Fix some shadow issues

which brings us a bit closer to enabling -Wshadow

BUG=webrtc:42223409

No-Iwyu: hard enough to review without IWYU already
Change-Id: I8620fc62e16aa4b15647d67760e952922ec8e202
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/381161
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#44148}
diff --git a/api/video_codecs/video_encoder.cc b/api/video_codecs/video_encoder.cc
index 4072643..fcbb250 100644
--- a/api/video_codecs/video_encoder.cc
+++ b/api/video_codecs/video_encoder.cc
@@ -160,11 +160,11 @@
     if (!fractions.empty()) {
       first = false;
       oss << "[ ";
-      for (size_t i = 0; i < fractions.size(); ++i) {
-        if (i > 0) {
+      for (size_t j = 0; j < fractions.size(); ++j) {
+        if (j > 0) {
           oss << ", ";
         }
-        oss << (static_cast<double>(fractions[i]) / kMaxFramerateFraction);
+        oss << (static_cast<double>(fractions[j]) / kMaxFramerateFraction);
       }
       oss << "] ";
     }
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 5294c19..88a3e7a 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -800,8 +800,8 @@
       std::optional<RtpState> fec_state =
           rtp_streams_[i].fec_generator->GetRtpState();
       if (fec_state) {
-        uint32_t ssrc = rtp_config_.flexfec.ssrc;
-        rtp_states[ssrc] = *fec_state;
+        uint32_t fec_ssrc = rtp_config_.flexfec.ssrc;
+        rtp_states[fec_ssrc] = *fec_state;
       }
     }
   }
diff --git a/media/base/codec.cc b/media/base/codec.cc
index 212447b..ac749e2 100644
--- a/media/base/codec.cc
+++ b/media/base/codec.cc
@@ -154,31 +154,31 @@
           codec_parameters.parameters == codec_capability.parameters);
 }
 
-bool Codec::GetParam(const std::string& name, std::string* out) const {
-  webrtc::CodecParameterMap::const_iterator iter = params.find(name);
+bool Codec::GetParam(const std::string& key, std::string* out) const {
+  webrtc::CodecParameterMap::const_iterator iter = params.find(key);
   if (iter == params.end())
     return false;
   *out = iter->second;
   return true;
 }
 
-bool Codec::GetParam(const std::string& name, int* out) const {
-  webrtc::CodecParameterMap::const_iterator iter = params.find(name);
+bool Codec::GetParam(const std::string& key, int* out) const {
+  webrtc::CodecParameterMap::const_iterator iter = params.find(key);
   if (iter == params.end())
     return false;
   return rtc::FromString(iter->second, out);
 }
 
-void Codec::SetParam(const std::string& name, const std::string& value) {
-  params[name] = value;
+void Codec::SetParam(const std::string& key, const std::string& value) {
+  params[key] = value;
 }
 
-void Codec::SetParam(const std::string& name, int value) {
-  params[name] = rtc::ToString(value);
+void Codec::SetParam(const std::string& key, int value) {
+  params[key] = rtc::ToString(value);
 }
 
-bool Codec::RemoveParam(const std::string& name) {
-  return params.erase(name) == 1;
+bool Codec::RemoveParam(const std::string& key) {
+  return params.erase(key) == 1;
 }
 
 void Codec::AddFeedbackParam(const FeedbackParam& param) {
diff --git a/media/base/codec.h b/media/base/codec.h
index 9258e35..2bc2fb9 100644
--- a/media/base/codec.h
+++ b/media/base/codec.h
@@ -138,12 +138,12 @@
   // generated, done e.g. by setCodecPreferences or setParameters.
   bool MatchesRtpCodec(const webrtc::RtpCodec& capability) const;
 
-  // Find the parameter for `name` and write the value to `out`.
-  bool GetParam(const std::string& name, std::string* out) const;
-  bool GetParam(const std::string& name, int* out) const;
+  // Find the parameter for `key` and write the value to `out`.
+  bool GetParam(const std::string& key, std::string* out) const;
+  bool GetParam(const std::string& key, int* out) const;
 
