Revert "Remove CodecInst pt.1"
This reverts commit 056f9738bf7a3d16da45398239656e165c4e0851.
Reason for revert: breaks downstream
Original change's description:
> Remove CodecInst pt.1
>
> Update audio_coding tests to not use CodecInst.
>
> Bug: webrtc:7626
> Change-Id: I880fb8d72d7d0a915d274e67feb6106f023697c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/112594
> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25879}
TBR=solenberg@webrtc.org,kwiberg@webrtc.org
Change-Id: I51d666969bcd63e2b7cb7d669ec2f59b5f8f9dde
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7626
Reviewed-on: https://webrtc-review.googlesource.com/c/112906
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25881}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 34df39f..fc5a803 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1343,6 +1343,8 @@
"test/opus_test.cc",
"test/opus_test.h",
"test/target_delay_unittest.cc",
+ "test/utility.cc",
+ "test/utility.h",
]
deps = [
":audio_coding",
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 8cb5baa..bb970c1 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -287,6 +287,113 @@
_channelCritSect.Leave();
}
+int16_t Channel::Stats(CodecInst& codecInst,
+ ACMTestPayloadStats& payloadStats) {
+ _channelCritSect.Enter();
+ int n;
+ payloadStats.payloadType = -1;
+ for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+ if (_payloadStats[n].payloadType == codecInst.pltype) {
+ memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
+ break;
+ }
+ }
+ if (payloadStats.payloadType == -1) {
+ _channelCritSect.Leave();
+ return -1;
+ }
+ for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
+ if (payloadStats.frameSizeStats[n].frameSizeSample == 0) {
+ _channelCritSect.Leave();
+ return 0;
+ }
+ payloadStats.frameSizeStats[n].usageLenSec =
+ (double)payloadStats.frameSizeStats[n].totalEncodedSamples /
+ (double)codecInst.plfreq;
+
+ payloadStats.frameSizeStats[n].rateBitPerSec =
+ payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
+ payloadStats.frameSizeStats[n].usageLenSec;
+ }
+ _channelCritSect.Leave();
+ return 0;
+}
+
+void Channel::Stats(uint32_t* numPackets) {
+ _channelCritSect.Enter();
+ int k;
+ int n;
+ memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
+ for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
+ if (_payloadStats[k].payloadType == -1) {
+ break;
+ }
+ numPackets[k] = 0;
+ for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
+ if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
+ break;
+ }
+ numPackets[k] += _payloadStats[k].frameSizeStats[n].numPackets;
+ }
+ }
+ _channelCritSect.Leave();
+}
+
+void Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
+ _channelCritSect.Enter();
+
+ int k;
+ int n;
+ memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
+ for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
+ if (_payloadStats[k].payloadType == -1) {
+ break;
+ }
+ payloadType[k] = (uint8_t)_payloadStats[k].payloadType;
+ payloadLenByte[k] = 0;
+ for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
+ if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
+ break;
+ }
+ payloadLenByte[k] +=
+ (uint16_t)_payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
+ }
+ }
+
+ _channelCritSect.Leave();
+}
+
+void Channel::PrintStats(CodecInst& codecInst) {
+ ACMTestPayloadStats payloadStats;
+ Stats(codecInst, payloadStats);
+ printf("%s %d kHz\n", codecInst.plname, codecInst.plfreq / 1000);
+ printf("=====================================================\n");
+ if (payloadStats.payloadType == -1) {
+ printf("No Packets are sent with payload-type %d (%s)\n\n",
+ codecInst.pltype, codecInst.plname);
+ return;
+ }
+ for (int k = 0; k < MAX_NUM_FRAMESIZES; k++) {
+ if (payloadStats.frameSizeStats[k].frameSizeSample == 0) {
+ break;
+ }
+ printf("Frame-size.................... %d samples\n",
+ payloadStats.frameSizeStats[k].frameSizeSample);
+ printf("Average Rate.................. %.0f bits/sec\n",
+ payloadStats.frameSizeStats[k].rateBitPerSec);
+ printf("Maximum Payload-Size.......... %" PRIuS " Bytes\n",
+ payloadStats.frameSizeStats[k].maxPayloadLen);
+ printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
+ ((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
+ (double)codecInst.plfreq) /
+ (double)payloadStats.frameSizeStats[k].frameSizeSample);
+ printf("Number of Packets............. %u\n",
+ (unsigned int)payloadStats.frameSizeStats[k].numPackets);
+ printf("Duration...................... %0.3f sec\n\n",
+ payloadStats.frameSizeStats[k].usageLenSec);
+ }
+}
+
uint32_t Channel::LastInTimestamp() {
uint32_t timestamp;
_channelCritSect.Enter();
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index eb197c6..fd161fb 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -58,6 +58,14 @@
void ResetStats();
+ int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
+
+ void Stats(uint32_t* numPackets);
+
+ void Stats(uint8_t* payloadType, uint32_t* payloadLenByte);
+
+ void PrintStats(CodecInst& codecInst);
+
void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
uint32_t LastInTimestamp();
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 70319c0..2408366 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -14,9 +14,12 @@
#include <stdlib.h>
#include <memory>
+#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/utility.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@@ -50,12 +53,14 @@
}
void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int in_sample_rate,
- int payload_type, SdpAudioFormat format) {
+ std::string in_file_name, int sample_rate, size_t channels) {
+ struct CodecInst sendCodec;
+ int codecNo;
+
// Open input file
const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
- _pcmFile.Open(file_name, in_sample_rate, "rb");
- if (format.num_channels == 2) {
+ _pcmFile.Open(file_name, sample_rate, "rb");
+ if (channels == 2) {
_pcmFile.ReadStereo(true);
}
// Set test length to 500 ms (50 blocks of 10 ms each).
@@ -63,9 +68,16 @@
// Fast-forward 1 second (100 blocks) since the file starts with silence.
_pcmFile.FastForward(100);
+ // Set the codec for the current test.
+ codecNo = codeId;
+
+ EXPECT_EQ(0, acm->Codec(codecNo, &sendCodec));
+
+ sendCodec.channels = channels;
+
acm->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder(
- payload_type, format, absl::nullopt));
- _packetization = new TestPacketization(rtpStream, format.clockrate_hz);
+ sendCodec.pltype, CodecInstToSdp(sendCodec), absl::nullopt));
+ _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
EXPECT_EQ(0, acm->RegisterTransportCallback(_packetization));
_acm = acm;
@@ -100,39 +112,30 @@
}
void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, size_t channels, int file_num) {
+ std::string out_file_name, size_t channels) {
+ struct CodecInst recvCodec = CodecInst();
+ int noOfCodecs;
EXPECT_EQ(0, acm->InitializeReceiver());
- if (channels == 1) {
- acm->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {0, {"PCMU", 8000, 1}},
- {8, {"PCMA", 8000, 1}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {120, {"OPUS", 48000, 2}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
- } else {
- ASSERT_EQ(channels, 2u);
- acm->SetReceiveCodecs({{111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {110, {"PCMU", 8000, 2}},
- {118, {"PCMA", 8000, 2}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+ noOfCodecs = acm->NumberOfCodecs();
+ for (int i = 0; i < noOfCodecs; i++) {
+ EXPECT_EQ(0, acm->Codec(i, &recvCodec));
+ if (recvCodec.channels == channels)
+ EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
+ CodecInstToSdp(recvCodec)));
+ // Forces mono/stereo for Opus.
