[-Wshadow] - Fix some warnings.
First CL to try to understand the extent of the cleanup needed in order to remove -Wno-shadow and follow Chromium on enabling this diagnostic. Bug: webrtc:13219 Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35134}
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@ -1171,24 +1171,24 @@ ParsedRtcEventLog::ParseStatus ParsedRtcEventLog::ParseStream(
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const int64_t timestamp_us = incoming.rtcp.timestamp.us();
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const uint8_t* packet_begin = incoming.rtcp.raw_data.data();
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const uint8_t* packet_end = packet_begin + incoming.rtcp.raw_data.size();
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auto status = StoreRtcpBlocks(
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auto store_rtcp_status = StoreRtcpBlocks(
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timestamp_us, packet_begin, packet_end, &incoming_sr_, &incoming_rr_,
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&incoming_xr_, &incoming_remb_, &incoming_nack_, &incoming_fir_,
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&incoming_pli_, &incoming_bye_, &incoming_transport_feedback_,
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&incoming_loss_notification_);
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RTC_RETURN_IF_ERROR(status);
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RTC_RETURN_IF_ERROR(store_rtcp_status);
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}
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for (const auto& outgoing : outgoing_rtcp_packets_) {
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const int64_t timestamp_us = outgoing.rtcp.timestamp.us();
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const uint8_t* packet_begin = outgoing.rtcp.raw_data.data();
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const uint8_t* packet_end = packet_begin + outgoing.rtcp.raw_data.size();
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auto status = StoreRtcpBlocks(
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auto store_rtcp_status = StoreRtcpBlocks(
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timestamp_us, packet_begin, packet_end, &outgoing_sr_, &outgoing_rr_,
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&outgoing_xr_, &outgoing_remb_, &outgoing_nack_, &outgoing_fir_,
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&outgoing_pli_, &outgoing_bye_, &outgoing_transport_feedback_,
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&outgoing_loss_notification_);
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RTC_RETURN_IF_ERROR(status);
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RTC_RETURN_IF_ERROR(store_rtcp_status);
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}
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// Store first and last timestamp events that might happen before the call is
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@ -3165,12 +3165,12 @@ ParsedRtcEventLog::StoreAudioNetworkAdaptationEvent(
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runtime_config.frame_length_ms = signed_frame_length_ms;
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}
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if (uplink_packet_loss_fraction_values[i].has_value()) {
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float uplink_packet_loss_fraction;
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float uplink_packet_loss_fraction2;
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RTC_PARSE_CHECK_OR_RETURN(ParsePacketLossFractionFromProtoFormat(
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rtc::checked_cast<uint32_t>(
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uplink_packet_loss_fraction_values[i].value()),
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&uplink_packet_loss_fraction));
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runtime_config.uplink_packet_loss_fraction = uplink_packet_loss_fraction;
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&uplink_packet_loss_fraction2));
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runtime_config.uplink_packet_loss_fraction = uplink_packet_loss_fraction2;
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}
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if (enable_fec_values[i].has_value()) {
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runtime_config.enable_fec =
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@ -272,9 +272,9 @@ void RtcEventLogSession::WriteVideoRecvConfigs(size_t video_recv_streams,
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} while (SsrcUsed(ssrc, incoming_extensions_));
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RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
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incoming_extensions_.emplace_back(ssrc, extensions);
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auto event = gen_.NewVideoReceiveStreamConfig(ssrc, extensions);
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event_log->Log(event->Copy());
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video_recv_config_list_.push_back(std::move(event));
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auto new_event = gen_.NewVideoReceiveStreamConfig(ssrc, extensions);
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event_log->Log(new_event->Copy());
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video_recv_config_list_.push_back(std::move(new_event));
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}
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}
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@ -1567,19 +1567,19 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) {
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EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
