Fixes ClangTidy errors in audio/

These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.

Bug: webrtc:10410
Change-Id: I1b46653b91bce012afabfa0f2d249718e6de2df8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127626
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27139}

audio/audio_receive_stream_unittest.cc

@@ -36,7 +36,6 @@ namespace {
 using testing::_;
 using testing::FloatEq;
 using testing::Return;
-using testing::ReturnRef;
 
 AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
   AudioDecodingCallStats audio_decode_stats;
@@ -64,8 +63,9 @@ const double kTotalOutputDuration = 0.5;
 
 const CallReceiveStatistics kCallStats = {345, 678, 901, 234,
                                           -12, 567, 890, 123};
-const std::pair<int, SdpAudioFormat> kReceiveCodec =
-    {123, {"codec_name_recv", 96000, 0}};
+const std::pair<int, SdpAudioFormat> kReceiveCodec = {
+    123,
+    {"codec_name_recv", 96000, 0}};
 const NetworkStatistics kNetworkStats = {
     123, 456, false, 789012, 3456, 123, 456, 789, 0,  {}, 789,
     12,  345, 678,   901,    0,    -1,  -1,  -1,  -1, -1, 0};

audio/audio_transport_impl.cc

@@ -144,7 +144,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
 
   // Measure audio level of speech after all processing.
   double sample_duration = static_cast<double>(number_of_frames) / sample_rate;
-  audio_level_.ComputeLevel(*audio_frame.get(), sample_duration);
+  audio_level_.ComputeLevel(*audio_frame, sample_duration);
 
   // Copy frame and push to each sending stream. The copy is required since an
   // encoding task will be posted internally to each stream.
@@ -157,7 +157,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
       auto it = sending_streams_.begin();
       while (++it != sending_streams_.end()) {
         std::unique_ptr<AudioFrame> audio_frame_copy(new AudioFrame());
-        audio_frame_copy->CopyFrom(*audio_frame.get());
+        audio_frame_copy->CopyFrom(*audio_frame);
         (*it)->SendAudioData(std::move(audio_frame_copy));
       }
       // Send the original frame to the first stream w/o copying.

audio/channel_send.cc

@@ -76,11 +76,10 @@ class TransportFeedbackProxy;
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
 
-class ChannelSend
-    : public ChannelSendInterface,
-      public AudioPacketizationCallback,  // receive encoded packets from the
-                                          // ACM
-      public TargetTransferRateObserver {
+class ChannelSend : public ChannelSendInterface,
+                    public AudioPacketizationCallback,  // receive encoded
+                                                        // packets from the ACM
+                    public TargetTransferRateObserver {
  public:
   // TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
   // declaration.
@@ -603,7 +602,7 @@ int32_t ChannelSend::SendMediaTransportAudio(
       // Sample count isn't conveniently available from the AudioCodingModule,
       // and needs some refactoring to wire up in a good way. For now, left as
       // zero.
-      /*sample_count=*/0,
+      /*samples_per_channel=*/0,
 
       /*sequence_number=*/media_transport_sequence_number_,
       MediaTransportFrameTypeForWebrtcFrameType(frameType), payloadType,

audio/test/media_transport_test.cc

@@ -34,9 +34,7 @@ namespace webrtc {
 namespace test {
 
 namespace {
-using testing::_;
 using testing::NiceMock;
-using testing::Return;
 
 constexpr int kPayloadTypeOpus = 17;
 constexpr int kSamplingFrequency = 48000;
@@ -113,7 +111,7 @@ TEST(AudioWithMediaTransport, DeliversAudio) {
 
   webrtc::internal::AudioReceiveStream receive_stream(
       Clock::GetRealTimeClock(),
-      /*rtp_stream_receiver_controller=*/nullptr,
+      /*receiver_controller=*/nullptr,
       /*packet_router=*/nullptr, receive_process_thread.get(), receive_config,
       audio_state, null_event_log.get());