSources::Entry::Entry() = default;
ContributingSources::Entry::Entry(int64_t timestamp_ms,
absl::optional<uint8_t> audio_level_arg)
: last_seen_ms(timestamp_ms), audio_level(audio_level_arg) {}
} // namespace webrtc

15
modules/rtp_rtcp/source/contributing_sources.h
View File

@ -32,18 +32,25 @@ class ContributingSources {
ContributingSources();
~ContributingSources();
// TODO(bugs.webrtc.org/3333): Needs to be extended with audio-level, to
// support RFC6465.
void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs);
void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level);
// Returns contributing sources seen the last 10 s.
std::vector<RtpSource> GetSources(int64_t now_ms) const;
private:
struct Entry {
Entry();
Entry(int64_t timestamp_ms, absl::optional<uint8_t> audio_level);
int64_t last_seen_ms;
absl::optional<uint8_t> audio_level;
};
void DeleteOldEntries(int64_t now_ms);
// Indexed by csrc.
std::map<uint32_t, int64_t> last_seen_ms_;
std::map<uint32_t, Entry> active_csrcs_;
absl::optional<int64_t> next_pruning_ms_;
};

34
modules/rtp_rtcp/source/contributing_sources_unittest.cc
View File

@ -30,7 +30,7 @@ TEST(ContributingSourcesTest, RecordSources) {
ContributingSources csrcs;
constexpr uint32_t kCsrcs[] = {kCsrc1, kCsrc2};
constexpr int64_t kTime1 = 10;
csrcs.Update(kTime1, kCsrcs);
csrcs.Update(kTime1, kCsrcs, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime1),
UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
@ -45,12 +45,12 @@ TEST(ContributingSourcesTest, UpdateSources) {
constexpr uint32_t kCsrcs2[] = {kCsrc2, kCsrc3};
constexpr int64_t kTime1 = 10;
constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
csrcs.Update(kTime1, kCsrcs1);
csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime1),
UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
csrcs.Update(kTime2, kCsrcs2);
csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime2),
UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
@ -65,12 +65,12 @@ TEST(ContributingSourcesTest, ReturnRecentOnly) {
constexpr int64_t kTime1 = 10;
constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
constexpr int64_t kTime3 = kTime1 + 12 * rtc::kNumMillisecsPerSec;
csrcs.Update(kTime1, kCsrcs1);
csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime1),
UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
csrcs.Update(kTime2, kCsrcs2);
csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime3),
UnorderedElementsAre(RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
@ -84,18 +84,18 @@ TEST(ContributingSourcesTest, PurgeOldSources) {
constexpr int64_t kTime1 = 10;
constexpr int64_t kTime2 = kTime1 + 10 * rtc::kNumMillisecsPerSec;
constexpr int64_t kTime3 = kTime1 + 20 * rtc::kNumMillisecsPerSec;
csrcs.Update(kTime1, kCsrcs1);
csrcs.Update(kTime1, kCsrcs1, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime2),
UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
csrcs.Update(kTime2, kCsrcs2);
csrcs.Update(kTime2, kCsrcs2, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime2),
UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
RtpSource(kTime2, kCsrc3, RtpSourceType::CSRC)));
csrcs.Update(kTime3, kCsrcs2);
csrcs.Update(kTime3, kCsrcs2, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime3),
UnorderedElementsAre(RtpSource(kTime3, kCsrc2, RtpSourceType::CSRC),
@ -108,4 +108,22 @@ TEST(ContributingSourcesTest, PurgeOldSources) {
RtpSource(kTime3, kCsrc3, RtpSourceType::CSRC)));
}
TEST(ContributingSourcesTest, AudioLevel) {
ContributingSources csrcs;
constexpr uint32_t kCsrcs[] = {kCsrc1, kCsrc2};
constexpr int64_t kTime1 = 10;
csrcs.Update(kTime1, kCsrcs, 47);
EXPECT_THAT(
csrcs.GetSources(kTime1),
UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC, 47),
RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC, 47)));
constexpr uint32_t kCsrcsSubset[] = {kCsrc1};
csrcs.Update(kTime1 + 1, kCsrcsSubset, absl::nullopt);
EXPECT_THAT(
csrcs.GetSources(kTime1 + 1),
UnorderedElementsAre(RtpSource(kTime1 + 1, kCsrc1, RtpSourceType::CSRC),
RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC, 47)));
}
} // namespace webrtc

3
video/rtp_video_stream_receiver.cc
View File

@ -320,7 +320,8 @@ void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
last_received_rtp_system_time_ms_ = now_ms;
std::vector<uint32_t> csrcs = packet.Csrcs();
contributing_sources_.Update(now_ms, csrcs);
contributing_sources_.Update(now_ms, csrcs,
/* audio level */absl::nullopt);
}
// Periodically log the RTP header of incoming packets.
if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {