diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 8bb54d6f2d..e7cee58f14 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -1118,7 +1118,7 @@ void ChannelSend::ProcessAndEncodeAudio(
 
 void ChannelSend::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
   RTC_DCHECK_GT(audio_input->samples_per_channel_, 0);
-  RTC_DCHECK_LE(audio_input->num_channels_, 2);
+  RTC_DCHECK_LE(audio_input->num_channels_, 8);
 
   // Measure time between when the audio frame is added to the task queue and
   // when the task is actually executed. Goal is to keep track of unwanted
diff --git a/modules/audio_mixer/DEPS b/modules/audio_mixer/DEPS
index dcee4a37d3..46f29bccf8 100644
--- a/modules/audio_mixer/DEPS
+++ b/modules/audio_mixer/DEPS
@@ -1,5 +1,6 @@
 include_rules = [
   "+audio/utility/audio_frame_operations.h",
+  "+audio/utility/channel_mixer.h",
   "+call",
   "+common_audio",
   "+modules/audio_coding",
diff --git a/modules/audio_mixer/audio_frame_manipulator.cc b/modules/audio_mixer/audio_frame_manipulator.cc
index 78c11b19c4..3100271cfb 100644
--- a/modules/audio_mixer/audio_frame_manipulator.cc
+++ b/modules/audio_mixer/audio_frame_manipulator.cc
@@ -11,6 +11,7 @@
 #include "modules/audio_mixer/audio_frame_manipulator.h"
 
 #include "audio/utility/audio_frame_operations.h"
+#include "audio/utility/channel_mixer.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -56,16 +57,36 @@ void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) {
 
 void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
   RTC_DCHECK_GE(target_number_of_channels, 1);
-  if (frame->num_channels_ == target_number_of_channels) {
+  // TODO(bugs.webrtc.org/10783): take channel layout into account as well.
+  if (frame->num_channels() == target_number_of_channels) {
     return;
   }
-  if (frame->num_channels_ > target_number_of_channels) {
-    AudioFrameOperations::DownmixChannels(target_number_of_channels, frame);
-  } else if (frame->num_channels_ < target_number_of_channels) {
-    AudioFrameOperations::UpmixChannels(target_number_of_channels, frame);
+
+  // Use legacy components for the most simple cases (mono <-> stereo) to ensure
+  // that native WebRTC clients are not affected when support for multi-channel
+  // audio is added to Chrome.
+  // TODO(bugs.webrtc.org/10783): utilize channel mixer for mono/stereo as well.
+  if (target_number_of_channels < 3 && frame->num_channels() < 3) {
+    if (frame->num_channels() > target_number_of_channels) {
+      AudioFrameOperations::DownmixChannels(target_number_of_channels, frame);
+    } else {
+      AudioFrameOperations::UpmixChannels(target_number_of_channels, frame);
+    }
+  } else {
+    // Use generic channel mixer when the number of channels for input our
+    // output is larger than two. E.g. stereo -> 5.1 channel up-mixing.
+    // TODO(bugs.webrtc.org/10783): ensure that actual channel layouts are used
+    // instead of guessing based on number of channels.
+    const ChannelLayout output_layout(
+        GuessChannelLayout(target_number_of_channels));
+    ChannelMixer mixer(GuessChannelLayout(frame->num_channels()),
+                       output_layout);
+    mixer.Transform(frame);
+    RTC_DCHECK_EQ(frame->channel_layout(), output_layout);
   }
-  RTC_DCHECK_EQ(frame->num_channels_, target_number_of_channels)
-      << "Wrong number of channels, " << frame->num_channels_ << " vs "
+  RTC_DCHECK_EQ(frame->num_channels(), target_number_of_channels)
+      << "Wrong number of channels, " << frame->num_channels() << " vs "
       << target_number_of_channels;
 }
+
 }  // namespace webrtc