Remove test code related to unused variables

`input_aecdump` and `processed_capture_samples` are never used,
so the code related to those variables, can be removed.

Bug: none
Change-Id: I4f0b7db01f1e218f62e732ab9a247dd0b9708666
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356542
Reviewed-by: Per Ã…hgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47105}
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.cc b/modules/audio_processing/test/aec_dump_based_simulator.cc
index 7af6d41..f78ab46 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -19,7 +19,6 @@
 #include <iostream>
 #include <memory>
 #include <optional>
-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
 #include <string>
 #include <utility>
 
@@ -79,18 +78,6 @@
   return true;
 }
 
-// Selectively reads the next proto-buf message from dump-file or string input.
-// Returns a bool indicating whether a new message was available.
-bool ReadNextMessage(bool use_dump_file,
-                     FILE* dump_input_file,
-                     std::stringstream& input,
-                     audioproc::Event& event_msg) {
-  if (use_dump_file) {
-    return ReadMessageFromFile(dump_input_file, &event_msg);
-  }
-  return ReadMessageFromString(&input, &event_msg);
-}
-
 }  // namespace
 
 AecDumpBasedSimulator::AecDumpBasedSimulator(
@@ -256,19 +243,12 @@
         CheckedDivExact(sample_rate_hz, kChunksPerSecond), 1));
   }
 
-  const bool use_dump_file = !settings_.aec_dump_input_string.has_value();
-  std::stringstream input;
-  if (use_dump_file) {
-    dump_input_file_ =
-        OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
-  } else {
-    input << settings_.aec_dump_input_string.value();
-  }
+  dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
 
   audioproc::Event event_msg;
   int capture_frames_since_init = 0;
   int init_index = 0;
-  while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
+  while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
     SelectivelyToggleDataDumping(init_index, capture_frames_since_init);
     HandleEvent(event_msg, capture_frames_since_init, init_index);
 
@@ -281,28 +261,19 @@
               *settings_.init_to_process >= init_index);
   }
 
-  if (use_dump_file) {
-    fclose(dump_input_file_);
-  }
+  fclose(dump_input_file_);
 
   DetachAecDump();
 }
 
 void AecDumpBasedSimulator::Analyze() {
-  const bool use_dump_file = !settings_.aec_dump_input_string.has_value();
-  std::stringstream input;
-  if (use_dump_file) {
-    dump_input_file_ =
-        OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
-  } else {
-    input << settings_.aec_dump_input_string.value();
-  }
+  dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
 
   audioproc::Event event_msg;
   int num_capture_frames = 0;
   int num_render_frames = 0;
   int init_index = 0;
-  while (ReadNextMessage(use_dump_file, dump_input_file_, input, event_msg)) {
+  while (ReadMessageFromFile(dump_input_file_, &event_msg)) {
     if (event_msg.type() == audioproc::Event::INIT) {
       ++init_index;
       constexpr float kNumFramesPerSecond = 100.f;
@@ -323,9 +294,7 @@
     }
   }
 
-  if (use_dump_file) {
-    fclose(dump_input_file_);
-  }
+  fclose(dump_input_file_);
 }
 
 void AecDumpBasedSimulator::HandleEvent(const audioproc::Event& event_msg,
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 66e2713..aa1af17 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -222,11 +222,7 @@
     applied_input_volume_ = ap_->recommended_stream_analog_level();
   }
 
-  if (buffer_memory_writer_) {
-    RTC_CHECK(!buffer_file_writer_);
-    buffer_memory_writer_->Write(*out_buf_);
-  } else if (buffer_file_writer_) {
-    RTC_CHECK(!buffer_memory_writer_);
+  if (buffer_file_writer_) {
     buffer_file_writer_->Write(*out_buf_);
   }
 
@@ -373,9 +369,6 @@
                       static_cast<size_t>(out_config_.num_channels()),
                       settings_.wav_output_format));
     buffer_file_writer_.reset(new ChannelBufferWavWriter(std::move(out_file)));
-  } else if (settings_.aec_dump_input_string.has_value()) {
-    buffer_memory_writer_ = std::make_unique<ChannelBufferVectorWriter>(
-        settings_.processed_capture_samples);
   }
 
   if (settings_.linear_aec_output_filename) {
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 3ac6023..1bf5c1b 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -21,7 +21,6 @@
 #include <vector>
 
