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Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo

Browse Source 
It's propagated from ReceiveStatistics up to VoiceReceiverInfo,
and then not used. It's not part of the standard stats.

Bug: None
Change-Id: I90ce6a72e3ca846adbbba5d3023fef18a2169018
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149164
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28933}
This commit is contained in:
Niels Möller 2019-08-16 11:05:42 +02:00 committed by Commit Bot
parent b689af4c99
commit 224c69d527
13 changed files with 1 additions and 22 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
audio/audio_receive_stream.cc
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@ -196,7 +196,6 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
call_stats.last_packet_received_timestamp_ms;
stats.codec_name = receive_codec->second.name;
stats.codec_payload_type = receive_codec->first;
stats.ext_seqnum = call_stats.extendedMax;
int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
if (clockrate_khz > 0) {
stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;

3
audio/audio_receive_stream_unittest.cc
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@ -63,7 +63,7 @@ const unsigned int kSpeechOutputLevel = 99;
const double kTotalOutputEnergy = 0.25;
const double kTotalOutputDuration = 0.5;
const CallReceiveStatistics kCallStats = {678, 901, 234, -12, 567, 890, 123};
const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 890, 123};
const std::pair<int, SdpAudioFormat> kReceiveCodec = {
123,
{"codec_name_recv", 96000, 0}};
@ -271,7 +271,6 @@ TEST(AudioReceiveStreamTest, GetStats) {
stats.packets_rcvd);
EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
EXPECT_EQ(kReceiveCodec.second.name, stats.codec_name);
EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
EXPECT_EQ(
kCallStats.jitterSamples / (kReceiveCodec.second.clockrate_hz / 1000),
stats.jitter_ms);

1
audio/audio_send_stream.cc
View File

@ -444,7 +444,6 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
if (block.source_SSRC == stats.local_ssrc) {
stats.packets_lost = block.cumulative_num_packets_lost;
stats.fraction_lost = Q8ToFloat(block.fraction_lost);
stats.ext_seqnum = block.extended_highest_sequence_number;
// Convert timestamps to milliseconds.
if (spec.format.clockrate_hz / 1000 > 0) {
stats.jitter_ms =

2
audio/audio_send_stream_unittest.cc
View File

@ -416,8 +416,6 @@ TEST(AudioSendStreamTest, GetStats) {
EXPECT_EQ(kReportBlock.cumulative_num_packets_lost, stats.packets_lost);
EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
EXPECT_EQ(kIsacFormat.name, stats.codec_name);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
stats.ext_seqnum);
EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
(kIsacFormat.clockrate_hz / 1000)),
stats.jitter_ms);

1
audio/channel_receive.cc
View File

@ -733,7 +733,6 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
}
stats.cumulativeLost = statistics.packets_lost;
stats.extendedMax = statistics.extended_highest_sequence_number;
stats.jitterSamples = statistics.jitter;
// --- RTT

1
audio/channel_receive.h
View File

@ -52,7 +52,6 @@ class RtpRtcp;
struct CallReceiveStatistics {
unsigned int cumulativeLost;
unsigned int extendedMax;
unsigned int jitterSamples;
int64_t rttMs;
size_t bytesReceived;

1
audio/test/audio_stats_test.cc
View File

@ -100,7 +100,6 @@ class NoLossTest : public AudioEndToEndTest {
// Match these stats between caller and receiver.
EXPECT_EQ(send_stats.local_ssrc, recv_stats.remote_ssrc);
EXPECT_EQ(*send_stats.codec_payload_type, *recv_stats.codec_payload_type);
EXPECT_TRUE(rtc::SafeEq(send_stats.ext_seqnum, recv_stats.ext_seqnum));
}
};
} // namespace

1
call/audio_receive_stream.h
View File

@ -43,7 +43,6 @@ class AudioReceiveStream {
uint32_t packets_lost = 0;
std::string codec_name;
absl::optional<int> codec_payload_type;
uint32_t ext_seqnum = 0;
uint32_t jitter_ms = 0;
uint32_t jitter_buffer_ms = 0;
uint32_t jitter_buffer_preferred_ms = 0;