-  void SetParam(const std::string& name, const std::string& value);
-  void SetParam(const std::string& name, int value);
+  void SetParam(const std::string& key, const std::string& value);
+  void SetParam(const std::string& key, int value);
 
   // It is safe to input a non-existent parameter.
   // Returns true if the parameter existed, false if it did not exist.
diff --git a/media/base/stream_params.cc b/media/base/stream_params.cc
index ac9daee..e076f6c 100644
--- a/media/base/stream_params.cc
+++ b/media/base/stream_params.cc
@@ -173,13 +173,13 @@
   }
 }
 
-void StreamParams::GetPrimarySsrcs(std::vector<uint32_t>* ssrcs) const {
+void StreamParams::GetPrimarySsrcs(std::vector<uint32_t>* primary_ssrcs) const {
   const SsrcGroup* sim_group = get_ssrc_group(kSimSsrcGroupSemantics);
   if (sim_group == NULL) {
-    ssrcs->push_back(first_ssrc());
+    primary_ssrcs->push_back(first_ssrc());
   } else {
-    ssrcs->insert(ssrcs->end(), sim_group->ssrcs.begin(),
-                  sim_group->ssrcs.end());
+    primary_ssrcs->insert(primary_ssrcs->end(), sim_group->ssrcs.begin(),
+                          sim_group->ssrcs.end());
   }
 }
 
diff --git a/media/base/stream_params.h b/media/base/stream_params.h
index e15789b..7dd6ed7 100644
--- a/media/base/stream_params.h
+++ b/media/base/stream_params.h
@@ -164,7 +164,7 @@
 
   // Convenience to get all the SIM SSRCs if there are SIM ssrcs, or
   // the first SSRC otherwise.
-  void GetPrimarySsrcs(std::vector<uint32_t>* ssrcs) const;
+  void GetPrimarySsrcs(std::vector<uint32_t>* primary_ssrcs) const;
 
   // Convenience to get all the secondary SSRCs for the given primary ssrcs
   // of a particular semantic.
diff --git a/media/engine/simulcast_encoder_adapter.cc b/media/engine/simulcast_encoder_adapter.cc
index 51022c3..6770f13 100644
--- a/media/engine/simulcast_encoder_adapter.cc
+++ b/media/engine/simulcast_encoder_adapter.cc
@@ -374,8 +374,8 @@
   // Singlecast or simulcast with simulcast-capable underlaying encoder.
   if (total_streams_count_ == 1 || !separate_encoders_needed) {
     RTC_LOG(LS_INFO) << "[SEA] InitEncode: Single-encoder mode";
-    int ret = encoder_context->encoder().InitEncode(&codec_, settings);
-    if (ret >= 0) {
+    int result = encoder_context->encoder().InitEncode(&codec_, settings);
+    if (result >= 0) {
       stream_contexts_.emplace_back(
           /*parent=*/nullptr, std::move(encoder_context),
           /*framerate_controller=*/nullptr, /*stream_idx=*/0, codec_.width,
@@ -427,13 +427,13 @@
                      << stream_idx << ", active: "
                      << (codec_.simulcastStream[stream_idx].active ? "true"
                                                                    : "false");
-    int ret = encoder_context->encoder().InitEncode(&stream_codec, settings);
-    if (ret < 0) {
+    int result = encoder_context->encoder().InitEncode(&stream_codec, settings);
+    if (result < 0) {
       encoder_context.reset();
       Release();
       RTC_LOG(LS_ERROR) << "[SEA] InitEncode: failed with error code: "
                         << WebRtcVideoCodecErrorToString(ret);
-      return ret;
+      return result;
     }
 
