diff --git a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
index 208a50c7c2..df67ce2914 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
@@ -21,9 +21,6 @@
 
 namespace webrtc {
 
-// The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo
-#define MAX_PAYLOAD_SIZE_BYTE 7680
-
 // General codec specific defines
 const int kIsacWbDefaultRate = 32000;
 const int kIsacSwbDefaultRate = 56000;
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 72de1b1c3d..ac0bc0b270 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -157,7 +157,6 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(
 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
 
 int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
-  uint8_t stream[2 * MAX_PAYLOAD_SIZE_BYTE];  // Make room for 1 RED payload.
   AudioEncoder::EncodedInfo encoded_info;
   uint8_t previous_pltype;
 
@@ -179,11 +178,13 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   last_rtp_timestamp_ = rtp_timestamp;
   first_frame_ = false;
 
-  encoded_info = audio_encoder->Encode(rtp_timestamp, input_data.audio,
-                                       input_data.length_per_channel,
-                                       sizeof(stream), stream);
+  encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());
+  encoded_info = audio_encoder->Encode(
+      rtp_timestamp, input_data.audio, input_data.length_per_channel,
+      encode_buffer_.size(), encode_buffer_.data());
+  encode_buffer_.SetSize(encoded_info.encoded_bytes);
   bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);
-  if (encoded_info.encoded_bytes == 0 && !encoded_info.send_even_if_empty) {
+  if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
     // Not enough data.
     return 0;
   }
@@ -192,11 +193,11 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   RTPFragmentationHeader my_fragmentation;
   ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
   FrameType frame_type;
-  if (encoded_info.encoded_bytes == 0 && encoded_info.send_even_if_empty) {
+  if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
     frame_type = kFrameEmpty;
     encoded_info.payload_type = previous_pltype;
   } else {
-    DCHECK_GT(encoded_info.encoded_bytes, 0u);
+    DCHECK_GT(encode_buffer_.size(), 0u);
     frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
   }
 
@@ -205,7 +206,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
     if (packetization_callback_) {
       packetization_callback_->SendData(
           frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
-          stream, encoded_info.encoded_bytes,
+          encode_buffer_.data(), encode_buffer_.size(),
           my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
                                                        : nullptr);
     }
@@ -216,7 +217,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
     }
   }
   previous_pltype_ = encoded_info.payload_type;
-  return static_cast<int32_t>(encoded_info.encoded_bytes);
+  return static_cast<int32_t>(encode_buffer_.size());
 }
 
 /////////////////////////////////////////
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 842aca076b..fd93901205 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
@@ -265,6 +266,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   int UpdateUponReceivingCodec(int index);
 
   const rtc::scoped_ptr<CriticalSectionWrapper> acm_crit_sect_;
+  rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
   int id_;  // TODO(henrik.lundin) Make const.
   uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
   uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index e5371d0b23..74f65a99f4 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -1619,6 +1619,10 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
   EXPECT_CALL(mock_encoder, NumChannels())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&encoder, &AudioEncoderMutablePcmU::NumChannels));
+  EXPECT_CALL(mock_encoder, MaxEncodedBytes())
+      .Times(AtLeast(1))
+      .WillRepeatedly(
+          Invoke(&encoder, &AudioEncoderMutablePcmU::MaxEncodedBytes));
   EXPECT_CALL(mock_encoder, EncodeInternal(_, _, _, _))
       .Times(AtLeast(1))
       .WillRepeatedly(