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Remove mixing_test.cc.

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BUG=webrtc:4690, webrtc:7238, webrtc:7364

Review-Url: https://codereview.webrtc.org/2756263002
Cr-Commit-Position: refs/heads/master@{#17308}
This commit is contained in:
solenberg 2017-03-20 05:16:01 -07:00 committed by Commit bot
parent 61c703118c
commit afcf378459
2 changed files with 0 additions and 287 deletions
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1
webrtc/voice_engine/BUILD.gn
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@ -269,7 +269,6 @@ if (rtc_include_tests) {
"test/auto_test/standard/dtmf_test.cc",
"test/auto_test/standard/file_before_streaming_test.cc",
"test/auto_test/standard/file_test.cc",
"test/auto_test/standard/mixing_test.cc",
"test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
"test/auto_test/standard/rtp_rtcp_extensions.cc",
"test/auto_test/standard/rtp_rtcp_test.cc",

286
webrtc/voice_engine/test/auto_test/standard/mixing_test.cc
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@ -1,286 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdio.h>
#include <string>
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
namespace webrtc {
namespace {
const int16_t kLimiterHeadroom = 29204; // == -1 dbFS
const int16_t kInt16Max = 0x7fff;
const int kPayloadType = 105;
const int kInSampleRateHz = 16000; // Input file taken as 16 kHz by default.
const int kRecSampleRateHz = 16000; // Recorded with 16 kHz L16.
const int kTestDurationMs = 3000;
const CodecInst kCodecL16 = {kPayloadType, "L16", 16000, 160, 1, 256000};
const CodecInst kCodecOpus = {kPayloadType, "opus", 48000, 960, 1, 32000};
} // namespace
class MixingTest : public AfterInitializationFixture {
protected:
MixingTest()
: output_filename_(test::OutputPath() + "mixing_test_output.pcm") {
}
void SetUp() {
transport_ = new LoopBackTransport(voe_network_, 0);
}
void TearDown() {
delete transport_;
}
// Creates and mixes |num_remote_streams| which play a file "as microphone"
// with |num_local_streams| which play a file "locally", using a constant
// amplitude of |input_value|. The local streams manifest as "anonymous"
// mixing participants, meaning they will be mixed regardless of the number
// of participants. (A stream is a VoiceEngine "channel").
//
// The mixed output is verified to always fall between |max_output_value| and
// |min_output_value|, after a startup phase.
//
// |num_remote_streams_using_mono| of the remote streams use mono, with the
// remainder using stereo.
void RunMixingTest(int num_remote_streams,
int num_local_streams,
int num_remote_streams_using_mono,
bool real_audio,
int16_t input_value,
int16_t max_output_value,
int16_t min_output_value,
const CodecInst& codec_inst) {
ASSERT_LE(num_remote_streams_using_mono, num_remote_streams);
if (real_audio) {
input_filename_ = test::ResourcePath("voice_engine/audio_long16", "pcm");
} else {
input_filename_ = test::OutputPath() + "mixing_test_input.pcm";
GenerateInputFile(input_value);
}
std::vector<int> local_streams(num_local_streams);
for (size_t i = 0; i < local_streams.size(); ++i) {
local_streams[i] = voe_base_->CreateChannel();
EXPECT_NE(-1, local_streams[i]);
}
StartLocalStreams(local_streams);
TEST_LOG("Playing %d local streams.\n", num_local_streams);
std::vector<int> remote_streams(num_remote_streams);
for (size_t i = 0; i < remote_streams.size(); ++i) {
remote_streams[i] = voe_base_->CreateChannel();
EXPECT_NE(-1, remote_streams[i]);
}
StartRemoteStreams(remote_streams, num_remote_streams_using_mono,
codec_inst);
TEST_LOG("Playing %d remote streams.\n", num_remote_streams);
// Give it plenty of time to get started.
SleepMs(1000);
// Start recording the mixed output and wait.
