diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index b3d4e1d..c5bd2e7 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -39,6 +39,9 @@
     "../base:rtc_task_queue",
     "../call:call_interfaces",
     "../common_audio",
+    "../modules/audio_coding:audio_encoder_factory_interface",
+    "../modules/audio_coding:builtin_audio_encoder_factory",
+    "../modules/audio_coding:cng",
     "../modules/audio_device",
     "../modules/audio_processing",
     "../modules/bitrate_controller:bitrate_controller",
diff --git a/audio/DEPS b/audio/DEPS
index 1bace55..da9e8e1 100644
--- a/audio/DEPS
+++ b/audio/DEPS
@@ -15,3 +15,15 @@
   "+webrtc/system_wrappers",
   "+webrtc/voice_engine",
 ]
+
+specific_include_rules = {
+  "audio_send_stream.cc": [
+    "+webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h",
+  ],
+  # TODO(ossu): Remove this exception when builtin_audio_encoder_factory.h
+  # has moved to api/, or when the proper mocks have been made.
+  "audio_send_stream_unittest.cc": [
+    "+webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
+  ],
+}
+
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index a45c74f..28541ca 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -11,16 +11,20 @@
 #include "webrtc/audio/audio_send_stream.h"
 
 #include <string>
+#include <utility>
+#include <vector>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
+#include "webrtc/base/function_view.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
+#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
@@ -32,21 +36,22 @@
 
 namespace webrtc {
 
-namespace {
-
-constexpr char kOpusCodecName[] = "opus";
-
-bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
-  return (STR_CASE_CMP(codec.plname, ref_name) == 0);
-}
-}  // namespace
-
 namespace internal {
 // TODO(elad.alon): Subsequent CL will make these values experiment-dependent.
 constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
 constexpr size_t kPacketLossRateMinNumAckedPackets = 50;
 constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40;
 
+namespace {
+void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
+                 rtc::FunctionView<void(AudioEncoder*)> lambda) {
+  channel_proxy->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+    RTC_DCHECK(encoder_ptr);
+    lambda(encoder_ptr->get());
+  });
+}
+}  // namespace
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -56,52 +61,28 @@
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
     : worker_queue_(worker_queue),
-      config_(config),
+      config_(Config(nullptr)),
       audio_state_(audio_state),
+      event_log_(event_log),
       bitrate_allocator_(bitrate_allocator),
       transport_(transport),
       packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
                            kPacketLossRateMinNumAckedPackets,
                            kRecoverablePacketLossRateMinNumAckedPairs) {
-  LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
-  RTC_DCHECK_NE(config_.voe_channel_id, -1);
+  LOG(LS_INFO) << "AudioSendStream: " << config.ToString();
+  RTC_DCHECK_NE(config.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(transport);
   RTC_DCHECK(transport->send_side_cc());
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
-  channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
-  channel_proxy_->SetRtcEventLog(event_log);
+  channel_proxy_ = voe_impl->GetChannelProxy(config.voe_channel_id);
+  channel_proxy_->SetRtcEventLog(event_log_);
   channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
   channel_proxy_->SetRTCPStatus(true);
-  channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
-  channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
-  // TODO(solenberg): Config NACK history window (which is a packet count),
-  // using the actual packet size for the configured codec.
-  channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
-                                config_.rtp.nack.rtp_history_ms / 20);
-
-  channel_proxy_->RegisterExternalTransport(config.send_transport);
   transport_->send_side_cc()->RegisterPacketFeedbackObserver(this);
 
-  for (const auto& extension : config.rtp.extensions) {
-    if (extension.uri == RtpExtension::kAudioLevelUri) {
-      channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
-    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
-      channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
-      transport->send_side_cc()->EnablePeriodicAlrProbing(true);
-      bandwidth_observer_.reset(transport->send_side_cc()
-                                    ->GetBitrateController()
-                                    ->CreateRtcpBandwidthObserver());
-    } else {
-      RTC_NOTREACHED() << "Registering unsupported RTP extension.";
-    }
-  }
-  channel_proxy_->RegisterSenderCongestionControlObjects(
-      transport, bandwidth_observer_.get());
-  if (!SetupSendCodec()) {
-    LOG(LS_ERROR) << "Failed to set up send codec state.";
-  }
+  ConfigureStream(this, config, true);
 
   pacer_thread_checker_.DetachFromThread();
 }
@@ -116,17 +97,102 @@
   channel_proxy_->SetRtcpRttStats(nullptr);
 }
 
+void AudioSendStream::Reconfigure(
+    const webrtc::AudioSendStream::Config& new_config) {
+  ConfigureStream(this, new_config, false);
+}
+
+void AudioSendStream::ConfigureStream(
+    webrtc::internal::AudioSendStream* stream,
+    const webrtc::AudioSendStream::Config& new_config,
+    bool first_time) {
+  LOG(LS_INFO) << "AudioSendStream::Configuring: " << new_config.ToString();
+  const auto& channel_proxy = stream->channel_proxy_;
+  const auto& old_config = stream->config_;
+
+  if (first_time || old_config.rtp.ssrc != new_config.rtp.ssrc) {
+    channel_proxy->SetLocalSSRC(new_config.rtp.ssrc);
+  }
+  if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
+    channel_proxy->SetRTCP_CNAME(new_config.rtp.c_name);
+  }
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  if (first_time || old_config.rtp.nack.rtp_history_ms !=
+                        new_config.rtp.nack.rtp_history_ms) {
+    channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
+                                 new_config.rtp.nack.rtp_history_ms / 20);
+  }
+
+  if (first_time ||
+      new_config.send_transport != old_config.send_transport) {
+    if (old_config.send_transport) {
+      channel_proxy->DeRegisterExternalTransport();
+    }
+
+    channel_proxy->RegisterExternalTransport(new_config.send_transport);
+  }
+
+  // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
+  // reserved for padding and MUST NOT be used as a local identifier.
+  // So it should be safe to use 0 here to indicate "not configured".
+  struct ExtensionIds {
+    int audio_level = 0;
+    int transport_sequence_number = 0;
+  };
+
+  auto find_extension_ids = [](const std::vector<RtpExtension>& extensions) {
+    ExtensionIds ids;
+    for (const auto& extension : extensions) {
+      if (extension.uri == RtpExtension::kAudioLevelUri) {
+        ids.audio_level = extension.id;
+      } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
+        ids.transport_sequence_number = extension.id;
+      }
+    }
+    return ids;
+  };
+
+  const ExtensionIds old_ids = find_extension_ids(old_config.rtp.extensions);
+  const ExtensionIds new_ids = find_extension_ids(new_config.rtp.extensions);
+  // Audio level indication
+  if (first_time || new_ids.audio_level != old_ids.audio_level) {
+    channel_proxy->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
+                                                     new_ids.audio_level);
+  }
+  // Transport sequence number
+  if (first_time ||
+      new_ids.transport_sequence_number != old_ids.transport_sequence_number) {
+    if (old_ids.transport_sequence_number) {
+      channel_proxy->ResetSenderCongestionControlObjects();
+      stream->bandwidth_observer_.reset();
+    }
+
+    if (new_ids.transport_sequence_number != 0) {
+      channel_proxy->EnableSendTransportSequenceNumber(
+          new_ids.transport_sequence_number);
+      stream->transport_->send_side_cc()->EnablePeriodicAlrProbing(true);
+      stream->bandwidth_observer_.reset(stream->transport_->send_side_cc()
+                                            ->GetBitrateController()
+                                            ->CreateRtcpBandwidthObserver());
+    }
+
+    channel_proxy->RegisterSenderCongestionControlObjects(
+        stream->transport_, stream->bandwidth_observer_.get());
+  }
+
+  if (!ReconfigureSendCodec(stream, new_config)) {
+    LOG(LS_ERROR) << "Failed to set up send codec state.";
+  }
+
+  ReconfigureBitrateObserver(stream, new_config);
+  stream->config_ = new_config;
+}
+
 void AudioSendStream::Start() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
-    RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps);
-    rtc::Event thread_sync_event(false /* manual_reset */, false);
-    worker_queue_->PostTask([this, &thread_sync_event] {
-      bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps,
-                                      config_.max_bitrate_bps, 0, true);
-      thread_sync_event.Set();
-    });
-    thread_sync_event.Wait(rtc::Event::kForever);
+    ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps);
   }
 
   ScopedVoEInterface<VoEBase> base(voice_engine());
@@ -138,12 +204,7 @@
 
 void AudioSendStream::Stop() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  rtc::Event thread_sync_event(false /* manual_reset */, false);
-  worker_queue_->PostTask([this, &thread_sync_event] {
-    bitrate_allocator_->RemoveObserver(this);
-    thread_sync_event.Set();
-  });
-  thread_sync_event.Wait(rtc::Event::kForever);
+  RemoveBitrateObserver();
 
   ScopedVoEInterface<VoEBase> base(voice_engine());
   int error = base->StopSend(config_.voe_channel_id);
@@ -183,11 +244,10 @@
   //                  implementation.
   stats.aec_quality_min = -1;
 
-  webrtc::CodecInst codec_inst = {0};
-  if (channel_proxy_->GetSendCodec(&codec_inst)) {
-    RTC_DCHECK_NE(codec_inst.pltype, -1);
-    stats.codec_name = codec_inst.plname;
-    stats.codec_payload_type = rtc::Optional<int>(codec_inst.pltype);
+  if (config_.send_codec_spec) {
+    const auto& spec = *config_.send_codec_spec;
+    stats.codec_name = spec.format.name;
+    stats.codec_payload_type = rtc::Optional<int>(spec.payload_type);
 
     // Get data from the last remote RTCP report.
     for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
@@ -196,10 +256,10 @@
         stats.packets_lost = block.cumulative_num_packets_lost;
         stats.fraction_lost = Q8ToFloat(block.fraction_lost);
         stats.ext_seqnum = block.extended_highest_sequence_number;
-        // Convert samples to milliseconds.
-        if (codec_inst.plfreq / 1000 > 0) {
+        // Convert timestamps to milliseconds.
+        if (spec.format.clockrate_hz / 1000 > 0) {
           stats.jitter_ms =
-              block.interarrival_jitter / (codec_inst.plfreq / 1000);
+              block.interarrival_jitter / (spec.format.clockrate_hz / 1000);
         }
         break;
       }
@@ -324,116 +384,193 @@
 }
 
 // Apply current codec settings to a single voe::Channel used for sending.
-bool AudioSendStream::SetupSendCodec() {
-  // Disable VAD and FEC unless we know the other side wants them.
-  channel_proxy_->SetVADStatus(false);
-  channel_proxy_->SetCodecFECStatus(false);
+bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
+                                     const Config& new_config) {
+  RTC_DCHECK(new_config.send_codec_spec);
+  const auto& spec = *new_config.send_codec_spec;
+  std::unique_ptr<AudioEncoder> encoder =
+      new_config.encoder_factory->MakeAudioEncoder(spec.payload_type,
+                                                   spec.format);
 
-  // We disable audio network adaptor here. This will on one hand make sure that
-  // audio network adaptor is disabled by default, and on the other allow audio
-  // network adaptor to be reconfigured, since SetReceiverFrameLengthRange can
-  // be only called when audio network adaptor is disabled.
-  channel_proxy_->DisableAudioNetworkAdaptor();
-
-  const auto& send_codec_spec = config_.send_codec_spec;
-
-  // We set the codec first, since the below extra configuration is only applied
-  // to the "current" codec.
-
-  // If codec is already configured, we do not it again.
-  // TODO(minyue): check if this check is really needed, or can we move it into
-  // |codec->SetSendCodec|.
-  webrtc::CodecInst current_codec = {0};
-  if (!channel_proxy_->GetSendCodec(&current_codec) ||
-      (send_codec_spec.codec_inst != current_codec)) {
-    if (!channel_proxy_->SetSendCodec(send_codec_spec.codec_inst)) {
-      LOG(LS_WARNING) << "SetSendCodec() failed.";
-      return false;
-    }
+  if (!encoder) {
+    LOG(LS_ERROR) << "Unable to create encoder for " << spec.format;
+    return false;
+  }
+  // If a bitrate has been specified for the codec, use it over the
+  // codec's default.
+  if (spec.target_bitrate_bps) {
+    encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps);
   }
 
-  // Codec internal FEC. Treat any failure as fatal internal error.
-  if (send_codec_spec.enable_codec_fec) {
-    if (!channel_proxy_->SetCodecFECStatus(true)) {
-      LOG(LS_WARNING) << "SetCodecFECStatus() failed.";
-      return false;
-    }
-  }
-
-  // DTX and maxplaybackrate are only set if current codec is Opus.
-  if (IsCodec(send_codec_spec.codec_inst, kOpusCodecName)) {
-    if (!channel_proxy_->SetOpusDtx(send_codec_spec.enable_opus_dtx)) {
-      LOG(LS_WARNING) << "SetOpusDtx() failed.";
-      return false;
-    }
-
-    // If opus_max_playback_rate <= 0, the default maximum playback rate
-    // (48 kHz) will be used.
-    if (send_codec_spec.opus_max_playback_rate > 0) {
-      if (!channel_proxy_->SetOpusMaxPlaybackRate(
-              send_codec_spec.opus_max_playback_rate)) {
-        LOG(LS_WARNING) << "SetOpusMaxPlaybackRate() failed.";
-        return false;
-      }
-    }
-
-    if (config_.audio_network_adaptor_config) {
-      // Audio network adaptor is only allowed for Opus currently.
-      // |SetReceiverFrameLengthRange| needs to be called before
-      // |EnableAudioNetworkAdaptor|.
-      channel_proxy_->SetReceiverFrameLengthRange(send_codec_spec.min_ptime_ms,
-                                                  send_codec_spec.max_ptime_ms);
-      channel_proxy_->EnableAudioNetworkAdaptor(
-          *config_.audio_network_adaptor_config);
+  // Enable ANA if configured (currently only used by Opus).
+  if (new_config.audio_network_adaptor_config) {
+    if (encoder->EnableAudioNetworkAdaptor(
+            *new_config.audio_network_adaptor_config, stream->event_log_)) {
       LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
-                   << config_.rtp.ssrc;
+                   << new_config.rtp.ssrc;
+    } else {
+      RTC_NOTREACHED();
     }
   }
 
-  // Set the CN payloadtype and the VAD status.
-  if (send_codec_spec.cng_payload_type != -1) {
-    // The CN payload type for 8000 Hz clockrate is fixed at 13.
-    if (send_codec_spec.cng_plfreq != 8000) {
-      webrtc::PayloadFrequencies cn_freq;
-      switch (send_codec_spec.cng_plfreq) {
-        case 16000:
-          cn_freq = webrtc::kFreq16000Hz;
-          break;
-        case 32000:
-          cn_freq = webrtc::kFreq32000Hz;
-          break;
-        default:
-          RTC_NOTREACHED();
-          return false;
-      }
-      if (!channel_proxy_->SetSendCNPayloadType(
-          send_codec_spec.cng_payload_type, cn_freq)) {
-        LOG(LS_WARNING) << "SetSendCNPayloadType() failed.";
-        // TODO(ajm): This failure condition will be removed from VoE.
-        // Restore the return here when we update to a new enough webrtc.
-        //
-        // Not returning false because the SetSendCNPayloadType will fail if
-        // the channel is already sending.
-        // This can happen if the remote description is applied twice, for
-        // example in the case of ROAP on top of JSEP, where both side will
-        // send the offer.
-      }
-    }
-
-    // Only turn on VAD if we have a CN payload type that matches the
-    // clockrate for the codec we are going to use.
-    if (send_codec_spec.cng_plfreq == send_codec_spec.codec_inst.plfreq &&
-        send_codec_spec.codec_inst.channels == 1) {
-      // TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
-      // interaction between VAD and Opus FEC.
-      if (!channel_proxy_->SetVADStatus(true)) {
-        LOG(LS_WARNING) << "SetVADStatus() failed.";
-        return false;
-      }
-    }
+  // Wrap the encoder in a an AudioEncoderCNG, if VAD is enabled.
+  if (spec.cng_payload_type) {
+    AudioEncoderCng::Config cng_config;
+    cng_config.num_channels = encoder->NumChannels();
+    cng_config.payload_type = *spec.cng_payload_type;
+    cng_config.speech_encoder = std::move(encoder);
+    cng_config.vad_mode = Vad::kVadNormal;
+    encoder.reset(new AudioEncoderCng(std::move(cng_config)));
   }
+
+  stream->channel_proxy_->SetEncoder(new_config.send_codec_spec->payload_type,
+                                     std::move(encoder));
   return true;
 }
 
+bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream,
+                                           const Config& new_config) {
+  const auto& old_config = stream->config_;
+  if (new_config.send_codec_spec == old_config.send_codec_spec) {
+    return true;
+  }
+
+  // If we have no encoder, or the format or payload type's changed, create a
+  // new encoder.
+  if (!old_config.send_codec_spec ||
+      new_config.send_codec_spec->format !=
+          old_config.send_codec_spec->format ||
+      new_config.send_codec_spec->payload_type !=
+          old_config.send_codec_spec->payload_type) {
+    return SetupSendCodec(stream, new_config);
+  }
+
+  // Should never move a stream from fully configured to unconfigured.
+  RTC_CHECK(new_config.send_codec_spec);
+
+  const rtc::Optional<int>& new_target_bitrate_bps =
+      new_config.send_codec_spec->target_bitrate_bps;
+  // If a bitrate has been specified for the codec, use it over the
+  // codec's default.
+  if (new_target_bitrate_bps &&
+      new_target_bitrate_bps !=
+          old_config.send_codec_spec->target_bitrate_bps) {
+    CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
+      encoder->OnReceivedTargetAudioBitrate(*new_target_bitrate_bps);
+    });
+  }
+
+  ReconfigureANA(stream, new_config);
+  ReconfigureCNG(stream, new_config);
+
+  return true;
+}
+
+void AudioSendStream::ReconfigureANA(AudioSendStream* stream,
+                                     const Config& new_config) {
+  if (new_config.audio_network_adaptor_config ==
+      stream->config_.audio_network_adaptor_config) {
+    return;
+  }
+  if (new_config.audio_network_adaptor_config) {
+    CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
+      if (encoder->EnableAudioNetworkAdaptor(
+              *new_config.audio_network_adaptor_config, stream->event_log_)) {
+        LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
+                     << new_config.rtp.ssrc;
+      } else {
+        RTC_NOTREACHED();
+      }
+    });
+  } else {
+    CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
+      encoder->DisableAudioNetworkAdaptor();
+    });
+    LOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
+                 << new_config.rtp.ssrc;
+  }
+}
+
+void AudioSendStream::ReconfigureCNG(AudioSendStream* stream,
+                                     const Config& new_config) {
+  if (new_config.send_codec_spec->cng_payload_type ==
+      stream->config_.send_codec_spec->cng_payload_type) {
+    return;
+  }
+
+  // Wrap or unwrap the encoder in an AudioEncoderCNG.
+  stream->channel_proxy_->ModifyEncoder(
+      [&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+        std::unique_ptr<AudioEncoder> old_encoder(std::move(*encoder_ptr));
+        auto sub_encoders = old_encoder->ReclaimContainedEncoders();
+        if (!sub_encoders.empty()) {
+          // Replace enc with its sub encoder. We need to put the sub
+          // encoder in a temporary first, since otherwise the old value
+          // of enc would be destroyed before the new value got assigned,
+          // which would be bad since the new value is a part of the old
+          // value.
+          auto tmp = std::move(sub_encoders[0]);
+          old_encoder = std::move(tmp);
+        }
+        if (new_config.send_codec_spec->cng_payload_type) {
+          AudioEncoderCng::Config config;
+          config.speech_encoder = std::move(old_encoder);
+          config.num_channels = config.speech_encoder->NumChannels();
+          config.payload_type = *new_config.send_codec_spec->cng_payload_type;
+          config.vad_mode = Vad::kVadNormal;
+          encoder_ptr->reset(new AudioEncoderCng(std::move(config)));
+        } else {
+          *encoder_ptr = std::move(old_encoder);
+        }
+      });
+}
+
+void AudioSendStream::ReconfigureBitrateObserver(
+    AudioSendStream* stream,
+    const webrtc::AudioSendStream::Config& new_config) {
+  // Since the Config's default is for both of these to be -1, this test will
+  // allow us to configure the bitrate observer if the new config has bitrate
+  // limits set, but would only have us call RemoveBitrateObserver if we were
+  // previously configured with bitrate limits.
+  if (stream->config_.min_bitrate_bps == new_config.min_bitrate_bps &&
+      stream->config_.max_bitrate_bps == new_config.max_bitrate_bps) {
+    return;
+  }
+
+  if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1) {
+    stream->ConfigureBitrateObserver(new_config.min_bitrate_bps,
+                                     new_config.max_bitrate_bps);
+  } else {
+    stream->RemoveBitrateObserver();
+  }
+}
+
+void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
+                                               int max_bitrate_bps) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_GE(max_bitrate_bps, min_bitrate_bps);
+  rtc::Event thread_sync_event(false /* manual_reset */, false);
+  worker_queue_->PostTask([&] {
+    // We may get a callback immediately as the observer is registered, so make
+    // sure the bitrate limits in config_ are up-to-date.
+    config_.min_bitrate_bps = min_bitrate_bps;
+    config_.max_bitrate_bps = max_bitrate_bps;
+    bitrate_allocator_->AddObserver(this, min_bitrate_bps, max_bitrate_bps, 0,
+                                    true);
+    thread_sync_event.Set();
+  });
+  thread_sync_event.Wait(rtc::Event::kForever);
+}
+
+void AudioSendStream::RemoveBitrateObserver() {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  rtc::Event thread_sync_event(false /* manual_reset */, false);
+  worker_queue_->PostTask([this, &thread_sync_event] {
+    bitrate_allocator_->RemoveObserver(this);
+    thread_sync_event.Set();
+  });
+  thread_sync_event.Wait(rtc::Event::kForever);
+}
+
 }  // namespace internal
 }  // namespace webrtc
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 567799c..56e099f 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -48,6 +48,8 @@
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
+  void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
+
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
@@ -75,14 +77,29 @@
  private:
   VoiceEngine* voice_engine() const;
 
-  bool SetupSendCodec();
+  // These are all static to make it less likely that (the old) config_ is
+  // accessed unintentionally.
+  static void ConfigureStream(AudioSendStream* stream,
+                              const Config& new_config,
+                              bool first_time);
+  static bool SetupSendCodec(AudioSendStream* stream, const Config& new_config);
+  static bool ReconfigureSendCodec(AudioSendStream* stream,
+                                   const Config& new_config);
+  static void ReconfigureANA(AudioSendStream* stream, const Config& new_config);
+  static void ReconfigureCNG(AudioSendStream* stream, const Config& new_config);
+  static void ReconfigureBitrateObserver(AudioSendStream* stream,
+                                         const Config& new_config);
+
+  void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps);
+  void RemoveBitrateObserver();
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker pacer_thread_checker_;
   rtc::TaskQueue* worker_queue_;
-  const webrtc::AudioSendStream::Config config_;
+  webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
+  RtcEventLog* const event_log_;
 
   BitrateAllocator* const bitrate_allocator_;
   RtpTransportControllerSendInterface* const transport_;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 91201d6..3fbbb62 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "webrtc/audio/audio_send_stream.h"
@@ -17,6 +18,8 @@
 #include "webrtc/base/task_queue.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h"
@@ -35,6 +38,7 @@
 using testing::_;
 using testing::Eq;
 using testing::Ne;
+using testing::Invoke;
 using testing::Return;
 
 const int kChannelId = 1;
@@ -56,6 +60,14 @@
 const int kTelephoneEventCode = 45;
 const int kTelephoneEventDuration = 6789;
 const CodecInst kIsacCodec = {103, "isac", 16000, 320, 1, 32000};
+constexpr int kIsacPayloadType = 103;
+const SdpAudioFormat kIsacFormat = {"isac", 16000, 1};
+const SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
+const SdpAudioFormat kG722Format = {"g722", 8000, 1};
+const AudioCodecSpec kCodecSpecs[] = {
+    {kIsacFormat, {16000, 1, 32000, 10000, 32000}},
+    {kOpusFormat, {48000, 1, 32000, 6000, 510000}},
+    {kG722Format, {16000, 1, 64000}}};
 
 class MockLimitObserver : public BitrateAllocator::LimitObserver {
  public:
@@ -69,6 +81,47 @@
   MOCK_CONST_METHOD0(AudioLevelFullRange, int16_t());
 };
 
+std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
+    int payload_type,
+    const SdpAudioFormat& format) {
+  for (const auto& spec : kCodecSpecs) {
+    if (format == spec.format) {
+      std::unique_ptr<MockAudioEncoder> encoder(new MockAudioEncoder);
+      ON_CALL(*encoder.get(), SampleRateHz())
+          .WillByDefault(Return(spec.info.sample_rate_hz));
+      ON_CALL(*encoder.get(), NumChannels())
+          .WillByDefault(Return(spec.info.num_channels));
+      ON_CALL(*encoder.get(), RtpTimestampRateHz())
+          .WillByDefault(Return(spec.format.clockrate_hz));
+      return encoder;
+    }
+  }
+  return nullptr;
+}
+
+rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
+  rtc::scoped_refptr<MockAudioEncoderFactory> factory =
+      new rtc::RefCountedObject<MockAudioEncoderFactory>();
+  ON_CALL(*factory.get(), GetSupportedEncoders())
+      .WillByDefault(Return(std::vector<AudioCodecSpec>(
+          std::begin(kCodecSpecs), std::end(kCodecSpecs))));
+  ON_CALL(*factory.get(), QueryAudioEncoder(_))
+      .WillByDefault(Invoke([](const SdpAudioFormat& format) {
+        for (const auto& spec : kCodecSpecs) {
+          if (format == spec.format) {
+            return rtc::Optional<AudioCodecInfo>(spec.info);
+          }
+        }
+        return rtc::Optional<AudioCodecInfo>();
+      }));
+  ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
+      .WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
+                               std::unique_ptr<AudioEncoder>* return_value) {
+        *return_value = SetupAudioEncoderMock(payload_type, format);
+      }));
+  return factory;
+}
+
 struct ConfigHelper {
   class FakeRtpTransportController
       : public RtpTransportControllerSendInterface {
@@ -97,7 +150,7 @@
     SendSideCongestionController send_side_cc_;
   };
 
-  explicit ConfigHelper(bool audio_bwe_enabled)
+  ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
       : stream_config_(nullptr),
         fake_transport_(&event_log_),
         bitrate_allocator_(&limit_observer_),
@@ -124,8 +177,13 @@
           return channel_proxy_;
         }));
 
-    SetupMockForSetupSendCodec();
+    SetupMockForSetupSendCodec(expect_set_encoder_call);
 
+    // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
+    // calls from the default ctor behavior.
+    stream_config_.send_codec_spec =
+        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+            {kIsacPayloadType, kIsacFormat});
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
@@ -136,16 +194,18 @@
       stream_config_.rtp.extensions.push_back(
           RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                        kTransportSequenceNumberId));
-      stream_config_.send_codec_spec.transport_cc_enabled = true;
+      stream_config_.send_codec_spec->transport_cc_enabled = true;
     }
-    // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
-    // calls from the default ctor behavior.
-    stream_config_.send_codec_spec.codec_inst = kIsacCodec;
+    stream_config_.encoder_factory = SetupEncoderFactoryMock();
     stream_config_.min_bitrate_bps = 10000;
     stream_config_.max_bitrate_bps = 65000;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
+  MockAudioEncoderFactory& mock_encoder_factory() {
+    return *static_cast<MockAudioEncoderFactory*>(
+        stream_config_.encoder_factory.get());
+  }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   RtpTransportControllerSendInterface* transport() { return &fake_transport_; }
@@ -176,6 +236,8 @@
                                        &fake_transport_, Eq(nullptr)))
           .Times(1);
     }
+    EXPECT_CALL(*channel_proxy_, SetBitrate(_, _))
+        .Times(1);
     EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects())
         .Times(1);
     EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr)).Times(1);
@@ -188,17 +250,13 @@
         .Times(1);  // Destructor resets the rtt stats.
   }
 
-  void SetupMockForSetupSendCodec() {
-    EXPECT_CALL(*channel_proxy_, SetVADStatus(false))
-        .WillOnce(Return(true));
-    EXPECT_CALL(*channel_proxy_, SetCodecFECStatus(false))
-        .WillOnce(Return(true));
-    EXPECT_CALL(*channel_proxy_, DisableAudioNetworkAdaptor());
-    // Let |GetSendCodec| return false for the first time to indicate that no
-    // send codec has been set.
-    EXPECT_CALL(*channel_proxy_, GetSendCodec(_)).WillOnce(Return(false));
-    EXPECT_CALL(*channel_proxy_, SetSendCodec(_)).WillOnce(Return(true));
+  void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
+    if (expect_set_encoder_call) {
+      EXPECT_CALL(*channel_proxy_, SetEncoderForMock(_, _))
+          .WillOnce(Return(true));
+    }
   }
+
   RtcpRttStats* rtcp_rtt_stats() { return &rtcp_rtt_stats_; }
 
   void SetupMockForSendTelephoneEvent() {
@@ -230,8 +288,6 @@
         .WillRepeatedly(Return(kCallStats));
     EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
         .WillRepeatedly(Return(report_blocks));
-    EXPECT_CALL(*channel_proxy_, GetSendCodec(_))
-        .WillRepeatedly(DoAll(SetArgPointee<0>(kIsacCodec), Return(true)));
     EXPECT_CALL(voice_engine_, transmit_mixer())
         .WillRepeatedly(Return(&transmit_mixer_));
     EXPECT_CALL(voice_engine_, audio_processing())
@@ -281,16 +337,13 @@
   config.voe_channel_id = kChannelId;
   config.min_bitrate_bps = 12000;
   config.max_bitrate_bps = 34000;
-  config.send_codec_spec.nack_enabled = true;
-  config.send_codec_spec.transport_cc_enabled = false;
-  config.send_codec_spec.enable_codec_fec = true;
-  config.send_codec_spec.enable_opus_dtx = false;
-  config.send_codec_spec.opus_max_playback_rate = 32000;
-  config.send_codec_spec.cng_payload_type = 42;
-  config.send_codec_spec.cng_plfreq = 56;
-  config.send_codec_spec.min_ptime_ms = 20;
-  config.send_codec_spec.max_ptime_ms = 60;
-  config.send_codec_spec.codec_inst = kIsacCodec;
+  config.send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+          {kIsacPayloadType, kIsacFormat});
+  config.send_codec_spec->nack_enabled = true;
+  config.send_codec_spec->transport_cc_enabled = false;
+  config.send_codec_spec->cng_payload_type = rtc::Optional<int>(42);
+  config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory();
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
   EXPECT_EQ(
@@ -299,15 +352,14 @@
       "{rtp_history_ms: 0}, c_name: foo_name}, send_transport: null, "
       "voe_channel_id: 1, min_bitrate_bps: 12000, max_bitrate_bps: 34000, "
       "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
-      "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
-      "32000, cng_payload_type: 42, cng_plfreq: 56, min_ptime: 20, max_ptime: "
-      "60, codec_inst: {pltype: 103, plname: \"isac\", plfreq: 16000, pacsize: "
-      "320, channels: 1, rate: 32000}}}",
+      "cng_payload_type: 42, payload_type: 103, "
+      "format: {name: isac, clockrate_hz: 16000, num_channels: 1, "
+      "parameters: {}}}}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -315,7 +367,7 @@
 }
 
 TEST(AudioSendStreamTest, SendTelephoneEvent) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -327,7 +379,7 @@
 }
 
 TEST(AudioSendStreamTest, SetMuted) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -337,7 +389,7 @@
 }
 
 TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
-  ConfigHelper helper(true);
+  ConfigHelper helper(true, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -345,7 +397,7 @@
 }
 
 TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -353,7 +405,7 @@
 }
 
 TEST(AudioSendStreamTest, GetStats) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -384,7 +436,7 @@
 }
 
 TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -402,43 +454,23 @@
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
 
-TEST(AudioSendStreamTest, SendCodecAppliesConfigParams) {
-  ConfigHelper helper(false);
+TEST(AudioSendStreamTest, SendCodecAppliesNetworkAdaptor) {
+  ConfigHelper helper(false, true);
   auto stream_config = helper.config();
-  const CodecInst kOpusCodec = {111, "opus", 48000, 960, 2, 64000};
-  stream_config.send_codec_spec.codec_inst = kOpusCodec;
-  stream_config.send_codec_spec.enable_codec_fec = true;
-  stream_config.send_codec_spec.enable_opus_dtx = true;
-  stream_config.send_codec_spec.opus_max_playback_rate = 12345;
-  stream_config.send_codec_spec.cng_plfreq = 16000;
-  stream_config.send_codec_spec.cng_payload_type = 105;
-  stream_config.send_codec_spec.min_ptime_ms = 10;
-  stream_config.send_codec_spec.max_ptime_ms = 60;
+  stream_config.send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>({0, kOpusFormat});
   stream_config.audio_network_adaptor_config =
       rtc::Optional<std::string>("abced");
-  EXPECT_CALL(*helper.channel_proxy(), SetCodecFECStatus(true))
-      .WillOnce(Return(true));
-  EXPECT_CALL(
-      *helper.channel_proxy(),
-      SetOpusDtx(stream_config.send_codec_spec.enable_opus_dtx))
-      .WillOnce(Return(true));
-  EXPECT_CALL(
-      *helper.channel_proxy(),
-      SetOpusMaxPlaybackRate(
-          stream_config.send_codec_spec.opus_max_playback_rate))
-      .WillOnce(Return(true));
-  EXPECT_CALL(*helper.channel_proxy(),
-              SetSendCNPayloadType(
-                  stream_config.send_codec_spec.cng_payload_type,
-                  webrtc::kFreq16000Hz))
-      .WillOnce(Return(true));
-  EXPECT_CALL(
-      *helper.channel_proxy(),
-      SetReceiverFrameLengthRange(stream_config.send_codec_spec.min_ptime_ms,
-                                  stream_config.send_codec_spec.max_ptime_ms));
-  EXPECT_CALL(
-      *helper.channel_proxy(),
-      EnableAudioNetworkAdaptor(*stream_config.audio_network_adaptor_config));
+
+  EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _))
+      .WillOnce(Invoke([](int payload_type, const SdpAudioFormat& format,
+                          std::unique_ptr<AudioEncoder>* return_value) {
+        auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
+        EXPECT_CALL(*mock_encoder.get(), EnableAudioNetworkAdaptor(_, _))
+            .WillOnce(Return(true));
+        *return_value = std::move(mock_encoder);
+      }));
+
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -446,24 +478,37 @@
 }
 