     // Intercept frame encode complete callback only for upper streams, where
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index f45e1b5..9d0d7b8 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2491,16 +2491,16 @@
     default_recv_base_minimum_delay_ms_ = delay_ms;
     ssrcs = unsignaled_recv_ssrcs_;
   }
-  for (uint32_t ssrc : ssrcs) {
-    const auto it = recv_streams_.find(ssrc);
+  for (uint32_t recv_ssrc : ssrcs) {
+    const auto it = recv_streams_.find(recv_ssrc);
     if (it == recv_streams_.end()) {
       RTC_LOG(LS_WARNING) << "SetBaseMinimumPlayoutDelayMs: no recv stream "
-                          << ssrc;
+                          << recv_ssrc;
       return false;
     }
     it->second->SetBaseMinimumPlayoutDelayMs(delay_ms);
     RTC_LOG(LS_INFO) << "SetBaseMinimumPlayoutDelayMs() to " << delay_ms
-                     << " for recv stream with ssrc " << ssrc;
+                     << " for recv stream with ssrc " << recv_ssrc;
   }
   return true;
 }
diff --git a/media/sctp/dcsctp_transport.cc b/media/sctp/dcsctp_transport.cc
index 65aba51..3bd8211 100644
--- a/media/sctp/dcsctp_transport.cc
+++ b/media/sctp/dcsctp_transport.cc
@@ -375,11 +375,11 @@
       ready_to_send_data_ = false;
       return RTCError(RTCErrorType::RESOURCE_EXHAUSTED);
     default:
-      absl::string_view message = dcsctp::ToString(error);
+      absl::string_view error_message = dcsctp::ToString(error);
       RTC_LOG(LS_ERROR) << debug_name_
                         << "->SendData(...): send() failed with error "
-                        << message << ".";
-      return RTCError(RTCErrorType::NETWORK_ERROR, message);
+                        << error_message << ".";
+      return RTCError(RTCErrorType::NETWORK_ERROR, error_message);
   }
 }
 
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 8ce64bd..28e5590 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -735,7 +735,6 @@
                                       int payload_type,
                                       int payload_size,
                                       int timestamp_offset) {
-  constexpr int kRedHeaderLength = 4;
   const size_t size =
       payload_size + 1 + (num_payloads - 1) * (payload_size + kRedHeaderLength);
   std::vector<uint8_t> payload(size, 0);
diff --git a/modules/audio_coding/neteq/red_payload_splitter.cc b/modules/audio_coding/neteq/red_payload_splitter.cc
index a3db686..4c753f5 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -114,8 +114,8 @@
       PacketList new_packets;  // An empty list to store the split packets in.
       for (size_t i = 0; i != new_headers.size(); ++i) {
         const auto& new_header = new_headers[i];
-        size_t payload_length = new_header.payload_length;
-        if (payload_ptr + payload_length >
+        size_t block_length = new_header.payload_length;
+        if (payload_ptr + block_length >
             red_packet.payload.data() + red_packet.payload.size()) {
           // The block lengths in the RED headers do not match the overall
           // packet length. Something is corrupt. Discard this and the remaining
@@ -131,9 +131,9 @@
         new_packet.sequence_number = red_packet.sequence_number;
         new_packet.priority.red_level =
             dchecked_cast<int>((new_headers.size() - 1) - i);
-        new_packet.payload.SetData(payload_ptr, payload_length);
+        new_packet.payload.SetData(payload_ptr, block_length);
         new_packets.push_front(std::move(new_packet));
-        payload_ptr += payload_length;
+        payload_ptr += block_length;
       }
       // Insert new packets into original list, before the element pointed to by
       // iterator `it`.
diff --git a/modules/pacing/pacing_controller.cc b/modules/pacing/pacing_controller.cc
index 8a43fa7..b480b95 100644
--- a/modules/pacing/pacing_controller.cc
+++ b/modules/pacing/pacing_controller.cc
@@ -415,8 +415,8 @@
         keepalive_data_sent +=
             DataSize::Bytes(packet->payload_size() + packet->padding_size());
         packet_sender_->SendPacket(std::move(packet), PacedPacketInfo());
-        for (auto& packet : packet_sender_->FetchFec()) {
-          EnqueuePacket(std::move(packet));
+        for (auto& fec_packet : packet_sender_->FetchFec()) {
+          EnqueuePacket(std::move(fec_packet));
         }
       }
     }
diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc
index 7fcef93..3e45d5f 100644
--- a/modules/pacing/packet_router.cc
+++ b/modules/pacing/packet_router.cc
@@ -222,8 +222,8 @@
     last_send_module_ = rtp_module;
   }
 