EXPECT_EQ(0, voe_file_->StartRecordingPlayout(-1 /* record meeting */,
output_filename_.c_str()));
SleepMs(kTestDurationMs);
while (GetFileDurationMs(output_filename_.c_str()) < kTestDurationMs) {
SleepMs(200);
}
EXPECT_EQ(0, voe_file_->StopRecordingPlayout(-1));
StopLocalStreams(local_streams);
StopRemoteStreams(remote_streams);
if (!real_audio) {
VerifyMixedOutput(max_output_value, min_output_value);
}
}
private:
// Generate input file with constant values equal to |input_value|. The file
// will be twice the duration of the test.
void GenerateInputFile(int16_t input_value) {
FILE* input_file = fopen(input_filename_.c_str(), "wb");
ASSERT_TRUE(input_file != NULL);
for (int i = 0; i < kInSampleRateHz / 1000 * (kTestDurationMs * 2); i++) {
ASSERT_EQ(1u, fwrite(&input_value, sizeof(input_value), 1, input_file));
}
ASSERT_EQ(0, fclose(input_file));
}
void VerifyMixedOutput(int16_t max_output_value, int16_t min_output_value) {
// Verify the mixed output.
FILE* output_file = fopen(output_filename_.c_str(), "rb");
ASSERT_TRUE(output_file != NULL);
int16_t output_value = 0;
int samples_read = 0;
while (fread(&output_value, sizeof(output_value), 1, output_file) == 1) {
samples_read++;
std::ostringstream trace_stream;
trace_stream << samples_read << " samples read";
SCOPED_TRACE(trace_stream.str());
ASSERT_LE(output_value, max_output_value);
ASSERT_GE(output_value, min_output_value);
}
// Ensure we've at least recorded half as much file as the duration of the
// test. We have to use a relaxed tolerance here due to filesystem flakiness
// on the bots.
ASSERT_GE((samples_read * 1000.0) / kRecSampleRateHz, kTestDurationMs);
// Ensure we read the entire file.
ASSERT_NE(0, feof(output_file));
ASSERT_EQ(0, fclose(output_file));
}
// Start up local streams ("anonymous" participants).
void StartLocalStreams(const std::vector<int>& streams) {
for (size_t i = 0; i < streams.size(); ++i) {
EXPECT_EQ(0, voe_base_->StartPlayout(streams[i]));
EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(streams[i],
input_filename_.c_str(), true));
}
}
void StopLocalStreams(const std::vector<int>& streams) {
for (size_t i = 0; i < streams.size(); ++i) {
EXPECT_EQ(0, voe_base_->StopPlayout(streams[i]));
EXPECT_EQ(0, voe_base_->DeleteChannel(streams[i]));
}
}
// Start up remote streams ("normal" participants).
void StartRemoteStreams(const std::vector<int>& streams,
int num_remote_streams_using_mono,
const CodecInst& codec_inst) {
for (int i = 0; i < num_remote_streams_using_mono; ++i) {
// Add some delay between starting up the channels in order to give them
// different energies in the "real audio" test and hopefully exercise
// more code paths.
SleepMs(50);
StartRemoteStream(streams[i], codec_inst, 1234 + 2 * i);
}
// The remainder of the streams will use stereo.
CodecInst codec_inst_stereo = codec_inst;
codec_inst_stereo.channels = 2;
codec_inst_stereo.pltype++;
for (size_t i = num_remote_streams_using_mono; i < streams.size(); ++i) {
StartRemoteStream(streams[i], codec_inst_stereo, 1234 + 2 * i);
}
}
// Start up a single remote stream.
void StartRemoteStream(int stream, const CodecInst& codec_inst, int port) {
EXPECT_EQ(0, voe_codec_->SetRecPayloadType(stream, codec_inst));
EXPECT_EQ(0, voe_network_->RegisterExternalTransport(stream, *transport_));
EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(
stream, static_cast<unsigned int>(stream)));
transport_->AddChannel(stream, stream);
EXPECT_EQ(0, voe_base_->StartPlayout(stream));
EXPECT_EQ(0, voe_codec_->SetSendCodec(stream, codec_inst));
EXPECT_EQ(0, voe_base_->StartSend(stream));
EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(stream,
input_filename_.c_str(), true));
}
void StopRemoteStreams(const std::vector<int>& streams) {
for (size_t i = 0; i < streams.size(); ++i) {
EXPECT_EQ(0, voe_base_->StopSend(streams[i]));
EXPECT_EQ(0, voe_base_->StopPlayout(streams[i]));
EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(streams[i]));
EXPECT_EQ(0, voe_base_->DeleteChannel(streams[i]));
}
}
int GetFileDurationMs(const char* file_name) {
FILE* fid = fopen(file_name, "rb");
EXPECT_FALSE(fid == NULL);
fseek(fid, 0, SEEK_END);
int size = ftell(fid);
EXPECT_NE(-1, size);
fclose(fid);
// Divided by 2 due to 2 bytes/sample.
