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Update NetEq network statistics in neteq_unittest.

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NetEqNetworkStatistics has been updated some time ago. A bit exactness test in neteq unittests is still using the old NetEqNetworkStatistics.

New neteq4_network_stats.dat generated by running TestBitExactness with flag "genref"

BUG=

Review URL: https://codereview.webrtc.org/1522103002

Cr-Commit-Position: refs/heads/master@{#11052}
This commit is contained in:
minyue
2015-12-16 07:36:04 -08:00
committed by Commit bot
parent 44307630d3
commit 5f026d03af
6 changed files with 192 additions and 89 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
resources/audio_coding/neteq4_network_stats.dat.sha1
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@ -1 +1 @@
e5e2d0ff26d16339cf0f37a3512bfa2d390a9a9a
2cf380a05ee07080bd72471e8ec7777a39644ec9

2
resources/audio_coding/neteq4_rtcp_stats.dat.sha1
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@ -1 +1 @@
948753a2087fbb5b74a3ea0b1aef8593c9c30b10
b8880bf9fed2487efbddcb8d94b9937a29ae521d

15
webrtc/modules/audio_coding/neteq/neteq_tests.gypi
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@ -39,6 +39,21 @@
'defines': [
],
}, # neteq_rtpplay
{
'target_name': 'neteq_unittest_proto',
'type': 'static_library',
'sources': [
'neteq_unittest.proto',
],
'variables': {
'proto_in_dir': '.',
# Workaround to protect against gyp's pathname relativization when
# this file is included by modules.gyp.
'proto_out_protected': 'webrtc/audio_coding/neteq',
'proto_out_dir': '<(proto_out_protected)',
},
'includes': ['../../../build/protoc.gypi',],
},
],
}],
],

231
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
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@ -33,24 +33,87 @@
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
#else
#include "webrtc/audio_coding/neteq/neteq_unittest.pb.h"
#endif
#endif
DEFINE_bool(gen_ref, false, "Generate reference files.");
namespace webrtc {
namespace {
static bool IsAllZero(const int16_t* buf, size_t buf_length) {
bool IsAllZero(const int16_t* buf, size_t buf_length) {
bool all_zero = true;
for (size_t n = 0; n < buf_length && all_zero; ++n)
all_zero = buf[n] == 0;
return all_zero;
}
static bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
bool all_non_zero = true;
for (size_t n = 0; n < buf_length && all_non_zero; ++n)
all_non_zero = buf[n] != 0;
return all_non_zero;
}
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
stats->set_packet_discard_rate(stats_raw.packet_discard_rate);
stats->set_expand_rate(stats_raw.expand_rate);
stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
stats->set_preemptive_rate(stats_raw.preemptive_rate);
stats->set_accelerate_rate(stats_raw.accelerate_rate);
stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
stats->set_clockdrift_ppm(stats_raw.clockdrift_ppm);
stats->set_added_zero_samples(stats_raw.added_zero_samples);
stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
}
void Convert(const webrtc::RtcpStatistics& stats_raw,
webrtc::neteq_unittest::RtcpStatistics* stats) {
stats->set_fraction_lost(stats_raw.fraction_lost);
stats->set_cumulative_lost(stats_raw.cumulative_lost);
stats->set_extended_max_sequence_number(
stats_raw.extended_max_sequence_number);
stats->set_jitter(stats_raw.jitter);
}
void WriteMessage(FILE* file, const std::string& message) {
int32_t size = message.length();
ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
if (size <= 0)
return;
ASSERT_EQ(static_cast<size_t>(size),
fwrite(message.data(), sizeof(char), size, file));
}
void ReadMessage(FILE* file, std::string* message) {
int32_t size;
ASSERT_EQ(1u, fread(&size, sizeof(size), 1, file));
if (size <= 0)
return;
rtc::scoped_ptr<char[]> buffer(new char[size]);
ASSERT_EQ(static_cast<size_t>(size),
fread(buffer.get(), sizeof(char), size, file));
message->assign(buffer.get(), size);
}
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
} // namespace
namespace webrtc {
class RefFiles {
public:
RefFiles(const std::string& input_file, const std::string& output_file);
@ -128,92 +191,84 @@ void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
}
}
void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats) {
if (output_fp_) {
ASSERT_EQ(1u, fwrite(&stats, sizeof(NetEqNetworkStatistics), 1,
output_fp_));
}
void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats_raw) {
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
if (!output_fp_)
return;
neteq_unittest::NetEqNetworkStatistics stats;
Convert(stats_raw, &stats);
std::string stats_string;
ASSERT_TRUE(stats.SerializeToString(&stats_string));
WriteMessage(output_fp_, stats_string);
#else
FAIL() << "Writing to reference file requires Proto Buffer.";
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
}
void RefFiles::ReadFromFileAndCompare(
const NetEqNetworkStatistics& stats) {
// TODO(minyue): Update resource/audio_coding/neteq_network_stats.dat and
// resource/audio_coding/neteq_network_stats_win32.dat.
struct NetEqNetworkStatisticsOld {
uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
uint16_t preferred_buffer_size_ms; // Target buffer size in ms.
uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
// jitter; 0 otherwise.
uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
uint16_t packet_discard_rate; // Late loss rate in Q14.
uint16_t expand_rate; // Fraction (of original stream) of synthesized
// audio inserted through expansion (in Q14).
uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration
// (in Q14).
int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
// (positive or negative).
int added_zero_samples; // Number of zero samples added in "off" mode.
};
if (input_fp_) {
// Read from ref file.
size_t stat_size = sizeof(NetEqNetworkStatisticsOld);
NetEqNetworkStatisticsOld ref_stats;
ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
// Compare
ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms);
ASSERT_EQ(stats.preferred_buffer_size_ms,
ref_stats.preferred_buffer_size_ms);
ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found);
ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate);
ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate);
ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate);
ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate);
ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate);
ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm);
ASSERT_EQ(stats.added_zero_samples,
static_cast<size_t>(ref_stats.added_zero_samples));
ASSERT_EQ(stats.secondary_decoded_rate, 0);
ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate);
}
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
if (!input_fp_)
return;
std::string stats_string;
ReadMessage(input_fp_, &stats_string);
neteq_unittest::NetEqNetworkStatistics ref_stats;
ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
// Compare
ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms());
ASSERT_EQ(stats.preferred_buffer_size_ms,
ref_stats.preferred_buffer_size_ms());
ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found());
ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate());
ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate());
ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate());
ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate());
ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate());
ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm());
ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples());
ASSERT_EQ(stats.secondary_decoded_rate, 0);
ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate());
#else
FAIL() << "Reading from reference file requires Proto Buffer.";
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
}
void RefFiles::WriteToFile(const RtcpStatistics& stats) {
if (output_fp_) {
ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.cumulative_lost),
sizeof(stats.cumulative_lost), 1, output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.extended_max_sequence_number),
sizeof(stats.extended_max_sequence_number), 1,
output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
output_fp_));
}
void RefFiles::WriteToFile(const RtcpStatistics& stats_raw) {
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
if (!output_fp_)
return;
neteq_unittest::RtcpStatistics stats;
Convert(stats_raw, &stats);
std::string stats_string;
ASSERT_TRUE(stats.SerializeToString(&stats_string));
WriteMessage(output_fp_, stats_string);
#else
FAIL() << "Writing to reference file requires Proto Buffer.";
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
}
void RefFiles::ReadFromFileAndCompare(
const RtcpStatistics& stats) {
if (input_fp_) {
// Read from ref file.
RtcpStatistics ref_stats;
ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
sizeof(ref_stats.fraction_lost), 1, input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.cumulative_lost),
sizeof(ref_stats.cumulative_lost), 1, input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.extended_max_sequence_number),
sizeof(ref_stats.extended_max_sequence_number), 1,
input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
input_fp_));
// Compare
ASSERT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
ASSERT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost);
ASSERT_EQ(ref_stats.extended_max_sequence_number,
stats.extended_max_sequence_number);
ASSERT_EQ(ref_stats.jitter, stats.jitter);
}
void RefFiles::ReadFromFileAndCompare(const RtcpStatistics& stats) {
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
if (!input_fp_)
return;
std::string stats_string;
ReadMessage(input_fp_, &stats_string);
neteq_unittest::RtcpStatistics ref_stats;
ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
// Compare
ASSERT_EQ(stats.fraction_lost, ref_stats.fraction_lost());
ASSERT_EQ(stats.cumulative_lost, ref_stats.cumulative_lost());
ASSERT_EQ(stats.extended_max_sequence_number,
ref_stats.extended_max_sequence_number());
ASSERT_EQ(stats.jitter, ref_stats.jitter());
#else
FAIL() << "Reading from reference file requires Proto Buffer.";
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
}
class NetEqDecodingTest : public ::testing::Test {
@ -234,10 +289,12 @@ class NetEqDecodingTest : public ::testing::Test {
void LoadDecoders();
void OpenInputFile(const std::string &rtp_file);
void Process(size_t* out_len);
void DecodeAndCompare(const std::string& rtp_file,
const std::string& ref_file,
const std::string& stat_ref_file,
const std::string& rtcp_ref_file);
static void PopulateRtpInfo(int frame_index,
int timestamp,
WebRtcRTPHeader* rtp_info);
@ -453,15 +510,15 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
*payload_len = 1; // Only noise level, no spectral parameters.
}
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
#define IF_ALL_CODECS(x) x
#define MAYBE_TestBitExactness TestBitExactness
#else
#define IF_ALL_CODECS(x) DISABLED_##x
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
#endif
TEST_F(NetEqDecodingTest,
DISABLED_ON_IOS(DISABLED_ON_ANDROID(IF_ALL_CODECS(TestBitExactness)))) {
TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal_new.rtp";
// Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm

29
webrtc/modules/audio_coding/neteq/neteq_unittest.proto Normal file
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@ -0,0 +1,29 @@
syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.neteq_unittest;
message NetEqNetworkStatistics {
optional uint32 current_buffer_size_ms = 1;
optional uint32 preferred_buffer_size_ms = 2;
optional uint32 jitter_peaks_found = 3;
optional uint32 packet_loss_rate = 4;
optional uint32 packet_discard_rate = 5;
optional uint32 expand_rate = 6;
optional uint32 speech_expand_rate = 7;
optional uint32 preemptive_rate = 8;
optional uint32 accelerate_rate = 9;
optional uint32 secondary_decoded_rate = 10;
optional int32 clockdrift_ppm = 11;
optional uint64 added_zero_samples = 12;
optional int32 mean_waiting_time_ms = 13;
optional int32 median_waiting_time_ms = 14;
optional int32 min_waiting_time_ms = 15;
optional int32 max_waiting_time_ms = 16;
}
message RtcpStatistics {
optional uint32 fraction_lost = 1;
optional uint32 cumulative_lost = 2;
optional uint32 extended_max_sequence_number = 3;
optional uint32 jitter = 4;
}

2
webrtc/modules/modules.gyp
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@ -398,10 +398,12 @@
['enable_protobuf==1', {
'defines': [
'WEBRTC_AUDIOPROC_DEBUG_DUMP',
'WEBRTC_NETEQ_UNITTEST_BITEXACT',
],
'dependencies': [
'audioproc_protobuf_utils',
'audioproc_unittest_proto',
'neteq_unittest_proto',
],
'sources': [
'audio_processing/audio_processing_impl_locking_unittest.cc',