+ if (!strcmp(recvCodec.plname, "opus")) {
+ recvCodec.channels = channels;
+ EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
+ CodecInstToSdp(recvCodec)));
+ }
}
int playSampFreq;
std::string file_name;
rtc::StringBuilder file_stream;
- file_stream << webrtc::test::OutputPath() << out_file_name << file_num
- << ".pcm";
+ file_stream << webrtc::test::OutputPath() << out_file_name
+ << static_cast<int>(codeId) << ".pcm";
file_name = file_stream.str();
_rtpStream = rtpStream;
@@ -222,45 +225,85 @@
}
}
-EncodeDecodeTest::EncodeDecodeTest() = default;
+EncodeDecodeTest::EncodeDecodeTest(int test_mode) {
+ // There used to be different test modes. The only one still supported is the
+ // "autotest" mode.
+ RTC_CHECK_EQ(0, test_mode);
+}
void EncodeDecodeTest::Perform() {
- const std::map<int, SdpAudioFormat> send_codecs = {{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {0, {"PCMU", 8000, 1}},
- {8, {"PCMA", 8000, 1}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}}};
- int file_num = 0;
- for (const auto& send_codec : send_codecs) {
- RTPFile rtpFile;
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+ int numCodecs = 1;
+ int codePars[3]; // Frequency, packet size, rate.
+ int numPars[52]; // Number of codec parameters sets (freq, pacsize, rate)
+ // to test, for a given codec.
- std::string fileName = webrtc::test::TempFilename(
- webrtc::test::OutputPath(), "encode_decode_rtp");
- rtpFile.Open(fileName.c_str(), "wb+");
- rtpFile.WriteHeader();
- Sender sender;
- sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000,
- send_codec.first, send_codec.second);
- sender.Run();
- sender.Teardown();
- rtpFile.Close();
+ codePars[0] = 0;
+ codePars[1] = 0;
+ codePars[2] = 0;
- rtpFile.Open(fileName.c_str(), "rb");
- rtpFile.ReadHeader();
- Receiver receiver;
- receiver.Setup(acm.get(), &rtpFile, "encodeDecode_out", 1, file_num);
- receiver.Run();
- receiver.Teardown();
- rtpFile.Close();
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+ struct CodecInst sendCodecTmp;
+ numCodecs = acm->NumberOfCodecs();
- file_num++;
+ for (int n = 0; n < numCodecs; n++) {
+ EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
+ if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "telephone-event")) {
+ numPars[n] = 0;
+ } else if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "cn")) {
+ numPars[n] = 0;
+ } else if (absl::EqualsIgnoreCase(sendCodecTmp.plname, "red")) {
+ numPars[n] = 0;
+ } else if (sendCodecTmp.channels == 2) {
+ numPars[n] = 0;
+ } else {
+ numPars[n] = 1;
+ }
}
+
+ // Loop over all mono codecs:
+ for (int codeId = 0; codeId < numCodecs; codeId++) {
+ // Only encode using real mono encoders, not telephone-event and cng.
+ for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
+ // Encode all data to file.
+ std::string fileName = EncodeToFile(1, codeId, codePars);
+
+ RTPFile rtpFile;
+ rtpFile.Open(fileName.c_str(), "rb");
+
+ _receiver.codeId = codeId;
+
+ rtpFile.ReadHeader();
+ _receiver.Setup(acm.get(), &rtpFile, "encodeDecode_out", 1);
+ _receiver.Run();
+ _receiver.Teardown();
+ rtpFile.Close();
+ }
+ }
+}
+
+std::string EncodeDecodeTest::EncodeToFile(int fileType,
+ int codeId,
+ int* codePars) {
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+ RTPFile rtpFile;
+ std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+ "encode_decode_rtp");
+ rtpFile.Open(fileName.c_str(), "wb+");
+ rtpFile.WriteHeader();
+
+ // Store for auto_test and logging.
+ _sender.codeId = codeId;
+
+ _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
+ if (acm->SendCodec()) {
+ _sender.Run();
+ }
+ _sender.Teardown();
+ rtpFile.Close();
+
+ return fileName;
}
} // namespace webrtc
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index df6ee5f..9132d71 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -47,12 +47,13 @@
public:
Sender();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int in_sample_rate,
- int payload_type, SdpAudioFormat format);
+ std::string in_file_name, int sample_rate, size_t channels);
void Teardown();
void Run();
bool Add10MsData();
+ uint8_t codeId;
+
protected:
AudioCodingModule* _acm;
@@ -67,12 +68,15 @@
Receiver();
virtual ~Receiver() {};
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, size_t channels, int file_num);
+ std::string out_file_name, size_t channels);
void Teardown();
void Run();
virtual bool IncomingPacket();
bool PlayoutData();
+ //for auto_test and logging
+ uint8_t codeId;
+
private:
PCMFile _pcmFile;
int16_t* _playoutBuffer;
@@ -92,8 +96,17 @@
class EncodeDecodeTest {
public:
- EncodeDecodeTest();
+ explicit EncodeDecodeTest(int test_mode);
void Perform();
+
+ uint16_t _playoutFreq;
+
+ private:
+ std::string EncodeToFile(int fileType, int codeId, int* codePars);
+
+ protected:
+ Sender _sender;
+ Receiver _receiver;
};
} // namespace webrtc
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 6f87659..a1629fd 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -30,7 +30,6 @@
RTPStream* rtpStream,
std::string out_file_name,
int channels,
- int file_num,
int loss_rate,
int burst_length) {
loss_rate_ = loss_rate;
@@ -38,7 +37,7 @@
burst_lost_counter_ = burst_length_; // To prevent first packet gets lost.
rtc::StringBuilder ss;
ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
- Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num);
+ Receiver::Setup(acm, rtpStream, ss.str(), channels);
}
bool ReceiverWithPacketLoss::IncomingPacket() {
@@ -90,11 +89,10 @@
void SenderWithFEC::Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string in_file_name,
- int payload_type,
- SdpAudioFormat format,
+ int sample_rate,
+ int channels,
int expected_loss_rate) {
- Sender::Setup(acm, rtpStream, in_file_name, format.clockrate_hz, payload_type,
- format);
+ Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
EXPECT_TRUE(SetFEC(true));
EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
}
@@ -125,6 +123,8 @@
in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
: "audio_coding/teststereo32kHz"),
sample_rate_hz_(32000),
+ sender_(new SenderWithFEC),
+ receiver_(new ReceiverWithPacketLoss),
expected_loss_rate_(expected_loss_rate),
actual_loss_rate_(actual_loss_rate),
burst_length_(burst_length) {}
@@ -133,32 +133,40 @@
#ifndef WEBRTC_CODEC_OPUS
return;
#else
- RTPFile rtpFile;
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
- SdpAudioFormat send_format = SdpAudioFormat("opus", 48000, 2);
- if (channels_ == 2) {
- send_format.parameters = {{"stereo", "1"}};
- }
+ AudioCodingModule::Config config;
+ config.decoder_factory = CreateBuiltinAudioDecoderFactory();
+ std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(config));
+ int codec_id = acm->Codec("opus", 48000, channels_);
+
+ RTPFile rtpFile;
std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
"packet_loss_test");
+
+ // Encode to file
rtpFile.Open(fileName.c_str(), "wb+");
rtpFile.WriteHeader();
- SenderWithFEC sender;
- sender.Setup(acm.get(), &rtpFile, in_file_name_, 120, send_format,
+
+ sender_->codeId = codec_id;
+
+ sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
expected_loss_rate_);
- sender.Run();
- sender.Teardown();
+ if (acm->SendCodec()) {
+ sender_->Run();
+ }
+ sender_->Teardown();
rtpFile.Close();
+ // Decode to file
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
- ReceiverWithPacketLoss receiver;
- receiver.Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 15,
+
+ receiver_->codeId = codec_id;
+
+ receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
actual_loss_rate_, burst_length_);
- receiver.Run();
- receiver.Teardown();
+ receiver_->Run();
+ receiver_->Teardown();
rtpFile.Close();
#endif
}
diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index b26f6ec..6018301 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -11,6 +11,7 @@
#ifndef MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
#define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#include <memory>
#include <string>
#include "modules/audio_coding/test/EncodeDecodeTest.h"
@@ -23,7 +24,6 @@
RTPStream* rtpStream,
std::string out_file_name,
int channels,
- int file_num,
int loss_rate,
int burst_length);
bool IncomingPacket() override;
@@ -43,8 +43,8 @@
void Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string in_file_name,
- int payload_type,
- SdpAudioFormat format,
+ int sample_rate,
+ int channels,
int expected_loss_rate);
bool SetPacketLossRate(int expected_loss_rate);
bool SetFEC(bool enable_fec);
@@ -65,6 +65,8 @@
int channels_;
std::string in_file_name_;
int sample_rate_hz_;
+ std::unique_ptr<SenderWithFEC> sender_;
+ std::unique_ptr<ReceiverWithPacketLoss> receiver_;
int expected_loss_rate_;
int actual_loss_rate_;
int burst_length_;
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index aad80e8..0099b2a 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -14,11 +14,13 @@
#include <limits>
#include <string>
-#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "common_types.h" // NOLINT(build/include)
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "modules/include/module_common_types.h"
+#include "modules/audio_coding/test/utility.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/strings/string_builder.h"
@@ -33,11 +35,6 @@
// The test loops through all available mono codecs, encode at "a" sends over
// the channel, and decodes at "b".