-  for (auto& packet : rtp_module->FetchFecPackets()) {
-    pending_fec_packets_.push_back(std::move(packet));
+  for (auto& fec_packet : rtp_module->FetchFecPackets()) {
+    pending_fec_packets_.push_back(std::move(fec_packet));
   }
 }
 
diff --git a/modules/rtp_rtcp/source/rtp_packetizer_av1.cc b/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
index 5575a63..c559e5d 100644
--- a/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
+++ b/modules/rtp_rtcp/source/rtp_packetizer_av1.cc
@@ -252,12 +252,12 @@
          obu_offset + limits.max_payload_len < obu.size;
          obu_offset += limits.max_payload_len) {
       packets.emplace_back(/*first_obu_index=*/obu_index);
-      Packet& packet = packets.back();
-      packet.num_obu_elements = 1;
-      packet.first_obu_offset = obu_offset;
+      Packet& middle_packet = packets.back();
+      middle_packet.num_obu_elements = 1;
+      middle_packet.first_obu_offset = obu_offset;
       int middle_fragment_size = limits.max_payload_len;
-      packet.last_obu_size = middle_fragment_size;
-      packet.packet_size = middle_fragment_size;
+      middle_packet.last_obu_size = middle_fragment_size;
+      middle_packet.packet_size = middle_fragment_size;
     }
 
     // Add the last fragment of the obu.
@@ -282,11 +282,11 @@
       last_fragment_size -= semi_last_fragment_size;
 
       packets.emplace_back(/*first_obu_index=*/obu_index);
-      Packet& packet = packets.back();
-      packet.num_obu_elements = 1;
-      packet.first_obu_offset = obu_offset;
-      packet.last_obu_size = semi_last_fragment_size;
-      packet.packet_size = semi_last_fragment_size;
+      Packet& second_last_packet = packets.back();
+      second_last_packet.num_obu_elements = 1;
+      second_last_packet.first_obu_offset = obu_offset;
+      second_last_packet.last_obu_size = semi_last_fragment_size;
+      second_last_packet.packet_size = semi_last_fragment_size;
       obu_offset += semi_last_fragment_size;
     }
     packets.emplace_back(/*first_obu_index=*/obu_index);
diff --git a/modules/video_coding/packet_buffer.cc b/modules/video_coding/packet_buffer.cc
index 718c4a8..6555aa5 100644
--- a/modules/video_coding/packet_buffer.cc
+++ b/modules/video_coding/packet_buffer.cc
@@ -391,13 +391,13 @@
         // Use uint16_t type to handle sequence number wrap around case.
         uint16_t num_packets = end_seq_num - start_seq_num;
         found_frames.reserve(found_frames.size() + num_packets);
-        for (uint16_t i = start_seq_num; i != end_seq_num; ++i) {
-          std::unique_ptr<Packet>& packet = buffer_[i % buffer_.size()];
+        for (uint16_t j = start_seq_num; j != end_seq_num; ++j) {
+          std::unique_ptr<Packet>& packet = buffer_[j % buffer_.size()];
           RTC_DCHECK(packet);
-          RTC_DCHECK_EQ(i, packet->seq_num());
+          RTC_DCHECK_EQ(j, packet->seq_num());
           // Ensure frame boundary flags are properly set.
-          packet->video_header.is_first_packet_in_frame = (i == start_seq_num);
-          packet->video_header.is_last_packet_in_frame = (i == seq_num);
+          packet->video_header.is_first_packet_in_frame = (j == start_seq_num);
+          packet->video_header.is_last_packet_in_frame = (j == seq_num);
           found_frames.push_back(std::move(packet));
         }
 
