diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc index 257c01850e..f78e8146dd 100644 --- a/talk/media/webrtc/fakewebrtccall.cc +++ b/talk/media/webrtc/fakewebrtccall.cc @@ -30,6 +30,7 @@ #include #include "talk/media/base/rtputils.h" +#include "webrtc/base/checks.h" #include "webrtc/base/gunit.h" namespace cricket { @@ -54,7 +55,7 @@ FakeVideoSendStream::FakeVideoSendStream( 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 96fff4224c..4281528d26 100644 --- a/talk/media/webrtc/fakewebrtccommon.h +++ b/talk/media/webrtc/fakewebrtccommon.h @@ -54,12 +54,6 @@ namespace cricket { #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 bfe22c433a..f8ede9a987 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 dba7d634ad..24ef846058 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 @@ static const webrtc::NetworkStatistics kNetStats = { 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 @@ class FakeWebRtcVoiceEngine 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 bcaeb89ea4..285ed01966 100644 --- a/talk/media/webrtc/webrtcvideocapturer.cc +++ b/talk/media/webrtc/webrtcvideocapturer.cc @@ -264,7 +264,7 @@ bool WebRtcVideoCapturer::SetApplyRotation(bool enable) { // 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 fbb5fc212e..4417e19425 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 @@ class WebRtcSimulcastEncoderFactory 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 @@ WebRtcVideoEngine2::~WebRtcVideoEngine2() { } 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 @@ bool WebRtcVideoEngine2::SetDefaultEncoderConfig( 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 @@ void WebRtcVideoEngine2::SetLogging(int min_sev, const char* filter) { 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::FindCodec(const VideoCodec& in) { 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 @@ std::vector WebRtcVideoEngine2::GetSupportedCodecs() const { // 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(payload_type), codecs[i].name, codecs[i].max_width, @@ -941,7 +941,7 @@ bool WebRtcVideoChannel2::SetSendCodecs(const std::vector& codecs) { 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 @@ bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) { 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 @@ bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp, 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 @@ void WebRtcVideoChannel2::FillBandwidthEstimationStats( 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 @@ void WebRtcVideoChannel2::OnReadyToSend(bool ready) { 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 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( 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 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder( // 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 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions( 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::SetDimensions( void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() { rtc::CritScope cs(&lock_); - assert(stream_ != NULL); + DCHECK(stream_ != NULL); stream_->Start(); sending_ = true; } @@ -2261,7 +2261,7 @@ WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder( // 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 @@ bool WebRtcVideoChannel2::VideoCodecSettings::operator!=( std::vector WebRtcVideoChannel2::MapCodecs(const std::vector& codecs) { - assert(!codecs.empty()); + DCHECK(!codecs.empty()); std::vector video_codecs; std::map payload_used; @@ -2468,14 +2468,14 @@ WebRtcVideoChannel2::MapCodecs(const std::vector& codecs) { 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 @@ WebRtcVideoChannel2::MapCodecs(const std::vector& codecs) { // 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::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 6811155912..86700bbdce 100644 --- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc +++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc @@ -109,7 +109,7 @@ class WebRtcVideoEngine2Test : public ::testing::Test { WebRtcVideoEngine2Test(WebRtcVoiceEngine* voice_engine) : engine_(voice_engine) { std::vector 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 @@ class WebRtcVideoEngine2Test : public ::testing::Test { } } - assert(codec_set); + DCHECK(codec_set); } protected: @@ -139,7 +139,7 @@ class WebRtcVideoEngine2Test : public ::testing::Test { 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 @@ class WebRtcVideoChannel2Test : public WebRtcVideoEngine2Test, } 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 @@ class WebRtcVideoChannel2SimulcastTest : public WebRtcVideoEngine2SimulcastTest, 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 @@ class WebRtcVideoChannel2SimulcastTest : public WebRtcVideoEngine2SimulcastTest, ASSERT_TRUE(channel_->SetSendCodecs(codecs)); std::vector 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 d936c9a420..baae3de3ef 100644 --- a/talk/media/webrtc/webrtcvoiceengine.cc +++ b/talk/media/webrtc/webrtcvoiceengine.cc @@ -338,7 +338,7 @@ static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) { 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 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() { } // 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::GetOutputVolume(int* level) { } 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 @@ void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) { 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 @@ void WebRtcVoiceEngine::CallbackOnError(int channel_num, int err_code) { 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 @@ bool WebRtcVoiceEngine::FindChannelAndSsrc( // 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 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer // 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 @@ bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) { 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 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32 ssrc) { 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 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { 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 93d8c519e1..7737f3413a 100644 --- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc +++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc @@ -102,7 +102,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test { 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); }