1
call/audio_send_stream.h
View File

@ -53,7 +53,6 @@ class AudioSendStream {
float fraction_lost = -1.0f;
std::string codec_name;
absl::optional<int> codec_payload_type;
int32_t ext_seqnum = -1;
int32_t jitter_ms = -1;
int64_t rtt_ms = -1;
int16_t audio_level = 0;

2
media/base/media_channel.h
View File

@ -466,7 +466,6 @@ struct MediaReceiverInfo {
struct VoiceSenderInfo : public MediaSenderInfo {
VoiceSenderInfo();
~VoiceSenderInfo();
int ext_seqnum = 0;
int jitter_ms = 0;
// Current audio level, expressed linearly [0,32767].
int audio_level = 0;
@ -482,7 +481,6 @@ struct VoiceSenderInfo : public MediaSenderInfo {
struct VoiceReceiverInfo : public MediaReceiverInfo {
VoiceReceiverInfo();
~VoiceReceiverInfo();
int ext_seqnum = 0;
int jitter_ms = 0;
int jitter_buffer_ms = 0;
int jitter_buffer_preferred_ms = 0;

2
media/engine/webrtc_voice_engine.cc
View File

@ -2169,7 +2169,6 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
sinfo.fraction_lost = stats.fraction_lost;
sinfo.codec_name = stats.codec_name;
sinfo.codec_payload_type = stats.codec_payload_type;
sinfo.ext_seqnum = stats.ext_seqnum;
sinfo.jitter_ms = stats.jitter_ms;
sinfo.rtt_ms = stats.rtt_ms;
sinfo.audio_level = stats.audio_level;
@ -2212,7 +2211,6 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
rinfo.packets_lost = stats.packets_lost;
rinfo.codec_name = stats.codec_name;
rinfo.codec_payload_type = stats.codec_payload_type;
rinfo.ext_seqnum = stats.ext_seqnum;
rinfo.jitter_ms = stats.jitter_ms;
rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;

5
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -572,7 +572,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
stats.fraction_lost = 34.56f;
stats.codec_name = "codec_name_send";
stats.codec_payload_type = 42;
stats.ext_seqnum = 789;
stats.jitter_ms = 12;
stats.rtt_ms = 345;
stats.audio_level = 678;
@ -607,7 +606,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
EXPECT_EQ(info.codec_name, stats.codec_name);
EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum);
EXPECT_EQ(info.jitter_ms, stats.jitter_ms);
EXPECT_EQ(info.rtt_ms, stats.rtt_ms);
EXPECT_EQ(info.audio_level, stats.audio_level);
@ -649,7 +647,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
stats.packets_lost = 101;
stats.codec_name = "codec_name_recv";
stats.codec_payload_type = 42;
stats.ext_seqnum = 678;
stats.jitter_ms = 901;
stats.jitter_buffer_ms = 234;
stats.jitter_buffer_preferred_ms = 567;
@ -692,8 +689,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::Test {
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<unsigned int>(info.ext_seqnum),
stats.ext_seqnum);
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_ms), stats.jitter_ms);
EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_ms),
stats.jitter_buffer_ms);

2
pc/stats_collector_unittest.cc
View File

@ -535,7 +535,6 @@ void InitVoiceSenderInfo(cricket::VoiceSenderInfo* voice_sender_info,
voice_sender_info->fraction_lost = 103;
voice_sender_info->jitter_ms = 104;
voice_sender_info->packets_lost = 105;
voice_sender_info->ext_seqnum = 106;
voice_sender_info->audio_level = 107;
voice_sender_info->apm_statistics.echo_return_loss = 108;
voice_sender_info->apm_statistics.echo_return_loss_enhancement = 109;
@ -568,7 +567,6 @@ void InitVoiceReceiverInfo(cricket::VoiceReceiverInfo* voice_receiver_info) {
voice_receiver_info->bytes_rcvd = 110;
voice_receiver_info->packets_rcvd = 111;
voice_receiver_info->packets_lost = 114;
voice_receiver_info->ext_seqnum = 115;
voice_receiver_info->jitter_ms = 116;
voice_receiver_info->jitter_buffer_ms = 117;
voice_receiver_info->jitter_buffer_preferred_ms = 118;