diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
index 257c018..f78e814 100644
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ b/talk/media/webrtc/fakewebrtccall.cc
@@ -30,6 +30,7 @@
 #include <algorithm>
 
 #include "talk/media/base/rtputils.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 
 namespace cricket {
@@ -54,7 +55,7 @@
       config_(config),
       codec_settings_set_(false),
       num_swapped_frames_(0) {
-  assert(config.encoder_settings.encoder != NULL);
+  DCHECK(config.encoder_settings.encoder != NULL);
   ReconfigureVideoEncoder(encoder_config);
 }
 
diff --git a/talk/media/webrtc/fakewebrtccommon.h b/talk/media/webrtc/fakewebrtccommon.h
index 96fff42..4281528 100644
--- a/talk/media/webrtc/fakewebrtccommon.h
+++ b/talk/media/webrtc/fakewebrtccommon.h
@@ -54,12 +54,6 @@
 #define WEBRTC_BOOL_FUNC(method, args) bool method args override
 
 #define WEBRTC_VOID_FUNC(method, args) void method args override
-
-#define WEBRTC_CHECK_CHANNEL(channel) \
-  if (channels_.find(channel) == channels_.end()) return -1;
-
-#define WEBRTC_ASSERT_CHANNEL(channel) \
-  ASSERT(channels_.find(channel) != channels_.end());
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCCOMMON_H_
diff --git a/talk/media/webrtc/fakewebrtcvideoengine.h b/talk/media/webrtc/fakewebrtcvideoengine.h
index bfe22c4..f8ede9a 100644
--- a/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -45,12 +45,6 @@
 
 namespace cricket {
 
-#define WEBRTC_CHECK_CAPTURER(capturer) \
-  if (capturers_.find(capturer) == capturers_.end()) return -1;
-
-#define WEBRTC_ASSERT_CAPTURER(capturer) \
-  ASSERT(capturers_.find(capturer) != capturers_.end());
-
 static const int kMinVideoBitrate = 100;
 static const int kStartVideoBitrate = 300;
 static const int kMaxVideoBitrate = 1000;
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index dba7d63..24ef846 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -38,6 +38,7 @@
 #include "talk/media/webrtc/fakewebrtccommon.h"
 #include "talk/media/webrtc/webrtcvoe.h"
 #include "webrtc/base/basictypes.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ -83,6 +84,12 @@
     7654,  // int addedSamples;
 };  // These random but non-trivial numbers are used for testing.
 
+#define WEBRTC_CHECK_CHANNEL(channel) \
+  if (channels_.find(channel) == channels_.end()) return -1;
+
+#define WEBRTC_ASSERT_CHANNEL(channel) \
+  DCHECK(channels_.find(channel) != channels_.end());
+
 // Verify the header extension ID, if enabled, is within the bounds specified in
 // [RFC5285]: 1-14 inclusive.
 #define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \
@@ -355,7 +362,7 @@
     return channels_[channel]->packets.empty();
   }
   void TriggerCallbackOnError(int channel_num, int err_code) {
-    ASSERT(observer_ != NULL);
+    DCHECK(observer_ != NULL);
     observer_->CallbackOnError(channel_num, err_code);
   }
   void set_playout_fail_channel(int channel) {
diff --git a/talk/media/webrtc/webrtcvideocapturer.cc b/talk/media/webrtc/webrtcvideocapturer.cc
index bcaeb89..285ed01 100644
--- a/talk/media/webrtc/webrtcvideocapturer.cc
+++ b/talk/media/webrtc/webrtcvideocapturer.cc
@@ -264,7 +264,7 @@
   // Can't take lock here as this will cause deadlock with
   // OnIncomingCapturedFrame. In fact, the whole method, including methods it
   // calls, can't take lock.
-  assert(module_);
+  DCHECK(module_);
 
   const std::string group_name =
       webrtc::field_trial::FindFullName("WebRTC-CVO");
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index fbb5fc2..4417e19 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -52,7 +52,7 @@
 
