From af512281b1b35759155b4151e8bf83467c28533c Mon Sep 17 00:00:00 2001 From: Philipp Hancke Date: Fri, 14 Oct 2022 09:32:58 +0200 Subject: [PATCH] audio: make packets lost a signed integer MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit as it is defined in RFC 3550. This avoids implicit casts between signed and unsigned definitions. BUG=webrtc:8626 Change-Id: I919b7c38ede1aa8d32f8e31b55660f540e5f5a6b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279240 Reviewed-by: Jakob Ivarsson‎ Reviewed-by: Harald Alvestrand Commit-Queue: Philipp Hancke Cr-Commit-Position: refs/heads/main@{#38522} --- audio/channel_receive.h | 2 +- audio/test/audio_stats_test.cc | 2 +- call/audio_receive_stream.h | 2 +- media/engine/webrtc_voice_engine_unittest.cc | 3 +-- 4 files changed, 4 insertions(+), 5 deletions(-) diff --git a/audio/channel_receive.h b/audio/channel_receive.h index c3eca29006..b47a4b5b97 100644 --- a/audio/channel_receive.h +++ b/audio/channel_receive.h @@ -51,7 +51,7 @@ class RtpPacketReceived; class RtpRtcp; struct CallReceiveStatistics { - unsigned int cumulativeLost; + int cumulativeLost; unsigned int jitterSamples; int64_t payload_bytes_rcvd = 0; int64_t header_and_padding_bytes_rcvd = 0; diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc index febcb066fd..c637bff94e 100644 --- a/audio/test/audio_stats_test.cc +++ b/audio/test/audio_stats_test.cc @@ -68,7 +68,7 @@ class NoLossTest : public AudioEndToEndTest { receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true); EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_rcvd); EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd); - EXPECT_EQ(0u, recv_stats.packets_lost); + EXPECT_EQ(0, recv_stats.packets_lost); EXPECT_EQ("opus", send_stats.codec_name); // recv_stats.jitter_ms // recv_stats.jitter_buffer_ms diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 5d3c38fb05..f383277323 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -38,7 +38,7 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface { uint32_t packets_rcvd = 0; uint64_t fec_packets_received = 0; uint64_t fec_packets_discarded = 0; - uint32_t packets_lost = 0; + int32_t packets_lost = 0; uint64_t packets_discarded = 0; uint32_t nacks_sent = 0; std::string codec_name; diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index f61cfb9e49..9644fbdae2 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -707,8 +707,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam { stats.header_and_padding_bytes_rcvd); EXPECT_EQ(rtc::checked_cast(info.packets_rcvd), stats.packets_rcvd); - EXPECT_EQ(rtc::checked_cast(info.packets_lost), - stats.packets_lost); + EXPECT_EQ(info.packets_lost, stats.packets_lost); EXPECT_EQ(info.codec_name, stats.codec_name); EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type); EXPECT_EQ(rtc::checked_cast(info.jitter_ms), stats.jitter_ms);