-#define CHECK_ERROR(f) \
- do { \
- EXPECT_GE(f, 0) << "Error Calling API"; \
- } while (0)
-
namespace {
const size_t kVariableSize = std::numeric_limits<size_t>::max();
}
@@ -104,7 +101,7 @@
payload_size_ = 0;
}
-TestAllCodecs::TestAllCodecs()
+TestAllCodecs::TestAllCodecs(int test_mode)
: acm_a_(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
acm_b_(AudioCodingModule::Create(
@@ -113,6 +110,8 @@
test_count_(0),
packet_size_samples_(0),
packet_size_bytes_(0) {
+ // test_mode = 0 for silent test (auto test)
+ test_mode_ = test_mode;
}
TestAllCodecs::~TestAllCodecs() {
@@ -127,28 +126,23 @@
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
infile_a_.Open(file_name, 32000, "rb");
+ if (test_mode_ == 0) {
+ RTC_LOG(LS_INFO) << "---------- TestAllCodecs ----------";
+ }
+
acm_a_->InitializeReceiver();
acm_b_->InitializeReceiver();
- acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {0, {"PCMU", 8000, 1}},
- {110, {"PCMU", 8000, 2}},
- {8, {"PCMA", 8000, 1}},
- {118, {"PCMA", 8000, 2}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ uint8_t num_encoders = acm_a_->NumberOfCodecs();
+ CodecInst my_codec_param;
+ for (uint8_t n = 0; n < num_encoders; n++) {
+ acm_b_->Codec(n, &my_codec_param);
+ if (!strcmp(my_codec_param.plname, "opus")) {
+ my_codec_param.channels = 1;
+ }
+ acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
+ CodecInstToSdp(my_codec_param));
+ }
// Create and connect the channel
channel_a_to_b_ = new TestPack;
@@ -157,6 +151,9 @@
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
char codec_g722[] = "G722";
@@ -174,6 +171,9 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#ifdef WEBRTC_CODEC_ILBC
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
char codec_ilbc[] = "ILBC";
@@ -188,6 +188,9 @@
outfile_b_.Close();
#endif
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
char codec_isac[] = "ISAC";
@@ -202,6 +205,9 @@
outfile_b_.Close();
#endif
#ifdef WEBRTC_CODEC_ISAC
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
@@ -214,6 +220,9 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
char codec_l16[] = "L16";
@@ -226,7 +235,9 @@
RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
-
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
@@ -238,7 +249,9 @@
RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
-
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
@@ -246,7 +259,9 @@
RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
Run(channel_a_to_b_);
outfile_b_.Close();
-
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
char codec_pcma[] = "PCMA";
@@ -262,7 +277,9 @@
Run(channel_a_to_b_);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
Run(channel_a_to_b_);
-
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
char codec_pcmu[] = "PCMU";
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
Run(channel_a_to_b_);
@@ -278,6 +295,9 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#ifdef WEBRTC_CODEC_OPUS
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ }
test_count_++;
OpenOutFile(test_count_);
char codec_opus[] = "OPUS";
@@ -297,6 +317,24 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+
+ /* Print out all codecs that were not tested in the run */
+ printf("The following codecs was not included in the test:\n");
+#ifndef WEBRTC_CODEC_ILBC
+ printf(" iLBC\n");
+#endif
+#ifndef WEBRTC_CODEC_ISAC
+ printf(" ISAC float\n");
+#endif
+#ifndef WEBRTC_CODEC_ISACFX
+ printf(" ISAC fix\n");
+#endif
+
+ printf("\nTo complete the test, listen to the %d number of output files.\n",
+ test_count_);
+ }
}
// Register Codec to use in the test
@@ -316,21 +354,21 @@
int rate,
int packet_size,
size_t extra_byte) {
+ if (test_mode_ != 0) {
+ // Print out codec and settings.
+ printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
+ sampling_freq_hz, rate, packet_size);
+ }
+
// Store packet-size in samples, used to validate the received packet.
// If G.722, store half the size to compensate for the timestamp bug in the
// RFC for G.722.
// If iSAC runs in adaptive mode, packet size in samples can change on the
// fly, so we exclude this test by setting |packet_size_samples_| to -1.