 #define UNIMPLEMENTED                                                 \
   LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \
-  ASSERT(false)
+  RTC_NOTREACHED()
 
 namespace cricket {
 namespace {
@@ -105,7 +105,7 @@
 
   webrtc::VideoEncoder* CreateVideoEncoder(
       webrtc::VideoCodecType type) override {
-    ASSERT(factory_ != NULL);
+    DCHECK(factory_ != NULL);
     // If it's a codec type we can simulcast, create a wrapped encoder.
     if (type == webrtc::kVideoCodecVP8) {
       return new webrtc::SimulcastEncoderAdapter(
@@ -558,14 +558,14 @@
 }
 
 void WebRtcVideoEngine2::SetCallFactory(WebRtcCallFactory* call_factory) {
-  assert(!initialized_);
+  DCHECK(!initialized_);
   call_factory_ = call_factory;
 }
 
 bool WebRtcVideoEngine2::Init(rtc::Thread* worker_thread) {
   LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
   worker_thread_ = worker_thread;
-  ASSERT(worker_thread_ != NULL);
+  DCHECK(worker_thread_ != NULL);
 
   initialized_ = true;
   return true;
@@ -605,7 +605,7 @@
 WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
     const VideoOptions& options,
     VoiceMediaChannel* voice_channel) {
-  assert(initialized_);
+  DCHECK(initialized_);
   LOG(LS_INFO) << "CreateChannel: "
                << (voice_channel != NULL ? "With" : "Without")
                << " voice channel. Options: " << options.ToString();
@@ -635,20 +635,20 @@
   LOG(LS_VERBOSE) << "SetLogging: " << min_sev << '"' << filter << '"';
   // if min_sev == -1, we keep the current log level.
   if (min_sev < 0) {
-    assert(min_sev == -1);
+    DCHECK(min_sev == -1);
     return;
   }
 }
 
 void WebRtcVideoEngine2::SetExternalDecoderFactory(
     WebRtcVideoDecoderFactory* decoder_factory) {
-  assert(!initialized_);
+  DCHECK(!initialized_);
   external_decoder_factory_ = decoder_factory;
 }
 
 void WebRtcVideoEngine2::SetExternalEncoderFactory(
     WebRtcVideoEncoderFactory* encoder_factory) {
-  assert(!initialized_);
+  DCHECK(!initialized_);
   if (external_encoder_factory_ == encoder_factory)
     return;
 
@@ -694,7 +694,7 @@
 bool WebRtcVideoEngine2::CanSendCodec(const VideoCodec& requested,
                                       const VideoCodec& current,
                                       VideoCodec* out) {
-  assert(out != NULL);
+  DCHECK(out != NULL);
 
   if (requested.width != requested.height &&
       (requested.height == 0 || requested.width == 0)) {
@@ -760,7 +760,7 @@
     // we only support up to 8 external payload types.
     const int kExternalVideoPayloadTypeBase = 120;
     size_t payload_type = kExternalVideoPayloadTypeBase + i;
-    assert(payload_type < 128);
+    DCHECK(payload_type < 128);
     VideoCodec codec(static_cast<int>(payload_type),
                      codecs[i].name,
                      codecs[i].max_width,
@@ -941,7 +941,7 @@
            send_streams_.begin();
        it != send_streams_.end();
        ++it) {
-    assert(it->second != NULL);
+    DCHECK(it->second != NULL);
     it->second->SetCodec(supported_codecs.front());
   }
 
@@ -1061,7 +1061,7 @@
                                 send_rtp_extensions_);
 
   uint32 ssrc = sp.first_ssrc();
-  assert(ssrc != 0);
+  DCHECK(ssrc != 0);
   send_streams_[ssrc] = stream;
 
   if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
@@ -1136,7 +1136,7 @@
     return false;
 
   uint32 ssrc = sp.first_ssrc();
-  assert(ssrc != 0);  // TODO(pbos): Is this ever valid?
+  DCHECK(ssrc != 0);  // TODO(pbos): Is this ever valid?
 
