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stats: more consistent use of has_value() for optionals

Browse source 
replacing
  if (optional) { ...}
with the more explicit
  if (optional.has_value()) { ... }

No functional changes.

BUG=None

Change-Id: I005fd3df307880b07cfda0cbe435efb0e0717a88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281362
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38544}
This commit is contained in:
Philipp Hancke 2022-11-02 16:08:16 +01:00 committed by WebRTC LUCI CQ
parent 3e7e15d240
commit 42e5ed38a7
1 changed files with 19 additions and 19 deletions
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38
pc/rtc_stats_collector.cc
View file

@ -380,7 +380,7 @@ std::string GetCodecIdAndMaybeCreateCodecStats(
std::make_unique<RTCCodecStats>(codec_id, timestamp_us));
codec_stats->payload_type = payload_type;
codec_stats->mime_type = codec_params.mime_type();
if (codec_params.clock_rate) {
if (codec_params.clock_rate.has_value()) {
codec_stats->clock_rate = static_cast<uint32_t>(*codec_params.clock_rate);
}
if (codec_params.num_channels) {
@ -419,17 +419,17 @@ void SetInboundRTPStreamStatsFromMediaReceiverInfo(
static_cast<int32_t>(media_receiver_info.packets_lost);
inbound_stats->jitter_buffer_delay =
media_receiver_info.jitter_buffer_delay_seconds;
if (media_receiver_info.jitter_buffer_target_delay_seconds) {
if (media_receiver_info.jitter_buffer_target_delay_seconds.has_value()) {
inbound_stats->jitter_buffer_target_delay =
*media_receiver_info.jitter_buffer_target_delay_seconds;
}
if (media_receiver_info.jitter_buffer_minimum_delay_seconds) {
if (media_receiver_info.jitter_buffer_minimum_delay_seconds.has_value()) {
inbound_stats->jitter_buffer_minimum_delay =
*media_receiver_info.jitter_buffer_minimum_delay_seconds;
}
inbound_stats->jitter_buffer_emitted_count =
media_receiver_info.jitter_buffer_emitted_count;
if (media_receiver_info.nacks_sent) {
if (media_receiver_info.nacks_sent.has_value()) {
inbound_stats->nack_count = *media_receiver_info.nacks_sent;
}
}
@ -482,11 +482,11 @@ std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
voice_receiver_info.total_output_duration;
// `fir_count`, `pli_count` and `sli_count` are only valid for video and are
// purposefully left undefined for audio.
if (voice_receiver_info.last_packet_received_timestamp_ms) {
if (voice_receiver_info.last_packet_received_timestamp_ms.has_value()) {
inbound_audio->last_packet_received_timestamp = static_cast<double>(
*voice_receiver_info.last_packet_received_timestamp_ms);
}
if (voice_receiver_info.estimated_playout_ntp_timestamp_ms) {
if (voice_receiver_info.estimated_playout_ntp_timestamp_ms.has_value()) {
// TODO(bugs.webrtc.org/10529): Fix time origin.
inbound_audio->estimated_playout_timestamp = static_cast<double>(
*voice_receiver_info.estimated_playout_ntp_timestamp_ms);
@ -551,7 +551,7 @@ CreateRemoteOutboundAudioStreamStats(
stats->remote_timestamp = static_cast<double>(
voice_receiver_info.last_sender_report_remote_timestamp_ms.value());
stats->reports_sent = voice_receiver_info.sender_reports_reports_count;
if (voice_receiver_info.round_trip_time) {
if (voice_receiver_info.round_trip_time.has_value()) {
stats->round_trip_time =
voice_receiver_info.round_trip_time->seconds<double>();
}
@ -607,7 +607,7 @@ void SetInboundRTPStreamStatsFromVideoReceiverInfo(
if (video_receiver_info.framerate_decoded > 0) {
inbound_video->frames_per_second = video_receiver_info.framerate_decoded;
}
if (video_receiver_info.qp_sum) {