- int clockrate_hz = sampling_freq_hz;
- size_t num_channels = 1;
- if (absl::EqualsIgnoreCase(codec_name, "G722")) {
+ if (!strcmp(codec_name, "G722")) {
packet_size_samples_ = packet_size / 2;
- clockrate_hz = sampling_freq_hz / 2;
- } else if (absl::EqualsIgnoreCase(codec_name, "ISAC") && (rate == -1)) {
+ } else if (!strcmp(codec_name, "ISAC") && (rate == -1)) {
packet_size_samples_ = -1;
- } else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) {
- packet_size_samples_ = packet_size;
- num_channels = 2;
} else {
packet_size_samples_ = packet_size;
}
@@ -364,9 +402,16 @@
}
ASSERT_TRUE(my_acm != NULL);
+ // Get all codec parameters before registering
+ CodecInst my_codec_param;
+ CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
+ sampling_freq_hz, 1));
+ my_codec_param.rate = rate;
+ my_codec_param.pacsize = packet_size;
+
auto factory = CreateBuiltinAudioEncoderFactory();
constexpr int payload_type = 17;
- SdpAudioFormat format = { codec_name, clockrate_hz, num_channels };
+ SdpAudioFormat format = CodecInstToSdp(my_codec_param);
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
my_acm->SetEncoder(
@@ -440,4 +485,11 @@
outfile_b_.Open(filename, 32000, "wb");
}
+void TestAllCodecs::DisplaySendReceiveCodec() {
+ CodecInst my_codec_param;
+ printf("%s -> ", acm_a_->SendCodec()->plname);
+ acm_b_->ReceiveCodec(&my_codec_param);
+ printf("%s\n", my_codec_param.plname);
+}
+
} // namespace webrtc
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index 3125efe..669a118 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -13,7 +13,7 @@
#include <memory>
-#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
namespace webrtc {
@@ -48,7 +48,7 @@
class TestAllCodecs {
public:
- TestAllCodecs();
+ explicit TestAllCodecs(int test_mode);
~TestAllCodecs();
void Perform();
@@ -67,7 +67,9 @@
void Run(TestPack* channel);
void OpenOutFile(int test_number);
+ void DisplaySendReceiveCodec();
+ int test_mode_;
std::unique_ptr<AudioCodingModule> acm_a_;
std::unique_ptr<AudioCodingModule> acm_b_;
TestPack* channel_a_to_b_;
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index ac51666..8bb3971 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -10,9 +10,8 @@
#include "modules/audio_coding/test/TestRedFec.h"
-#include <utility>
+#include <assert.h>
-#include "absl/strings/match.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_decoder_factory_template.h"
@@ -25,11 +24,12 @@
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/utility.h"
#include "rtc_base/strings/string_builder.h"
-#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
namespace webrtc {
@@ -173,7 +173,6 @@
auto encoder = encoder_factory_->MakeAudioEncoder(payload_type, codec_format,
absl::nullopt);
EXPECT_NE(encoder, nullptr);
- std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
if (!absl::EqualsIgnoreCase(codec_format.name, "opus")) {
if (vad_mode.has_value()) {
AudioEncoderCngConfig config;
@@ -182,22 +181,22 @@
config.payload_type = cn_payload_type;
config.vad_mode = vad_mode.value();
encoder = CreateComfortNoiseEncoder(std::move(config));
- receive_codecs.emplace(
- std::make_pair(cn_payload_type,
- SdpAudioFormat("CN", codec_format.clockrate_hz, 1)));
+ EXPECT_EQ(true,
+ other_acm->RegisterReceiveCodec(
+ cn_payload_type, {"CN", codec_format.clockrate_hz, 1}));
}
if (use_red) {
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type;
config.speech_encoder = std::move(encoder);
encoder = absl::make_unique<AudioEncoderCopyRed>(std::move(config));
- receive_codecs.emplace(
- std::make_pair(red_payload_type,
- SdpAudioFormat("red", codec_format.clockrate_hz, 1)));
+ EXPECT_EQ(true,
+ other_acm->RegisterReceiveCodec(
+ red_payload_type, {"red", codec_format.clockrate_hz, 1}));
}
}
acm->SetEncoder(std::move(encoder));
- other_acm->SetReceiveCodecs(receive_codecs);
+ EXPECT_EQ(true, other_acm->RegisterReceiveCodec(payload_type, codec_format));
}
void TestRedFec::Run() {
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index dd4be6a..bf8e189 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -15,8 +15,9 @@
#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "modules/include/module_common_types.h"
+#include "modules/audio_coding/test/utility.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@@ -31,6 +32,7 @@
last_in_timestamp_(0),
total_bytes_(0),
payload_size_(0),
+ codec_mode_(kNotSet),
lost_packet_(false) {}
TestPackStereo::~TestPackStereo() {}
@@ -96,7 +98,7 @@
lost_packet_ = lost;
}
-TestStereo::TestStereo()
+TestStereo::TestStereo(int test_mode)
: acm_a_(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
acm_b_(AudioCodingModule::Create(
@@ -106,6 +108,8 @@
pack_size_samp_(0),
pack_size_bytes_(0),
counter_(0) {
+ // test_mode = 0 for silent test (auto test)
+ test_mode_ = test_mode;
}
TestStereo::~TestStereo() {
@@ -138,25 +142,27 @@
EXPECT_EQ(0, acm_a_->InitializeReceiver());
EXPECT_EQ(0, acm_b_->InitializeReceiver());
- acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}},
- {104, {"ISAC", 32000, 1}},
- {107, {"L16", 8000, 1}},
- {108, {"L16", 16000, 1}},
- {109, {"L16", 32000, 1}},
- {111, {"L16", 8000, 2}},
- {112, {"L16", 16000, 2}},
- {113, {"L16", 32000, 2}},
- {0, {"PCMU", 8000, 1}},
- {110, {"PCMU", 8000, 2}},
- {8, {"PCMA", 8000, 1}},
- {118, {"PCMA", 8000, 2}},
- {102, {"ILBC", 8000, 1}},
- {9, {"G722", 8000, 1}},
- {119, {"G722", 8000, 2}},
- {120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
- {13, {"CN", 8000, 1}},
- {98, {"CN", 16000, 1}},
- {99, {"CN", 32000, 1}}});
+ // Register all available codes as receiving codecs.
+ uint8_t num_encoders = acm_a_->NumberOfCodecs();
+ CodecInst my_codec_param;
+ for (uint8_t n = 0; n < num_encoders; n++) {
+ EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
+ EXPECT_EQ(true, acm_b_->RegisterReceiveCodec(
+ my_codec_param.pltype, CodecInstToSdp(my_codec_param)));
+ }
+
+ // Test that unregister all receive codecs works.
+ for (uint8_t n = 0; n < num_encoders; n++) {
+ EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
+ EXPECT_EQ(0, acm_b_->UnregisterReceiveCodec(my_codec_param.pltype));
+ }
+
+ // Register all available codes as receiving codecs once more.
+ for (uint8_t n = 0; n < num_encoders; n++) {
+ EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
+ EXPECT_EQ(true, acm_b_->RegisterReceiveCodec(
+ my_codec_param.pltype, CodecInstToSdp(my_codec_param)));
+ }
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
@@ -165,6 +171,9 @@
char codec_pcma_temp[] = "PCMA";
RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2);
+ if (test_mode_ != 0) {
+ printf("\n");
+ }
//
// Test Stereo-To-Stereo for all codecs.
@@ -174,6 +183,11 @@
// All codecs are tested for all allowed sampling frequencies, rates and
// packet sizes.
+ if (test_mode_ != 0) {
+ printf("===========================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ }
channel_a2b_->set_codec_mode(kStereo);
test_cntr_++;
OpenOutFile(test_cntr_);
@@ -192,6 +206,11 @@
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
+ if (test_mode_ != 0) {
+ printf("===========================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ }
channel_a2b_->set_codec_mode(kStereo);
test_cntr_++;
OpenOutFile(test_cntr_);
@@ -206,6 +225,11 @@
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
+ if (test_mode_ != 0) {
+ printf("===========================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
@@ -218,6 +242,11 @@
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
+ if (test_mode_ != 0) {
+ printf("===========================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
@@ -226,6 +255,11 @@
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#ifdef PCMA_AND_PCMU
+ if (test_mode_ != 0) {
+ printf("===========================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ }
channel_a2b_->set_codec_mode(kStereo);
audio_channels = 2;
codec_channels = 2;
@@ -244,8 +278,13 @@
Run(channel_a2b_, audio_channels, codec_channels);
RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
- out_file_.Close();
+ out_file_.Close();
+ if (test_mode_ != 0) {
+ printf("===========================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
char codec_pcmu[] = "PCMU";
@@ -264,6 +303,11 @@
out_file_.Close();
#endif
#ifdef WEBRTC_CODEC_OPUS
+ if (test_mode_ != 0) {
+ printf("===========================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-stereo\n");
+ }
channel_a2b_->set_codec_mode(kStereo);
audio_channels = 2;
codec_channels = 2;
@@ -296,6 +340,11 @@
audio_channels = 1;
codec_channels = 2;
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
OpenOutFile(test_cntr_);
@@ -303,25 +352,43 @@
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
-
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
-
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#ifdef PCMA_AND_PCMU
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
OpenOutFile(test_cntr_);
@@ -332,6 +399,12 @@
out_file_.Close();
#endif
#ifdef WEBRTC_CODEC_OPUS
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Mono-to-stereo\n");
+ }
+
// Keep encode and decode in stereo.
test_cntr_++;
channel_a2b_->set_codec_mode(kStereo);
@@ -353,30 +426,54 @@
channel_a2b_->set_codec_mode(kMono);
// Run stereo audio and mono codec.