 // VAD is applied when codec is mono and the CNG frequency matches the codec
-// sample rate.
+// clock rate.
 TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, false);
   auto stream_config = helper.config();
-  const CodecInst kG722Codec = {9, "g722", 8000, 160, 1, 16000};
-  stream_config.send_codec_spec.codec_inst = kG722Codec;
-  stream_config.send_codec_spec.cng_plfreq = 8000;
-  stream_config.send_codec_spec.cng_payload_type = 105;
-  EXPECT_CALL(*helper.channel_proxy(), SetVADStatus(true))
-      .WillOnce(Return(true));
+  stream_config.send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>({9, kG722Format});
+  stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105);
+  using ::testing::Invoke;
+  std::unique_ptr<AudioEncoder> stolen_encoder;
+  EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
+      .WillOnce(
+          Invoke([&stolen_encoder](int payload_type,
+                                   std::unique_ptr<AudioEncoder>* encoder) {
+            stolen_encoder = std::move(*encoder);
+            return true;
+          }));
+
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
       helper.rtcp_rtt_stats());
+
+  // We cannot truly determine if the encoder created is an AudioEncoderCng.  It
+  // is the only reasonable implementation that will return something from
+  // ReclaimContainedEncoders, though.
+  ASSERT_TRUE(stolen_encoder);
+  EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty());
 }
 
 TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -475,7 +520,7 @@
 }
 
 TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
-  ConfigHelper helper(false);
+  ConfigHelper helper(false, true);
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
@@ -484,5 +529,27 @@
   send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
 }
 
+// Test that AudioSendStream doesn't recreate the encoder unnecessarily.
+TEST(AudioSendStreamTest, DontRecreateEncoder) {
+  ConfigHelper helper(false, false);
+  // WillOnce is (currently) the default used by ConfigHelper if asked to set an
+  // expectation for SetEncoder. Since this behavior is essential for this test
+  // to be correct, it's instead set-up manually here. Otherwise a simple change
+  // to ConfigHelper (say to WillRepeatedly) would silently make this test
+  // useless.
+  EXPECT_CALL(*helper.channel_proxy(), SetEncoderForMock(_, _))
+      .WillOnce(Return(true));
+
+  auto stream_config = helper.config();
+  stream_config.send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>({9, kG722Format});
+  stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105);
+  internal::AudioSendStream send_stream(
+      stream_config, helper.audio_state(), helper.worker_queue(),
+      helper.transport(), helper.bitrate_allocator(), helper.event_log(),
+      helper.rtcp_rtt_stats());
+  send_stream.Reconfigure(stream_config);
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/audio/test/low_bandwidth_audio_test.cc b/audio/test/low_bandwidth_audio_test.cc
index 98cfa70..f9e6e2b 100644
--- a/audio/test/low_bandwidth_audio_test.cc
+++ b/audio/test/low_bandwidth_audio_test.cc
@@ -97,8 +97,12 @@
 void AudioQualityTest::ModifyAudioConfigs(
   AudioSendStream::Config* send_config,
   std::vector<AudioReceiveStream::Config>* receive_configs) {
-  send_config->send_codec_spec.codec_inst = webrtc::CodecInst{
-      test::CallTest::kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
+  // Large bitrate by default.
+  const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
+                                              {{"stereo", "1"}});
+  send_config->send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+          {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
 }
 
 void AudioQualityTest::PerformTest() {
@@ -130,14 +134,15 @@
 class Mobile2GNetworkTest : public AudioQualityTest {
   void ModifyAudioConfigs(AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override {
-    send_config->send_codec_spec.codec_inst = CodecInst{
-        test::CallTest::kAudioSendPayloadType,  // pltype
-        "OPUS",                                 // plname
-        48000,                                  // plfreq
-        2880,                                   // pacsize
-        1,                                      // channels
-        6000                                    // rate bits/sec
-    };
+    send_config->send_codec_spec =
+        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+            {test::CallTest::kAudioSendPayloadType,
+             {"OPUS",
+              48000,
+              2,
+              {{"maxaveragebitrate", "6000"},
+               {"ptime", "60"},
+               {"stereo", "1"}}}});
   }
 
   FakeNetworkPipe::Config GetNetworkPipeConfig() override {
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 47847cc..4111881 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -29,6 +29,7 @@
     "../api/audio_codecs:audio_codecs_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
+    "../modules/audio_coding:audio_encoder_factory_interface",
     "../modules/audio_coding:audio_encoder_interface",
   ]
 }
@@ -131,6 +132,7 @@
       "../base:rtc_base_approved",
       "../logging:rtc_event_log_api",
       "../modules/audio_coding",
+      "../modules/audio_coding:builtin_audio_encoder_factory",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/rtp_rtcp",
       "../system_wrappers",
diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc
index 6091462..0bc555b 100644
--- a/call/audio_send_stream.cc
+++ b/call/audio_send_stream.cc
@@ -12,21 +12,6 @@
 
 #include <string>
 
-namespace {
-
-std::string ToString(const webrtc::CodecInst& codec_inst) {
-  std::stringstream ss;
-  ss << "{pltype: " << codec_inst.pltype;
-  ss << ", plname: \"" << codec_inst.plname << "\"";
-  ss << ", plfreq: " << codec_inst.plfreq;
-  ss << ", pacsize: " << codec_inst.pacsize;
-  ss << ", channels: " << codec_inst.channels;
-  ss << ", rate: " << codec_inst.rate;
-  ss << '}';
-  return ss.str();
-}
-}  // namespace
-
 namespace webrtc {
 
 AudioSendStream::Stats::Stats() = default;
@@ -44,7 +29,8 @@
   ss << ", voe_channel_id: " << voe_channel_id;
   ss << ", min_bitrate_bps: " << min_bitrate_bps;
   ss << ", max_bitrate_bps: " << max_bitrate_bps;
-  ss << ", send_codec_spec: " << send_codec_spec.ToString();
+  ss << ", send_codec_spec: "
+     << (send_codec_spec ? send_codec_spec->ToString() : "<unset>");
   ss << '}';
   return ss.str();
 }
@@ -70,24 +56,20 @@
   return ss.str();
 }
 
-AudioSendStream::Config::SendCodecSpec::SendCodecSpec() {
-  webrtc::CodecInst empty_inst = {0};
-  codec_inst = empty_inst;
-  codec_inst.pltype = -1;
-}
+AudioSendStream::Config::SendCodecSpec::SendCodecSpec(
+    int payload_type,
+    const SdpAudioFormat& format)
+    : payload_type(payload_type), format(format) {}
+AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;
 
 std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
   std::stringstream ss;
   ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
   ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
-  ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false");
-  ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false");
-  ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
-  ss << ", cng_payload_type: " << cng_payload_type;
-  ss << ", cng_plfreq: " << cng_plfreq;
-  ss << ", min_ptime: " << min_ptime_ms;
-  ss << ", max_ptime: " << max_ptime_ms;
-  ss << ", codec_inst: " << ::ToString(codec_inst);
+  ss << ", cng_payload_type: "
+     << (cng_payload_type ? std::to_string(*cng_payload_type) : "<unset>");
+  ss << ", payload_type: " << payload_type;
+  ss << ", format: " << format;
   ss << '}';
   return ss.str();
 }
@@ -96,12 +78,9 @@
     const AudioSendStream::Config::SendCodecSpec& rhs) const {
   if (nack_enabled == rhs.nack_enabled &&
       transport_cc_enabled == rhs.transport_cc_enabled &&
-      enable_codec_fec == rhs.enable_codec_fec &&
-      enable_opus_dtx == rhs.enable_opus_dtx &&
-      opus_max_playback_rate == rhs.opus_max_playback_rate &&
       cng_payload_type == rhs.cng_payload_type &&
-      cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
-      min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
+      payload_type == rhs.payload_type && format == rhs.format &&
+      target_bitrate_bps == rhs.target_bitrate_bps) {
     return true;
   }
   return false;
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 4291430..5363d13 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -15,10 +15,11 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/config.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -102,7 +103,8 @@
     rtc::Optional<std::string> audio_network_adaptor_config;
 
     struct SendCodecSpec {
-      SendCodecSpec();
+      SendCodecSpec(int payload_type, const SdpAudioFormat& format);
+      ~SendCodecSpec();
       std::string ToString() const;
 
       bool operator==(const SendCodecSpec& rhs) const;
@@ -110,19 +112,22 @@
         return !(*this == rhs);
       }
 
+      int payload_type;
+      SdpAudioFormat format;
       bool nack_enabled = false;
       bool transport_cc_enabled = false;
-      bool enable_codec_fec = false;
-      bool enable_opus_dtx = false;
-      int opus_max_playback_rate = 0;
-      int cng_payload_type = -1;
-      int cng_plfreq = -1;
-      int max_ptime_ms = -1;
-      int min_ptime_ms = -1;
-      webrtc::CodecInst codec_inst;
-    } send_codec_spec;
+      rtc::Optional<int> cng_payload_type;
+      // If unset, use the encoder's default target bitrate.
+      rtc::Optional<int> target_bitrate_bps;
+    };
+
+    rtc::Optional<SendCodecSpec> send_codec_spec;
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
   };
 
+  // Reconfigure the stream according to the Configuration.
+  virtual void Reconfigure(const Config& config) = 0;
+
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 7ade528..6d89574 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -19,6 +19,7 @@
 #include "webrtc/call/call.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
@@ -232,8 +233,10 @@
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
   audio_send_config.rtp.ssrc = kAudioSendSsrc;
-  audio_send_config.send_codec_spec.codec_inst =
-      CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
+  audio_send_config.send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+          {kAudioSendPayloadType, {"ISAC", 16000, 1}});
+  audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
   AudioSendStream* audio_send_stream =
       sender_call_->CreateAudioSendStream(audio_send_config);
 
diff --git a/media/BUILD.gn b/media/BUILD.gn
index b6529bf..a86f86a 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -105,6 +105,8 @@
     "../base:rtc_base_approved",
     "../call:call_interfaces",
     "../common_video:common_video",
+    "../modules/audio_coding:audio_encoder_factory_interface",
+    "../modules/audio_coding:builtin_audio_encoder_factory",
     "../p2p",
   ]
 
diff --git a/media/engine/fakewebrtccall.cc b/media/engine/fakewebrtccall.cc
index b539665..d323ef6 100644
--- a/media/engine/fakewebrtccall.cc
+++ b/media/engine/fakewebrtccall.cc
@@ -26,6 +26,11 @@
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
+void FakeAudioSendStream::Reconfigure(
+    const webrtc::AudioSendStream::Config& config) {
+  config_ = config;
+}
+
 const webrtc::AudioSendStream::Config&
     FakeAudioSendStream::GetConfig() const {
   return config_;
diff --git a/media/engine/fakewebrtccall.h b/media/engine/fakewebrtccall.h
index 1ddf95a..77cae6b 100644
--- a/media/engine/fakewebrtccall.h
+++ b/media/engine/fakewebrtccall.h
@@ -55,6 +55,8 @@
 
  private:
   // webrtc::AudioSendStream implementation.
+  void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
+
   void Start() override { sending_ = true; }
   void Stop() override { sending_ = false; }
 
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 99ada08..379001e 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -37,7 +37,7 @@
 #include "webrtc/media/engine/payload_type_mapper.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
@@ -66,27 +66,9 @@
 #error "Set WEBRTC_INTELLIGIBILITY_ENHANCER to either 0 or 1"
 #endif
 
-// Codec parameters for Opus.
-// draft-spittka-payload-rtp-opus-03
-
-// Recommended bitrates:
-// 8-12 kb/s for NB speech,
-// 16-20 kb/s for WB speech,
-// 28-40 kb/s for FB speech,
-// 48-64 kb/s for FB mono music, and
-// 64-128 kb/s for FB stereo music.
-// The current implementation applies the following values to mono signals,
-// and multiplies them by 2 for stereo.
-const int kOpusBitrateNbBps = 12000;
-const int kOpusBitrateWbBps = 20000;
-const int kOpusBitrateFbBps = 32000;
-
-// Opus bitrate should be in the range between 6000 and 510000.
+// For SendSideBwe, Opus bitrate should be in the range between 6000 and 32000.
 const int kOpusMinBitrateBps = 6000;
-const int kOpusMaxBitrateBps = 510000;
-
-// iSAC bitrate should be <= 56000.
-const int kIsacMaxBitrateBps = 56000;
+const int kOpusBitrateFbBps = 32000;
 
 // Default audio dscp value.
 // See http://tools.ietf.org/html/rfc2474 for details.
@@ -125,8 +107,15 @@
 // Dumps an AudioCodec in RFC 2327-ish format.
 std::string ToString(const AudioCodec& codec) {
   std::stringstream ss;
-  ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
-     << " (" << codec.id << ")";
+  ss << codec.name << "/" << codec.clockrate << "/" << codec.channels;
+  if (!codec.params.empty()) {
+    ss << " {";
+    for (const auto& param : codec.params) {
+      ss << " " << param.first << "=" << param.second;
+    }
+    ss << " }";
+  }
+  ss << " (" << codec.id << ")";
   return ss.str();
 }
 
@@ -134,10 +123,6 @@
   return (_stricmp(codec.name.c_str(), ref_name) == 0);
 }
 
-bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
-  return (_stricmp(codec.plname, ref_name) == 0);
-}
-
 bool FindCodec(const std::vector<AudioCodec>& codecs,
                const AudioCodec& codec,
                AudioCodec* found_codec) {
@@ -165,12 +150,6 @@
   return it == payload_types.end();
 }
 
-// Return true if codec.params[feature] == "1", false otherwise.
-bool IsCodecFeatureEnabled(const AudioCodec& codec, const char* feature) {
-  int value;
-  return codec.GetParam(feature, &value) && value == 1;
-}
-
 rtc::Optional<std::string> GetAudioNetworkAdaptorConfig(
     const AudioOptions& options) {
   if (options.audio_network_adaptor && *options.audio_network_adaptor &&
@@ -182,85 +161,6 @@
   return rtc::Optional<std::string>();
 }
 
-// Returns integer parameter params[feature] if it is defined. Returns
-// |default_value| otherwise.
-int GetCodecFeatureInt(const AudioCodec& codec,
-                       const char* feature,
-                       int default_value) {
-  int value = 0;
-  if (codec.GetParam(feature, &value)) {
-    return value;
-  }
-  return default_value;
-}
-
-// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
-// otherwise. If the value (either from params or codec.bitrate) <=0, use the
-// default configuration. If the value is beyond feasible bit rate of Opus,
-// clamp it. Returns the Opus bit rate for operation.
-int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
-  int bitrate = 0;
-  bool use_param = true;
-  if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
-    bitrate = codec.bitrate;
-    use_param = false;
-  }
-  if (bitrate <= 0) {
-    if (max_playback_rate <= 8000) {
-      bitrate = kOpusBitrateNbBps;
-    } else if (max_playback_rate <= 16000) {
-      bitrate = kOpusBitrateWbBps;
-    } else {
-      bitrate = kOpusBitrateFbBps;
-    }
-
-    if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
-      bitrate *= 2;
-    }
-  } else if (bitrate < kOpusMinBitrateBps || bitrate > kOpusMaxBitrateBps) {
-    bitrate = (bitrate < kOpusMinBitrateBps) ? kOpusMinBitrateBps
-                                             : kOpusMaxBitrateBps;
-    std::string rate_source =
-        use_param ? "Codec parameter \"maxaveragebitrate\"" :
-            "Supplied Opus bitrate";
-    LOG(LS_WARNING) << rate_source
-                    << " is invalid and is replaced by: "
-                    << bitrate;
-  }
-  return bitrate;
-}
-
-void GetOpusConfig(const AudioCodec& codec,
-                   webrtc::CodecInst* voe_codec,
-                   bool* enable_codec_fec,
-                   int* max_playback_rate,
-                   bool* enable_codec_dtx,
-                   int* min_ptime_ms,
-                   int* max_ptime_ms) {
-  *enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
-  *enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
-  *max_playback_rate = GetCodecFeatureInt(codec, kCodecParamMaxPlaybackRate,
-                                          kOpusDefaultMaxPlaybackRate);
-  *max_ptime_ms =
-      GetCodecFeatureInt(codec, kCodecParamMaxPTime, kOpusDefaultMaxPTime);
-  *min_ptime_ms =
-      GetCodecFeatureInt(codec, kCodecParamMinPTime, kOpusDefaultMinPTime);
-  if (*max_ptime_ms < *min_ptime_ms) {
-    // If min ptime or max ptime defined by codec parameter is wrong, we use
-    // the default values.
-    *max_ptime_ms = kOpusDefaultMaxPTime;
-    *min_ptime_ms = kOpusDefaultMinPTime;
-  }
-
-  // If OPUS, change what we send according to the "stereo" codec
-  // parameter, and not the "channels" parameter.  We set
-  // voe_codec.channels to 2 if "stereo=1" and 1 otherwise.  If
-  // the bitrate is not specified, i.e. is <= zero, we set it to the
-  // appropriate default value for mono or stereo Opus.
-  voe_codec->channels = IsCodecFeatureEnabled(codec, kCodecParamStereo) ? 2 : 1;
-  voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
-}
-
 webrtc::AudioState::Config MakeAudioStateConfig(
     VoEWrapper* voe_wrapper,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
@@ -274,283 +174,41 @@
   return config;
 }
 
