diff --git a/media/BUILD.gn b/media/BUILD.gn
index 5cf4175a25..cdae182406 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -282,7 +282,6 @@ rtc_static_library("rtc_audio_video") {
     "../modules/video_coding",
     "../modules/video_coding:video_codec_interface",
     "../modules/video_coding:video_coding_utility",
-    "../pc:rtc_pc_base",
     "../rtc_base",
     "../rtc_base:audio_format_to_string",
     "../rtc_base:checks",
@@ -540,8 +539,6 @@ if (rtc_include_tests) {
       "../modules/video_coding:video_codec_interface",
       "../modules/video_coding:webrtc_vp8",
       "../p2p:p2p_test_utils",
-      "../pc:rtc_pc",
-      "../pc:rtc_pc_base",
       "../rtc_base",
       "../rtc_base:checks",
       "../rtc_base:gunit_helpers",
diff --git a/media/DEPS b/media/DEPS
index e3e03c54fc..1e13c18b16 100644
--- a/media/DEPS
+++ b/media/DEPS
@@ -11,7 +11,6 @@ include_rules = [
   "+modules/video_coding",
   "+modules/video_coding/utility",
   "+p2p",
-  "+pc",
   "+sound",
   "+system_wrappers",
   "+usrsctplib",
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 55a2826ca9..0635c9f3db 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -609,10 +609,10 @@ webrtc::AudioState* WebRtcVoiceEngine::audio_state() {
   return audio_state_.get();
 }
 
-AudioCodecs WebRtcVoiceEngine::CollectCodecs(
+std::vector<AudioCodec> WebRtcVoiceEngine::CollectCodecs(
     const std::vector<webrtc::AudioCodecSpec>& specs) const {
   PayloadTypeMapper mapper;
-  AudioCodecs out;
+  std::vector<AudioCodec> out;
 
   // Only generate CN payload types for these clockrates:
   std::map<int, bool, std::greater<int>> generate_cn = {
@@ -622,7 +622,7 @@ AudioCodecs WebRtcVoiceEngine::CollectCodecs(
       {8000, false}, {16000, false}, {32000, false}, {48000, false}};
 
   auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
-                              AudioCodecs* out) {
+                              std::vector<AudioCodec>* out) {
     absl::optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
     if (opt_codec) {
       if (out) {
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index eb9f3ad4f5..aaa2778a21 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -22,10 +22,9 @@
 #include "api/task_queue/task_queue_factory.h"
 #include "call/audio_state.h"
 #include "call/call.h"
+#include "media/base/media_engine.h"
 #include "media/base/rtp_utils.h"
 #include "media/engine/apm_helpers.h"
-#include "modules/audio_processing/include/audio_processing.h"
-#include "pc/channel.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
 #include "rtc_base/experiments/audio_allocation_settings.h"
@@ -99,7 +98,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   webrtc::AudioProcessing* apm() const;
   webrtc::AudioState* audio_state();
 
-  AudioCodecs CollectCodecs(
+  std::vector<AudioCodec> CollectCodecs(
       const std::vector<webrtc::AudioCodecSpec>& specs) const;
 
   rtc::ThreadChecker signal_thread_checker_;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 65f3c7b33c..3cbad9fa62 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -28,7 +28,6 @@
 #include "media/engine/webrtc_voice_engine.h"
 #include "modules/audio_device/include/mock_audio_device.h"
 #include "modules/audio_processing/include/mock_audio_processing.h"
-#include "pc/channel.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/byte_order.h"
 #include "rtc_base/numerics/safe_conversions.h"