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels);
+
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
-
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
-
+ if (test_mode_ != 0) {
+ printf("==============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels);
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
#ifdef PCMA_AND_PCMU
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels);
@@ -386,11 +483,26 @@
out_file_.Close();
#endif
#ifdef WEBRTC_CODEC_OPUS
+ if (test_mode_ != 0) {
+ printf("===============================================================\n");
+ printf("Test number: %d\n", test_cntr_ + 1);
+ printf("Test type: Stereo-to-mono\n");
+ }
test_cntr_++;
OpenOutFile(test_cntr_);
// Encode and decode in mono.
RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels);
- acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
+ CodecInst opus_codec_param;
+ for (uint8_t n = 0; n < num_encoders; n++) {
+ EXPECT_EQ(0, acm_b_->Codec(n, &opus_codec_param));
+ if (!strcmp(opus_codec_param.plname, "opus")) {
+ opus_codec_param.channels = 1;
+ EXPECT_EQ(true,
+ acm_b_->RegisterReceiveCodec(opus_codec_param.pltype,
+ CodecInstToSdp(opus_codec_param)));
+ break;
+ }
+ }
Run(channel_a2b_, audio_channels, codec_channels);
// Encode in stereo, decode in mono.
@@ -404,22 +516,65 @@
// Decode in mono.
test_cntr_++;
OpenOutFile(test_cntr_);
+ if (test_mode_ != 0) {
+ // Print out codec and settings
+ printf(
+ "Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+ " Decode: mono\n",
+ test_cntr_);
+ }
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
// Decode in stereo.
test_cntr_++;
OpenOutFile(test_cntr_);
- acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
+ if (test_mode_ != 0) {
+ // Print out codec and settings
+ printf(
+ "Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+ " Decode: stereo\n",
+ test_cntr_);
+ }
+ opus_codec_param.channels = 2;
+ EXPECT_EQ(true,
+ acm_b_->RegisterReceiveCodec(opus_codec_param.pltype,
+ CodecInstToSdp(opus_codec_param)));
Run(channel_a2b_, audio_channels, 2);
out_file_.Close();
// Decode in mono.
test_cntr_++;
OpenOutFile(test_cntr_);
- acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
+ if (test_mode_ != 0) {
+ // Print out codec and settings
+ printf(
+ "Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+ " Decode: mono\n",
+ test_cntr_);
+ }
+ opus_codec_param.channels = 1;
+ EXPECT_EQ(true,
+ acm_b_->RegisterReceiveCodec(opus_codec_param.pltype,
+ CodecInstToSdp(opus_codec_param)));
Run(channel_a2b_, audio_channels, codec_channels);
out_file_.Close();
+
#endif
+ // Print out which codecs were tested, and which were not, in the run.
+ if (test_mode_ != 0) {
+ printf("\nThe following codecs was INCLUDED in the test:\n");
+ printf(" G.722\n");
+ printf(" PCM16\n");
+ printf(" G.711\n");
+#ifdef WEBRTC_CODEC_OPUS
+ printf(" Opus\n");
+#endif
+ printf(
+ "\nTo complete the test, listen to the %d number of output "
+ "files.\n",
+ test_cntr_);
+ }
+
// Delete the file pointers.
delete in_file_stereo_;
delete in_file_mono_;
@@ -439,6 +594,12 @@
int rate,
int pack_size,
int channels) {
+ if (test_mode_ != 0) {
+ // Print out codec and settings
+ printf("Codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
+ sampling_freq_hz, rate, pack_size);
+ }
+
// Store packet size in samples, used to validate the received packet
pack_size_samp_ = pack_size;
@@ -558,7 +719,7 @@
}
}
- // Run receive side of ACM
+ // Run received side of ACM
bool muted;
EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
ASSERT_FALSE(muted);
@@ -597,4 +758,17 @@
out_file_.Open(file_name, 32000, "wb");
}
+void TestStereo::DisplaySendReceiveCodec() {
+ auto send_codec = acm_a_->SendCodec();
+ if (test_mode_ != 0) {
+ ASSERT_TRUE(send_codec);
+ printf("%s -> ", send_codec->plname);
+ }
+ CodecInst receive_codec;
+ acm_b_->ReceiveCodec(&receive_codec);
+ if (test_mode_ != 0) {
+ printf("%s\n", receive_codec.plname);
+ }
+}
+
} // namespace webrtc
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index da10bf1..0d80631 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -15,7 +15,7 @@
#include <memory>
-#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
#define PCMA_AND_PCMU
@@ -58,7 +58,7 @@
class TestStereo {
public:
- TestStereo();
+ explicit TestStereo(int test_mode);
~TestStereo();
void Perform();
@@ -79,6 +79,9 @@
int out_channels,
int percent_loss = 0);
void OpenOutFile(int16_t test_number);
+ void DisplaySendReceiveCodec();
+
+ int test_mode_;
std::unique_ptr<AudioCodingModule> acm_a_;
std::unique_ptr<AudioCodingModule> acm_b_;
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index 8e16280..09c69f9 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -23,8 +23,8 @@
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
#include "rtc_base/strings/string_builder.h"
-#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
namespace webrtc {
@@ -94,9 +94,8 @@
channel_->SetIsStereo(encoder->NumChannels() > 1);
acm_send_->SetEncoder(std::move(encoder));
- std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
- acm_receive_->SetReceiveCodecs(receive_codecs);
-
+ EXPECT_EQ(true,
+ acm_receive_->RegisterReceiveCodec(payload_type, codec_format));
return added_comfort_noise;
}
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index f2358e7..68b2c1e 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -16,6 +16,7 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "common_audio/vad/include/vad.h"
+#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/test/Channel.h"
diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 750656f..85926f1 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -25,8 +25,12 @@
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
+// This parameter is used to describe how to run the tests. It is normally
+// set to 0, and all tests are run in quite mode.
+#define ACM_TEST_MODE 0
+
TEST(AudioCodingModuleTest, TestAllCodecs) {
- webrtc::TestAllCodecs().Perform();
+ webrtc::TestAllCodecs(ACM_TEST_MODE).Perform();
}
#if defined(WEBRTC_ANDROID)
@@ -34,7 +38,7 @@
#else
TEST(AudioCodingModuleTest, TestEncodeDecode) {
#endif
- webrtc::EncodeDecodeTest().Perform();
+ webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
}
TEST(AudioCodingModuleTest, TestRedFec) {
@@ -46,7 +50,7 @@
#else
TEST(AudioCodingModuleTest, TestIsac) {
#endif
- webrtc::ISACTest().Perform();
+ webrtc::ISACTest(ACM_TEST_MODE).Perform();
}
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
@@ -66,7 +70,7 @@
#else
TEST(AudioCodingModuleTest, TestStereo) {
#endif
- webrtc::TestStereo().Perform();
+ webrtc::TestStereo(ACM_TEST_MODE).Perform();
}
TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index 3af114e..2079a94 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -16,9 +16,15 @@
#include <memory>
+#ifdef WIN32
+#include <Windows.h>
+#endif
+
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@@ -59,26 +65,25 @@
const int payload_type1,
const SdpAudioFormat& format2,
const int payload_type2) {
-
//--- Set A codecs
_acmA->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
- _acmA->SetReceiveCodecs({{payload_type2, format2}});
+ EXPECT_EQ(true, _acmA->RegisterReceiveCodec(payload_type2, format2));
//--- Set ref-A codecs
_acmRefA->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
- _acmRefA->SetReceiveCodecs({{payload_type2, format2}});
+ EXPECT_EQ(true, _acmRefA->RegisterReceiveCodec(payload_type2, format2));
//--- Set B codecs
_acmB->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
- _acmB->SetReceiveCodecs({{payload_type1, format1}});
+ EXPECT_EQ(true, _acmB->RegisterReceiveCodec(payload_type1, format1));
//--- Set ref-B codecs
_acmRefB->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
- _acmRefB->SetReceiveCodecs({{payload_type1, format1}});
+ EXPECT_EQ(true, _acmRefB->RegisterReceiveCodec(payload_type1, format1));
uint16_t frequencyHz;
@@ -179,13 +184,14 @@
if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
_acmB->SetEncoder(encoder_factory->MakeAudioEncoder(
payload_type2, format2, absl::nullopt));
+ EXPECT_TRUE(_acmB->SendCodec());
}
// Initialize receiver on side A.