-class WebRtcVoiceCodecs final {
- public:
-  // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
-  // list and add a test which verifies VoE supports the listed codecs.
-  static std::vector<AudioCodec> SupportedSendCodecs() {
-    std::vector<AudioCodec> result;
-    // Iterate first over our preferred codecs list, so that the results are
-    // added in order of preference.
-    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
-      const CodecPref* pref = &kCodecPrefs[i];
-      for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
-        // Change the sample rate of G722 to 8000 to match SDP.
-        MaybeFixupG722(&voe_codec, 8000);
-        // Skip uncompressed formats.
-        if (IsCodec(voe_codec, kL16CodecName)) {
-          continue;
-        }
-
-        if (!IsCodec(voe_codec, pref->name) ||
-            pref->clockrate != voe_codec.plfreq ||
-            pref->channels != voe_codec.channels) {
-          // Not a match.
-          continue;
-        }
-
-        AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
-                         voe_codec.rate, voe_codec.channels);
-        LOG(LS_INFO) << "Adding supported codec: " << ToString(codec);
-        if (IsCodec(codec, kIsacCodecName)) {
-          // Indicate auto-bitrate in signaling.
-          codec.bitrate = 0;
-        }
-        if (IsCodec(codec, kOpusCodecName)) {
-          // Only add fmtp parameters that differ from the spec.
-          if (kPreferredMinPTime != kOpusDefaultMinPTime) {
-            codec.params[kCodecParamMinPTime] =
-                rtc::ToString(kPreferredMinPTime);
-          }
-          if (kPreferredMaxPTime != kOpusDefaultMaxPTime) {
-            codec.params[kCodecParamMaxPTime] =
-                rtc::ToString(kPreferredMaxPTime);
-          }
-          codec.SetParam(kCodecParamUseInbandFec, 1);
-          codec.AddFeedbackParam(
-              FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
-
-          // TODO(hellner): Add ptime, sprop-stereo, and stereo
-          // when they can be set to values other than the default.
-        }
-        result.push_back(codec);
-      }
-    }
-    return result;
-  }
-
-  static bool ToCodecInst(const AudioCodec& in,
-                          webrtc::CodecInst* out) {
-    for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
-      // Change the sample rate of G722 to 8000 to match SDP.
-      MaybeFixupG722(&voe_codec, 8000);
-      AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
-                       voe_codec.rate, voe_codec.channels);
-      bool multi_rate = IsCodecMultiRate(voe_codec);
-      // Allow arbitrary rates for ISAC to be specified.
-      if (multi_rate) {
-        // Set codec.bitrate to 0 so the check for codec.Matches() passes.
-        codec.bitrate = 0;
-      }
-      if (codec.Matches(in)) {
-        if (out) {
-          // Fixup the payload type.
-          voe_codec.pltype = in.id;
-
-          // Set bitrate if specified.
-          if (multi_rate && in.bitrate != 0) {
-            voe_codec.rate = in.bitrate;
-          }
-
-          // Reset G722 sample rate to 16000 to match WebRTC.
-          MaybeFixupG722(&voe_codec, 16000);
-
-          *out = voe_codec;
-        }
-        return true;
-      }
-    }
-    return false;
-  }
-
-  static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
-    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
-      if (IsCodec(codec, kCodecPrefs[i].name) &&
-          kCodecPrefs[i].clockrate == codec.plfreq) {
-        return kCodecPrefs[i].is_multi_rate;
-      }
-    }
-    return false;
-  }
-
-  static int MaxBitrateBps(const webrtc::CodecInst& codec) {
-    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
-      if (IsCodec(codec, kCodecPrefs[i].name) &&
-          kCodecPrefs[i].clockrate == codec.plfreq) {
-        return kCodecPrefs[i].max_bitrate_bps;
-      }
-    }
-    return 0;
-  }
-
-  static rtc::ArrayView<const int> GetPacketSizesMs(
-      const webrtc::CodecInst& codec) {
-    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
-      if (IsCodec(codec, kCodecPrefs[i].name)) {
-        size_t num_packet_sizes = kMaxNumPacketSize;
-        for (int index = 0; index < kMaxNumPacketSize; index++) {
-          if (kCodecPrefs[i].packet_sizes_ms[index] == 0) {
-            num_packet_sizes = index;
-            break;
-          }
-        }
-        return rtc::ArrayView<const int>(kCodecPrefs[i].packet_sizes_ms,
-                                         num_packet_sizes);
-      }
-    }
-    return rtc::ArrayView<const int>();
-  }
-
-  // If the AudioCodec param kCodecParamPTime is set, then we will set it to
-  // codec pacsize if it's valid, or we will pick the next smallest value we
-  // support.
-  // TODO(Brave): Query supported packet sizes from ACM when the API is ready.
-  static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
-    for (const CodecPref& codec_pref : kCodecPrefs) {
-      if ((IsCodec(*codec, codec_pref.name) &&
-          codec_pref.clockrate == codec->plfreq) ||
-          IsCodec(*codec, kG722CodecName)) {
-        int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
-        if (packet_size_ms) {
-          // Convert unit from milli-seconds to samples.
-          codec->pacsize = (codec->plfreq / 1000) * packet_size_ms;
-          return true;
-        }
-      }
-    }
-    return false;
-  }
-
-  static const AudioCodec* GetPreferredCodec(
-      const std::vector<AudioCodec>& codecs,
-      webrtc::CodecInst* out) {
-    RTC_DCHECK(out);
-    // Select the preferred send codec (the first non-telephone-event/CN codec).
-    for (const AudioCodec& codec : codecs) {
-      if (IsCodec(codec, kDtmfCodecName) || IsCodec(codec, kCnCodecName)) {
-        // Skip telephone-event/CN codecs - they will be handled later.
-        continue;
-      }
-
-      // We'll use the first codec in the list to actually send audio data.
-      // Be sure to use the payload type requested by the remote side.
-      // Ignore codecs we don't know about. The negotiation step should prevent
-      // this, but double-check to be sure.
-      if (!ToCodecInst(codec, out)) {
-        LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
-        continue;
-      }
-      return &codec;
-    }
-    return nullptr;
-  }
-
- private:
-  static const int kMaxNumPacketSize = 6;
-  struct CodecPref {
-    const char* name;
-    int clockrate;
-    size_t channels;
-    int payload_type;
-    bool is_multi_rate;
-    int packet_sizes_ms[kMaxNumPacketSize];
-    int max_bitrate_bps;
-  };
-  // Note: keep the supported packet sizes in ascending order.
-  static const CodecPref kCodecPrefs[14];
-
-  static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
-    int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
-    for (int packet_size_ms : codec_pref.packet_sizes_ms) {
-      if (packet_size_ms && packet_size_ms <= ptime_ms) {
-        selected_packet_size_ms = packet_size_ms;
-      }
-    }
-    return selected_packet_size_ms;
-  }
-
-  // Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
-  // which says that G722 should be advertised as 8 kHz although it is a 16 kHz
-  // codec.
-  static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
-    if (IsCodec(*voe_codec, kG722CodecName)) {
-      // If the DCHECK triggers, the codec definition in WebRTC VoiceEngine
-      // has changed, and this special case is no longer needed.
-      RTC_DCHECK(voe_codec->plfreq != new_plfreq);
-      voe_codec->plfreq = new_plfreq;
-    }
-  }
-};
-
-const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[14] = {
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-    {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60, 120},
-     kOpusMaxBitrateBps},
-#else
-    {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps},
-#endif
-    {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps},
-    {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrateBps},
-    // G722 should be advertised as 8000 Hz because of the RFC "bug".
-    {kG722CodecName, 8000, 1, 9, false, {10, 20, 30, 40, 50, 60}},
-    {kIlbcCodecName, 8000, 1, 102, false, {20, 30, 40, 60}},
-    {kPcmuCodecName, 8000, 1, 0, false, {10, 20, 30, 40, 50, 60}},
-    {kPcmaCodecName, 8000, 1, 8, false, {10, 20, 30, 40, 50, 60}},
-    {kCnCodecName, 32000, 1, 106, false, {}},
-    {kCnCodecName, 16000, 1, 105, false, {}},
-    {kCnCodecName, 8000, 1, 13, false, {}},
-    {kDtmfCodecName, 48000, 1, 110, false, {}},
-    {kDtmfCodecName, 32000, 1, 112, false, {}},
-    {kDtmfCodecName, 16000, 1, 113, false, {}},
-    {kDtmfCodecName, 8000, 1, 126, false, {}}
-};
-
 // |max_send_bitrate_bps| is the bitrate from "b=" in SDP.
 // |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters.
 rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
                                       rtc::Optional<int> rtp_max_bitrate_bps,
-                                      const webrtc::CodecInst& codec_inst) {
+                                      const webrtc::AudioCodecSpec& spec) {
   // If application-configured bitrate is set, take minimum of that and SDP
   // bitrate.
   const int bps = rtp_max_bitrate_bps
                       ? MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
                       : max_send_bitrate_bps;
-  const int codec_rate = codec_inst.rate;
-
   if (bps <= 0) {
-    return rtc::Optional<int>(codec_rate);
+    return rtc::Optional<int>(spec.info.default_bitrate_bps);
   }
 
-  if (codec_inst.pltype == -1) {
-    return rtc::Optional<int>(codec_rate);
-    ;
-  }
-
-  if (WebRtcVoiceCodecs::IsCodecMultiRate(codec_inst)) {
-    // If codec is multi-rate then just set the bitrate.
-    return rtc::Optional<int>(
-        std::min(bps, WebRtcVoiceCodecs::MaxBitrateBps(codec_inst)));
-  }
-
-  if (bps < codec_inst.rate) {
+  if (bps < spec.info.min_bitrate_bps) {
     // If codec is not multi-rate and |bps| is less than the fixed bitrate then
     // fail. If codec is not multi-rate and |bps| exceeds or equal the fixed
     // bitrate then ignore.
-    LOG(LS_ERROR) << "Failed to set codec " << codec_inst.plname
+    LOG(LS_ERROR) << "Failed to set codec " << spec.format.name
                   << " to bitrate " << bps << " bps"
-                  << ", requires at least " << codec_inst.rate << " bps.";
+                  << ", requires at least " << spec.info.min_bitrate_bps
+                  << " bps.";
     return rtc::Optional<int>();
   }
-  return rtc::Optional<int>(codec_rate);
+
+  if (spec.info.HasFixedBitrate()) {
+    return rtc::Optional<int>(spec.info.default_bitrate_bps);
+  } else {
+    // If codec is multi-rate then just set the bitrate.
+    return rtc::Optional<int>(std::min(bps, spec.info.max_bitrate_bps));
+  }
 }
 
 }  // namespace
 
-bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
-                                    webrtc::CodecInst* out) {
-  return WebRtcVoiceCodecs::ToCodecInst(in, out);
-}
-
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
@@ -565,7 +223,10 @@
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     VoEWrapper* voe_wrapper)
-    : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
+    : adm_(adm),
+      encoder_factory_(webrtc::CreateBuiltinAudioEncoderFactory()),
+      decoder_factory_(decoder_factory),
+      voe_wrapper_(voe_wrapper) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   RTC_DCHECK(voe_wrapper);
@@ -575,13 +236,13 @@
 
   // Load our audio codec list.
   LOG(LS_INFO) << "Supported send codecs in order of preference:";
-  send_codecs_ = WebRtcVoiceCodecs::SupportedSendCodecs();
+  send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders());
   for (const AudioCodec& codec : send_codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
   LOG(LS_INFO) << "Supported recv codecs in order of preference:";
-  recv_codecs_ = CollectRecvCodecs();
+  recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders());
   for (const AudioCodec& codec : recv_codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
@@ -1056,11 +717,10 @@
   return transmit_mixer_;
 }
 
-AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
+AudioCodecs WebRtcVoiceEngine::CollectCodecs(
+    const std::vector<webrtc::AudioCodecSpec>& specs) const {
   PayloadTypeMapper mapper;
   AudioCodecs out;
-  const std::vector<webrtc::AudioCodecSpec>& specs =
-      decoder_factory_->GetSupportedDecoders();
 
   // Only generate CN payload types for these clockrates:
   std::map<int, bool, std::greater<int>> generate_cn = {{ 8000,  false },
@@ -1140,12 +800,14 @@
       webrtc::AudioTransport* voe_audio_transport,
       uint32_t ssrc,
       const std::string& c_name,
-      const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec,
+      const rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
+          send_codec_spec,
       const std::vector<webrtc::RtpExtension>& extensions,
       int max_send_bitrate_bps,
       const rtc::Optional<std::string>& audio_network_adaptor_config,
       webrtc::Call* call,
-      webrtc::Transport* send_transport)
+      webrtc::Transport* send_transport,
+      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
         config_(send_transport),
@@ -1157,13 +819,20 @@
     // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
     // RTC_DCHECK(voe_audio_transport);
     RTC_DCHECK(call);
+    RTC_DCHECK(encoder_factory);
     config_.rtp.ssrc = ssrc;
     config_.rtp.c_name = c_name;
     config_.voe_channel_id = ch;
     config_.rtp.extensions = extensions;
     config_.audio_network_adaptor_config = audio_network_adaptor_config;
+    config_.encoder_factory = encoder_factory;
     rtp_parameters_.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
-    RecreateAudioSendStream(send_codec_spec);
+
+    if (send_codec_spec) {
+      UpdateSendCodecSpec(*send_codec_spec);
+    }
+
+    stream_ = call_->CreateAudioSendStream(config_);
   }
 
   ~WebRtcAudioSendStream() override {
@@ -1172,56 +841,45 @@
     call_->DestroyAudioSendStream(stream_);
   }
 
-  void RecreateAudioSendStream(
+  void SetSendCodecSpec(
       const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
-    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    send_codec_spec_ = send_codec_spec;
-    config_.rtp.nack.rtp_history_ms =
-        send_codec_spec_.nack_enabled ? kNackRtpHistoryMs : 0;
-    config_.send_codec_spec = send_codec_spec_;
-    auto send_rate = ComputeSendBitrate(
-        max_send_bitrate_bps_, rtp_parameters_.encodings[0].max_bitrate_bps,
-        send_codec_spec.codec_inst);
-    if (send_rate) {
-      // Apply a send rate that abides by |max_send_bitrate_bps_| and
-      // |rtp_parameters_| when possible. Otherwise use the codec rate.
-      config_.send_codec_spec.codec_inst.rate = *send_rate;
-    }
-    RecreateAudioSendStream();
+    UpdateSendCodecSpec(send_codec_spec);
+    ReconfigureAudioSendStream();
   }
 
-  void RecreateAudioSendStream(
-      const std::vector<webrtc::RtpExtension>& extensions) {
+  void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     config_.rtp.extensions = extensions;
-    RecreateAudioSendStream();
+    ReconfigureAudioSendStream();
   }
 
-  void RecreateAudioSendStream(
+  void SetAudioNetworkAdaptorConfig(
       const rtc::Optional<std::string>& audio_network_adaptor_config) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (config_.audio_network_adaptor_config == audio_network_adaptor_config) {
       return;
     }
     config_.audio_network_adaptor_config = audio_network_adaptor_config;
-    RecreateAudioSendStream();
+    UpdateAllowedBitrateRange();
+    ReconfigureAudioSendStream();
   }
 
   bool SetMaxSendBitrate(int bps) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    auto send_rate =
-        ComputeSendBitrate(bps, rtp_parameters_.encodings[0].max_bitrate_bps,
-                           send_codec_spec_.codec_inst);
+    RTC_DCHECK(config_.send_codec_spec);
+    RTC_DCHECK(audio_codec_spec_);
+    auto send_rate = ComputeSendBitrate(
+        bps, rtp_parameters_.encodings[0].max_bitrate_bps, *audio_codec_spec_);
+
     if (!send_rate) {
       return false;
     }
 
     max_send_bitrate_bps_ = bps;
 
