diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.cc b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
index 442ddc1e4b..e91161e33c 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.cc
@@ -82,6 +82,10 @@ bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
   return false;
 }
 
+int AudioDecoder::SampleRateHz() const {
+  return -1;
+}
+
 AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
   switch (type) {
     case 0:  // TODO(hlundin): Both iSAC and Opus return 0 for speech.
diff --git a/webrtc/modules/audio_coding/codecs/audio_decoder.h b/webrtc/modules/audio_coding/codecs/audio_decoder.h
index 580ddbf74f..c77a069fc3 100644
--- a/webrtc/modules/audio_coding/codecs/audio_decoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -93,6 +93,14 @@ class AudioDecoder {
   // Returns true if the packet has FEC and false otherwise.
   virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
 
+  // Returns the actual sample rate of the decoder's output.
+  // NOTE: For now, this has a default implementation that returns an unusable
+  // value (-1). That default implementation will go away soon, and at the same
+  // time callers will start relying on the return value, so make sure you
+  // override it with something that returns a correct value!
+  // TODO(kwiberg): Remove the default implementation.
+  virtual int SampleRateHz() const;
+
   virtual size_t Channels() const = 0;
 
  protected:
diff --git a/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc b/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
index deb1b1f4bd..48b2f5d437 100644
--- a/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
+++ b/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc
@@ -74,7 +74,7 @@ NamedDecoderConstructor decoder_constructors[] = {
     {"isac",
      [](const SdpAudioFormat& format) {
        return format.clockrate_hz == 16000 && format.num_channels == 1
-                  ? Unique(new AudioDecoderIsacFix)
+                  ? Unique(new AudioDecoderIsacFix(format.clockrate_hz))
                   : nullptr;
      }},
 #elif defined(WEBRTC_CODEC_ISAC)
@@ -82,14 +82,15 @@ NamedDecoderConstructor decoder_constructors[] = {
      [](const SdpAudioFormat& format) {
        return (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
                       format.num_channels == 1
-                  ? Unique(new AudioDecoderIsac)
+                  ? Unique(new AudioDecoderIsac(format.clockrate_hz))
                   : nullptr;
      }},
 #endif
     {"l16",
      [](const SdpAudioFormat& format) {
        return format.num_channels >= 1
-                  ? Unique(new AudioDecoderPcm16B(format.num_channels))
+                  ? Unique(new AudioDecoderPcm16B(format.clockrate_hz,
+                                                  format.num_channels))
                   : nullptr;
      }},
 #ifdef WEBRTC_CODEC_G722
@@ -136,7 +137,15 @@ class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
       const SdpAudioFormat& format) override {
     for (const auto& dc : decoder_constructors) {
       if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
-        return std::unique_ptr<AudioDecoder>(dc.constructor(format));
+        std::unique_ptr<AudioDecoder> dec = dc.constructor(format);
+        if (dec) {
+          const int expected_sample_rate_hz =
+              STR_CASE_CMP(format.name.c_str(), "g722") == 0
+                  ? 2 * format.clockrate_hz
+                  : format.clockrate_hz;
+          RTC_CHECK_EQ(expected_sample_rate_hz, dec->SampleRateHz());
+        }
+        return dec;
       }
     }
     return nullptr;
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index 9757b4a010..af164c4bb7 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -16,6 +16,10 @@ namespace webrtc {
 
 void AudioDecoderPcmU::Reset() {}
 
+int AudioDecoderPcmU::SampleRateHz() const {
+  return 8000;
+}
+
 size_t AudioDecoderPcmU::Channels() const {
   return num_channels_;
 }
@@ -25,7 +29,7 @@ int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeU(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
@@ -40,6 +44,10 @@ int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
 