if (((secPassed % 7) == 6) && (msecPassed == 0))
EXPECT_EQ(0, _acmA->InitializeReceiver());
// Re-register codec on side A.
if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
- _acmA->SetReceiveCodecs({{payload_type2, format2}});
+ EXPECT_EQ(true, _acmA->RegisterReceiveCodec(payload_type2, format2));
}
}
}
diff --git a/modules/audio_coding/test/TwoWayCommunication.h b/modules/audio_coding/test/TwoWayCommunication.h
index 7d0cdb9..a679732 100644
--- a/modules/audio_coding/test/TwoWayCommunication.h
+++ b/modules/audio_coding/test/TwoWayCommunication.h
@@ -18,6 +18,7 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index 339d419..c332fe0 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -14,9 +14,20 @@
#include <stdio.h>
#include <string.h>
+#ifdef _WIN32
+#include <windows.h>
+#elif defined(WEBRTC_LINUX)
+#include <time.h>
+#else
+#include <sys/time.h>
+#include <time.h>
+#endif
+
#include "absl/strings/match.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/test/utility.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/sleep.h"
@@ -32,10 +43,15 @@
namespace {
-constexpr int kISAC16kPayloadType = 103;
-constexpr int kISAC32kPayloadType = 104;
-const SdpAudioFormat kISAC16kFormat = { "ISAC", 16000, 1 };
-const SdpAudioFormat kISAC32kFormat = { "ISAC", 32000, 1 };
+AudioEncoderIsacFloat::Config MakeConfig(const CodecInst& ci) {
+ EXPECT_THAT(ci.plname, StrCaseEq("ISAC"));
+ EXPECT_THAT(ci.plfreq, AnyOf(Eq(16000), Eq(32000)));
+ EXPECT_THAT(ci.channels, Eq(1u));
+ AudioEncoderIsacFloat::Config config;
+ config.sample_rate_hz = ci.plfreq;
+ EXPECT_THAT(config.IsOk(), Eq(true));
+ return config;
+}
AudioEncoderIsacFloat::Config TweakConfig(
AudioEncoderIsacFloat::Config config,
@@ -61,96 +77,43 @@
} // namespace
-ISACTest::ACMTestTimer::ACMTestTimer() : _msec(0), _sec(0), _min(0), _hour(0) {
- return;
-}
-
-ISACTest::ACMTestTimer::~ACMTestTimer() {
- return;
-}
-
-void ISACTest::ACMTestTimer::Reset() {
- _msec = 0;
- _sec = 0;
- _min = 0;
- _hour = 0;
- return;
-}
-void ISACTest::ACMTestTimer::Tick10ms() {
- _msec += 10;
- Adjust();
- return;
-}
-
-void ISACTest::ACMTestTimer::Tick1ms() {
- _msec++;
- Adjust();
- return;
-}
-
-void ISACTest::ACMTestTimer::Tick100ms() {
- _msec += 100;
- Adjust();
- return;
-}
-
-void ISACTest::ACMTestTimer::Tick1sec() {
- _sec++;
- Adjust();
- return;
-}
-
-void ISACTest::ACMTestTimer::CurrentTimeHMS(char* currTime) {
- sprintf(currTime, "%4lu:%02u:%06.3f", _hour, _min,
- (double)_sec + (double)_msec / 1000.);
- return;
-}
-
-void ISACTest::ACMTestTimer::CurrentTime(unsigned long& h,
- unsigned char& m,
- unsigned char& s,
- unsigned short& ms) {
- h = _hour;
- m = _min;
- s = _sec;
- ms = _msec;
- return;
-}
-
-void ISACTest::ACMTestTimer::Adjust() {
- unsigned int n;
- if (_msec >= 1000) {
- n = _msec / 1000;
- _msec -= (1000 * n);
- _sec += n;
- }
- if (_sec >= 60) {
- n = _sec / 60;
- _sec -= (n * 60);
- _min += n;
- }
- if (_min >= 60) {
- n = _min / 60;
- _min -= (n * 60);
- _hour += n;
- }
-}
-
-ISACTest::ISACTest()
+ISACTest::ISACTest(int testMode)
: _acmA(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
_acmB(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {}
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
+ _testMode(testMode) {}
ISACTest::~ISACTest() {}
void ISACTest::Setup() {
+ int codecCntr;
+ CodecInst codecParam;
+
+ for (codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs();
+ codecCntr++) {
+ EXPECT_EQ(0, AudioCodingModule::Codec(codecCntr, &codecParam));
+ if (absl::EqualsIgnoreCase(codecParam.plname, "ISAC") &&
+ codecParam.plfreq == 16000) {
+ memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
+ _idISAC16kHz = codecCntr;
+ }
+ if (absl::EqualsIgnoreCase(codecParam.plname, "ISAC") &&
+ codecParam.plfreq == 32000) {
+ memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
+ _idISAC32kHz = codecCntr;
+ }
+ }
+
// Register both iSAC-wb & iSAC-swb in both sides as receiver codecs.
- std::map<int, SdpAudioFormat> receive_codecs =
- {{kISAC16kPayloadType, kISAC16kFormat},
- {kISAC32kPayloadType, kISAC32kFormat}};
- _acmA->SetReceiveCodecs(receive_codecs);
- _acmB->SetReceiveCodecs(receive_codecs);
+ EXPECT_EQ(true, _acmA->RegisterReceiveCodec(_paramISAC16kHz.pltype,
+ CodecInstToSdp(_paramISAC16kHz)));
+ EXPECT_EQ(true, _acmA->RegisterReceiveCodec(_paramISAC32kHz.pltype,
+ CodecInstToSdp(_paramISAC32kHz)));
+ EXPECT_EQ(true, _acmB->RegisterReceiveCodec(_paramISAC16kHz.pltype,
+ CodecInstToSdp(_paramISAC16kHz)));
+ EXPECT_EQ(true, _acmB->RegisterReceiveCodec(_paramISAC32kHz.pltype,
+ CodecInstToSdp(_paramISAC32kHz)));
//--- Set A-to-B channel
_channel_A2B.reset(new Channel);
@@ -165,14 +128,10 @@
file_name_swb_ =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
_inFileA.Open(file_name_swb_, 32000, "rb");
// Set test length to 500 ms (50 blocks of 10 ms each).