-    if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
-      // Recreate AudioSendStream with new bit rate.
-      config_.send_codec_spec.codec_inst.rate = *send_rate;
-      RecreateAudioSendStream();
+    if (send_rate != config_.send_codec_spec->target_bitrate_bps) {
+      config_.send_codec_spec->target_bitrate_bps = send_rate;
+      ReconfigureAudioSendStream();
     }
     return true;
   }
@@ -1337,11 +995,15 @@
     if (!ValidateRtpParameters(parameters)) {
       return false;
     }
-    auto send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
-                                        parameters.encodings[0].max_bitrate_bps,
-                                        send_codec_spec_.codec_inst);
-    if (!send_rate) {
-      return false;
+
+    rtc::Optional<int> send_rate;
+    if (audio_codec_spec_) {
+      send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
+                                     parameters.encodings[0].max_bitrate_bps,
+                                     *audio_codec_spec_);
+      if (!send_rate) {
+        return false;
+      }
     }
 
     const rtc::Optional<int> old_rtp_max_bitrate =
@@ -1350,9 +1012,12 @@
     rtp_parameters_ = parameters;
 
     if (rtp_parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) {
-      // Recreate AudioSendStream with new bit rate.
-      config_.send_codec_spec.codec_inst.rate = *send_rate;
-      RecreateAudioSendStream();
+      // Reconfigure AudioSendStream with new bit rate.
+      if (send_rate) {
+        config_.send_codec_spec->target_bitrate_bps = send_rate;
+      }
+      UpdateAllowedBitrateRange();
+      ReconfigureAudioSendStream();
     } else {
       // parameters.encodings[0].active could have changed.
       UpdateSendState();
@@ -1372,18 +1037,17 @@
     }
   }
 
-  void RecreateAudioSendStream() {
+  void UpdateAllowedBitrateRange() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    if (stream_) {
-      call_->DestroyAudioSendStream(stream_);
-      stream_ = nullptr;
-    }
-    RTC_DCHECK(!stream_);
-    if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
+    const bool is_opus =
+        config_.send_codec_spec &&
+        !STR_CASE_CMP(config_.send_codec_spec->format.name.c_str(),
+                      kOpusCodecName);
+    if (is_opus && webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
       config_.min_bitrate_bps = kOpusMinBitrateBps;
 
       // This means that when RtpParameters is reset, we may change the
-      // encoder's bit rate immediately (through call_->CreateAudioSendStream),
+      // encoder's bit rate immediately (through ReconfigureAudioSendStream()),
       // meanwhile change the cap to the output of BWE.
       config_.max_bitrate_bps =
           rtp_parameters_.encodings[0].max_bitrate_bps
@@ -1393,48 +1057,65 @@
       // TODO(mflodman): Keep testing this and set proper values.
       // Note: This is an early experiment currently only supported by Opus.
       if (send_side_bwe_with_overhead_) {
-        auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs(
-            config_.send_codec_spec.codec_inst);
-        if (!packet_sizes_ms.empty()) {
-          int max_packet_size_ms =
-              *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
+        const int max_packet_size_ms =
+            WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
 
-          // Audio network adaptor will just use 20ms and 60ms frame lengths.
-          // The adaptor will only be active for the Opus encoder.
-          if (config_.audio_network_adaptor_config &&
-              IsCodec(config_.send_codec_spec.codec_inst, kOpusCodecName)) {
-#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
-            max_packet_size_ms = 120;
-#else
-            max_packet_size_ms = 60;
-#endif
-          }
+        // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+        constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
 
-          // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
-          constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
+        int min_overhead_bps =
+            kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;
 
-          int min_overhead_bps =
-              kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;
+        // We assume that |config_.max_bitrate_bps| before the next line is
+        // a hard limit on the payload bitrate, so we add min_overhead_bps to
+        // it to ensure that, when overhead is deducted, the payload rate
+        // never goes beyond the limit.
+        // Note: this also means that if a higher overhead is forced, we
+        // cannot reach the limit.
+        // TODO(minyue): Reconsider this when the signaling to BWE is done
+        // through a dedicated API.
+        config_.max_bitrate_bps += min_overhead_bps;
 
-          // We assume that |config_.max_bitrate_bps| before the next line is
-          // a hard limit on the payload bitrate, so we add min_overhead_bps to
-          // it to ensure that, when overhead is deducted, the payload rate
-          // never goes beyond the limit.
-          // Note: this also means that if a higher overhead is forced, we
-          // cannot reach the limit.
-          // TODO(minyue): Reconsider this when the signaling to BWE is done
-          // through a dedicated API.
-          config_.max_bitrate_bps += min_overhead_bps;
-
-          // In contrast to max_bitrate_bps, we let min_bitrate_bps always be
-          // reachable.
-          config_.min_bitrate_bps += min_overhead_bps;
-        }
+        // In contrast to max_bitrate_bps, we let min_bitrate_bps always be
+        // reachable.
+        config_.min_bitrate_bps += min_overhead_bps;
       }
     }
-    stream_ = call_->CreateAudioSendStream(config_);
-    RTC_CHECK(stream_);
-    UpdateSendState();
+  }
+
+  void UpdateSendCodecSpec(
+      const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
+    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+    config_.rtp.nack.rtp_history_ms =
+        send_codec_spec.nack_enabled ? kNackRtpHistoryMs : 0;
+    config_.send_codec_spec =
+        rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>(
+            send_codec_spec);
+    auto info =
+        config_.encoder_factory->QueryAudioEncoder(send_codec_spec.format);
+    RTC_DCHECK(info);
+    // If a specific target bitrate has been set for the stream, use that as
+    // the new default bitrate when computing send bitrate.
+    if (send_codec_spec.target_bitrate_bps) {
+      info->default_bitrate_bps = std::max(
+          info->min_bitrate_bps,
+          std::min(info->max_bitrate_bps, *send_codec_spec.target_bitrate_bps));
+    }
+
+    audio_codec_spec_.emplace(
+        webrtc::AudioCodecSpec{send_codec_spec.format, *info});
+
+    config_.send_codec_spec->target_bitrate_bps = ComputeSendBitrate(
+        max_send_bitrate_bps_, rtp_parameters_.encodings[0].max_bitrate_bps,
+        *audio_codec_spec_);
+
+    UpdateAllowedBitrateRange();
+  }
+
+  void ReconfigureAudioSendStream() {
+    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+    RTC_DCHECK(stream_);
+    stream_->Reconfigure(config_);
   }
 
   rtc::ThreadChecker worker_thread_checker_;
@@ -1455,7 +1136,7 @@
   bool muted_ = false;
   int max_send_bitrate_bps_;
   webrtc::RtpParameters rtp_parameters_;
-  webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
+  rtc::Optional<webrtc::AudioCodecSpec> audio_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
 };
@@ -1651,7 +1332,7 @@
   if (send_rtp_extensions_ != filtered_extensions) {
     send_rtp_extensions_.swap(filtered_extensions);
     for (auto& it : send_streams_) {
-      it.second->RecreateAudioSendStream(send_rtp_extensions_);
+      it.second->SetRtpExtensions(send_rtp_extensions_);
     }
   }
 
@@ -1818,10 +1499,10 @@
     return false;
   }
 
-  rtc::Optional<std::string> audio_network_adatptor_config =
+  rtc::Optional<std::string> audio_network_adaptor_config =
       GetAudioNetworkAdaptorConfig(options_);
   for (auto& it : send_streams_) {
-    it.second->RecreateAudioSendStream(audio_network_adatptor_config);
+    it.second->SetAudioNetworkAdaptorConfig(audio_network_adaptor_config);
   }
 
   LOG(LS_INFO) << "Set voice channel options. Current options: "
@@ -1938,86 +1619,66 @@
     }
   }
 
-  // Scan through the list to figure out the codec to use for sending, along
-  // with the proper configuration for VAD, CNG, NACK and Opus-specific
-  // parameters.
-  // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
-  webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
+  // Scan through the list to figure out the codec to use for sending.
+  rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> send_codec_spec;
   webrtc::Call::Config::BitrateConfig bitrate_config;
-  {
-    send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
+  rtc::Optional<webrtc::AudioCodecInfo> voice_codec_info;
+  for (const AudioCodec& voice_codec : codecs) {
+    if (!(IsCodec(voice_codec, kCnCodecName) ||
+          IsCodec(voice_codec, kDtmfCodecName) ||
+          IsCodec(voice_codec, kRedCodecName))) {
+      webrtc::SdpAudioFormat format(voice_codec.name, voice_codec.clockrate,
+                                    voice_codec.channels, voice_codec.params);
 
-    // Find send codec (the first non-telephone-event/CN codec).
-    const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
-        codecs, &send_codec_spec.codec_inst);
-    if (!codec) {
-      LOG(LS_WARNING) << "Received empty list of codecs.";
-      return false;
-    }
-
-    send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
-    send_codec_spec.nack_enabled = HasNack(*codec);
-    bitrate_config = GetBitrateConfigForCodec(*codec);
-
-    // For Opus as the send codec, we are to determine inband FEC, maximum
-    // playback rate, and opus internal dtx.
-    if (IsCodec(*codec, kOpusCodecName)) {
-      GetOpusConfig(*codec, &send_codec_spec.codec_inst,
-                    &send_codec_spec.enable_codec_fec,
-                    &send_codec_spec.opus_max_playback_rate,
-                    &send_codec_spec.enable_opus_dtx,
-                    &send_codec_spec.min_ptime_ms,
-                    &send_codec_spec.max_ptime_ms);
-    }
-
-    // Set packet size if the AudioCodec param kCodecParamPTime is set.
-    int ptime_ms = 0;
-    if (codec->GetParam(kCodecParamPTime, &ptime_ms)) {
-      if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(
-          &send_codec_spec.codec_inst, ptime_ms)) {
-        LOG(LS_WARNING) << "Failed to set packet size for codec "
-                        << send_codec_spec.codec_inst.plname;
-        return false;
-      }
-    }
-
-    // Loop through the codecs list again to find the CN codec.
-    // TODO(solenberg): Break out into a separate function?
-    for (const AudioCodec& cn_codec : codecs) {
-      // Ignore codecs we don't know about. The negotiation step should prevent
-      // this, but double-check to be sure.
-      webrtc::CodecInst voe_codec = {0};
-      if (!WebRtcVoiceEngine::ToCodecInst(cn_codec, &voe_codec)) {
-        LOG(LS_WARNING) << "Unknown codec " << ToString(cn_codec);
+      voice_codec_info = engine()->encoder_factory_->QueryAudioEncoder(format);
+      if (!voice_codec_info) {
+        LOG(LS_WARNING) << "Unknown codec " << ToString(voice_codec);
         continue;
       }
 
+      send_codec_spec =
+          rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>(
+              {voice_codec.id, format});
+      if (voice_codec.bitrate > 0) {
+        send_codec_spec->target_bitrate_bps =
+            rtc::Optional<int>(voice_codec.bitrate);
+      }
+      send_codec_spec->transport_cc_enabled = HasTransportCc(voice_codec);
+      send_codec_spec->nack_enabled = HasNack(voice_codec);
+      bitrate_config = GetBitrateConfigForCodec(voice_codec);
+      break;
+    }
+  }
+
+  if (!send_codec_spec) {
+    return false;
+  }
+
+  RTC_DCHECK(voice_codec_info);
+  if (voice_codec_info->allow_comfort_noise) {
+    // Loop through the codecs list again to find the CN codec.
+    // TODO(solenberg): Break out into a separate function?
+    for (const AudioCodec& cn_codec : codecs) {
       if (IsCodec(cn_codec, kCnCodecName) &&
-          cn_codec.clockrate == codec->clockrate) {
-        // Turn voice activity detection/comfort noise on if supported.
-        // Set the wideband CN payload type appropriately.
-        // (narrowband always uses the static payload type 13).
-        int cng_plfreq = -1;
+          cn_codec.clockrate == send_codec_spec->format.clockrate_hz) {
         switch (cn_codec.clockrate) {
           case 8000:
           case 16000:
           case 32000:
-            cng_plfreq = cn_codec.clockrate;
+            send_codec_spec->cng_payload_type = rtc::Optional<int>(cn_codec.id);
             break;
           default:
             LOG(LS_WARNING) << "CN frequency " << cn_codec.clockrate
                             << " not supported.";
-            continue;
+            break;
         }
-        send_codec_spec.cng_payload_type = cn_codec.id;
-        send_codec_spec.cng_plfreq = cng_plfreq;
         break;
       }
     }
 
     // Find the telephone-event PT exactly matching the preferred send codec.
     for (const AudioCodec& dtmf_codec : dtmf_codecs) {
-      if (dtmf_codec.clockrate == codec->clockrate) {
+      if (dtmf_codec.clockrate == send_codec_spec->format.clockrate_hz) {
         dtmf_payload_type_ = rtc::Optional<int>(dtmf_codec.id);
         dtmf_payload_freq_ = dtmf_codec.clockrate;
         break;
@@ -2029,7 +1690,7 @@
     send_codec_spec_ = std::move(send_codec_spec);
     // Apply new settings to all streams.
     for (const auto& kv : send_streams_) {
-      kv.second->RecreateAudioSendStream(send_codec_spec_);
+      kv.second->SetSendCodecSpec(*send_codec_spec_);
     }
   } else {
     // If the codec isn't changing, set the start bitrate to -1 which means
@@ -2040,12 +1701,12 @@
 
   // Check if the transport cc feedback or NACK status has changed on the
   // preferred send codec, and in that case reconfigure all receive streams.
-  if (recv_transport_cc_enabled_ != send_codec_spec_.transport_cc_enabled ||
-      recv_nack_enabled_ != send_codec_spec_.nack_enabled) {
+  if (recv_transport_cc_enabled_ != send_codec_spec_->transport_cc_enabled ||
+      recv_nack_enabled_ != send_codec_spec_->nack_enabled) {
     LOG(LS_INFO) << "Recreate all the receive streams because the send "
                     "codec has changed.";
-    recv_transport_cc_enabled_ = send_codec_spec_.transport_cc_enabled;
-    recv_nack_enabled_ = send_codec_spec_.nack_enabled;
+    recv_transport_cc_enabled_ = send_codec_spec_->transport_cc_enabled;
+    recv_nack_enabled_ = send_codec_spec_->nack_enabled;
     for (auto& kv : recv_streams_) {
       kv.second->RecreateAudioReceiveStream(recv_transport_cc_enabled_,
                                             recv_nack_enabled_);
@@ -2168,7 +1829,7 @@
   WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
       channel, audio_transport, ssrc, sp.cname, send_codec_spec_,
       send_rtp_extensions_, max_send_bitrate_bps_, audio_network_adaptor_config,
-      call_, this);
+      call_, this, engine()->encoder_factory_);
   send_streams_.insert(std::make_pair(ssrc, stream));
 
   // At this point the stream's local SSRC has been updated. If it is the first
diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h
index 4d71c89..20b975c 100644
--- a/media/engine/webrtcvoiceengine.h
+++ b/media/engine/webrtcvoiceengine.h
@@ -29,6 +29,7 @@
 #include "webrtc/media/engine/apm_helpers.h"
 #include "webrtc/media/engine/webrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/pc/channel.h"
 
@@ -51,9 +52,6 @@
 class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
   friend class WebRtcVoiceMediaChannel;
  public:
-  // Exposed for the WVoE/MC unit test.
-  static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
-
   WebRtcVoiceEngine(
       webrtc::AudioDeviceModule* adm,
       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
@@ -114,13 +112,15 @@
   webrtc::AudioProcessing* apm();
   webrtc::voe::TransmitMixer* transmit_mixer();
 
-  AudioCodecs CollectRecvCodecs() const;
+  AudioCodecs CollectCodecs(
+      const std::vector<webrtc::AudioCodecSpec>& specs) const;
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
   // The audio device manager.
   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_;
+  rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_;
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_;
   // Reference to the APM, owned by VoE.
   webrtc::AudioProcessing* apm_ = nullptr;
@@ -291,7 +291,8 @@
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
-  webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
+  rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>
+      send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
index dfb49cf..735f36e 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -41,7 +41,7 @@
 
 const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1);
 const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1);
-const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 64000, 2);
+const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 32000, 2);
 const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
 const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
 const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
@@ -362,34 +362,16 @@
     SetSendParameters(send_parameters);
   }
 
-  void CheckSendCodec(int32_t ssrc,
-                      const char expected_name[],
-                      int expected_channels,
-                      int expected_bitrate) {
-    const auto& codec = GetSendStreamConfig(ssrc).send_codec_spec.codec_inst;
-    EXPECT_STREQ(expected_name, codec.plname);
-    EXPECT_EQ(expected_channels, codec.channels);
-    EXPECT_EQ(expected_bitrate, codec.rate);
+  void CheckSendCodecBitrate(int32_t ssrc,
+                             const char expected_name[],
+                             int expected_bitrate) {
+    const auto& spec = GetSendStreamConfig(ssrc).send_codec_spec;
+    EXPECT_EQ(expected_name, spec->format.name);
+    EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps);
   }
 
-  int GetOpusMaxPlaybackRate(int32_t ssrc) {
-    return GetSendStreamConfig(ssrc).send_codec_spec.opus_max_playback_rate;
-  }
-
-  bool GetOpusDtx(int32_t ssrc) {
-    return GetSendStreamConfig(ssrc).send_codec_spec.enable_opus_dtx;
-  }
-
-  bool GetCodecFec(int32_t ssrc) {
-    return GetSendStreamConfig(ssrc).send_codec_spec.enable_codec_fec;
-  }
-
-  int GetCodecBitrate(int32_t ssrc) {
-    return GetSendStreamConfig(ssrc).send_codec_spec.codec_inst.rate;
-  }
-
-  int GetCodecPacSize(int32_t ssrc) {
-    return GetSendStreamConfig(ssrc).send_codec_spec.codec_inst.pacsize;
+  rtc::Optional<int> GetCodecBitrate(int32_t ssrc) {
+    return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps;
   }
 
   const rtc::Optional<std::string>& GetAudioNetworkAdaptorConfig(int32_t ssrc) {
@@ -718,19 +700,6 @@
   EXPECT_EQ("", config.sync_group);
 }
 