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 39e8784..1c6e272 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -420,7 +420,7 @@
 
   for (const auto& kv : candidates_by_transport_name) {
     const std::string& transport_name = kv.first;
-    const cricket::Candidates& candidates = kv.second;
+    const cricket::Candidates& transport_candidates = kv.second;
     cricket::JsepTransport* jsep_transport =
         GetJsepTransportByName(transport_name);
     if (!jsep_transport) {
@@ -428,7 +428,7 @@
           << "Not removing candidate because the JsepTransport doesn't exist.";
       continue;
     }
-    for (const cricket::Candidate& candidate : candidates) {
+    for (const cricket::Candidate& candidate : transport_candidates) {
       cricket::DtlsTransportInternal* dtls =
           candidate.component() == cricket::ICE_CANDIDATE_COMPONENT_RTP
               ? jsep_transport->rtp_dtls_transport()
diff --git a/pc/media_session.cc b/pc/media_session.cc
index 82722a2..b8be225 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -762,7 +762,6 @@
       current_content = &current_description->contents()[msection_index];
       // Media type must match unless this media section is being recycled.
     }
-    RTCError error;
     switch (media_description_options.type) {
       case MEDIA_TYPE_AUDIO:
       case MEDIA_TYPE_VIDEO:
@@ -958,7 +957,6 @@
     }
     RtpHeaderExtensions header_extensions = RtpHeaderExtensionsFromCapabilities(
         UnstoppedRtpHeaderExtensionCapabilities(header_extensions_in));
-    RTCError error;
     switch (media_description_options.type) {
       case MEDIA_TYPE_AUDIO:
       case MEDIA_TYPE_VIDEO:
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index dc196ff..101a5b7 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -953,15 +953,17 @@
       RTC_DCHECK_EQ(s, &certificate_stats);
       break;
     }
-    RTCCertificateStats* certificate_stats =
+    RTCCertificateStats* current_certificate_stats =
         new RTCCertificateStats(certificate_stats_id, timestamp);
-    certificate_stats->fingerprint = s->fingerprint;
-    certificate_stats->fingerprint_algorithm = s->fingerprint_algorithm;
-    certificate_stats->base64_certificate = s->base64_certificate;
+    current_certificate_stats->fingerprint = s->fingerprint;
+    current_certificate_stats->fingerprint_algorithm = s->fingerprint_algorithm;
+    current_certificate_stats->base64_certificate = s->base64_certificate;
     if (prev_certificate_stats)
-      prev_certificate_stats->issuer_certificate_id = certificate_stats->id();
-    report->AddStats(std::unique_ptr<RTCCertificateStats>(certificate_stats));
-    prev_certificate_stats = certificate_stats;
+      prev_certificate_stats->issuer_certificate_id =
+          current_certificate_stats->id();
+    report->AddStats(
+        std::unique_ptr<RTCCertificateStats>(current_certificate_stats));
+    prev_certificate_stats = current_certificate_stats;
   }
 }
 
