diff --git a/api/BUILD.gn b/api/BUILD.gn
index 14faff8034..ce6662743f 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -238,7 +238,7 @@ rtc_source_set("audio_options_api") {
   ]
 
   deps = [
-    "../rtc_base:rtc_base_approved",
+    "../rtc_base:stringutils",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/api/audio_options.cc b/api/audio_options.cc
index c196d7d9df..a4a37d219a 100644
--- a/api/audio_options.cc
+++ b/api/audio_options.cc
@@ -10,9 +10,120 @@
 
 #include "api/audio_options.h"
 
+#include "rtc_base/strings/string_builder.h"
+
 namespace cricket {
+namespace {
+
+template <class T>
+void ToStringIfSet(rtc::SimpleStringBuilder* result,
+                   const char* key,
+                   const absl::optional<T>& val) {
+  if (val) {
+    (*result) << key << ": " << *val << ", ";
+  }
+}
+
+template <typename T>
+void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
+  if (o) {
+    *s = o;
+  }
+}
+
+}  // namespace
 
 AudioOptions::AudioOptions() = default;
 AudioOptions::~AudioOptions() = default;
 
+void AudioOptions::SetAll(const AudioOptions& change) {
+  SetFrom(&echo_cancellation, change.echo_cancellation);
+#if defined(WEBRTC_IOS)
+  SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
+#endif
+  SetFrom(&auto_gain_control, change.auto_gain_control);
+  SetFrom(&noise_suppression, change.noise_suppression);
+  SetFrom(&highpass_filter, change.highpass_filter);
+  SetFrom(&stereo_swapping, change.stereo_swapping);
+  SetFrom(&audio_jitter_buffer_max_packets,
+          change.audio_jitter_buffer_max_packets);
+  SetFrom(&audio_jitter_buffer_fast_accelerate,
+          change.audio_jitter_buffer_fast_accelerate);
+  SetFrom(&typing_detection, change.typing_detection);
+  SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
+  SetFrom(&experimental_agc, change.experimental_agc);
+  SetFrom(&extended_filter_aec, change.extended_filter_aec);
+  SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
+  SetFrom(&experimental_ns, change.experimental_ns);
+  SetFrom(&residual_echo_detector, change.residual_echo_detector);
+  SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
+  SetFrom(&tx_agc_digital_compression_gain,
+          change.tx_agc_digital_compression_gain);
+  SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
+  SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
+  SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
+  SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
+}
+
+bool AudioOptions::operator==(const AudioOptions& o) const {
+  return echo_cancellation == o.echo_cancellation &&
+#if defined(WEBRTC_IOS)
+         ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
+#endif
+         auto_gain_control == o.auto_gain_control &&
+         noise_suppression == o.noise_suppression &&
+         highpass_filter == o.highpass_filter &&
+         stereo_swapping == o.stereo_swapping &&
+         audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
+         audio_jitter_buffer_fast_accelerate ==
+             o.audio_jitter_buffer_fast_accelerate &&
+         typing_detection == o.typing_detection &&
+         aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
+         experimental_agc == o.experimental_agc &&
+         extended_filter_aec == o.extended_filter_aec &&
+         delay_agnostic_aec == o.delay_agnostic_aec &&
+         experimental_ns == o.experimental_ns &&
+         residual_echo_detector == o.residual_echo_detector &&
+         tx_agc_target_dbov == o.tx_agc_target_dbov &&
+         tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
+         tx_agc_limiter == o.tx_agc_limiter &&
+         combined_audio_video_bwe == o.combined_audio_video_bwe &&
+         audio_network_adaptor == o.audio_network_adaptor &&
+         audio_network_adaptor_config == o.audio_network_adaptor_config;
+}
+
+std::string AudioOptions::ToString() const {
+  char buffer[1024];
+  rtc::SimpleStringBuilder result(buffer);
+  result << "AudioOptions {";
+  ToStringIfSet(&result, "aec", echo_cancellation);
+#if defined(WEBRTC_IOS)
+  ToStringIfSet(&result, "ios_force_software_aec_HACK",
+                ios_force_software_aec_HACK);
+#endif
+  ToStringIfSet(&result, "agc", auto_gain_control);
+  ToStringIfSet(&result, "ns", noise_suppression);
+  ToStringIfSet(&result, "hf", highpass_filter);
+  ToStringIfSet(&result, "swap", stereo_swapping);
+  ToStringIfSet(&result, "audio_jitter_buffer_max_packets",
+                audio_jitter_buffer_max_packets);
+  ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
+                audio_jitter_buffer_fast_accelerate);
+  ToStringIfSet(&result, "typing", typing_detection);
+  ToStringIfSet(&result, "comfort_noise", aecm_generate_comfort_noise);
+  ToStringIfSet(&result, "experimental_agc", experimental_agc);
+  ToStringIfSet(&result, "extended_filter_aec", extended_filter_aec);
+  ToStringIfSet(&result, "delay_agnostic_aec", delay_agnostic_aec);
+  ToStringIfSet(&result, "experimental_ns", experimental_ns);
+  ToStringIfSet(&result, "residual_echo_detector", residual_echo_detector);
+  ToStringIfSet(&result, "tx_agc_target_dbov", tx_agc_target_dbov);
+  ToStringIfSet(&result, "tx_agc_digital_compression_gain",
+                tx_agc_digital_compression_gain);
+  ToStringIfSet(&result, "tx_agc_limiter", tx_agc_limiter);
+  ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
+  ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
+  result << "}";
+  return result.str();
+}
+
 }  // namespace cricket
diff --git a/api/audio_options.h b/api/audio_options.h
index aefc7a1739..aecff41429 100644
--- a/api/audio_options.h
+++ b/api/audio_options.h
@@ -14,7 +14,6 @@
 #include <string>
 