 void AudioDecoderPcmA::Reset() {}
 
+int AudioDecoderPcmA::SampleRateHz() const {
+  return 8000;
+}
+
 size_t AudioDecoderPcmA::Channels() const {
   return num_channels_;
 }
@@ -49,7 +57,7 @@ int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret = WebRtcG711_DecodeA(encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 7a627e757c..7fdc3591b3 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -24,6 +24,7 @@ class AudioDecoderPcmU final : public AudioDecoder {
   }
   void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
   size_t Channels() const override;
 
  protected:
@@ -45,6 +46,7 @@ class AudioDecoderPcmA final : public AudioDecoder {
   }
   void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
   size_t Channels() const override;
 
  protected:
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
index 7676e90d9e..379293b748 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -35,7 +35,7 @@ int AudioDecoderG722::DecodeInternal(const uint8_t* encoded,
                                      int sample_rate_hz,
                                      int16_t* decoded,
                                      SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 16000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   size_t ret =
       WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
@@ -53,6 +53,10 @@ int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
   return static_cast<int>(2 * encoded_len / Channels());
 }
 
+int AudioDecoderG722::SampleRateHz() const {
+  return 16000;
+}
+
 size_t AudioDecoderG722::Channels() const {
   return 1;
 }
@@ -74,7 +78,7 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
                                            int sample_rate_hz,
                                            int16_t* decoded,
                                            SpeechType* speech_type) {
-  RTC_DCHECK_EQ(sample_rate_hz, 16000);
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
   int16_t temp_type = 1;  // Default is speech.
   // De-interleave the bit-stream into two separate payloads.
   uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
@@ -100,6 +104,10 @@ int AudioDecoderG722Stereo::DecodeInternal(const uint8_t* encoded,
   return static_cast<int>(ret);
 }
 
+int AudioDecoderG722Stereo::SampleRateHz() const {
+  return 16000;
+}
+
 size_t AudioDecoderG722Stereo::Channels() const {
   return 2;
 }
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
index 1837ffabe2..ccca73d1a1 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -25,6 +25,7 @@ class AudioDecoderG722 final : public AudioDecoder {
   bool HasDecodePlc() const override;
   void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
   size_t Channels() const override;
 
  protected:
@@ -44,6 +45,8 @@ class AudioDecoderG722Stereo final : public AudioDecoder {
   AudioDecoderG722Stereo();
   ~AudioDecoderG722Stereo() override;
   void Reset() override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
 
  protected:
   int DecodeInternal(const uint8_t* encoded,
@@ -51,7 +54,6 @@ class AudioDecoderG722Stereo final : public AudioDecoder {
                      int sample_rate_hz,
                      int16_t* decoded,
                      SpeechType* speech_type) override;
-  size_t Channels() const override;
 
  private:
   // Splits the stereo-interleaved payload in |encoded| into separate payloads
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
index 9ae0e1a95e..dab5805f98 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -49,6 +49,10 @@ void AudioDecoderIlbc::Reset() {
   WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
 }
 
+int AudioDecoderIlbc::SampleRateHz() const {
+  return 8000;
+}
+
 size_t AudioDecoderIlbc::Channels() const {
   return 1;
 }
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
index 036c11fac4..1083479ad8 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -25,6 +25,7 @@ class AudioDecoderIlbc final : public AudioDecoder {
   bool HasDecodePlc() const override;
   size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
   void Reset() override;
+  int SampleRateHz() const override;
   size_t Channels() const override;
 
  protected:
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
index b1907bbb39..264cca2bf3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -14,18 +14,24 @@
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 
 namespace webrtc {
 