 #include "absl/base/nullability.h"
-#include "absl/strings/string_view.h"
 #include "api/audio/audio_processing.h"
 #include "api/scoped_refptr.h"
 #include "common_audio/channel_buffer.h"
@@ -152,8 +151,6 @@
   std::optional<std::string> call_order_input_filename;
   std::optional<std::string> call_order_output_filename;
   std::optional<std::string> aec_settings_filename;
-  std::optional<absl::string_view> aec_dump_input_string;
-  std::vector<float>* processed_capture_samples = nullptr;
   bool analysis_only = false;
   std::optional<int> dump_start_frame;
   std::optional<int> dump_end_frame;
@@ -244,7 +241,6 @@
   size_t num_reverse_process_stream_calls_ = 0;
   std::unique_ptr<ChannelBufferWavWriter> buffer_file_writer_;
   std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_file_writer_;
-  std::unique_ptr<ChannelBufferVectorWriter> buffer_memory_writer_;
   std::unique_ptr<WavWriter> linear_aec_output_file_writer_;
   ApiCallStatistics api_call_statistics_;
   std::ofstream residual_echo_likelihood_graph_writer_;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index a6b26f1..589fa1a 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -565,11 +565,6 @@
         "Error: The aec dump file cannot be specified "
         "together with input wav files!\n");
 
-    ReportConditionalErrorAndExit(
-        !!settings.aec_dump_input_string,
-        "Error: The aec dump input string cannot be specified "
-        "together with input wav files!\n");
-
     ReportConditionalErrorAndExit(!!settings.artificial_nearend_filename,
                                   "Error: The artificial nearend cannot be "
                                   "specified together with input wav files!\n");
@@ -585,13 +580,8 @@
         "must be specified if the reverse output wav filename is specified!\n");
   } else {
     ReportConditionalErrorAndExit(
-        !settings.aec_dump_input_filename && !settings.aec_dump_input_string,
-        "Error: Either the aec dump input file, the wav "
-        "input file or the aec dump input string must be specified!\n");
-    ReportConditionalErrorAndExit(
-        settings.aec_dump_input_filename && settings.aec_dump_input_string,
-        "Error: The aec dump input file cannot be specified together with the "
-        "aec dump input string!\n");
+        !settings.aec_dump_input_filename,
+        "Error: The aec dump input file must be specified!\n");
   }
 
   ReportConditionalErrorAndExit(settings.use_aec && !(*settings.use_aec) &&
@@ -857,7 +847,7 @@
   RTC_CHECK(audio_processing);
 
   std::unique_ptr<AudioProcessingSimulator> processor;
-  if (settings.aec_dump_input_filename || settings.aec_dump_input_string) {
+  if (settings.aec_dump_input_filename) {
     processor = std::make_unique<AecDumpBasedSimulator>(
         settings, std::move(audio_processing));
   } else {
diff --git a/modules/audio_processing/test/protobuf_utils.cc b/modules/audio_processing/test/protobuf_utils.cc
index fb0c082..227966e 100644
--- a/modules/audio_processing/test/protobuf_utils.cc
+++ b/modules/audio_processing/test/protobuf_utils.cc
@@ -14,35 +14,12 @@
 #include <cstdint>
 #include <cstdio>
 #include <memory>
-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
 
 #include "rtc_base/protobuf_utils.h"
 #include "rtc_base/system/arch.h"
 
 namespace webrtc {
 
-namespace {
-// Allocates new memory in the memory owned by the unique_ptr to fit the raw
-// message and returns the number of bytes read when having a string stream as
-// input.
-size_t ReadMessageBytesFromString(std::stringstream* input,
-                                  std::unique_ptr<uint8_t[]>* bytes) {
-  int32_t size = 0;
-  input->read(reinterpret_cast<char*>(&size), sizeof(int32_t));
-  int32_t size_read = input->gcount();
-  if (size_read != sizeof(int32_t))
-    return 0;
-  if (size <= 0)
-    return 0;
-
-  *bytes = std::make_unique<uint8_t[]>(size);
-  input->read(reinterpret_cast<char*>(bytes->get()),
-              size * sizeof((*bytes)[0]));
-  size_read = input->gcount();
-  return size_read == size ? size : 0;
-}
-}  // namespace
-
 size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) {
 // The "wire format" for the size is little-endian. Assume we're running on
 // a little-endian machine.
@@ -70,15 +47,4 @@
   return msg->ParseFromArray(bytes.get(), size);
 }
 
