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Event logs - separate audio_level and voice_activity

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Bug: webrtc:8111
Change-Id: I44d81c5b4f5b854e8accd84521fbbd7b50228903
Reviewed-on: https://webrtc-review.googlesource.com/c/109571
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25589}
This commit is contained in:
Elad Alon 2018-11-09 21:46:43 +01:00 committed by Commit Bot
parent 466620b326
commit eb809f30d1
3 changed files with 66 additions and 18 deletions
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42
logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
View File

@ -239,13 +239,6 @@ rtclog2::IceCandidatePairEvent::IceCandidatePairEventType ConvertToProtoFormat(
return rtclog2::IceCandidatePairEvent::UNKNOWN_CHECK_TYPE;
}
uint8_t ConvertAudioLevelToProtoFormat(bool voice_activity,
uint8_t audio_level) {
RTC_DCHECK_EQ(audio_level & static_cast<uint8_t>(0x80), 0);
constexpr uint8_t kVoiceActivityBit = 0x80;
return audio_level | (voice_activity ? kVoiceActivityBit : 0);
}
// Copies all RTCP blocks except APP, SDES and unknown from |packet| to
// |buffer|. |buffer| must have space for |IP_PACKET_SIZE| bytes. |packet| must
// be at most |IP_PACKET_SIZE| bytes long.
@ -338,8 +331,10 @@ void EncodeRtcpPacket(rtc::ArrayView<const EventType*> batch,
static_assert(sizeof(std::string::value_type) == sizeof(uint8_t), "");
const size_t buffer_length = RemoveNonWhitelistedRtcpBlocks(
event->packet_, reinterpret_cast<uint8_t*>(&scrubed_packets[i][0]));
if (buffer_length < event->packet_.size()) {
scrubed_packets[i].resize(buffer_length);
}
}
proto_batch->set_raw_packet_blobs(EncodeBlobs(scrubed_packets));
}
@ -405,17 +400,19 @@ void EncodeRtpPacket(const std::vector<const EventType*>& batch,
}
}
// TODO(eladalon): Separate audio level from voice activity.
absl::optional<uint64_t> base_audio_level;
absl::optional<uint64_t> base_voice_activity;
{
bool voice_activity;
uint8_t audio_level;
if (base_event->header_.template GetExtension<AudioLevel>(&voice_activity,
&audio_level)) {
proto_batch->set_audio_level(
ConvertAudioLevelToProtoFormat(voice_activity, audio_level));
base_audio_level =
ConvertAudioLevelToProtoFormat(voice_activity, audio_level);
RTC_DCHECK_LE(audio_level, 0x7Fu);
base_audio_level = audio_level;
proto_batch->set_audio_level(audio_level);
base_voice_activity = voice_activity;
proto_batch->set_voice_activity(voice_activity);
}
}
@ -587,7 +584,8 @@ void EncodeRtpPacket(const std::vector<const EventType*>& batch,
uint8_t audio_level;
if (event->header_.template GetExtension<AudioLevel>(&voice_activity,
&audio_level)) {
values[i] = ConvertAudioLevelToProtoFormat(voice_activity, audio_level);
RTC_DCHECK_LE(audio_level, 0x7Fu);
values[i] = audio_level;
} else {
values[i].reset();
}
@ -596,6 +594,24 @@ void EncodeRtpPacket(const std::vector<const EventType*>& batch,
if (!encoded_deltas.empty()) {
proto_batch->set_audio_level_deltas(encoded_deltas);
}
// voice_activity (RTP extension)
for (size_t i = 0; i < values.size(); ++i) {
const EventType* event = batch[i + 1];
bool voice_activity;
uint8_t audio_level;
if (event->header_.template GetExtension<AudioLevel>(&voice_activity,
&audio_level)) {
RTC_DCHECK_LE(audio_level, 0x7Fu);
values[i] = voice_activity;
} else {
values[i].reset();
}
}
encoded_deltas = EncodeDeltas(base_voice_activity, values);
if (!encoded_deltas.empty()) {
proto_batch->set_voice_activity_deltas(encoded_deltas);
}
}
} // namespace