   rtc::CritScope stream_lock(&stream_crit_);
   // Remove running stream if this was a default stream.
@@ -1326,7 +1326,7 @@
 bool WebRtcVideoChannel2::SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
   LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
                << (capturer != NULL ? "(capturer)" : "NULL");
-  assert(ssrc != 0);
+  DCHECK(ssrc != 0);
   {
     rtc::CritScope stream_lock(&stream_crit_);
     if (send_streams_.find(ssrc) == send_streams_.end()) {
@@ -1419,7 +1419,7 @@
 bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
   LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
                   << (mute ? "mute" : "unmute");
-  assert(ssrc != 0);
+  DCHECK(ssrc != 0);
   rtc::CritScope stream_lock(&stream_crit_);
   if (send_streams_.find(ssrc) == send_streams_.end()) {
     LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
@@ -1700,7 +1700,7 @@
     return;
 
   if (format_.width == 0) {  // Dropping frames.
-    assert(format_.height == 0);
+    DCHECK(format_.height == 0);
     LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
     return;
   }
@@ -1870,7 +1870,7 @@
 
   // This shouldn't happen, we should not be trying to create something we don't
   // support.
-  assert(false);
+  DCHECK(false);
   return AllocatedEncoder(NULL, webrtc::kVideoCodecUnknown, false);
 }
 
@@ -2009,7 +2009,7 @@
   last_dimensions_.height = height;
   last_dimensions_.is_screencast = is_screencast;
 
-  assert(!parameters_.encoder_config.streams.empty());
+  DCHECK(!parameters_.encoder_config.streams.empty());
 
   VideoCodecSettings codec_settings;
   parameters_.codec_settings.Get(&codec_settings);
@@ -2035,7 +2035,7 @@
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() {
   rtc::CritScope cs(&lock_);
-  assert(stream_ != NULL);
+  DCHECK(stream_ != NULL);
   stream_->Start();
   sending_ = true;
 }
@@ -2261,7 +2261,7 @@
 
   // This shouldn't happen, we should not be trying to create something we don't
   // support.
-  assert(false);
+  DCHECK(false);
   return AllocatedDecoder(NULL, webrtc::kVideoCodecUnknown, false);
 }
 
@@ -2443,7 +2443,7 @@
 
 std::vector<WebRtcVideoChannel2::VideoCodecSettings>
 WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) {
-  assert(!codecs.empty());
+  DCHECK(!codecs.empty());
 
   std::vector<VideoCodecSettings> video_codecs;
   std::map<int, bool> payload_used;
@@ -2468,14 +2468,14 @@
     switch (in_codec.GetCodecType()) {
       case VideoCodec::CODEC_RED: {
         // RED payload type, should not have duplicates.
-        assert(fec_settings.red_payload_type == -1);
+        DCHECK(fec_settings.red_payload_type == -1);
         fec_settings.red_payload_type = in_codec.id;
         continue;
       }
 
       case VideoCodec::CODEC_ULPFEC: {
         // ULPFEC payload type, should not have duplicates.
-        assert(fec_settings.ulpfec_payload_type == -1);
+        DCHECK(fec_settings.ulpfec_payload_type == -1);
         fec_settings.ulpfec_payload_type = in_codec.id;
         continue;
       }
@@ -2504,7 +2504,7 @@
 
