From 13f54b2c5684119a1b31f172b3745cbec7229ebb Mon Sep 17 00:00:00 2001 From: hbos Date: Tue, 28 Feb 2017 06:56:04 -0800 Subject: [PATCH] Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine. As per https://github.com/w3c/webrtc-stats/pull/168. NOTRY due to broken linux_ubsan_vptr, all other tests passed. BUG=webrtc:7061 NOTRY=True Review-Url: https://codereview.webrtc.org/2718383002 Cr-Commit-Position: refs/heads/master@{#16907} --- webrtc/api/stats/rtcstats_objects.h | 4 ++-- webrtc/pc/rtcstats_integrationtest.cc | 4 ++-- webrtc/pc/rtcstatscollector.cc | 2 +- webrtc/pc/rtcstatscollector_unittest.cc | 8 ++++---- webrtc/stats/rtcstats_objects.cc | 12 ++++++------ 5 files changed, 15 insertions(+), 15 deletions(-) diff --git a/webrtc/api/stats/rtcstats_objects.h b/webrtc/api/stats/rtcstats_objects.h index baf0d28386..cc01b5e551 100644 --- a/webrtc/api/stats/rtcstats_objects.h +++ b/webrtc/api/stats/rtcstats_objects.h @@ -87,12 +87,12 @@ class RTCCodecStats final : public RTCStats { ~RTCCodecStats() override; RTCStatsMember payload_type; - RTCStatsMember codec; + RTCStatsMember mime_type; RTCStatsMember clock_rate; // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061 RTCStatsMember channels; // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061 - RTCStatsMember parameters; + RTCStatsMember sdp_fmtp_line; // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061 RTCStatsMember implementation; }; diff --git a/webrtc/pc/rtcstats_integrationtest.cc b/webrtc/pc/rtcstats_integrationtest.cc index 34adb4cb13..1176b54aee 100644 --- a/webrtc/pc/rtcstats_integrationtest.cc +++ b/webrtc/pc/rtcstats_integrationtest.cc @@ -334,10 +334,10 @@ class RTCStatsReportVerifier { const RTCCodecStats& codec) { RTCStatsVerifier verifier(report_, &codec); verifier.TestMemberIsDefined(codec.payload_type); - verifier.TestMemberIsDefined(codec.codec); + verifier.TestMemberIsDefined(codec.mime_type); verifier.TestMemberIsPositive(codec.clock_rate); verifier.TestMemberIsUndefined(codec.channels); - verifier.TestMemberIsUndefined(codec.parameters); + verifier.TestMemberIsUndefined(codec.sdp_fmtp_line); verifier.TestMemberIsUndefined(codec.implementation); return verifier.ExpectAllMembersSuccessfullyTested(); } diff --git a/webrtc/pc/rtcstatscollector.cc b/webrtc/pc/rtcstatscollector.cc index abb4ab94c7..36a4bdef7e 100644 --- a/webrtc/pc/rtcstatscollector.cc +++ b/webrtc/pc/rtcstatscollector.cc @@ -177,7 +177,7 @@ std::unique_ptr CodecStatsFromRtpCodecParameters( RTCCodecStatsIDFromDirectionMediaAndPayload(inbound, audio, payload_type), timestamp_us)); codec_stats->payload_type = payload_type; - codec_stats->codec = codec_params.mime_type(); + codec_stats->mime_type = codec_params.mime_type(); if (codec_params.clock_rate) { codec_stats->clock_rate = static_cast(*codec_params.clock_rate); } diff --git a/webrtc/pc/rtcstatscollector_unittest.cc b/webrtc/pc/rtcstatscollector_unittest.cc index eea33e0abb..db26096c3b 100644 --- a/webrtc/pc/rtcstatscollector_unittest.cc +++ b/webrtc/pc/rtcstatscollector_unittest.cc @@ -826,25 +826,25 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) { RTCCodecStats expected_inbound_audio_codec( "RTCCodec_InboundAudio_1", report->timestamp_us()); expected_inbound_audio_codec.payload_type = 1; - expected_inbound_audio_codec.codec = "audio/opus"; + expected_inbound_audio_codec.mime_type = "audio/opus"; expected_inbound_audio_codec.clock_rate = 1337; RTCCodecStats expected_outbound_audio_codec( "RTCCodec_OutboundAudio_2", report->timestamp_us()); expected_outbound_audio_codec.payload_type = 2; - expected_outbound_audio_codec.codec = "audio/isac"; + expected_outbound_audio_codec.mime_type = "audio/isac"; expected_outbound_audio_codec.clock_rate = 1338; RTCCodecStats expected_inbound_video_codec( "RTCCodec_InboundVideo_3", report->timestamp_us()); expected_inbound_video_codec.payload_type = 3; - expected_inbound_video_codec.codec = "video/H264"; + expected_inbound_video_codec.mime_type = "video/H264"; expected_inbound_video_codec.clock_rate = 1339; RTCCodecStats expected_outbound_video_codec( "RTCCodec_OutboundVideo_4", report->timestamp_us()); expected_outbound_video_codec.payload_type = 4; - expected_outbound_video_codec.codec = "video/VP8"; + expected_outbound_video_codec.mime_type = "video/VP8"; expected_outbound_video_codec.clock_rate = 1340; ASSERT_TRUE(report->Get(expected_inbound_audio_codec.id())); diff --git a/webrtc/stats/rtcstats_objects.cc b/webrtc/stats/rtcstats_objects.cc index 68b33f58c2..9a7ebede6e 100644 --- a/webrtc/stats/rtcstats_objects.cc +++ b/webrtc/stats/rtcstats_objects.cc @@ -72,10 +72,10 @@ RTCCertificateStats::~RTCCertificateStats() { WEBRTC_RTCSTATS_IMPL(RTCCodecStats, RTCStats, "codec", &payload_type, - &codec, + &mime_type, &clock_rate, &channels, - ¶meters, + &sdp_fmtp_line, &implementation); RTCCodecStats::RTCCodecStats( @@ -87,10 +87,10 @@ RTCCodecStats::RTCCodecStats( std::string&& id, int64_t timestamp_us) : RTCStats(std::move(id), timestamp_us), payload_type("payloadType"), - codec("codec"), + mime_type("mimeType"), clock_rate("clockRate"), channels("channels"), - parameters("parameters"), + sdp_fmtp_line("sdpFmtpLine"), implementation("implementation") { } @@ -98,10 +98,10 @@ RTCCodecStats::RTCCodecStats( const RTCCodecStats& other) : RTCStats(other.id(), other.timestamp_us()), payload_type(other.payload_type), - codec(other.codec), + mime_type(other.mime_type), clock_rate(other.clock_rate), channels(other.channels), - parameters(other.parameters), + sdp_fmtp_line(other.sdp_fmtp_line), implementation(other.implementation) { }