@@ -1960,64 +1962,67 @@
     // There is one transport stats for each channel.
     for (const cricket::TransportChannelStats& channel_stats :
          transport_stats.channel_stats) {
-      auto transport_stats = std::make_unique<RTCTransportStats>(
+      auto channel_transport_stats = std::make_unique<RTCTransportStats>(
           RTCTransportStatsIDFromTransportChannel(transport_name,
                                                   channel_stats.component),
           timestamp);
-      transport_stats->packets_sent =
+      channel_transport_stats->packets_sent =
           channel_stats.ice_transport_stats.packets_sent;
-      transport_stats->packets_received =
+      channel_transport_stats->packets_received =
           channel_stats.ice_transport_stats.packets_received;
-      transport_stats->bytes_sent =
+      channel_transport_stats->bytes_sent =
           channel_stats.ice_transport_stats.bytes_sent;
-      transport_stats->bytes_received =
+      channel_transport_stats->bytes_received =
           channel_stats.ice_transport_stats.bytes_received;
-      transport_stats->dtls_state =
+      channel_transport_stats->dtls_state =
           DtlsTransportStateToRTCDtlsTransportState(channel_stats.dtls_state);
-      transport_stats->selected_candidate_pair_changes =
+      channel_transport_stats->selected_candidate_pair_changes =
           channel_stats.ice_transport_stats.selected_candidate_pair_changes;
-      transport_stats->ice_role =
+      channel_transport_stats->ice_role =
           IceRoleToRTCIceRole(channel_stats.ice_transport_stats.ice_role);
-      transport_stats->ice_local_username_fragment =
+      channel_transport_stats->ice_local_username_fragment =
           channel_stats.ice_transport_stats.ice_local_username_fragment;
-      transport_stats->ice_state = IceTransportStateToRTCIceTransportState(
-          channel_stats.ice_transport_stats.ice_state);
+      channel_transport_stats->ice_state =
+          IceTransportStateToRTCIceTransportState(
+              channel_stats.ice_transport_stats.ice_state);
       for (const cricket::ConnectionInfo& info :
            channel_stats.ice_transport_stats.connection_infos) {
         if (info.best_connection) {
-          transport_stats->selected_candidate_pair_id =
+          channel_transport_stats->selected_candidate_pair_id =
               RTCIceCandidatePairStatsIDFromConnectionInfo(info);
         }
       }
       if (channel_stats.component != cricket::ICE_CANDIDATE_COMPONENT_RTCP &&
           !rtcp_transport_stats_id.empty()) {
-        transport_stats->rtcp_transport_stats_id = rtcp_transport_stats_id;
+        channel_transport_stats->rtcp_transport_stats_id =
+            rtcp_transport_stats_id;
       }
       if (!local_certificate_id.empty())
-        transport_stats->local_certificate_id = local_certificate_id;
+        channel_transport_stats->local_certificate_id = local_certificate_id;
       if (!remote_certificate_id.empty())
-        transport_stats->remote_certificate_id = remote_certificate_id;
+        channel_transport_stats->remote_certificate_id = remote_certificate_id;
       // Crypto information
       if (channel_stats.ssl_version_bytes) {
         char bytes[5];
         snprintf(bytes, sizeof(bytes), "%04X", channel_stats.ssl_version_bytes);
-        transport_stats->tls_version = bytes;
+        channel_transport_stats->tls_version = bytes;
       }
 
       if (channel_stats.dtls_role) {
-        transport_stats->dtls_role =
+        channel_transport_stats->dtls_role =
             *channel_stats.dtls_role == SSL_CLIENT ? "client" : "server";
       } else {
-        transport_stats->dtls_role = "unknown";
+        channel_transport_stats->dtls_role = "unknown";
       }
 
-      transport_stats->dtls_cipher = channel_stats.tls_cipher_suite_name;
+      channel_transport_stats->dtls_cipher =
+          channel_stats.tls_cipher_suite_name;
       if (channel_stats.srtp_crypto_suite != kSrtpInvalidCryptoSuite &&
           SrtpCryptoSuiteToName(channel_stats.srtp_crypto_suite).length()) {
-        transport_stats->srtp_cipher =
+        channel_transport_stats->srtp_cipher =
             SrtpCryptoSuiteToName(channel_stats.srtp_crypto_suite);
       }
-      report->AddStats(std::move(transport_stats));
+      report->AddStats(std::move(channel_transport_stats));
     }
   }
 }
diff --git a/pc/sdp_munging_detector.cc b/pc/sdp_munging_detector.cc
index f096f4a..0c8e6fb 100644
--- a/pc/sdp_munging_detector.cc
+++ b/pc/sdp_munging_detector.cc
@@ -350,17 +350,20 @@
                            "last created description.";
     return SdpMungingType::kNumberOfContents;
   }
-  for (size_t i = 0; i < last_created_contents.size(); i++) {
+  for (size_t content_index = 0; content_index < last_created_contents.size();
+       content_index++) {
     // TODO: crbug.com/40567530 - more checks are needed here.
-    if (last_created_contents[i].mid() != contents_to_set[i].mid()) {
+    if (last_created_contents[content_index].mid() !=
+        contents_to_set[content_index].mid()) {
       RTC_LOG(LS_WARNING) << "SDP munging: mid does not match "
                              "last created description.";
       return SdpMungingType::kMid;
     }
 
     auto* last_created_media_description =
-        last_created_contents[i].media_description();
-    auto* media_description_to_set = contents_to_set[i].media_description();
+        last_created_contents[content_index].media_description();
+    auto* media_description_to_set =
+        contents_to_set[content_index].media_description();
     if (!(last_created_media_description && media_description_to_set)) {
       continue;
     }
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 96a1cb9..13a5d39 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -3232,36 +3232,38 @@
 
   for (auto&& transceivers_stable_state_pair : transceivers()->StableStates()) {
     auto transceiver = transceivers_stable_state_pair.first;
-    auto state = transceivers_stable_state_pair.second;
+    auto stable_state = transceivers_stable_state_pair.second;
 
