diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 5bb568e4cf..486dcb11ac 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -888,6 +888,9 @@ int ChannelReceive::GetRtpTimestampRateHz() const {
   // TODO(ossu): Zero clockrate can only happen if we've added an external
   // decoder for a format we don't support internally. Remove once that way of
   // adding decoders is gone!
+  // TODO(kwiberg): `decoder->second.clockrate_hz` is an RTP clockrate as it
+  // should, but `acm_receiver_.last_output_sample_rate_hz()` is a codec sample
+  // rate, which is not always the same thing.
   return (decoder && decoder->second.clockrate_hz != 0)
              ? decoder->second.clockrate_hz
              : acm_receiver_.last_output_sample_rate_hz();
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 1c8d88da6e..40196151af 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -73,7 +73,7 @@ absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
   if (!last_decoder_) {
     return absl::nullopt;
   }
-  return last_decoder_->second.clockrate_hz;
+  return last_decoder_->sample_rate_hz;
 }
 
 int AcmReceiver::last_output_sample_rate_hz() const {
@@ -89,7 +89,7 @@ int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
 
   int payload_type = rtp_header.payloadType;
   auto format = neteq_->GetDecoderFormat(payload_type);
-  if (format && absl::EqualsIgnoreCase(format->name, "red")) {
+  if (format && absl::EqualsIgnoreCase(format->sdp_format.name, "red")) {
     // This is a RED packet. Get the format of the audio codec.
     payload_type = incoming_payload[0] & 0x7f;
     format = neteq_->GetDecoderFormat(payload_type);
@@ -102,15 +102,17 @@ int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
 
   {
     rtc::CritScope lock(&crit_sect_);
-    if (absl::EqualsIgnoreCase(format->name, "cn")) {
-      if (last_decoder_ && last_decoder_->second.num_channels > 1) {
+    if (absl::EqualsIgnoreCase(format->sdp_format.name, "cn")) {
+      if (last_decoder_ && last_decoder_->num_channels > 1) {
         // This is a CNG and the audio codec is not mono, so skip pushing in
         // packets into NetEq.
         return 0;
       }
     } else {
-      RTC_DCHECK(format);
-      last_decoder_ = std::make_pair(payload_type, *format);
+      last_decoder_ = DecoderInfo{/*payload_type=*/payload_type,
+                                  /*sample_rate_hz=*/format->sample_rate_hz,
+                                  /*num_channels=*/format->num_channels,
+                                  /*sdp_format=*/std::move(format->sdp_format)};
     }
   }  // |crit_sect_| is released.
 
@@ -221,8 +223,8 @@ absl::optional<std::pair<int, SdpAudioFormat>> AcmReceiver::LastDecoder()
   if (!last_decoder_) {
     return absl::nullopt;
   }
-  RTC_DCHECK_NE(-1, last_decoder_->first);  // Payload type should be valid.
-  return last_decoder_;
+  RTC_DCHECK_NE(-1, last_decoder_->payload_type);
+  return std::make_pair(last_decoder_->payload_type, last_decoder_->sdp_format);
 }
 
 void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) const {
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index f07f8a951c..15126566ae 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -203,11 +203,17 @@ class AcmReceiver {
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
  private:
+  struct DecoderInfo {
+    int payload_type;
+    int sample_rate_hz;
+    int num_channels;
+    SdpAudioFormat sdp_format;
+  };
+
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
   rtc::CriticalSection crit_sect_;
-  absl::optional<std::pair<int, SdpAudioFormat>> last_decoder_
-      RTC_GUARDED_BY(crit_sect_);
+  absl::optional<DecoderInfo> last_decoder_ RTC_GUARDED_BY(crit_sect_);
   ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
   std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index c6af751757..b53b5ada9f 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -143,6 +143,13 @@ class NetEq {
 
   enum ReturnCodes { kOK = 0, kFail = -1 };
 
+  // Return type for GetDecoderFormat.
+  struct DecoderFormat {
+    int sample_rate_hz;
+    int num_channels;
+    SdpAudioFormat sdp_format;
+  };
+
   // Creates a new NetEq object, with parameters set in |config|. The |config|
   // object will only have to be valid for the duration of the call to this
   // method.
@@ -265,7 +272,7 @@ class NetEq {
 
   // Returns the decoder info for the given payload type. Returns empty if no
   // such payload type was registered.
-  virtual absl::optional<SdpAudioFormat> GetDecoderFormat(
+  virtual absl::optional<DecoderFormat> GetDecoderFormat(
       int payload_type) const = 0;
 
   // Flushes both the packet buffer and the sync buffer.
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 751fc457de..37036e34c4 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -392,21 +392,23 @@ int NetEqImpl::last_output_sample_rate_hz() const {
   return last_output_sample_rate_hz_;
 }
 
-absl::optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
+absl::optional<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat(
     int payload_type) const {
   rtc::CritScope lock(&crit_sect_);
   const DecoderDatabase::DecoderInfo* const di =
       decoder_database_->GetDecoderInfo(payload_type);
-  if (!di) {
-    return absl::nullopt;  // Payload type not registered.
+  if (di) {
+    const AudioDecoder* const decoder = di->GetDecoder();
+    // TODO(kwiberg): Why the special case for RED?
+    return DecoderFormat{
+        /*sample_rate_hz=*/di->IsRed() ? 8000 : di->SampleRateHz(),
+        /*num_channels=*/
+        decoder ? rtc::dchecked_cast<int>(decoder->Channels()) : 1,
+        /*sdp_format=*/di->GetFormat()};
+  } else {
+    // Payload type not registered.
+    return absl::nullopt;
   }
-
-  SdpAudioFormat format = di->GetFormat();
-  // TODO(solenberg): This is legacy but messed up - mixing RTP rate and SR.
-  format.clockrate_hz = di->IsRed() ? 8000 : di->SampleRateHz();
-  const AudioDecoder* const decoder = di->GetDecoder();
-  format.num_channels = decoder ? decoder->Channels() : 1;
-  return format;
 }
 
 void NetEqImpl::FlushBuffers() {
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 8ecb9b6f74..842869f9fc 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -182,7 +182,7 @@ class NetEqImpl : public webrtc::NetEq {
 
   int last_output_sample_rate_hz() const override;
 
-  absl::optional<SdpAudioFormat> GetDecoderFormat(
+  absl::optional<DecoderFormat> GetDecoderFormat(
       int payload_type) const override;
 
   // Flushes both the packet buffer and the sync buffer.