+// TODO(kwiberg): Remove the possibility of not specifying the sample rate at
+// object creation time.
 template <typename T>
 class AudioDecoderIsacT final : public AudioDecoder {
  public:
   AudioDecoderIsacT();
   explicit AudioDecoderIsacT(
       const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
+  explicit AudioDecoderIsacT(int sample_rate_hz);
+  AudioDecoderIsacT(int sample_rate_hz,
+                    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
   ~AudioDecoderIsacT() override;
 
   bool HasDecodePlc() const override;
@@ -37,6 +43,7 @@ class AudioDecoderIsacT final : public AudioDecoder {
                      uint32_t rtp_timestamp,
                      uint32_t arrival_timestamp) override;
   int ErrorCode() override;
+  int SampleRateHz() const override;
   size_t Channels() const override;
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
@@ -45,9 +52,12 @@ class AudioDecoderIsacT final : public AudioDecoder {
                      SpeechType* speech_type) override;
 
  private:
+  AudioDecoderIsacT(rtc::Optional<int> sample_rate_hz,
+                    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo);
+
   typename T::instance_type* isac_state_;
+  rtc::Optional<int> sample_rate_hz_;
   rtc::scoped_refptr<LockedIsacBandwidthInfo> bwinfo_;
-  int decoder_sample_rate_hz_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderIsacT);
 };
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
index 4998feac84..431632112a 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
@@ -19,12 +19,31 @@ namespace webrtc {
 
 template <typename T>
 AudioDecoderIsacT<T>::AudioDecoderIsacT()
-    : AudioDecoderIsacT(nullptr) {}
+    : AudioDecoderIsacT(rtc::Optional<int>(), nullptr) {}
 
 template <typename T>
 AudioDecoderIsacT<T>::AudioDecoderIsacT(
     const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
-    : bwinfo_(bwinfo), decoder_sample_rate_hz_(-1) {
+    : AudioDecoderIsacT(rtc::Optional<int>(), bwinfo) {}
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT(int sample_rate_hz)
+    : AudioDecoderIsacT(rtc::Optional<int>(sample_rate_hz), nullptr) {}
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT(
+    int sample_rate_hz,
+    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
+    : AudioDecoderIsacT(rtc::Optional<int>(sample_rate_hz), bwinfo) {}
+
+template <typename T>
+AudioDecoderIsacT<T>::AudioDecoderIsacT(
+    rtc::Optional<int> sample_rate_hz,
+    const rtc::scoped_refptr<LockedIsacBandwidthInfo>& bwinfo)
+    : sample_rate_hz_(sample_rate_hz), bwinfo_(bwinfo) {
+  RTC_CHECK(!sample_rate_hz || *sample_rate_hz == 16000 ||
+            *sample_rate_hz == 32000)
+      << "Unsupported sample rate " << *sample_rate_hz;
   RTC_CHECK_EQ(0, T::Create(&isac_state_));
   T::DecoderInit(isac_state_);
   if (bwinfo_) {
@@ -32,6 +51,9 @@ AudioDecoderIsacT<T>::AudioDecoderIsacT(
     T::GetBandwidthInfo(isac_state_, &bi);
     bwinfo_->Set(bi);
   }
+  if (sample_rate_hz_) {
+    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, *sample_rate_hz_));
+  }
 }
 
 template <typename T>
@@ -45,11 +67,13 @@ int AudioDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
                                          int sample_rate_hz,
                                          int16_t* decoded,
                                          SpeechType* speech_type) {
-  RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
-      << "Unsupported sample rate " << sample_rate_hz;
-  if (sample_rate_hz != decoder_sample_rate_hz_) {
-    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
-    decoder_sample_rate_hz_ = sample_rate_hz;
+  if (sample_rate_hz_) {
+    RTC_CHECK_EQ(*sample_rate_hz_, sample_rate_hz);
+  } else {
+    RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
+        << "Unsupported sample rate " << sample_rate_hz;
+    sample_rate_hz_ = rtc::Optional<int>(sample_rate_hz);
+    RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, *sample_rate_hz_));
   }
   int16_t temp_type = 1;  // Default is speech.
   int ret =
@@ -95,6 +119,12 @@ int AudioDecoderIsacT<T>::ErrorCode() {
   return T::GetErrorCode(isac_state_);
 }
 