-// Returns true on success, false on error or end of string stream.
-bool ReadMessageFromString(std::stringstream* input, MessageLite* msg) {
-  std::unique_ptr<uint8_t[]> bytes;
-  size_t size = ReadMessageBytesFromString(input, &bytes);
-  if (!size)
-    return false;
-
-  msg->Clear();
-  return msg->ParseFromArray(bytes.get(), size);
-}
-
 }  // namespace webrtc
diff --git a/modules/audio_processing/test/protobuf_utils.h b/modules/audio_processing/test/protobuf_utils.h
index 2685a4c..77f1491 100644
--- a/modules/audio_processing/test/protobuf_utils.h
+++ b/modules/audio_processing/test/protobuf_utils.h
@@ -15,7 +15,6 @@
 #include <cstdint>
 #include <cstdio>
 #include <memory>
-#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
 
 #include "rtc_base/protobuf_utils.h"
 
@@ -31,11 +30,6 @@
 // Returns true on success, false on error or end-of-file.
 bool ReadMessageFromFile(FILE* file, MessageLite* msg);
 
-// Returns true on success, false on error or end of string stream.
-bool ReadMessageFromString(
-    std::stringstream* input,  // no-presubmit-check TODO(webrtc:8982)
-    MessageLite* msg);
-
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_
diff --git a/modules/audio_processing/test/test_utils.cc b/modules/audio_processing/test/test_utils.cc
index d4b221d..24cf176 100644
--- a/modules/audio_processing/test/test_utils.cc
+++ b/modules/audio_processing/test/test_utils.cc
@@ -113,25 +113,6 @@
   file_->WriteSamples(&interleaved_[0], interleaved_.size());
 }
 
-ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
-    : output_(output) {
-  RTC_DCHECK(output_);
-}
-
-ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;
-
-void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) {
-  // Account for sample rate changes throughout a simulation.
-  interleaved_buffer_.resize(buffer.size());
-  InterleavedView<float> view(&interleaved_buffer_[0], buffer.num_frames(),
-                              buffer.num_channels());
-  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
-             view);
-  size_t old_size = output_->size();
-  output_->resize(old_size + interleaved_buffer_.size());
-  FloatToFloatS16(interleaved_buffer_.data(), interleaved_buffer_.size(),
-                  output_->data() + old_size);
-}
 
 FILE* OpenFile(absl::string_view filename, absl::string_view mode) {
   std::string filename_str(filename);
diff --git a/modules/audio_processing/test/test_utils.h b/modules/audio_processing/test/test_utils.h
index e979c85..114cd13 100644
--- a/modules/audio_processing/test/test_utils.h
+++ b/modules/audio_processing/test/test_utils.h
@@ -101,25 +101,6 @@
   std::vector<float> interleaved_;
 };
 
-// Takes a pointer to a vector. Allows appending the samples of channel buffers
-// to the given vector, by interleaving the samples and converting them to float
-// S16.
-class ChannelBufferVectorWriter final {
- public:
-  explicit ChannelBufferVectorWriter(std::vector<float>* output);
-  ChannelBufferVectorWriter(const ChannelBufferVectorWriter&) = delete;
-  ChannelBufferVectorWriter& operator=(const ChannelBufferVectorWriter&) =
-      delete;
-  ~ChannelBufferVectorWriter();
-
-  // Creates an interleaved copy of `buffer`, converts the samples to float S16
-  // and appends the result to output_.
-  void Write(const ChannelBuffer<float>& buffer);
-
- private:
-  std::vector<float> interleaved_buffer_;
-  std::vector<float>* output_;
-};
 
 // Exits on failure; do not use in unit tests.
 FILE* OpenFile(absl::string_view filename, absl::string_view mode);