8
logging/rtc_event_log/rtc_event_log2.proto
View File

@ -86,7 +86,9 @@ message IncomingRtpPackets {
optional int32 transmission_time_offset = 16;
optional uint32 absolute_send_time = 17;
optional uint32 video_rotation = 18;
// |audio_level| and |voice_activity| are always used in conjunction.
optional uint32 audio_level = 19;
optional bool voice_activity = 20;
// TODO(terelius): Add other header extensions like playout delay?
// Delta encodings.
@ -105,7 +107,9 @@ message IncomingRtpPackets {
optional bytes transmission_time_offset_deltas = 116;
optional bytes absolute_send_time_deltas = 117;
optional bytes video_rotation_deltas = 118;
// |audio_level| and |voice_activity| are always used in conjunction.
optional bytes audio_level_deltas = 119;
optional bytes voice_activity_deltas = 120;
}
message OutgoingRtpPackets {
@ -150,7 +154,9 @@ message OutgoingRtpPackets {
optional int32 transmission_time_offset = 16;
optional uint32 absolute_send_time = 17;
optional uint32 video_rotation = 18;
// |audio_level| and |voice_activity| are always used in conjunction.
optional uint32 audio_level = 19;
optional bool voice_activity = 20;
// TODO(terelius): Add other header extensions like playout delay?
// Delta encodings.
@ -169,7 +175,9 @@ message OutgoingRtpPackets {
optional bytes transmission_time_offset_deltas = 116;
optional bytes absolute_send_time_deltas = 117;
optional bytes video_rotation_deltas = 118;
// |audio_level| and |voice_activity| are always used in conjunction.
optional bytes audio_level_deltas = 119;
optional bytes voice_activity_deltas = 120;
}
message IncomingRtcpPackets {

32
logging/rtc_event_log/rtc_event_log_parser_new.cc
View File

@ -497,11 +497,16 @@ void StoreRtpPackets(
rtc::checked_cast<uint8_t>(proto.video_rotation()));
}
if (proto.has_audio_level()) {
RTC_CHECK(proto.has_voice_activity());
header.extension.hasAudioLevel = true;
header.extension.voiceActivity =
rtc::checked_cast<bool>(proto.voice_activity());
const uint8_t audio_level =
rtc::checked_cast<uint8_t>(proto.audio_level());
header.extension.voiceActivity = (audio_level >> 7) != 0;
header.extension.audioLevel = audio_level & 0x7Fu;
RTC_CHECK_LE(audio_level, 0x7Fu);
header.extension.audioLevel = audio_level;
} else {
RTC_CHECK(!proto.has_voice_activity());
}
(*rtp_packets_map)[header.ssrc].emplace_back(
proto.timestamp_ms() * 1000, header, proto.header_size(),
@ -620,6 +625,17 @@ void StoreRtpPackets(
RTC_CHECK_EQ(audio_level_values.size(), number_of_deltas);
}
// voice_activity (RTP extension)
std::vector<absl::optional<uint64_t>> voice_activity_values;
{
const absl::optional<uint64_t> base_voice_activity =
proto.has_voice_activity() ? proto.voice_activity()
: absl::optional<uint64_t>();
voice_activity_values = DecodeDeltas(proto.voice_activity_deltas(),
base_voice_activity, number_of_deltas);
RTC_CHECK_EQ(voice_activity_values.size(), number_of_deltas);
}
// Delta decoding
for (size_t i = 0; i < number_of_deltas; ++i) {
RTC_CHECK(timestamp_ms_values[i].has_value());
@ -670,11 +686,18 @@ void StoreRtpPackets(
rtc::checked_cast<uint8_t>(video_rotation_values[i].value()));
}
if (audio_level_values.size() > i && audio_level_values[i].has_value()) {
RTC_CHECK(voice_activity_values.size() > i &&
voice_activity_values[i].has_value());
header.extension.hasAudioLevel = true;
header.extension.voiceActivity =
rtc::checked_cast<bool>(voice_activity_values[i].value());
const uint8_t audio_level =
rtc::checked_cast<uint8_t>(audio_level_values[i].value());
header.extension.voiceActivity = (audio_level >> 7) != 0;
header.extension.audioLevel = audio_level & 0x7Fu;
RTC_CHECK_LE(audio_level, 0x7Fu);
header.extension.audioLevel = audio_level;
} else {
RTC_CHECK(voice_activity_values.size() <= i ||
!voice_activity_values[i].has_value());
}
(*rtp_packets_map)[header.ssrc].emplace_back(
timestamp_ms_values[i].value() * 1000, header, header.headerLength,
@ -2300,6 +2323,7 @@ void ParsedRtcEventLogNew::StoreAudioNetworkAdaptationEvent(
runtime_config.enable_dtx = proto.enable_dtx();
}
if (proto.has_num_channels()) {
// TODO(eladalon): Encode 1/2 -> 0/1, to improve
runtime_config.num_channels = proto.num_channels();
}
audio_network_adaptation_events_.emplace_back(1000 * proto.timestamp_ms(),