   // One of these codecs should have been a video codec. Only having FEC
   // parameters into this code is a logic error.
-  assert(!video_codecs.empty());
+  DCHECK(!video_codecs.empty());
 
   for (std::map<int, int>::const_iterator it = rtx_mapping.begin();
        it != rtx_mapping.end();
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index 6811155..86700bb 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -109,7 +109,7 @@
   WebRtcVideoEngine2Test(WebRtcVoiceEngine* voice_engine)
       : engine_(voice_engine) {
     std::vector<VideoCodec> engine_codecs = engine_.codecs();
-    assert(!engine_codecs.empty());
+    DCHECK(!engine_codecs.empty());
     bool codec_set = false;
     for (size_t i = 0; i < engine_codecs.size(); ++i) {
       if (engine_codecs[i].name == "red") {
@@ -128,7 +128,7 @@
       }
     }
 
-    assert(codec_set);
+    DCHECK(codec_set);
   }
 
  protected:
@@ -139,7 +139,7 @@
 
    private:
     webrtc::Call* CreateCall(const webrtc::Call::Config& config) override {
-      assert(fake_call_ == NULL);
+      DCHECK(fake_call_ == NULL);
       fake_call_ = new FakeCall(config);
       return fake_call_;
     }
@@ -835,7 +835,7 @@
   }
 
   webrtc::Call* CreateCall(const webrtc::Call::Config& config) override {
-    assert(fake_call_ == NULL);
+    DCHECK(fake_call_ == NULL);
     fake_call_ = new FakeCall(config);
     return fake_call_;
   }
@@ -2566,7 +2566,7 @@
 
  protected:
   webrtc::Call* CreateCall(const webrtc::Call::Config& config) override {
-    assert(fake_call_ == NULL);
+    DCHECK(fake_call_ == NULL);
     fake_call_ = new FakeCall(config);
     return fake_call_;
   }
@@ -2585,7 +2585,7 @@
     ASSERT_TRUE(channel_->SetSendCodecs(codecs));
 
     std::vector<uint32> ssrcs = MAKE_VECTOR(kSsrcs3);
-    assert(num_configured_streams <= ssrcs.size());
+    DCHECK(num_configured_streams <= ssrcs.size());
     ssrcs.resize(num_configured_streams);
 
     FakeVideoSendStream* stream =
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index d936c9a..baae3de 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -338,7 +338,7 @@
   if (IsCodec(*voe_codec, kG722CodecName)) {
     // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
     // has changed, and this special case is no longer needed.
-    ASSERT(voe_codec->plfreq != new_plfreq);
+    DCHECK(voe_codec->plfreq != new_plfreq);
     voe_codec->plfreq = new_plfreq;
   }
 }
@@ -600,14 +600,14 @@
   }
 
   // Test to see if the media processor was deregistered properly
-  ASSERT(SignalRxMediaFrame.is_empty());
-  ASSERT(SignalTxMediaFrame.is_empty());
+  DCHECK(SignalRxMediaFrame.is_empty());
+  DCHECK(SignalTxMediaFrame.is_empty());
 
   tracing_->SetTraceCallback(NULL);
 }
 
 bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
-  ASSERT(worker_thread == rtc::Thread::Current());
+  DCHECK(worker_thread == rtc::Thread::Current());
   LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
   bool res = InitInternal();
   if (res) {
@@ -1223,7 +1223,7 @@
 }
 
 bool WebRtcVoiceEngine::SetOutputVolume(int level) {
-  ASSERT(level >= 0 && level <= 255);
+  DCHECK(level >= 0 && level <= 255);
   if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
     LOG_RTCERR1(SetSpeakerVolume, level);
     return false;
@@ -1456,7 +1456,7 @@
   LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
                   << channel_num << ".";
   if (FindChannelAndSsrc(channel_num, &channel, &ssrc)) {
-    ASSERT(channel != NULL);
+    DCHECK(channel != NULL);
     channel->OnError(ssrc, err_code);
   } else {
     LOG(LS_ERROR) << "VoiceEngine channel " << channel_num
@@ -1466,14 +1466,14 @@
 
 bool WebRtcVoiceEngine::FindChannelAndSsrc(
     int channel_num, WebRtcVoiceMediaChannel** channel, uint32* ssrc) const {
-  ASSERT(channel != NULL && ssrc != NULL);
+  DCHECK(channel != NULL && ssrc != NULL);
 