if (video_receiver_info.qp_sum.has_value()) {
inbound_video->qp_sum = *video_receiver_info.qp_sum;
}
if (video_receiver_info.timing_frame_info.has_value()) {
@ -637,11 +637,11 @@ void SetInboundRTPStreamStatsFromVideoReceiverInfo(
inbound_video->min_playout_delay =
static_cast<double>(video_receiver_info.min_playout_delay_ms) /
rtc::kNumMillisecsPerSec;
if (video_receiver_info.last_packet_received_timestamp_ms) {
if (video_receiver_info.last_packet_received_timestamp_ms.has_value()) {
inbound_video->last_packet_received_timestamp = static_cast<double>(
*video_receiver_info.last_packet_received_timestamp_ms);
}
if (video_receiver_info.estimated_playout_ntp_timestamp_ms) {
if (video_receiver_info.estimated_playout_ntp_timestamp_ms.has_value()) {
// TODO(bugs.webrtc.org/10529): Fix time origin if needed.
inbound_video->estimated_playout_timestamp = static_cast<double>(
*video_receiver_info.estimated_playout_ntp_timestamp_ms);
@ -694,7 +694,7 @@ void SetOutboundRTPStreamStatsFromVoiceSenderInfo(
outbound_audio->mid = mid;
outbound_audio->media_type = "audio";
outbound_audio->kind = "audio";
if (voice_sender_info.target_bitrate &&
if (voice_sender_info.target_bitrate.has_value() &&
*voice_sender_info.target_bitrate > 0) {
outbound_audio->target_bitrate = *voice_sender_info.target_bitrate;
}
@ -739,10 +739,10 @@ void SetOutboundRTPStreamStatsFromVideoSenderInfo(
static_cast<uint32_t>(video_sender_info.firs_rcvd);
outbound_video->pli_count =
static_cast<uint32_t>(video_sender_info.plis_rcvd);
if (video_sender_info.qp_sum)
if (video_sender_info.qp_sum.has_value())
outbound_video->qp_sum = *video_sender_info.qp_sum;
if (video_sender_info.target_bitrate &&
video_sender_info.target_bitrate > 0) {
if (video_sender_info.target_bitrate.has_value() &&
*video_sender_info.target_bitrate > 0) {
outbound_video->target_bitrate = *video_sender_info.target_bitrate;
}
outbound_video->frames_encoded = video_sender_info.frames_encoded;
@ -781,7 +781,7 @@ void SetOutboundRTPStreamStatsFromVideoSenderInfo(
outbound_video->encoder_implementation =
video_sender_info.encoder_implementation_name;
}
if (video_sender_info.rid) {
if (video_sender_info.rid.has_value()) {
outbound_video->rid = *video_sender_info.rid;
}
}
@ -971,10 +971,10 @@ const std::string& ProduceIceCandidateStats(int64_t timestamp_us,
template <typename StatsType>
void SetAudioProcessingStats(StatsType* stats,
const AudioProcessingStats& apm_stats) {
if (apm_stats.echo_return_loss) {
if (apm_stats.echo_return_loss.has_value()) {
stats->echo_return_loss = *apm_stats.echo_return_loss;
}
if (apm_stats.echo_return_loss_enhancement) {
if (apm_stats.echo_return_loss_enhancement.has_value()) {
stats->echo_return_loss_enhancement =
*apm_stats.echo_return_loss_enhancement;
}
@ -1694,7 +1694,7 @@ void RTCStatsCollector::ProduceIceCandidateAndPairStats_n(
candidate_pair_stats->total_round_trip_time =
static_cast<double>(info.total_round_trip_time_ms) /
rtc::kNumMillisecsPerSec;
if (info.current_round_trip_time_ms) {
if (info.current_round_trip_time_ms.has_value()) {
candidate_pair_stats->current_round_trip_time =
static_cast<double>(*info.current_round_trip_time_ms) /
rtc::kNumMillisecsPerSec;
@ -1727,7 +1727,7 @@ void RTCStatsCollector::ProduceIceCandidateAndPairStats_n(
info.sent_ping_requests_total -
info.sent_ping_requests_before_first_response);
if (info.last_data_received) {
if (info.last_data_received.has_value()) {
candidate_pair_stats->last_packet_received_timestamp =
static_cast<double>(info.last_data_received->ms());
}