+template <typename T>
+int AudioDecoderIsacT<T>::SampleRateHz() const {
+  RTC_CHECK(sample_rate_hz_) << "Sample rate not set yet!";
+  return *sample_rate_hz_;
+}
+
 template <typename T>
 size_t AudioDecoderIsacT<T>::Channels() const {
   return 1;
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index f64e811afe..42abd0a0d6 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -87,6 +87,10 @@ bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
   return (fec == 1);
 }
 
+int AudioDecoderOpus::SampleRateHz() const {
+  return 48000;
+}
+
 size_t AudioDecoderOpus::Channels() const {
   return channels_;
 }
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index be48ca988e..c22204156b 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -27,6 +27,7 @@ class AudioDecoderOpus final : public AudioDecoder {
   int PacketDurationRedundant(const uint8_t* encoded,
                               size_t encoded_len) const override;
   bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
   size_t Channels() const override;
 
  protected:
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index 834c070073..dce5f4c516 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -15,13 +15,20 @@
 
 namespace webrtc {
 
-AudioDecoderPcm16B::AudioDecoderPcm16B(size_t num_channels)
-    : num_channels_(num_channels) {
+AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
+    : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
+  RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+             sample_rate_hz == 32000 || sample_rate_hz == 48000)
+      << "Unsupported sample rate " << sample_rate_hz;
   RTC_DCHECK_GE(num_channels, 1u);
 }
 
 void AudioDecoderPcm16B::Reset() {}
 
+int AudioDecoderPcm16B::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
 size_t AudioDecoderPcm16B::Channels() const {
   return num_channels_;
 }
@@ -31,9 +38,7 @@ int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
                                        int sample_rate_hz,
                                        int16_t* decoded,
                                        SpeechType* speech_type) {
-  RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
-             sample_rate_hz == 32000 || sample_rate_hz == 48000)
-      << "Unsupported sample rate " << sample_rate_hz;
+  RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
   size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
   *speech_type = ConvertSpeechType(1);
   return static_cast<int>(ret);
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
index 692cb94282..df94a6a6bd 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -18,9 +18,10 @@ namespace webrtc {
 
 class AudioDecoderPcm16B final : public AudioDecoder {
  public:
-  explicit AudioDecoderPcm16B(size_t num_channels);
+  AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels);
   void Reset() override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
   size_t Channels() const override;
 
  protected:
@@ -31,6 +32,7 @@ class AudioDecoderPcm16B final : public AudioDecoder {
                      SpeechType* speech_type) override;
 
  private:
+  const int sample_rate_hz_;
   const size_t num_channels_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoderPcm16B);
 };
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 1d0802e441..f83afccdf5 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -310,7 +310,7 @@ class AudioDecoderPcm16BTest : public AudioDecoderTest {
     codec_input_rate_hz_ = 16000;
     frame_size_ = 20 * codec_input_rate_hz_ / 1000;
     data_length_ = 10 * frame_size_;
-    decoder_ = new AudioDecoderPcm16B(1);
+    decoder_ = new AudioDecoderPcm16B(codec_input_rate_hz_, 1);
     assert(decoder_);
     AudioEncoderPcm16B::Config config;
     config.sample_rate_hz = codec_input_rate_hz_;
@@ -370,7 +370,7 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest {
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
     audio_encoder_.reset(new AudioEncoderIsac(config));
-    decoder_ = new AudioDecoderIsac();
+    decoder_ = new AudioDecoderIsac(codec_input_rate_hz_);
   }
 };
 
@@ -387,7 +387,7 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
     audio_encoder_.reset(new AudioEncoderIsac(config));
-    decoder_ = new AudioDecoderIsac();
+    decoder_ = new AudioDecoderIsac(codec_input_rate_hz_);
   }
 };
 
@@ -404,7 +404,7 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest {
     config.frame_size_ms =
         1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
     audio_encoder_.reset(new AudioEncoderIsacFix(config));
-    decoder_ = new AudioDecoderIsacFix();
+    decoder_ = new AudioDecoderIsacFix(codec_input_rate_hz_);
   }
 };