 #include "absl/types/optional.h"
-#include "rtc_base/stringencode.h"
 
 namespace cricket {
 
@@ -25,101 +24,12 @@ namespace cricket {
 struct AudioOptions {
   AudioOptions();
   ~AudioOptions();
-  void SetAll(const AudioOptions& change) {
-    SetFrom(&echo_cancellation, change.echo_cancellation);
-#if defined(WEBRTC_IOS)
-    SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
-#endif
-    SetFrom(&auto_gain_control, change.auto_gain_control);
-    SetFrom(&noise_suppression, change.noise_suppression);
-    SetFrom(&highpass_filter, change.highpass_filter);
-    SetFrom(&stereo_swapping, change.stereo_swapping);
-    SetFrom(&audio_jitter_buffer_max_packets,
-            change.audio_jitter_buffer_max_packets);
-    SetFrom(&audio_jitter_buffer_fast_accelerate,
-            change.audio_jitter_buffer_fast_accelerate);
-    SetFrom(&typing_detection, change.typing_detection);
-    SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
-    SetFrom(&experimental_agc, change.experimental_agc);
-    SetFrom(&extended_filter_aec, change.extended_filter_aec);
-    SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
-    SetFrom(&experimental_ns, change.experimental_ns);
-    SetFrom(&residual_echo_detector, change.residual_echo_detector);
-    SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
-    SetFrom(&tx_agc_digital_compression_gain,
-            change.tx_agc_digital_compression_gain);
-    SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
-    SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
-    SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
-    SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
-  }
+  void SetAll(const AudioOptions& change);
 
-  bool operator==(const AudioOptions& o) const {
-    return echo_cancellation == o.echo_cancellation &&
-#if defined(WEBRTC_IOS)
-           ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
-#endif
-           auto_gain_control == o.auto_gain_control &&
-           noise_suppression == o.noise_suppression &&
-           highpass_filter == o.highpass_filter &&
-           stereo_swapping == o.stereo_swapping &&
-           audio_jitter_buffer_max_packets ==
-               o.audio_jitter_buffer_max_packets &&
-           audio_jitter_buffer_fast_accelerate ==
-               o.audio_jitter_buffer_fast_accelerate &&
-           typing_detection == o.typing_detection &&
-           aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
-           experimental_agc == o.experimental_agc &&
-           extended_filter_aec == o.extended_filter_aec &&
-           delay_agnostic_aec == o.delay_agnostic_aec &&
-           experimental_ns == o.experimental_ns &&
-           residual_echo_detector == o.residual_echo_detector &&
-           tx_agc_target_dbov == o.tx_agc_target_dbov &&
-           tx_agc_digital_compression_gain ==
-               o.tx_agc_digital_compression_gain &&
-           tx_agc_limiter == o.tx_agc_limiter &&
-           combined_audio_video_bwe == o.combined_audio_video_bwe &&
-           audio_network_adaptor == o.audio_network_adaptor &&
-           audio_network_adaptor_config == o.audio_network_adaptor_config;
-  }
+  bool operator==(const AudioOptions& o) const;
   bool operator!=(const AudioOptions& o) const { return !(*this == o); }
 
-  std::string ToString() const {
-    std::ostringstream ost;
-    ost << "AudioOptions {";
-    ost << ToStringIfSet("aec", echo_cancellation);
-#if defined(WEBRTC_IOS)
-    ost << ToStringIfSet("ios_force_software_aec_HACK",
-                         ios_force_software_aec_HACK);
-#endif
-    ost << ToStringIfSet("agc", auto_gain_control);
-    ost << ToStringIfSet("ns", noise_suppression);
-    ost << ToStringIfSet("hf", highpass_filter);
-    ost << ToStringIfSet("swap", stereo_swapping);
-    ost << ToStringIfSet("audio_jitter_buffer_max_packets",
-                         audio_jitter_buffer_max_packets);
-    ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
-                         audio_jitter_buffer_fast_accelerate);
-    ost << ToStringIfSet("typing", typing_detection);
-    ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
-    ost << ToStringIfSet("experimental_agc", experimental_agc);
-    ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
-    ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
-    ost << ToStringIfSet("experimental_ns", experimental_ns);
-    ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
-    ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
-    ost << ToStringIfSet("tx_agc_digital_compression_gain",
-                         tx_agc_digital_compression_gain);
-    ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
-    ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
-    ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
-    // The adaptor config is a serialized proto buffer and therefore not human
-    // readable. So we comment out the following line.
-    // ost << ToStringIfSet("audio_network_adaptor_config",
-    //     audio_network_adaptor_config);
-    ost << "}";
-    return ost.str();
-  }
+  std::string ToString() const;
 
   // Audio processing that attempts to filter away the output signal from
   // later inbound pickup.
@@ -164,27 +74,6 @@ struct AudioOptions {
   absl::optional<bool> audio_network_adaptor;
   // Config string for audio network adaptor.
   absl::optional<std::string> audio_network_adaptor_config;
-
- private:
-  template <class T>
-  static std::string ToStringIfSet(const char* key,
-                                   const absl::optional<T>& val) {
-    std::string str;
-    if (val) {
-      str = key;
-      str += ": ";
-      str += val ? rtc::ToString(*val) : "";
-      str += ", ";
-    }
-    return str;
-  }
-
-  template <typename T>
-  static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
-    if (o) {
-      *s = o;
-    }
-  }
 };
 
 }  // namespace cricket