   *channel = NULL;
   *ssrc = 0;
   // Find corresponding channel and ssrc
   for (ChannelList::const_iterator it = channels_.begin();
       it != channels_.end(); ++it) {
-    ASSERT(*it != NULL);
+    DCHECK(*it != NULL);
     if ((*it)->FindSsrc(channel_num, ssrc)) {
       *channel = *it;
       return true;
@@ -1487,14 +1487,14 @@
 // obtain the voice engine's channel number.
 bool WebRtcVoiceEngine::FindChannelNumFromSsrc(
     uint32 ssrc, MediaProcessorDirection direction, int* channel_num) {
-  ASSERT(channel_num != NULL);
-  ASSERT(direction == MPD_RX || direction == MPD_TX);
+  DCHECK(channel_num != NULL);
+  DCHECK(direction == MPD_RX || direction == MPD_TX);
 
   *channel_num = -1;
   // Find corresponding channel for ssrc.
   for (ChannelList::const_iterator it = channels_.begin();
       it != channels_.end(); ++it) {
-    ASSERT(*it != NULL);
+    DCHECK(*it != NULL);
     if (direction & MPD_RX) {
       *channel_num = (*it)->GetReceiveChannelNum(ssrc);
     }
@@ -1804,9 +1804,9 @@
   // TODO(xians): Make sure Start() is called only once.
   void Start(AudioRenderer* renderer) {
     rtc::CritScope lock(&lock_);
-    ASSERT(renderer != NULL);
+    DCHECK(renderer != NULL);
     if (renderer_ != NULL) {
-      ASSERT(renderer_ == renderer);
+      DCHECK(renderer_ == renderer);
       return;
     }
 
@@ -2575,7 +2575,7 @@
       return false;
     }
   } else {  // SEND_NOTHING
-    ASSERT(send == SEND_NOTHING);
+    DCHECK(send == SEND_NOTHING);
     if (engine()->voe()->base()->StopSend(channel) == -1) {
       LOG_RTCERR1(StopSend, channel);
       return false;
@@ -2866,7 +2866,7 @@
   receive_channels_.erase(it);
 
   if (ssrc == default_receive_ssrc_) {
-    ASSERT(IsDefaultChannel(channel));
+    DCHECK(IsDefaultChannel(channel));
     // Recycle the default channel is for recv stream.
     if (playout_)
       SetPlayout(voe_channel(), false);
@@ -3546,15 +3546,15 @@
 
 void WebRtcVoiceMediaChannel::GetLastMediaError(
     uint32* ssrc, VoiceMediaChannel::Error* error) {
-  ASSERT(ssrc != NULL);
-  ASSERT(error != NULL);
+  DCHECK(ssrc != NULL);
+  DCHECK(error != NULL);
   FindSsrc(voe_channel(), ssrc);
   *error = WebRtcErrorToChannelError(GetLastEngineError());
 }
 
 bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
   rtc::CritScope lock(&receive_channels_cs_);
-  ASSERT(ssrc != NULL);
+  DCHECK(ssrc != NULL);
   if (channel_num == -1 && send_ != SEND_NOTHING) {
     // Sometimes the VoiceEngine core will throw error with channel_num = -1.
     // This means the error is not limited to a specific channel.  Signal the
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 93d8c51..7737f34 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -102,7 +102,7 @@
    public:
     explicit ChannelErrorListener(cricket::VoiceMediaChannel* channel)
         : ssrc_(0), error_(cricket::VoiceMediaChannel::ERROR_NONE) {
-      ASSERT(channel != NULL);
+      DCHECK(channel != NULL);
       channel->SignalMediaError.connect(
           this, &ChannelErrorListener::OnVoiceChannelError);
     }