-// Tests that the list of supported codecs is created properly and ordered
-// correctly (such that opus appears first).
-// TODO(ossu): This test should move into a separate builtin audio codecs
-// module.
-TEST_F(WebRtcVoiceEngineTestFake, CodecOrder) {
-  const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs();
-  ASSERT_FALSE(codecs.empty());
-  EXPECT_STRCASEEQ("opus", codecs[0].name.c_str());
-  EXPECT_EQ(48000, codecs[0].clockrate);
-  EXPECT_EQ(2, codecs[0].channels);
-  EXPECT_EQ(64000, codecs[0].bitrate);
-}
-
 TEST_F(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) {
   const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs();
   bool opus_found = false;
@@ -743,46 +712,6 @@
   EXPECT_TRUE(opus_found);
 }
 
-// Tests that we can find codecs by name or id, and that we interpret the
-// clockrate and bitrate fields properly.
-TEST_F(WebRtcVoiceEngineTestFake, FindCodec) {
-  cricket::AudioCodec codec;
-  webrtc::CodecInst codec_inst;
-  // Find PCMU with explicit clockrate and bitrate.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kPcmuCodec, &codec_inst));
-  // Find ISAC with explicit clockrate and 0 bitrate.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kIsacCodec, &codec_inst));
-  // Find telephone-event with explicit clockrate and 0 bitrate.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec1,
-                                                      &codec_inst));
-  // Find telephone-event with explicit clockrate and 0 bitrate.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec2,
-                                                      &codec_inst));
-  // Find ISAC with a different payload id.
-  codec = kIsacCodec;
-  codec.id = 127;
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
-  EXPECT_EQ(codec.id, codec_inst.pltype);
-  // Find PCMU with a 0 clockrate.
-  codec = kPcmuCodec;
-  codec.clockrate = 0;
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
-  EXPECT_EQ(codec.id, codec_inst.pltype);
-  EXPECT_EQ(8000, codec_inst.plfreq);
-  // Find PCMU with a 0 bitrate.
-  codec = kPcmuCodec;
-  codec.bitrate = 0;
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
-  EXPECT_EQ(codec.id, codec_inst.pltype);
-  EXPECT_EQ(64000, codec_inst.rate);
-  // Find ISAC with an explicit bitrate.
-  codec = kIsacCodec;
-  codec.bitrate = 32000;
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
-  EXPECT_EQ(codec.id, codec_inst.pltype);
-  EXPECT_EQ(32000, codec_inst.rate);
-}
-
 // Test that we set our inbound codecs properly, including changing PT.
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
   EXPECT_TRUE(SetupChannel());
@@ -935,9 +864,6 @@
   parameters.codecs.push_back(kOpusCodec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(GetRecvStream(kSsrcX).started());
-  webrtc::CodecInst gcodec;
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &gcodec));
-  EXPECT_EQ(kOpusCodec.id, gcodec.pltype);
 }
 
 // Test that we accept adding the same codec with a different payload type.
@@ -965,20 +891,17 @@
   // PCMU, default bitrate == 64000.
   TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000);
 
-  // opus, default bitrate == 64000.
-  TestMaxSendBandwidth(kOpusCodec, -1, true, 64000);
+  // opus, default bitrate == 32000 in mono.
+  TestMaxSendBandwidth(kOpusCodec, -1, true, 32000);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
   EXPECT_TRUE(SetupSendStream());
 
-  // Test that the bitrate of a multi-rate codec is always the maximum.
-
   // ISAC, default bitrate == 32000.
-  TestMaxSendBandwidth(kIsacCodec, 40000, true, 40000);
   TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000);
   // Rates above the max (56000) should be capped.
-  TestMaxSendBandwidth(kIsacCodec, 100000, true, 56000);
+  TestMaxSendBandwidth(kIsacCodec, 100000, true, 32000);
 
   // opus, default bitrate == 64000.
   TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000);
@@ -1041,8 +964,8 @@
 TEST_F(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) {
   EXPECT_TRUE(SetupSendStream());
 
-  // opus, default bitrate == 64000.
-  SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 64000);
+  // opus, default bitrate == 32000.
+  SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 32000);
   SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000);
   SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000);
   SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000);
@@ -1127,21 +1050,21 @@
   // Configure one stream to be limited by the stream config, another to be
   // limited by the global max, and the third one with no per-stream limit
   // (still subject to the global limit).
-  SetGlobalMaxBitrate(kOpusCodec, 64000);
-  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 48000));
-  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 96000));
+  SetGlobalMaxBitrate(kOpusCodec, 32000);
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 24000));
+  EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 48000));
   EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1));
 
-  EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0]));
-  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[1]));
-  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[2]));
+  EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0]));
+  EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[1]));
+  EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2]));
 
   // Remove the global cap; the streams should switch to their respective
   // maximums (or remain unchanged if there was no other limit on them.)
   SetGlobalMaxBitrate(kOpusCodec, -1);
-  EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[0]));
-  EXPECT_EQ(96000, GetCodecBitrate(kSsrcs4[1]));
-  EXPECT_EQ(64000, GetCodecBitrate(kSsrcs4[2]));
+  EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0]));
+  EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[1]));
+  EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2]));
 }
 
 // Test that GetRtpSendParameters returns the currently configured codecs.
@@ -1296,22 +1219,20 @@
   parameters.codecs.push_back(kPcmuCodec);
   parameters.codecs.push_back(kCn8000Codec);
   parameters.codecs[0].id = 96;
-  parameters.codecs[0].bitrate = 48000;
-  const int initial_num = call_.GetNumCreatedSendStreams();
+  parameters.codecs[0].bitrate = 22000;
   SetSendParameters(parameters);
-  EXPECT_EQ(initial_num + 1, call_.GetNumCreatedSendStreams());
-  const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-  EXPECT_EQ(96, send_codec_spec.codec_inst.pltype);
-  EXPECT_EQ(48000, send_codec_spec.codec_inst.rate);
-  EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
-  EXPECT_NE(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
-  EXPECT_EQ(-1, send_codec_spec.cng_payload_type);
+  const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+  EXPECT_EQ(96, send_codec_spec.payload_type);
+  EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps);
+  EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
+  EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000);
+  EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type);
   EXPECT_FALSE(channel_->CanInsertDtmf());
 }
 
-// Test that VoE Channel doesn't call SetSendCodec again if same codec is tried
-// to apply.
-TEST_F(WebRtcVoiceEngineTestFake, DontResetSetSendCodec) {
+// Test that WebRtcVoiceEngine reconfigures, rather than recreates its
+// AudioSendStream.
+TEST_F(WebRtcVoiceEngineTestFake, DontRecreateSendStream) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kIsacCodec);
@@ -1321,24 +1242,15 @@
   parameters.codecs[0].bitrate = 48000;
   const int initial_num = call_.GetNumCreatedSendStreams();
   SetSendParameters(parameters);
-  EXPECT_EQ(initial_num + 1, call_.GetNumCreatedSendStreams());
+  EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
   // Calling SetSendCodec again with same codec which is already set.
   // In this case media channel shouldn't send codec to VoE.
   SetSendParameters(parameters);
-  EXPECT_EQ(initial_num + 1, call_.GetNumCreatedSendStreams());
+  EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
 }
 
-// Verify that G722 is set with 16000 samples per second to WebRTC.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecG722) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kG722CodecSdp);
-  SetSendParameters(parameters);
-  const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-  EXPECT_STREQ("G722", gcodec.plname);
-  EXPECT_EQ(1, gcodec.channels);
-  EXPECT_EQ(16000, gcodec.plfreq);
-}
+// TODO(ossu): Revisit if these tests need to be here, now that these kinds of
+// tests should be available in AudioEncoderOpusTest.
 
 // Test that if clockrate is not 48000 for opus, we fail.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
@@ -1403,19 +1315,17 @@
   EXPECT_FALSE(channel_->SetSendParameters(parameters));
 }
 
-// Test that with bitrate=0 and no stereo,
-// channels and bitrate are 1 and 32000.
+// Test that with bitrate=0 and no stereo, bitrate is 32000.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kOpusCodec);
   parameters.codecs[0].bitrate = 0;
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 1, 32000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 32000);
 }
 
-// Test that with bitrate=0 and stereo=0,
-// channels and bitrate are 1 and 32000.
+// Test that with bitrate=0 and stereo=0, bitrate is 32000.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
@@ -1423,11 +1333,10 @@
   parameters.codecs[0].bitrate = 0;
   parameters.codecs[0].params["stereo"] = "0";
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 1, 32000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 32000);
 }
 
-// Test that with bitrate=invalid and stereo=0,
-// channels and bitrate are 1 and 32000.
+// Test that with bitrate=invalid and stereo=0, bitrate is 32000.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
@@ -1436,15 +1345,14 @@
   // bitrate that's out of the range between 6000 and 510000 will be clamped.
   parameters.codecs[0].bitrate = 5999;
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 1, 6000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 6000);
 
   parameters.codecs[0].bitrate = 510001;
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 1, 510000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 510000);
 }
 
-// Test that with bitrate=0 and stereo=1,
-// channels and bitrate are 2 and 64000.
+// Test that with bitrate=0 and stereo=1, bitrate is 64000.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
@@ -1452,11 +1360,10 @@
   parameters.codecs[0].bitrate = 0;
   parameters.codecs[0].params["stereo"] = "1";
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 2, 64000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 64000);
 }
 
-// Test that with bitrate=invalid and stereo=1,
-// channels and bitrate are 2 and 64000.
+// Test that with bitrate=invalid and stereo=1, bitrate is 64000.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
@@ -1465,31 +1372,29 @@
   // bitrate that's out of the range between 6000 and 510000 will be clamped.
   parameters.codecs[0].bitrate = 5999;
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 2, 6000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 6000);
 
   parameters.codecs[0].bitrate = 510001;
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 2, 510000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 510000);
 }
 
-// Test that with bitrate=N and stereo unset,
-// channels and bitrate are 1 and N.
+// Test that with bitrate=N and stereo unset, bitrate is N.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kOpusCodec);
   parameters.codecs[0].bitrate = 96000;
   SetSendParameters(parameters);
-  const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-  EXPECT_EQ(111, gcodec.pltype);
-  EXPECT_EQ(96000, gcodec.rate);
-  EXPECT_STREQ("opus", gcodec.plname);
-  EXPECT_EQ(1, gcodec.channels);
-  EXPECT_EQ(48000, gcodec.plfreq);
+  const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+  EXPECT_EQ(111, spec.payload_type);
+  EXPECT_EQ(96000, spec.target_bitrate_bps);
+  EXPECT_EQ("opus", spec.format.name);
+  EXPECT_EQ(2, spec.format.num_channels);
+  EXPECT_EQ(48000, spec.format.clockrate_hz);
 }
 
-// Test that with bitrate=N and stereo=0,
-// channels and bitrate are 1 and N.
+// Test that with bitrate=N and stereo=0, bitrate is N.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
@@ -1497,22 +1402,20 @@
   parameters.codecs[0].bitrate = 30000;
   parameters.codecs[0].params["stereo"] = "0";
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 1, 30000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 30000);
 }
 
-// Test that with bitrate=N and without any parameters,
-// channels and bitrate are 1 and N.
+// Test that with bitrate=N and without any parameters, bitrate is N.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
   parameters.codecs.push_back(kOpusCodec);
   parameters.codecs[0].bitrate = 30000;
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 1, 30000);
+  CheckSendCodecBitrate(kSsrcX, "opus", 30000);
 }
 
-// Test that with bitrate=N and stereo=1,
-// channels and bitrate are 2 and N.
+// Test that with bitrate=N and stereo=1, bitrate is N.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) {
   EXPECT_TRUE(SetupSendStream());
   cricket::AudioSendParameters parameters;
@@ -1520,30 +1423,7 @@
   parameters.codecs[0].bitrate = 30000;
   parameters.codecs[0].params["stereo"] = "1";
   SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 2, 30000);
-}
-
-// Test that bitrate will be overridden by the "maxaveragebitrate" parameter.
-// Also test that the "maxaveragebitrate" can't be set to values outside the
-// range of 6000 and 510000
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusMaxAverageBitrate) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 30000;
-  // Ignore if less than 6000.
-  parameters.codecs[0].params["maxaveragebitrate"] = "5999";
-  SetSendParameters(parameters);
-  EXPECT_EQ(6000, GetCodecBitrate(kSsrcX));
-
-  // Ignore if larger than 510000.
-  parameters.codecs[0].params["maxaveragebitrate"] = "510001";
-  SetSendParameters(parameters);
-  EXPECT_EQ(510000, GetCodecBitrate(kSsrcX));
-
-  parameters.codecs[0].params["maxaveragebitrate"] = "200000";
-  SetSendParameters(parameters);
-  EXPECT_EQ(200000, GetCodecBitrate(kSsrcX));
+  CheckSendCodecBitrate(kSsrcX, "opus", 30000);
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithBitrates) {
@@ -1679,83 +1559,6 @@
   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcZ).rtp.nack.rtp_history_ms);
 }
 
-// Test that without useinbandfec, Opus FEC is off.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFec) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  SetSendParameters(parameters);
-  EXPECT_FALSE(GetCodecFec(kSsrcX));
-}
-
-// Test that with useinbandfec=0, Opus FEC is off.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusDisableFec) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].params["useinbandfec"] = "0";
-  SetSendParameters(parameters);
-  CheckSendCodec(kSsrcX, "opus", 1, 32000);
-}
-
-// Test that with useinbandfec=1, Opus FEC is on.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFec) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].params["useinbandfec"] = "1";
-  SetSendParameters(parameters);
-  EXPECT_TRUE(GetCodecFec(kSsrcX));
-  CheckSendCodec(kSsrcX, "opus", 1, 32000);
-}
-
-// Test that with useinbandfec=1, stereo=1, Opus FEC is on.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusEnableFecStereo) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].params["stereo"] = "1";
-  parameters.codecs[0].params["useinbandfec"] = "1";
-  SetSendParameters(parameters);
-  EXPECT_TRUE(GetCodecFec(kSsrcX));
-  CheckSendCodec(kSsrcX, "opus", 2, 64000);
-}
-
-// Test that with non-Opus, codec FEC is off.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacNoFec) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kIsacCodec);
-  SetSendParameters(parameters);
-  EXPECT_FALSE(GetCodecFec(kSsrcX));
-}
-
-// Test the with non-Opus, even if useinbandfec=1, FEC is off.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacWithParamNoFec) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kIsacCodec);
-  parameters.codecs[0].params["useinbandfec"] = "1";
-  SetSendParameters(parameters);
-  EXPECT_FALSE(GetCodecFec(kSsrcX));
-}
-
-// Test that Opus FEC status can be changed.
-TEST_F(WebRtcVoiceEngineTestFake, ChangeOpusFecStatus) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  SetSendParameters(parameters);
-  EXPECT_FALSE(GetCodecFec(kSsrcX));
-
-  parameters.codecs[0].params["useinbandfec"] = "1";
-  SetSendParameters(parameters);
-  EXPECT_TRUE(GetCodecFec(kSsrcX));
-}
-
 TEST_F(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) {
   EXPECT_TRUE(SetupChannel());
   cricket::AudioSendParameters send_parameters;
@@ -1778,162 +1581,6 @@
       call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc);
 }
 
-// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
-TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateNb) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000);
-  SetSendParameters(parameters);
-  EXPECT_EQ(8000, GetOpusMaxPlaybackRate(kSsrcX));
-  EXPECT_EQ(12000, GetCodecBitrate(kSsrcX));
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1");
-  SetSendParameters(parameters);
-  EXPECT_EQ(24000, GetCodecBitrate(kSsrcX));
-}
-
-// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
-TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateMb) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8001);
-  SetSendParameters(parameters);
-  EXPECT_EQ(8001, GetOpusMaxPlaybackRate(kSsrcX));
-  EXPECT_EQ(20000, GetCodecBitrate(kSsrcX));
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1");
-  SetSendParameters(parameters);
-  EXPECT_EQ(40000, GetCodecBitrate(kSsrcX));
-}
-
-// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
-TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateWb) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 12001);
-  SetSendParameters(parameters);
-  EXPECT_EQ(12001, GetOpusMaxPlaybackRate(kSsrcX));
-  EXPECT_EQ(20000, GetCodecBitrate(kSsrcX));
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1");
-  SetSendParameters(parameters);
-  EXPECT_EQ(40000, GetCodecBitrate(kSsrcX));
-}
-
-// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
-TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateSwb) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 16001);
-  SetSendParameters(parameters);
-  EXPECT_EQ(16001, GetOpusMaxPlaybackRate(kSsrcX));
-  EXPECT_EQ(32000, GetCodecBitrate(kSsrcX));
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1");
-  SetSendParameters(parameters);
-  EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
-}
-
-// Test 24000 < maxplaybackrate triggers Opus full band mode.
-TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateFb) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].bitrate = 0;
-  parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 24001);
-  SetSendParameters(parameters);
-  EXPECT_EQ(24001, GetOpusMaxPlaybackRate(kSsrcX));
-  EXPECT_EQ(32000, GetCodecBitrate(kSsrcX));
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamStereo, "1");
-  SetSendParameters(parameters);
-  EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
-}
-
-// Test Opus that without maxplaybackrate, default playback rate is used.
-TEST_F(WebRtcVoiceEngineTestFake, DefaultOpusMaxPlaybackRate) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  SetSendParameters(parameters);
-  EXPECT_EQ(48000, GetOpusMaxPlaybackRate(kSsrcX));
-}
-
-// Test the with non-Opus, maxplaybackrate has no effect.
-TEST_F(WebRtcVoiceEngineTestFake, SetNonOpusMaxPlaybackRate) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kIsacCodec);
-  parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 32000);
-  SetSendParameters(parameters);
-  EXPECT_EQ(0, GetOpusMaxPlaybackRate(kSsrcX));
-}
-
-// Test maxplaybackrate can be set on two streams.
-TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  SetSendParameters(parameters);
-  EXPECT_EQ(48000, GetOpusMaxPlaybackRate(kSsrcX));
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamMaxPlaybackRate, 8000);
-  SetSendParameters(parameters);
-  EXPECT_EQ(8000, GetOpusMaxPlaybackRate(kSsrcX));
-
-  channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY));
-  EXPECT_EQ(8000, GetOpusMaxPlaybackRate(kSsrcY));
-}
-
-// Test that with usedtx=0, Opus DTX is off.
-TEST_F(WebRtcVoiceEngineTestFake, DisableOpusDtxOnOpus) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].params["usedtx"] = "0";
-  SetSendParameters(parameters);
-  EXPECT_FALSE(GetOpusDtx(kSsrcX));
-}
-
-// Test that with usedtx=1, Opus DTX is on.
-TEST_F(WebRtcVoiceEngineTestFake, EnableOpusDtxOnOpus) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].params["usedtx"] = "1";
-  SetSendParameters(parameters);
-  EXPECT_TRUE(GetOpusDtx(kSsrcX));
-}
-
-// Test that usedtx=1 works with stereo Opus.
-TEST_F(WebRtcVoiceEngineTestFake, EnableOpusDtxOnOpusStereo) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].params["usedtx"] = "1";
-  parameters.codecs[0].params["stereo"] = "1";
-  SetSendParameters(parameters);
-  EXPECT_TRUE(GetOpusDtx(kSsrcX));
-}
-
-// Test that usedtx=1 does not work with non Opus.
-TEST_F(WebRtcVoiceEngineTestFake, CannotEnableOpusDtxOnNonOpus) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kIsacCodec);
-  parameters.codecs[0].params["usedtx"] = "1";
-  SetSendParameters(parameters);
-  EXPECT_FALSE(GetOpusDtx(kSsrcX));
-}
-
 // Test that we can switch back and forth between Opus and ISAC with CN.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) {
   EXPECT_TRUE(SetupSendStream());
@@ -1942,9 +1589,9 @@
   opus_parameters.codecs.push_back(kOpusCodec);
   SetSendParameters(opus_parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(111, gcodec.pltype);
-    EXPECT_STREQ("opus", gcodec.plname);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(111, spec.payload_type);
+    EXPECT_STRCASEEQ("opus", spec.format.name.c_str());
   }
 
   cricket::AudioSendParameters isac_parameters;
@@ -1953,16 +1600,16 @@
   isac_parameters.codecs.push_back(kOpusCodec);
   SetSendParameters(isac_parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(103, gcodec.pltype);
-    EXPECT_STREQ("ISAC", gcodec.plname);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(103, spec.payload_type);
+    EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
   }
 
   SetSendParameters(opus_parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(111, gcodec.pltype);
-    EXPECT_STREQ("opus", gcodec.plname);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(111, spec.payload_type);
+    EXPECT_STRCASEEQ("opus", spec.format.name.c_str());
   }
 }
 
@@ -1973,88 +1620,58 @@
   parameters.codecs.push_back(kIsacCodec);  // bitrate == 32000
   SetSendParameters(parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(103, gcodec.pltype);
-    EXPECT_STREQ("ISAC", gcodec.plname);
-    EXPECT_EQ(32000, gcodec.rate);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(103, spec.payload_type);
+    EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
+    EXPECT_EQ(32000, spec.target_bitrate_bps);
   }
 
   parameters.codecs[0].bitrate = 0;         // bitrate == default
   SetSendParameters(parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(103, gcodec.pltype);
-    EXPECT_STREQ("ISAC", gcodec.plname);
-    EXPECT_EQ(32000, gcodec.rate);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(103, spec.payload_type);
+    EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
+    EXPECT_EQ(32000, spec.target_bitrate_bps);
   }
   parameters.codecs[0].bitrate = 28000;     // bitrate == 28000
   SetSendParameters(parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(103, gcodec.pltype);
-    EXPECT_STREQ("ISAC", gcodec.plname);
-    EXPECT_EQ(28000, gcodec.rate);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(103, spec.payload_type);
+    EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
+    EXPECT_EQ(28000, spec.target_bitrate_bps);
   }
 
   parameters.codecs[0] = kPcmuCodec;        // bitrate == 64000
   SetSendParameters(parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(0, gcodec.pltype);
-    EXPECT_STREQ("PCMU", gcodec.plname);
-    EXPECT_EQ(64000, gcodec.rate);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(0, spec.payload_type);
+    EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str());
+    EXPECT_EQ(64000, spec.target_bitrate_bps);
   }
 
   parameters.codecs[0].bitrate = 0;         // bitrate == default
   SetSendParameters(parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(0, gcodec.pltype);
-    EXPECT_STREQ("PCMU", gcodec.plname);
-    EXPECT_EQ(64000, gcodec.rate);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(0, spec.payload_type);
+    EXPECT_STREQ("PCMU", spec.format.name.c_str());
+    EXPECT_EQ(64000, spec.target_bitrate_bps);
   }
 
   parameters.codecs[0] = kOpusCodec;
   parameters.codecs[0].bitrate = 0;         // bitrate == default
   SetSendParameters(parameters);
   {
-    const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-    EXPECT_EQ(111, gcodec.pltype);
-    EXPECT_STREQ("opus", gcodec.plname);
-    EXPECT_EQ(32000, gcodec.rate);
+    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_EQ(111, spec.payload_type);
+    EXPECT_STREQ("opus", spec.format.name.c_str());
+    EXPECT_EQ(32000, spec.target_bitrate_bps);
   }
 }
 
-// Test that we could set packet size specified in kCodecParamPTime.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsPTimeAsPacketSize) {
-  EXPECT_TRUE(SetupSendStream());
-  cricket::AudioSendParameters parameters;
-  parameters.codecs.push_back(kOpusCodec);
-  parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 40); // Within range.
-  SetSendParameters(parameters);
-  EXPECT_EQ(1920, GetCodecPacSize(kSsrcX));  // Opus gets 40ms.
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 5); // Below range.
-  SetSendParameters(parameters);
-  EXPECT_EQ(480, GetCodecPacSize(kSsrcX));  // Opus gets 10ms.
-
-  parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 80); // Beyond range.
-  SetSendParameters(parameters);
-  EXPECT_EQ(2880, GetCodecPacSize(kSsrcX));  // Opus gets 60ms.
-
-  parameters.codecs[0] = kIsacCodec;  // Also try Isac, with unsupported size.
-  parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 40); // Within range.
-  SetSendParameters(parameters);
-  EXPECT_EQ(480, GetCodecPacSize(
-                     kSsrcX));  // Isac gets 30ms as the next smallest value.
-
-  parameters.codecs[0] = kG722CodecSdp;  // Try G722 @8kHz as negotiated in SDP.
-  parameters.codecs[0].SetParam(cricket::kCodecParamPTime, 40);
-  SetSendParameters(parameters);
-  EXPECT_EQ(640, GetCodecPacSize(
-                     kSsrcX));  // G722 gets 40ms @16kHz as defined in VoE.
-}
-
 // Test that we fail if no codecs are specified.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) {
   EXPECT_TRUE(SetupSendStream());
@@ -2073,9 +1690,9 @@
   parameters.codecs[0].id = 98;  // DTMF
   parameters.codecs[1].id = 96;
   SetSendParameters(parameters);
-  const auto& gcodec = GetSendStreamConfig(kSsrcX).send_codec_spec.codec_inst;
-  EXPECT_EQ(96, gcodec.pltype);
-  EXPECT_STREQ("ISAC", gcodec.plname);
+  const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+  EXPECT_EQ(96, spec.payload_type);
+  EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
   EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
@@ -2111,11 +1728,10 @@
   parameters.codecs[0].id = 98;  // wideband CN
   parameters.codecs[1].id = 96;
   SetSendParameters(parameters);
-  const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-  EXPECT_EQ(96, send_codec_spec.codec_inst.pltype);
-  EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
+  const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+  EXPECT_EQ(96, send_codec_spec.payload_type);
+  EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
   EXPECT_EQ(98, send_codec_spec.cng_payload_type);
-  EXPECT_EQ(webrtc::kFreq16000Hz, send_codec_spec.cng_plfreq);
 }
 
 // Test that we set VAD and DTMF types correctly as caller.
@@ -2132,13 +1748,11 @@
   parameters.codecs[2].id = 97;  // wideband CN
   parameters.codecs[4].id = 98;  // DTMF
   SetSendParameters(parameters);
-  const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-  EXPECT_EQ(96, send_codec_spec.codec_inst.pltype);
-  EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
-  EXPECT_EQ(1, send_codec_spec.codec_inst.channels);
-  EXPECT_EQ(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
+  const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+  EXPECT_EQ(96, send_codec_spec.payload_type);
+  EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
+  EXPECT_EQ(1, send_codec_spec.format.num_channels);
   EXPECT_EQ(97, send_codec_spec.cng_payload_type);
-  EXPECT_EQ(webrtc::kFreq16000Hz, send_codec_spec.cng_plfreq);
   EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
@@ -2159,13 +1773,11 @@
   EXPECT_TRUE(channel_->AddSendStream(
       cricket::StreamParams::CreateLegacy(kSsrcX)));
 
-  const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-  EXPECT_EQ(96, send_codec_spec.codec_inst.pltype);
-  EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
-  EXPECT_EQ(1, send_codec_spec.codec_inst.channels);
-  EXPECT_EQ(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
+  const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+  EXPECT_EQ(96, send_codec_spec.payload_type);
+  EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
+  EXPECT_EQ(1, send_codec_spec.format.num_channels);
   EXPECT_EQ(97, send_codec_spec.cng_payload_type);
-  EXPECT_EQ(webrtc::kFreq16000Hz, send_codec_spec.cng_plfreq);
   EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
@@ -2180,39 +1792,35 @@
   parameters.codecs[1].id = 97;
   SetSendParameters(parameters);
   {
-    const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-    EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
-    EXPECT_EQ(1, send_codec_spec.codec_inst.channels);
-    EXPECT_EQ(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
+    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
+    EXPECT_EQ(1, send_codec_spec.format.num_channels);
     EXPECT_EQ(97, send_codec_spec.cng_payload_type);
-    EXPECT_EQ(webrtc::kFreq16000Hz, send_codec_spec.cng_plfreq);
   }
   // Set PCMU(8K) and CN(16K). VAD should not be activated.
   parameters.codecs[0] = kPcmuCodec;
   SetSendParameters(parameters);
   {
-    const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-    EXPECT_STREQ("PCMU", send_codec_spec.codec_inst.plname);
-    EXPECT_NE(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
+    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
+    EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type);
   }
   // Set PCMU(8K) and CN(8K). VAD should be activated.
   parameters.codecs[1] = kCn8000Codec;
   SetSendParameters(parameters);
   {
-    const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-    EXPECT_STREQ("PCMU", send_codec_spec.codec_inst.plname);
-    EXPECT_EQ(1, send_codec_spec.codec_inst.channels);
-    EXPECT_EQ(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
+    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
+    EXPECT_EQ(1, send_codec_spec.format.num_channels);
     EXPECT_EQ(13, send_codec_spec.cng_payload_type);
-    EXPECT_EQ(webrtc::kFreq8000Hz, send_codec_spec.cng_plfreq);
   }
   // Set ISAC(16K) and CN(8K). VAD should not be activated.
   parameters.codecs[0] = kIsacCodec;
   SetSendParameters(parameters);
   {
-    const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-    EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
-    EXPECT_NE(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
+    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
+    EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type);
   }
 }
 
@@ -2230,13 +1838,11 @@
   parameters.codecs[2].id = 97;  // wideband CN
   parameters.codecs[4].id = 98;  // DTMF
   SetSendParameters(parameters);
-  const auto& send_codec_spec = GetSendStreamConfig(kSsrcX).send_codec_spec;
-  EXPECT_EQ(96, send_codec_spec.codec_inst.pltype);
-  EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
-  EXPECT_EQ(1, send_codec_spec.codec_inst.channels);
-  EXPECT_EQ(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
+  const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
+  EXPECT_EQ(96, send_codec_spec.payload_type);
+  EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
+  EXPECT_EQ(1, send_codec_spec.format.num_channels);
   EXPECT_EQ(97, send_codec_spec.cng_payload_type);
-  EXPECT_EQ(webrtc::kFreq16000Hz, send_codec_spec.cng_plfreq);
   EXPECT_TRUE(channel_->CanInsertDtmf());
 }
 
@@ -2394,12 +2000,10 @@
   for (uint32_t ssrc : kSsrcs4) {
     ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr);
     const auto& send_codec_spec =
-        call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
-    EXPECT_STREQ("ISAC", send_codec_spec.codec_inst.plname);
-    EXPECT_EQ(send_codec_spec.codec_inst.plfreq, send_codec_spec.cng_plfreq);
-    EXPECT_EQ(1, send_codec_spec.codec_inst.channels);
+        *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
+    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
+    EXPECT_EQ(1, send_codec_spec.format.num_channels);
     EXPECT_EQ(97, send_codec_spec.cng_payload_type);
-    EXPECT_EQ(webrtc::kFreq16000Hz, send_codec_spec.cng_plfreq);
   }
 
   // Change to PCMU(8K) and CN(16K).
@@ -2408,9 +2012,9 @@
   for (uint32_t ssrc : kSsrcs4) {
     ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr);
     const auto& send_codec_spec =
-        call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
-    EXPECT_STREQ("PCMU", send_codec_spec.codec_inst.plname);
-    EXPECT_EQ(-1, send_codec_spec.cng_payload_type);
+        *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
+    EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
+    EXPECT_EQ(rtc::Optional<int>(), send_codec_spec.cng_payload_type);
   }
 }
 
@@ -2606,12 +2210,9 @@
   SetSendParameters(send_parameters_);
   EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
             GetAudioNetworkAdaptorConfig(kSsrcX));
-  const int initial_num = call_.GetNumCreatedSendStreams();
   cricket::AudioOptions options;
   options.audio_network_adaptor = rtc::Optional<bool>(false);
   SetAudioSend(kSsrcX, true, nullptr, &options);
-  // AudioSendStream expected to be recreated.
-  EXPECT_EQ(initial_num + 1, call_.GetNumCreatedSendStreams());
   EXPECT_EQ(rtc::Optional<std::string>(), GetAudioNetworkAdaptorConfig(kSsrcX));
 }
 