-    if (state.did_set_fired_direction()) {
+    if (stable_state.did_set_fired_direction()) {
       // If this rollback triggers going from not receiving to receving again,
       // we need to fire "ontrack".
       bool previously_fired_direction_is_recv =
           transceiver->fired_direction().has_value() &&
           RtpTransceiverDirectionHasRecv(*transceiver->fired_direction());
       bool currently_fired_direction_is_recv =
-          state.fired_direction().has_value() &&
-          RtpTransceiverDirectionHasRecv(state.fired_direction().value());
+          stable_state.fired_direction().has_value() &&
+          RtpTransceiverDirectionHasRecv(
+              stable_state.fired_direction().value());
       if (!previously_fired_direction_is_recv &&
           currently_fired_direction_is_recv) {
         now_receiving_transceivers.push_back(transceiver);
       }
-      transceiver->internal()->set_fired_direction(state.fired_direction());
+      transceiver->internal()->set_fired_direction(
+          stable_state.fired_direction());
     }
 
-    if (state.remote_stream_ids()) {
+    if (stable_state.remote_stream_ids()) {
       std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
       std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
       SetAssociatedRemoteStreams(transceiver->internal()->receiver_internal(),
-                                 state.remote_stream_ids().value(),
+                                 stable_state.remote_stream_ids().value(),
                                  &added_streams, &removed_streams);
       all_added_streams.insert(all_added_streams.end(), added_streams.begin(),
                                added_streams.end());
       all_removed_streams.insert(all_removed_streams.end(),
                                  removed_streams.begin(),
                                  removed_streams.end());
-      if (!state.has_m_section() && !state.newly_created()) {
+      if (!stable_state.has_m_section() && !stable_state.newly_created()) {
         continue;
       }
     }
@@ -3277,7 +3279,7 @@
         transceiver->receiver()) {
       removed_receivers.push_back(transceiver->receiver());
     }
-    if (state.newly_created()) {
+    if (stable_state.newly_created()) {
       if (transceiver->internal()->reused_for_addtrack()) {
         transceiver->internal()->set_created_by_addtrack(true);
       } else {
@@ -3285,15 +3287,15 @@
         transceivers()->Remove(transceiver);
       }
     }
-    if (state.init_send_encodings()) {
+    if (stable_state.init_send_encodings()) {
       transceiver->internal()->sender_internal()->set_init_send_encodings(
-          state.init_send_encodings().value());
+          stable_state.init_send_encodings().value());
     }
     transceiver->internal()->sender_internal()->set_transport(nullptr);
     transceiver->internal()->receiver_internal()->set_transport(nullptr);
-    if (state.has_m_section()) {
-      transceiver->internal()->set_mid(state.mid());
-      transceiver->internal()->set_mline_index(state.mline_index());
+    if (stable_state.has_m_section()) {
+      transceiver->internal()->set_mid(stable_state.mid());
+      transceiver->internal()->set_mline_index(stable_state.mline_index());
     }
   }
   RTCError e = transport_controller_s()->RollbackTransports();
@@ -3694,7 +3696,6 @@
   }
 
   // Verify crypto settings.
-  std::string crypto_error;
   if (pc_->dtls_enabled()) {
     RTCError crypto_error = VerifyCrypto(
         sdesc->description(), pc_->dtls_enabled(), bundle_groups_by_mid);
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index 7085bc0..c4a24c3 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -1143,15 +1143,15 @@
   }
   if (fields.size() >= (current_position + 2) &&
       fields[current_position] == kAttributeCandidateRport) {
-    int port = 0;
-    if (!GetValueFromString(first_line, fields[++current_position], &port,
-                            error)) {
+    int related_port = 0;
+    if (!GetValueFromString(first_line, fields[++current_position],
+                            &related_port, error)) {
       return false;
     }
-    if (!IsValidPort(port)) {
+    if (!IsValidPort(related_port)) {
       return ParseFailed(first_line, "Invalid port number.", error);
     }
-    related_address.SetPort(port);
+    related_address.SetPort(related_port);
     ++current_position;
   }
 