return size * 1000 / kRecSampleRateHz / 2;
}
std::string input_filename_;
const std::string output_filename_;
LoopBackTransport* transport_;
};
// This test has no verification, but exercises additional code paths in a
// somewhat more realistic scenario using real audio. It can at least hunt for
// asserts and crashes.
TEST_F(MixingTest, MixManyChannelsForStress) {
RunMixingTest(10, 0, 10, true, 0, 0, 0, kCodecL16);
}
TEST_F(MixingTest, MixManyChannelsForStressOpus) {
RunMixingTest(10, 0, 10, true, 0, 0, 0, kCodecOpus);
}
// These tests assume a maximum of three mixed participants. We typically allow
// a +/- 10% range around the expected output level to account for distortion
// from coding and processing in the loopback chain.
TEST_F(MixingTest, FourChannelsWithOnlyThreeMixed) {
const int16_t kInputValue = 1000;
const int16_t kExpectedOutput = kInputValue * 3;
RunMixingTest(4, 0, 4, false, kInputValue, 1.1 * kExpectedOutput,
0.9 * kExpectedOutput, kCodecL16);
}
// Ensure the mixing saturation protection is working. We can do this because
// the mixing limiter is given some headroom, so the expected output is less
// than full scale.
TEST_F(MixingTest, VerifySaturationProtection) {
const int16_t kInputValue = 20000;
const int16_t kExpectedOutput = kLimiterHeadroom;
// If this isn't satisfied, we're not testing anything.
ASSERT_GT(kInputValue * 3, kInt16Max);
ASSERT_LT(1.1 * kExpectedOutput, kInt16Max);
RunMixingTest(3, 0, 3, false, kInputValue, 1.1 * kExpectedOutput,
0.9 * kExpectedOutput, kCodecL16);
}
TEST_F(MixingTest, SaturationProtectionHasNoEffectOnOneChannel) {
const int16_t kInputValue = kInt16Max;
const int16_t kExpectedOutput = kInt16Max;
// If this isn't satisfied, we're not testing anything.
ASSERT_GT(0.95 * kExpectedOutput, kLimiterHeadroom);
// Tighter constraints are required here to properly test this.
RunMixingTest(1, 0, 1, false, kInputValue, kExpectedOutput,
0.95 * kExpectedOutput, kCodecL16);
}
TEST_F(MixingTest, VerifyAnonymousAndNormalParticipantMixing) {
const int16_t kInputValue = 1000;
const int16_t kExpectedOutput = kInputValue * 2;
RunMixingTest(1, 1, 1, false, kInputValue, 1.1 * kExpectedOutput,
0.9 * kExpectedOutput, kCodecL16);
}
TEST_F(MixingTest, AnonymousParticipantsAreAlwaysMixed) {
const int16_t kInputValue = 1000;
const int16_t kExpectedOutput = kInputValue * 4;
RunMixingTest(3, 1, 3, false, kInputValue, 1.1 * kExpectedOutput,
0.9 * kExpectedOutput, kCodecL16);
}
TEST_F(MixingTest, VerifyStereoAndMonoMixing) {
const int16_t kInputValue = 1000;
const int16_t kExpectedOutput = kInputValue * 2;
RunMixingTest(2, 0, 1, false, kInputValue, 1.1 * kExpectedOutput,
// Lower than 0.9 due to observed flakiness on bots.
0.8 * kExpectedOutput, kCodecL16);
}
} // namespace webrtc