Fixing some case-sensitive codec name comparisons.
As specified in RFC 4288, Section 4.2, and RFC 4855, Section 3, these names should be case-insensitive. They already were being treated as case-insensitive in some other places. This bug was resulting in either a crash or no decoded video, depending on the platform. BUG=webrtc:6439, webrtc:7027 Review-Url: https://codereview.webrtc.org/2782273002 Cr-Commit-Position: refs/heads/master@{#17515}
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@ -245,7 +245,7 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
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received_sdp_munger_ = munger;
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}
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// Siimlar to the above, but this is run on SDP immediately after it's
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// Similar to the above, but this is run on SDP immediately after it's
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// generated.
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void SetGeneratedSdpMunger(
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std::function<void(cricket::SessionDescription*)> munger) {
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@ -2701,6 +2701,60 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
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delete SetCalleePcWrapperAndReturnCurrent(nullptr);
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}
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// Test that audio and video flow end-to-end when codec names don't use the
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// expected casing, given that they're supposed to be case insensitive. To test
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// this, all but one codec is removed from each media description, and its
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// casing is changed.
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//
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// In the past, this has regressed and caused crashes/black video, due to the
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// fact that code at some layers was doing case-insensitive comparisons and
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// code at other layers was not.
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TEST_F(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
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ASSERT_TRUE(CreatePeerConnectionWrappers());
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ConnectFakeSignaling();
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caller()->AddAudioVideoMediaStream();
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callee()->AddAudioVideoMediaStream();
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// Remove all but one audio/video codec (opus and VP8), and change the
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// casing of the caller's generated offer.
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caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) {
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cricket::AudioContentDescription* audio =
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GetFirstAudioContentDescription(description);
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ASSERT_NE(nullptr, audio);
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auto audio_codecs = audio->codecs();
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audio_codecs.erase(std::remove_if(audio_codecs.begin(), audio_codecs.end(),
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[](const cricket::AudioCodec& codec) {
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return codec.name != "opus";
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}),
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audio_codecs.end());
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ASSERT_EQ(1u, audio_codecs.size());
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audio_codecs[0].name = "OpUs";
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audio->set_codecs(audio_codecs);
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cricket::VideoContentDescription* video =
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GetFirstVideoContentDescription(description);
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ASSERT_NE(nullptr, video);
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auto video_codecs = video->codecs();
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video_codecs.erase(std::remove_if(video_codecs.begin(), video_codecs.end(),
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[](const cricket::VideoCodec& codec) {
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return codec.name != "VP8";
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}),
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video_codecs.end());
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ASSERT_EQ(1u, video_codecs.size());
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video_codecs[0].name = "vP8";
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video->set_codecs(video_codecs);
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});
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caller()->CreateAndSetAndSignalOffer();
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ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
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// Verify frames are still received end-to-end.
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ExpectNewFramesReceivedWithWait(
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kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
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kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
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kMaxWaitForFramesMs);
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}
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} // namespace
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#endif // if !defined(THREAD_SANITIZER)
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@ -18,6 +18,7 @@
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/trace_event.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/video_coding/include/video_codec_interface.h"
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#include "webrtc/system_wrappers/include/metrics.h"
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@ -50,15 +51,21 @@ const char* GetUmaPrefix(VideoEncoderConfig::ContentType content_type) {
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HistogramCodecType PayloadNameToHistogramCodecType(
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const std::string& payload_name) {
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if (payload_name == "VP8") {
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return kVideoVp8;
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} else if (payload_name == "VP9") {
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return kVideoVp9;
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} else if (payload_name == "H264") {
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return kVideoH264;
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} else {
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rtc::Optional<VideoCodecType> codecType =
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PayloadNameToCodecType(payload_name);
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if (!codecType) {
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return kVideoUnknown;
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}
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switch (*codecType) {
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case kVideoCodecVP8:
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return kVideoVp8;
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case kVideoCodecVP9:
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return kVideoVp9;
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case kVideoCodecH264:
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return kVideoH264;
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default:
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return kVideoUnknown;
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}
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}
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void UpdateCodecTypeHistogram(const std::string& payload_name) {
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@ -21,6 +21,7 @@
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#include "webrtc/base/logging.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/base/trace_event.h"
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#include "webrtc/common_types.h"
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#include "webrtc/common_video/h264/profile_level_id.h"
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
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@ -142,15 +143,8 @@ VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
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codec.plType = decoder.payload_type;
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strncpy(codec.plName, decoder.payload_name.c_str(), sizeof(codec.plName));
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if (decoder.payload_name == "VP8") {
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codec.codecType = kVideoCodecVP8;
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} else if (decoder.payload_name == "VP9") {
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codec.codecType = kVideoCodecVP9;
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} else if (decoder.payload_name == "H264") {
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codec.codecType = kVideoCodecH264;
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} else {
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codec.codecType = kVideoCodecGeneric;
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}
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codec.codecType =
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PayloadNameToCodecType(decoder.payload_name).value_or(kVideoCodecGeneric);
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if (codec.codecType == kVideoCodecVP8) {
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*(codec.VP8()) = VideoEncoder::GetDefaultVp8Settings();
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@ -263,9 +263,14 @@ std::string VideoSendStream::StreamStats::ToString() const {
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namespace {
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bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
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if (payload_name == "VP8" || payload_name == "VP9")
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rtc::Optional<VideoCodecType> codecType =
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PayloadNameToCodecType(payload_name);
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if (codecType &&
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(*codecType == kVideoCodecVP8 || *codecType == kVideoCodecVP9)) {
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return true;
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RTC_DCHECK(payload_name == "H264" || payload_name == "FAKE")
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}
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RTC_DCHECK((codecType && *codecType == kVideoCodecH264) ||
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payload_name == "FAKE")
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<< "unknown payload_name " << payload_name;
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return false;
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}
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