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EXPECT_EQ(absl::nullopt, send_codec_spec.red_payload_type);
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for (int i = 1; i < 32; i++) {
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parameters.codecs[0].params[""] += "/111";
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SetSendParameters(parameters);
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const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
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EXPECT_EQ(111, send_codec_spec.payload_type);
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EXPECT_STRCASEEQ("opus", send_codec_spec.format.name.c_str());
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EXPECT_EQ(112, send_codec_spec.red_payload_type);
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}
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parameters.codecs[0].params[""] += "/111";
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SetSendParameters(parameters);
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const auto& send_codec_spec2 = *GetSendStreamConfig(kSsrcX).send_codec_spec;
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EXPECT_EQ(111, send_codec_spec2.payload_type);
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EXPECT_STRCASEEQ("opus", send_codec_spec2.format.name.c_str());
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EXPECT_EQ(absl::nullopt, send_codec_spec2.red_payload_type);
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EXPECT_EQ(112, send_codec_spec2.red_payload_type);
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}
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parameters.codecs[0].params[""] += "/111";
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SetSendParameters(parameters);
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const auto& send_codec_spec3 = *GetSendStreamConfig(kSsrcX).send_codec_spec;
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EXPECT_EQ(111, send_codec_spec3.payload_type);
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EXPECT_STRCASEEQ("opus", send_codec_spec3.format.name.c_str());
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EXPECT_EQ(absl::nullopt, send_codec_spec3.red_payload_type);
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}
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// Test that WebRtcVoiceEngine reconfigures, rather than recreates its
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@ -1841,16 +1841,16 @@ TEST_F(ApmTest, Process) {
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const int kStatsAggregationFrameNum = 100; // 1 second.
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if (frame_count % kStatsAggregationFrameNum == 0) {
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// Get echo and delay metrics.
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AudioProcessingStats stats = apm_->GetStatistics();
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AudioProcessingStats stats2 = apm_->GetStatistics();
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// Echo metrics.
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const float echo_return_loss = stats.echo_return_loss.value_or(-1.0f);
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const float echo_return_loss = stats2.echo_return_loss.value_or(-1.0f);
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const float echo_return_loss_enhancement =
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stats.echo_return_loss_enhancement.value_or(-1.0f);
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stats2.echo_return_loss_enhancement.value_or(-1.0f);
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const float residual_echo_likelihood =
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stats.residual_echo_likelihood.value_or(-1.0f);
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stats2.residual_echo_likelihood.value_or(-1.0f);
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const float residual_echo_likelihood_recent_max =
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stats.residual_echo_likelihood_recent_max.value_or(-1.0f);
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stats2.residual_echo_likelihood_recent_max.value_or(-1.0f);
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if (!absl::GetFlag(FLAGS_write_apm_ref_data)) {
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const audioproc::Test::EchoMetrics& reference =
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@ -107,10 +107,10 @@ DesktopRect GetExcludedWindowPixelBounds(CGWindowID window, float dip_to_pixel_s
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CFArrayRef window_array = CGWindowListCreateDescriptionFromArray(window_id_array);
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if (CFArrayGetCount(window_array) > 0) {
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CFDictionaryRef window =
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CFDictionaryRef win =
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reinterpret_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(window_array, 0));
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CFDictionaryRef bounds_ref =
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reinterpret_cast<CFDictionaryRef>(CFDictionaryGetValue(window, kCGWindowBounds));
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reinterpret_cast<CFDictionaryRef>(CFDictionaryGetValue(win, kCGWindowBounds));
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CGRectMakeWithDictionaryRepresentation(bounds_ref, &rect);
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}
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@ -374,13 +374,13 @@ bool ScreenCapturerMac::CgBlit(const DesktopFrame& frame, const DesktopRegion& r
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// Copy the dirty region from the display buffer into our desktop buffer.