@@ -3684,95 +3285,39 @@
   }
 }
 
-// Tests that the library is configured with the codecs we want.
-TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
-  // TODO(ossu): These tests should move into a future "builtin audio codecs"
-  // module.
-
-  // Check codecs by name.
-#ifdef WEBRTC_CODEC_OPUS
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "OPUS", 48000, 0, 2), nullptr));
-#endif
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "ISAC", 16000, 0, 1), nullptr));
-#endif
-#if (defined(WEBRTC_CODEC_ISAC))
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "ISAC", 32000, 0, 1), nullptr));
-#endif
-#ifdef WEBRTC_CODEC_ILBC
-  // Check that name matching is case-insensitive.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "ILBC", 8000, 0, 1), nullptr));
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "iLBC", 8000, 0, 1), nullptr));
-#endif
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "CN", 32000, 0, 1), nullptr));
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "CN", 16000, 0, 1), nullptr));
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "telephone-event", 8000, 0, 1), nullptr));
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "telephone-event", 16000, 0, 1), nullptr));
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "telephone-event", 32000, 0, 1), nullptr));
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(96, "telephone-event", 48000, 0, 1), nullptr));
-  // Check codecs with an id by id.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 8000, 0, 1), nullptr));  // PCMU
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(8, "", 8000, 0, 1), nullptr));  // PCMA
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(9, "", 8000, 0, 1), nullptr));  // G722
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(13, "", 8000, 0, 1), nullptr));  // CN
-  // Check sample/bitrate matching.
-  EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "PCMU", 8000, 64000, 1), nullptr));
-  // Check that bad codecs fail.
-  EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(99, "ABCD", 0, 0, 1), nullptr));
-  EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(88, "", 0, 0, 1), nullptr));
-  EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 0, 0, 2), nullptr));
-  EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 5000, 0, 1), nullptr));
-  EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
-      cricket::AudioCodec(0, "", 0, 5000, 1), nullptr));
-
-  // Verify the payload id of common audio codecs, including CN, ISAC, and G722.
+// Verify the payload id of common audio codecs, including CN, ISAC, and G722.
+TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
   // TODO(ossu): Why are the payload types of codecs with non-static payload
   // type assignments checked here? It shouldn't really matter.
   cricket::WebRtcVoiceEngine engine(
       nullptr, webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr);
   for (const cricket::AudioCodec& codec : engine.send_codecs()) {
-    if (codec.name == "CN" && codec.clockrate == 16000) {
+    auto is_codec = [&codec](const char* name, int clockrate = 0) {
+      return STR_CASE_CMP(codec.name.c_str(), name) == 0 &&
+             (clockrate == 0 || codec.clockrate == clockrate);
+    };
+    if (is_codec("CN", 16000)) {
       EXPECT_EQ(105, codec.id);
-    } else if (codec.name == "CN" && codec.clockrate == 32000) {
+    } else if (is_codec("CN", 32000)) {
       EXPECT_EQ(106, codec.id);
-    } else if (codec.name == "ISAC" && codec.clockrate == 16000) {
+    } else if (is_codec("ISAC", 16000)) {
       EXPECT_EQ(103, codec.id);
-    } else if (codec.name == "ISAC" && codec.clockrate == 32000) {
+    } else if (is_codec("ISAC", 32000)) {
       EXPECT_EQ(104, codec.id);
-    } else if (codec.name == "G722" && codec.clockrate == 8000) {
+    } else if (is_codec("G722", 8000)) {
       EXPECT_EQ(9, codec.id);
-    } else if (codec.name == "telephone-event" && codec.clockrate == 8000) {
+    } else if (is_codec("telephone-event", 8000)) {
       EXPECT_EQ(126, codec.id);
     // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned.
     // Remove these checks once both send and receive side assigns payload types
     // dynamically.
-    } else if (codec.name == "telephone-event" && codec.clockrate == 16000) {
+    } else if (is_codec("telephone-event", 16000)) {
       EXPECT_EQ(113, codec.id);
-    } else if (codec.name == "telephone-event" && codec.clockrate == 32000) {
+    } else if (is_codec("telephone-event", 32000)) {
       EXPECT_EQ(112, codec.id);
-    } else if (codec.name == "telephone-event" && codec.clockrate == 48000) {
+    } else if (is_codec("telephone-event", 48000)) {
       EXPECT_EQ(110, codec.id);
-    } else if (codec.name == "opus") {
+    } else if (is_codec("opus")) {
       EXPECT_EQ(111, codec.id);
       ASSERT_TRUE(codec.params.find("minptime") != codec.params.end());
       EXPECT_EQ("10", codec.params.find("minptime")->second);
diff --git a/modules/audio_coding/codecs/mock/mock_audio_encoder.h b/modules/audio_coding/codecs/mock/mock_audio_encoder.h
index 4c1b5f7..279db37 100644
--- a/modules/audio_coding/codecs/mock/mock_audio_encoder.h
+++ b/modules/audio_coding/codecs/mock/mock_audio_encoder.h
@@ -47,6 +47,10 @@
   MOCK_METHOD1(OnReceivedUplinkPacketLossFraction,
                void(float uplink_packet_loss_fraction));
 
+  MOCK_METHOD2(EnableAudioNetworkAdaptor,
+               bool(const std::string& config_string,
+                    RtcEventLog* event_log));
+
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD3(EncodeImpl,
                EncodedInfo(uint32_t timestamp,
diff --git a/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h b/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h
new file mode 100644
index 0000000..1b13a18
--- /dev/null
+++ b/modules/audio_coding/codecs/mock/mock_audio_encoder_factory.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h"
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioEncoderFactory : public AudioEncoderFactory {
+ public:
+  MOCK_METHOD0(GetSupportedEncoders, std::vector<AudioCodecSpec>());
+  MOCK_METHOD1(QueryAudioEncoder,
+               rtc::Optional<AudioCodecInfo>(const SdpAudioFormat& format));
+
+  std::unique_ptr<AudioEncoder> MakeAudioEncoder(int payload_type,
+                                                 const SdpAudioFormat& format) {
+    std::unique_ptr<AudioEncoder> return_value;
+    MakeAudioEncoderMock(payload_type, format, &return_value);
+    return return_value;
+  }
+  MOCK_METHOD3(MakeAudioEncoderMock,
+               void(int payload_type,
+                    const SdpAudioFormat& format,
+                    std::unique_ptr<AudioEncoder>* return_value));
+
+  // Creates a MockAudioEncoderFactory with no formats and that may not be
+  // invoked to create a codec - useful for initializing a voice engine, for
+  // example.
+  static rtc::scoped_refptr<webrtc::MockAudioEncoderFactory>
+  CreateUnusedFactory() {
+    using testing::_;
+    using testing::AnyNumber;
+    using testing::Return;
+
+    rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> factory =
+        new rtc::RefCountedObject<webrtc::MockAudioEncoderFactory>;
+    ON_CALL(*factory.get(), GetSupportedEncoders())
+        .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
+    ON_CALL(*factory.get(), QueryAudioEncoder(_))
+        .WillByDefault(Return(rtc::Optional<AudioCodecInfo>()));
+
+    EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _)).Times(0);
+    return factory;
+  }
+
+  // Creates a MockAudioEncoderFactory with no formats that may be invoked to
+  // create a codec any number of times. It will, though, return nullptr on each
+  // call, since it supports no codecs.
+  static rtc::scoped_refptr<webrtc::MockAudioEncoderFactory>
+  CreateEmptyFactory() {
+    using testing::_;
+    using testing::AnyNumber;
+    using testing::Return;
+    using testing::SetArgPointee;
+
+    rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> factory =
+        new rtc::RefCountedObject<webrtc::MockAudioEncoderFactory>;
+    ON_CALL(*factory.get(), GetSupportedEncoders())
+        .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
+    ON_CALL(*factory.get(), QueryAudioEncoder(_))
+        .WillByDefault(Return(rtc::Optional<AudioCodecInfo>()));
+    ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
+        .WillByDefault(SetArgPointee<2>(nullptr));
+
+    EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
+        .Times(AnyNumber());
+    return factory;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_FACTORY_H_
diff --git a/test/BUILD.gn b/test/BUILD.gn
index d7c1f8a..9f40c09 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -407,6 +407,7 @@
     "../audio",
     "../base:rtc_base_approved",
     "../call",
+    "../modules/audio_coding:builtin_audio_encoder_factory",
     "../modules/audio_mixer:audio_mixer_impl",
     "../modules/audio_processing",
     "../video",
diff --git a/test/call_test.cc b/test/call_test.cc
index 6ec3fda..f5850e5 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -15,6 +15,7 @@
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
@@ -38,6 +39,7 @@
       num_audio_streams_(0),
       num_flexfec_streams_(0),
       decoder_factory_(CreateBuiltinAudioDecoderFactory()),
+      encoder_factory_(CreateBuiltinAudioEncoderFactory()),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
 
@@ -222,8 +224,10 @@
     audio_send_config_ = AudioSendStream::Config(send_transport);
     audio_send_config_.voe_channel_id = voe_send_.channel_id;
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
-    audio_send_config_.send_codec_spec.codec_inst =
-        CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
+    audio_send_config_.send_codec_spec =
+        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+            {kAudioSendPayloadType, {"OPUS", 48000, 2, {{"stereo", "1"}}}});
+    audio_send_config_.encoder_factory = encoder_factory_;
   }
 
   // TODO(brandtr): Update this when we support multistream protection.
diff --git a/test/call_test.h b/test/call_test.h
index b9523f5..39df343 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -123,6 +123,7 @@
   size_t num_audio_streams_;
   size_t num_flexfec_streams_;
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
   test::FakeVideoRenderer fake_renderer_;
 
  private:
diff --git a/test/mock_voe_channel_proxy.h b/test/mock_voe_channel_proxy.h
index 26645d4..bd6bb4c 100644
--- a/test/mock_voe_channel_proxy.h
+++ b/test/mock_voe_channel_proxy.h
@@ -30,6 +30,9 @@
   MOCK_METHOD2(SetEncoderForMock,
                bool(int payload_type,
                     std::unique_ptr<AudioEncoder>* encoder));
+  MOCK_METHOD1(
+      ModifyEncoder,
+      void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
   MOCK_METHOD1(SetRTCPStatus, void(bool enable));
   MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
   MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
@@ -68,11 +71,6 @@
   MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
   MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
   MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
-  MOCK_METHOD1(EnableAudioNetworkAdaptor,
-               void(const std::string& config_string));
-  MOCK_METHOD0(DisableAudioNetworkAdaptor, void());
-  MOCK_METHOD2(SetReceiverFrameLengthRange,
-               void(int min_frame_length_ms, int max_frame_length_ms));
   MOCK_METHOD2(GetAudioFrameWithInfo,
       AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
                                          AudioFrame* audio_frame));
@@ -86,14 +84,6 @@
   MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
   MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
   MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
-  MOCK_CONST_METHOD1(GetSendCodec, bool(CodecInst* codec_inst));
-  MOCK_METHOD1(SetVADStatus, bool(bool enable));
-  MOCK_METHOD1(SetCodecFECStatus, bool(bool enable));
-  MOCK_METHOD1(SetOpusDtx, bool(bool enable));
-  MOCK_METHOD1(SetOpusMaxPlaybackRate, bool(int frequency_hz));
-  MOCK_METHOD1(SetSendCodec, bool(const CodecInst& codec_inst));
-  MOCK_METHOD2(SetSendCNPayloadType,
-               bool(int type, PayloadFrequencies frequency));
   MOCK_METHOD1(SetReceiveCodecs,
                void(const std::map<int, SdpAudioFormat>& codecs));
   MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc
index 565506c..508ba13 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -1728,9 +1728,13 @@
     audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
     audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
   }
-  audio_send_config_.send_codec_spec.codec_inst =
-      CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
-  audio_send_config_.send_codec_spec.enable_opus_dtx = params_.audio.dtx;
+  audio_send_config_.send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+          {kAudioSendPayloadType,
+           {"OPUS", 48000, 2,
+            {{"usedtx", (params_.audio.dtx ? "1" : "0")},
+              {"stereo", "1"}}}});
+
   audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
 
   AudioReceiveStream::Config audio_config;
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 8c9b001..f56d1d8 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -1303,9 +1303,15 @@
   }
 
   audio_coding_->SetEncoder(std::move(encoder));
+  codec_manager_.UnsetCodecInst();
   return true;
 }
 
+void Channel::ModifyEncoder(
+    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
+  audio_coding_->ModifyEncoder(modifier);
+}
+
 int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterVoiceEngineObserver()");
@@ -1337,9 +1343,16 @@
 }
 
 int32_t Channel::GetSendCodec(CodecInst& codec) {
-  auto send_codec = codec_manager_.GetCodecInst();
-  if (send_codec) {
-    codec = *send_codec;
+  {
+    const CodecInst* send_codec = codec_manager_.GetCodecInst();
+    if (send_codec) {
+      codec = *send_codec;
+      return 0;
+    }
+  }
+  rtc::Optional<CodecInst> acm_send_codec = audio_coding_->SendCodec();
+  if (acm_send_codec) {
+    codec = *acm_send_codec;
     return 0;
   }
   return -1;
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index f114d72..d2d91ce 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -175,6 +175,8 @@
 
   // Send using this encoder, with this payload type.
   bool SetEncoder(int payload_type, std::unique_ptr<AudioEncoder> encoder);
+  void ModifyEncoder(
+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
 
   // API methods
 
diff --git a/voice_engine/channel_proxy.cc b/voice_engine/channel_proxy.cc
index 74234a2..25978c7 100644
--- a/voice_engine/channel_proxy.cc
+++ b/voice_engine/channel_proxy.cc
@@ -36,6 +36,12 @@
   return channel()->SetEncoder(payload_type, std::move(encoder));
 }
 
+void ChannelProxy::ModifyEncoder(
+    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  channel()->ModifyEncoder(modifier);
+}
+
 void ChannelProxy::SetRTCPStatus(bool enable) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   channel()->SetRTCPStatus(enable);
@@ -235,24 +241,6 @@
   channel()->SetRtcEventLog(event_log);
 }
 
-void ChannelProxy::EnableAudioNetworkAdaptor(const std::string& config_string) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  bool ret = channel()->EnableAudioNetworkAdaptor(config_string);
-  RTC_DCHECK(ret);
-;}
-
-void ChannelProxy::DisableAudioNetworkAdaptor() {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  channel()->DisableAudioNetworkAdaptor();
-}
-
-void ChannelProxy::SetReceiverFrameLengthRange(int min_frame_length_ms,
-                                               int max_frame_length_ms) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  channel()->SetReceiverFrameLengthRange(min_frame_length_ms,
-                                         max_frame_length_ms);
-}
-
 AudioMixer::Source::AudioFrameInfo ChannelProxy::GetAudioFrameWithInfo(
     int sample_rate_hz,
     AudioFrame* audio_frame) {
@@ -319,69 +307,6 @@
   return channel()->GetRecCodec(*codec_inst) == 0;
 }
 
-bool ChannelProxy::GetSendCodec(CodecInst* codec_inst) const {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return channel()->GetSendCodec(*codec_inst) == 0;
-}
-
-bool ChannelProxy::SetVADStatus(bool enable) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return channel()->SetVADStatus(enable, VADNormal, false) == 0;
-}
-
-bool ChannelProxy::SetCodecFECStatus(bool enable) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return channel()->SetCodecFECStatus(enable) == 0;
-}
-
-bool ChannelProxy::SetOpusDtx(bool enable) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return channel()->SetOpusDtx(enable) == 0;
-}
-
-bool ChannelProxy::SetOpusMaxPlaybackRate(int frequency_hz) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return channel()->SetOpusMaxPlaybackRate(frequency_hz) == 0;
-}
-
-bool ChannelProxy::SetSendCodec(const CodecInst& codec_inst) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  // Validation code copied from VoECodecImpl::SetSendCodec().
-  if ((STR_CASE_CMP(codec_inst.plname, "L16") == 0) &&
-      (codec_inst.pacsize >= 960)) {
-    return false;
-  }
-  if (!STR_CASE_CMP(codec_inst.plname, "CN") ||
-      !STR_CASE_CMP(codec_inst.plname, "TELEPHONE-EVENT") ||
-      !STR_CASE_CMP(codec_inst.plname, "RED")) {
-    return false;
-  }
-  if ((codec_inst.channels != 1) && (codec_inst.channels != 2)) {
-    return false;
-  }
-  if (!AudioCodingModule::IsCodecValid(codec_inst)) {
-    return false;
-  }
-  return channel()->SetSendCodec(codec_inst) == 0;
-}
-
-bool ChannelProxy::SetSendCNPayloadType(int type,
-                                        PayloadFrequencies frequency) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  // Validation code copied from VoECodecImpl::SetSendCNPayloadType().
-  if (type < 96 || type > 127) {
-    // Only allow dynamic range: 96 to 127
-    return false;
-  }
-  if ((frequency != kFreq16000Hz) && (frequency != kFreq32000Hz)) {
-    // It is not possible to modify the payload type for CN/8000.
-    // We only allow modification of the CN payload type for CN/16000
-    // and CN/32000.
-    return false;
-  }
-  return channel()->SetSendCNPayloadType(type, frequency) == 0;
-}
-
 void ChannelProxy::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
   // TODO(elad.alon): This fails in UT; fix and uncomment.
   // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7405
diff --git a/voice_engine/channel_proxy.h b/voice_engine/channel_proxy.h
index 829b094..50c1e5f 100644
--- a/voice_engine/channel_proxy.h
+++ b/voice_engine/channel_proxy.h
@@ -58,6 +58,8 @@
 
   virtual bool SetEncoder(int payload_type,
                           std::unique_ptr<AudioEncoder> encoder);
+  virtual void ModifyEncoder(
+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
 
   virtual void SetRTCPStatus(bool enable);
   virtual void SetLocalSSRC(uint32_t ssrc);
@@ -98,10 +100,6 @@
       GetAudioDecoderFactory() const;
   virtual void SetChannelOutputVolumeScaling(float scaling);
   virtual void SetRtcEventLog(RtcEventLog* event_log);
-  virtual void EnableAudioNetworkAdaptor(const std::string& config_string);
-  virtual void DisableAudioNetworkAdaptor();
-  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
-                                           int max_frame_length_ms);
   virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
       int sample_rate_hz,
       AudioFrame* audio_frame);
@@ -115,13 +113,6 @@
   virtual void SetMinimumPlayoutDelay(int delay_ms);
   virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
   virtual bool GetRecCodec(CodecInst* codec_inst) const;
-  virtual bool GetSendCodec(CodecInst* codec_inst) const;
-  virtual bool SetVADStatus(bool enable);
-  virtual bool SetCodecFECStatus(bool enable);
-  virtual bool SetOpusDtx(bool enable);
-  virtual bool SetOpusMaxPlaybackRate(int frequency_hz);
-  virtual bool SetSendCodec(const CodecInst& codec_inst);
-  virtual bool SetSendCNPayloadType(int type, PayloadFrequencies frequency);
   virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
   virtual void OnRecoverableUplinkPacketLossRate(
       float recoverable_packet_loss_rate);