@@ -187,9 +146,9 @@
while (!_inFileA.EndOfFile()) {
Run10ms();
}
-
- EXPECT_TRUE(_acmA->ReceiveFormat());
- EXPECT_TRUE(_acmB->ReceiveFormat());
+ CodecInst receiveCodec;
+ EXPECT_EQ(0, _acmA->ReceiveCodec(&receiveCodec));
+ EXPECT_EQ(0, _acmB->ReceiveCodec(&receiveCodec));
_inFileA.Close();
_outFileA.Close();
@@ -211,13 +170,45 @@
testNr++;
EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+ if (_testMode != 0) {
+ SetISACConfigDefault(wbISACConfig);
+ SetISACConfigDefault(swbISACConfig);
+
+ wbISACConfig.currentRateBitPerSec = -1;
+ swbISACConfig.currentRateBitPerSec = -1;
+ wbISACConfig.initRateBitPerSec = 13000;
+ wbISACConfig.initFrameSizeInMsec = 60;
+ swbISACConfig.initRateBitPerSec = 20000;
+ swbISACConfig.initFrameSizeInMsec = 30;
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+ SetISACConfigDefault(wbISACConfig);
+ SetISACConfigDefault(swbISACConfig);
+
+ wbISACConfig.currentRateBitPerSec = 20000;
+ swbISACConfig.currentRateBitPerSec = 48000;
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+ wbISACConfig.currentRateBitPerSec = 16000;
+ swbISACConfig.currentRateBitPerSec = 30000;
+ wbISACConfig.currentFrameSizeMsec = 60;
+ testNr++;
+ EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+ }
+
SetISACConfigDefault(wbISACConfig);
SetISACConfigDefault(swbISACConfig);
testNr++;
EncodeDecode(testNr, wbISACConfig, swbISACConfig);
testNr++;
- SwitchingSamplingRate(testNr, 4);
+ if (_testMode == 0) {
+ SwitchingSamplingRate(testNr, 4);
+ } else {
+ SwitchingSamplingRate(testNr, 80);
+ }
}
void ISACTest::Run10ms() {
@@ -254,16 +245,12 @@
_outFileB.Open(file_name_out, 32000, "wb");
// Side A is sending super-wideband, and side B is sending wideband.
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- swbISACConfig),
- kISAC32kPayloadType));
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- wbISACConfig),
- kISAC16kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ TweakConfig(MakeConfig(_paramISAC32kHz), swbISACConfig),
+ _paramISAC32kHz.pltype));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ TweakConfig(MakeConfig(_paramISAC16kHz), wbISACConfig),
+ _paramISAC16kHz.pltype));
bool adaptiveMode = false;
if ((swbISACConfig.currentRateBitPerSec == -1) ||
@@ -275,10 +262,30 @@
_channel_B2A->ResetStats();
char currentTime[500];
+ int64_t time_ms = rtc::TimeMillis();
while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
Run10ms();
_myTimer.Tick10ms();
_myTimer.CurrentTimeHMS(currentTime);
+
+ if ((adaptiveMode) && (_testMode != 0)) {
+ time_ms += 10;
+ int64_t time_left_ms = time_ms - rtc::TimeMillis();
+ if (time_left_ms > 0) {
+ SleepMs(time_left_ms);
+ }
+
+ EXPECT_TRUE(_acmA->SendCodec());
+ EXPECT_TRUE(_acmB->SendCodec());
+ }
+ }
+
+ if (_testMode != 0) {
+ printf("\n\nSide A statistics\n\n");
+ _channel_A2B->PrintStats(_paramISAC32kHz);
+
+ printf("\n\nSide B statistics\n\n");
+ _channel_B2A->PrintStats(_paramISAC16kHz);
}
_channel_A2B->ResetStats();
@@ -309,14 +316,10 @@
// Start with side A sending super-wideband and side B seding wideband.
// Toggle sending wideband/super-wideband in this test.
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
int numSendCodecChanged = 0;
_myTimer.Reset();
@@ -325,23 +328,21 @@
Run10ms();
_myTimer.Tick10ms();
_myTimer.CurrentTimeHMS(currentTime);
+ if (_testMode == 2)
+ printf("\r%s", currentTime);
if (_inFileA.EndOfFile()) {
if (_inFileA.SamplingFrequency() == 16000) {
// Switch side A to send super-wideband.
_inFileA.Close();
_inFileA.Open(file_name_swb_, 32000, "rb");
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
} else {
// Switch side A to send wideband.
_inFileA.Close();
_inFileA.Open(file_name_swb_, 32000, "rb");
- _acmA->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
+ _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
}
numSendCodecChanged++;
}
@@ -351,18 +352,14 @@
// Switch side B to send super-wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
- kISAC32kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
} else {
// Switch side B to send wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
- _acmB->SetEncoder(
- AudioEncoderIsacFloat::MakeAudioEncoder(
- *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
- kISAC16kPayloadType));
+ _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
+ MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
}
numSendCodecChanged++;
}
diff --git a/modules/audio_coding/test/iSACTest.h b/modules/audio_coding/test/iSACTest.h
index e000476..0b140b6 100644
--- a/modules/audio_coding/test/iSACTest.h
+++ b/modules/audio_coding/test/iSACTest.h
@@ -15,9 +15,14 @@
#include <memory>
+#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+#define NO_OF_CLIENTS 15
namespace webrtc {
@@ -32,37 +37,12 @@
class ISACTest {
public:
- ISACTest();
+ explicit ISACTest(int testMode);
~ISACTest();
void Perform();
private:
- class ACMTestTimer {
- public:
- ACMTestTimer();
- ~ACMTestTimer();
-
- void Reset();
- void Tick10ms();
- void Tick1ms();
- void Tick100ms();
- void Tick1sec();
- void CurrentTimeHMS(char* currTime);
- void CurrentTime(unsigned long& h,
- unsigned char& m,
- unsigned char& s,
- unsigned short& ms);
-
- private:
- void Adjust();
-
- unsigned short _msec;
- unsigned char _sec;
- unsigned char _min;
- unsigned long _hour;
- };
-
void Setup();
void Run10ms();
@@ -85,9 +65,15 @@
PCMFile _outFileA;
PCMFile _outFileB;
+ uint8_t _idISAC16kHz;
+ uint8_t _idISAC32kHz;
+ CodecInst _paramISAC16kHz;
+ CodecInst _paramISAC32kHz;
+
std::string file_name_swb_;
ACMTestTimer _myTimer;
+ int _testMode;
};
} // namespace webrtc
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 1e24e5d..954b183 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -13,9 +13,12 @@
#include <string>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h" // NOLINT(build/include)
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "modules/audio_coding/test/TestStereo.h"
+#include "modules/audio_coding/test/utility.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@@ -86,10 +89,13 @@
EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
// Register Opus stereo as receiving codec.
- constexpr int kOpusPayloadType = 120;
- const SdpAudioFormat kOpusFormatStereo("opus", 48000, 2, {{"stereo", "1"}});
- payload_type_ = kOpusPayloadType;
- acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatStereo}});
+ CodecInst opus_codec_param;
+ int codec_id = acm_receiver_->Codec("opus", 48000, 2);
+ EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
+ payload_type_ = opus_codec_param.pltype;
+ EXPECT_EQ(true,
+ acm_receiver_->RegisterReceiveCodec(
+ opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
@@ -153,8 +159,10 @@
OpenOutFile(test_cntr);
// Register Opus mono as receiving codec.
- const SdpAudioFormat kOpusFormatMono("opus", 48000, 2);
- acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatMono}});
+ opus_codec_param.channels = 1;
+ EXPECT_EQ(true,
+ acm_receiver_->RegisterReceiveCodec(
+ opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
// Run Opus with 2.5 ms frame size.