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uint8_t* out_ptr = frame.GetFrameDataAtPos(display_bounds.top_left());
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for (DesktopRegion::Iterator i(copy_region); !i.IsAtEnd(); i.Advance()) {
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for (DesktopRegion::Iterator it(copy_region); !it.IsAtEnd(); it.Advance()) {
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CopyRect(display_base_address,
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src_bytes_per_row,
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out_ptr,
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frame.stride(),
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DesktopFrame::kBytesPerPixel,
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i.rect());
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it.rect());
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}
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if (excluded_image) {
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@ -962,9 +962,9 @@ void StatsCollector::ExtractSessionInfo_s(SessionStats& session_stats) {
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}
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for (const auto& channel_iter : transport.stats.channel_stats) {
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StatsReport::Id id(
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StatsReport::Id channel_stats_id(
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StatsReport::NewComponentId(transport.name, channel_iter.component));
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StatsReport* channel_report = reports_.ReplaceOrAddNew(id);
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StatsReport* channel_report = reports_.ReplaceOrAddNew(channel_stats_id);
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channel_report->set_timestamp(stats_gathering_started_);
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channel_report->AddInt(StatsReport::kStatsValueNameComponent,
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channel_iter.component);
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@ -128,7 +128,7 @@ struct LargeNonTrivial {
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LargeNonTrivial(LargeNonTrivial&& m) {}
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~LargeNonTrivial() = default;
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void operator()(int& a) { a = 1; }
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void operator()(int& b) { b = 1; }
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};
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TEST(CallbackList, LargeNonTrivialTest) {
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@ -308,12 +308,12 @@ void AsyncHttpsProxySocket::ProcessInput(char* data, size_t* len) {
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if (data[pos++] != '\n')
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continue;
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size_t len = pos - start - 1;
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if ((len > 0) && (data[start + len - 1] == '\r'))
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--len;
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size_t length = pos - start - 1;
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if ((length > 0) && (data[start + length - 1] == '\r'))
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--length;
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data[start + len] = 0;
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ProcessLine(data + start, len);
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data[start + length] = 0;
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ProcessLine(data + start, length);
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start = pos;
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}
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@ -566,9 +566,9 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
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return;
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RTC_LOG(LS_VERBOSE) << "Bound on " << addr << ":" << port;
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} else if (atyp == 3) {
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uint8_t len;
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uint8_t length;
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std::string addr;
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if (!response.ReadUInt8(&len) || !response.ReadString(&addr, len) ||
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if (!response.ReadUInt8(&length) || !response.ReadString(&addr, length) ||
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!response.ReadUInt16(&port))
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return;
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RTC_LOG(LS_VERBOSE) << "Bound on " << addr << ":" << port;
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@ -846,12 +846,12 @@ void VirtualSocketServer::CancelConnects(VirtualSocket* socket) {
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MessageAddress* data = static_cast<MessageAddress*>(it->pdata);
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SocketAddress local_addr = socket->GetLocalAddress();
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// Lookup remote side.
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VirtualSocket* socket = LookupConnection(local_addr, data->addr);
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if (socket) {
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VirtualSocket* lookup_socket = LookupConnection(local_addr, data->addr);
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if (lookup_socket) {
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// Server socket, remote side is a socket retreived by
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// accept. Accepted sockets are not bound so we will not
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// find it by looking in the bindings table.
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Disconnect(socket);
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Disconnect(lookup_socket);
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RemoveConnection(local_addr, data->addr);
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} else {
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Disconnect(data->addr);
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@ -381,12 +381,12 @@ void ReceiveStatisticsProxy::UpdateHistograms(
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<< " " << media_bitrate_kbps << '\n';
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}
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int num_total_frames =
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int num_total_frames2 =
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stats.frame_counts.key_frames + stats.frame_counts.delta_frames;
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if (num_total_frames >= kMinRequiredSamples) {
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if (num_total_frames2 >= kMinRequiredSamples) {
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int num_key_frames = stats.frame_counts.key_frames;
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int key_frames_permille =
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(num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
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(num_key_frames * 1000 + num_total_frames2 / 2) / num_total_frames2;
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RTC_HISTOGRAM_COUNTS_SPARSE_1000(
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uma_prefix + ".KeyFramesReceivedInPermille" + uma_suffix,
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key_frames_permille);
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@ -394,12 +394,12 @@ void ReceiveStatisticsProxy::UpdateHistograms(
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<< " " << key_frames_permille << '\n';
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}
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absl::optional<int> qp = stats.qp_counter.Avg(kMinRequiredSamples);
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if (qp) {
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absl::optional<int> qp2 = stats.qp_counter.Avg(kMinRequiredSamples);
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if (qp2) {
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RTC_HISTOGRAM_COUNTS_SPARSE_200(
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uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, *qp);
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uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, *qp2);
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log_stream << uma_prefix << ".Decoded.Vp8.Qp" << uma_suffix << " "
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<< *qp << '\n';
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<< *qp2 << '\n';
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}
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}
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}
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