@@ -1327,7 +1327,6 @@
   if (data_desc->use_sctpmap()) {
     // draft-ietf-mmusic-sctp-sdp-04
     // a=sctpmap:sctpmap-number  protocol  [streams]
-    StringBuilder os;
     InitAttrLine(kAttributeSctpmap, &os);
     os << kSdpDelimiterColon << data_desc->port() << kSdpDelimiterSpace
        << kDefaultSctpmapProtocol << kSdpDelimiterSpace
@@ -1949,7 +1948,6 @@
   }
   if (media_desc->rtcp_fb_ack_ccfb()) {
     // RFC 8888 section 6
-    StringBuilder os;
     InitAttrLine(kAttributeRtcpFb, &os);
     os << kSdpDelimiterColon;
     os << "* ack ccfb";
diff --git a/rtc_base/openssl_stream_adapter.cc b/rtc_base/openssl_stream_adapter.cc
index 10235b0..f9b514e 100644
--- a/rtc_base/openssl_stream_adapter.cc
+++ b/rtc_base/openssl_stream_adapter.cc
@@ -758,8 +758,8 @@
       events_to_signal |= webrtc::SE_OPEN;
     } else {
       state_ = SSL_CONNECTING;
-      if (int err = BeginSSL()) {
-        Error("BeginSSL", err, 0, true);
+      if (int error = BeginSSL()) {
+        Error("BeginSSL", error, 0, true);
         return;
       }
     }
@@ -772,8 +772,8 @@
     if (state_ == SSL_NONE) {
       events_to_signal |= events & (webrtc::SE_READ | webrtc::SE_WRITE);
     } else if (state_ == SSL_CONNECTING) {
-      if (int err = ContinueSSL()) {
-        Error("ContinueSSL", err, 0, true);
+      if (int error = ContinueSSL()) {
+        Error("ContinueSSL", error, 0, true);
         return;
       }
     } else if (state_ == SSL_CONNECTED) {
diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc
index 47f53d4..1f0ae1e 100644
--- a/video/receive_statistics_proxy.cc
+++ b/video/receive_statistics_proxy.cc
@@ -197,10 +197,10 @@
                << key_frames_permille << '\n';
   }
 
-  std::optional<int> qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
-  if (qp) {
-    RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", *qp);
-    log_stream << "WebRTC.Video.Decoded.Vp8.Qp " << *qp << '\n';
+  std::optional<int> vp8_qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
+  if (vp8_qp) {
+    RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", *vp8_qp);
+    log_stream << "WebRTC.Video.Decoded.Vp8.Qp " << *vp8_qp << '\n';
   }
 
   std::optional<int> decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 3bf561f..9fb2caf 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -962,16 +962,16 @@
     }
     if (!pending_resolution.has_value() || !pending_resolution->empty()) {
       // Flush the buffered frames.
-      for (const auto& frame : buffered_encoded_frames_) {
+      for (const auto& buffered_frame : buffered_encoded_frames_) {
         RecordableEncodedFrame::EncodedResolution resolution{
-            frame->EncodedImage()._encodedWidth,
-            frame->EncodedImage()._encodedHeight};
-        if (IsKeyFrameAndUnspecifiedResolution(*frame)) {
+            buffered_frame->EncodedImage()._encodedWidth,
+            buffered_frame->EncodedImage()._encodedHeight};
+        if (IsKeyFrameAndUnspecifiedResolution(*buffered_frame)) {
           RTC_DCHECK(!pending_resolution->empty());
           resolution = *pending_resolution;
         }
         encoded_frame_buffer_function_(
-            WebRtcRecordableEncodedFrame(*frame, resolution));
+            WebRtcRecordableEncodedFrame(*buffered_frame, resolution));
       }
       buffered_encoded_frames_.clear();
     }