Run(channel_a2b_, audio_channels, 32000, 120);
diff --git a/modules/audio_coding/test/opus_test.h b/modules/audio_coding/test/opus_test.h
index c69f922..019e96b 100644
--- a/modules/audio_coding/test/opus_test.h
+++ b/modules/audio_coding/test/opus_test.h
@@ -17,6 +17,7 @@
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "modules/audio_coding/test/TestStereo.h"
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 8d82b6e..071a6d8 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -12,8 +12,10 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/utility.h"
#include "modules/include/module_common_types.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@@ -33,9 +35,8 @@
ASSERT_EQ(0, acm_->InitializeReceiver());
constexpr int pltype = 108;
- std::map<int, SdpAudioFormat> receive_codecs =
- {{pltype, {"L16", kSampleRateHz, 1}}};
- acm_->SetReceiveCodecs(receive_codecs);
+ ASSERT_EQ(true,
+ acm_->RegisterReceiveCodec(pltype, {"L16", kSampleRateHz, 1}));
rtp_info_.header.payloadType = pltype;
rtp_info_.header.timestamp = 0;
diff --git a/modules/audio_coding/test/utility.cc b/modules/audio_coding/test/utility.cc
new file mode 100644
index 0000000..53f8077
--- /dev/null
+++ b/modules/audio_coding/test/utility.cc
@@ -0,0 +1,299 @@
+/*
+ * Copyright (c) 2011 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 "utility.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "test/gtest.h"
+
+#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
+
+namespace webrtc {
+
+ACMTestTimer::ACMTestTimer() : _msec(0), _sec(0), _min(0), _hour(0) {
+ return;
+}
+
+ACMTestTimer::~ACMTestTimer() {
+ return;
+}
+
+void ACMTestTimer::Reset() {
+ _msec = 0;
+ _sec = 0;
+ _min = 0;
+ _hour = 0;
+ return;
+}
+void ACMTestTimer::Tick10ms() {
+ _msec += 10;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::Tick1ms() {
+ _msec++;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::Tick100ms() {
+ _msec += 100;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::Tick1sec() {
+ _sec++;
+ Adjust();
+ return;
+}
+
+void ACMTestTimer::CurrentTimeHMS(char* currTime) {
+ sprintf(currTime, "%4lu:%02u:%06.3f", _hour, _min,
+ (double)_sec + (double)_msec / 1000.);
+ return;
+}
+
+void ACMTestTimer::CurrentTime(unsigned long& h,
+ unsigned char& m,
+ unsigned char& s,
+ unsigned short& ms) {
+ h = _hour;
+ m = _min;
+ s = _sec;
+ ms = _msec;
+ return;
+}
+
+void ACMTestTimer::Adjust() {
+ unsigned int n;
+ if (_msec >= 1000) {
+ n = _msec / 1000;
+ _msec -= (1000 * n);
+ _sec += n;
+ }
+ if (_sec >= 60) {
+ n = _sec / 60;
+ _sec -= (n * 60);
+ _min += n;
+ }
+ if (_min >= 60) {
+ n = _min / 60;
+ _min -= (n * 60);
+ _hour += n;
+ }
+}
+
+int16_t ChooseCodec(CodecInst& codecInst) {
+ PrintCodecs();
+ // AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
+ uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+ int8_t codecID;
+ bool outOfRange = false;
+ char myStr[15] = "";
+ do {
+ printf("\nChoose a codec [0]: ");
+ EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+ codecID = atoi(myStr);
+ if ((codecID < 0) || (codecID >= noCodec)) {
+ printf("\nOut of range.\n");
+ outOfRange = true;
+ }
+ } while (outOfRange);
+
+ CHECK_ERROR(AudioCodingModule::Codec((uint8_t)codecID, &codecInst));
+ return 0;
+}
+
+void PrintCodecs() {
+ uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+
+ CodecInst codecInst;
+ printf("No Name [Hz] [bps]\n");
+ for (uint8_t codecCntr = 0; codecCntr < noCodec; codecCntr++) {
+ AudioCodingModule::Codec(codecCntr, &codecInst);
+ printf("%2d- %-18s %5d %6d\n", codecCntr, codecInst.plname,
+ codecInst.plfreq, codecInst.rate);
+ }
+}
+
+namespace test {
+
+CircularBuffer::CircularBuffer(uint32_t len)
+ : _buff(NULL),
+ _idx(0),
+ _buffIsFull(false),
+ _calcAvg(false),
+ _calcVar(false),
+ _sum(0),
+ _sumSqr(0) {
+ _buff = new double[len];
+ if (_buff == NULL) {
+ _buffLen = 0;
+ } else {
+ for (uint32_t n = 0; n < len; n++) {
+ _buff[n] = 0;
+ }
+ _buffLen = len;
+ }
+}
+
+CircularBuffer::~CircularBuffer() {
+ if (_buff != NULL) {
+ delete[] _buff;
+ _buff = NULL;
+ }
+}
+
+void CircularBuffer::Update(const double newVal) {
+ assert(_buffLen > 0);
+
+ // store the value that is going to be overwritten
+ double oldVal = _buff[_idx];
+ // record the new value
+ _buff[_idx] = newVal;
+ // increment the index, to point to where we would
+ // write next
+ _idx++;
+ // it is a circular buffer, if we are at the end
+ // we have to cycle to the beginning
+ if (_idx >= _buffLen) {
+ // flag that the buffer is filled up.
+ _buffIsFull = true;
+ _idx = 0;
+ }
+
+ // Update
+
+ if (_calcAvg) {
+ // for the average we have to update
+ // the sum
+ _sum += (newVal - oldVal);
+ }
+
+ if (_calcVar) {
+ // to calculate variance we have to update
+ // the sum of squares
+ _sumSqr += (double)(newVal - oldVal) * (double)(newVal + oldVal);
+ }
+}
+
+void CircularBuffer::SetArithMean(bool enable) {
+ assert(_buffLen > 0);
+
+ if (enable && !_calcAvg) {
+ uint32_t lim;
+ if (_buffIsFull) {
+ lim = _buffLen;
+ } else {
+ lim = _idx;
+ }
+ _sum = 0;
+ for (uint32_t n = 0; n < lim; n++) {
+ _sum += _buff[n];
+ }
+ }
+ _calcAvg = enable;
+}
+
+void CircularBuffer::SetVariance(bool enable) {
+ assert(_buffLen > 0);
+
+ if (enable && !_calcVar) {
+ uint32_t lim;
+ if (_buffIsFull) {
+ lim = _buffLen;
+ } else {
+ lim = _idx;
+ }
+ _sumSqr = 0;
+ for (uint32_t n = 0; n < lim; n++) {
+ _sumSqr += _buff[n] * _buff[n];
+ }
+ }
+ _calcAvg = enable;
+}
+
+int16_t CircularBuffer::ArithMean(double& mean) {
+ assert(_buffLen > 0);
+
+ if (_buffIsFull) {
+ mean = _sum / (double)_buffLen;
+ return 0;
+ } else {
+ if (_idx > 0) {
+ mean = _sum / (double)_idx;
+ return 0;
+ } else {
+ return -1;
+ }
+ }
+}
+
+int16_t CircularBuffer::Variance(double& var) {
+ assert(_buffLen > 0);
+
+ if (_buffIsFull) {
+ var = _sumSqr / (double)_buffLen;
+ return 0;
+ } else {
+ if (_idx > 0) {
+ var = _sumSqr / (double)_idx;
+ return 0;
+ } else {
+ return -1;
+ }
+ }
+}
+
+} // namespace test
+
+bool FixedPayloadTypeCodec(const char* payloadName) {
+ char fixPayloadTypeCodecs[NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE][32] = {
+ "PCMU", "PCMA", "GSM", "G723", "DVI4", "LPC", "PCMA",
+ "G722", "QCELP", "CN", "MPA", "G728", "G729"};
+
+ for (int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++) {
+ if (absl::EqualsIgnoreCase(payloadName, fixPayloadTypeCodecs[n])) {
+ return true;
+ }
+ }
+ return false;
+}
+
+void VADCallback::Reset() {
+ memset(_numFrameTypes, 0, sizeof(_numFrameTypes));
+}
+
+VADCallback::VADCallback() {
+ memset(_numFrameTypes, 0, sizeof(_numFrameTypes));
+}
+
+void VADCallback::PrintFrameTypes() {
+ printf("kEmptyFrame......... %d\n", _numFrameTypes[kEmptyFrame]);
+ printf("kAudioFrameSpeech... %d\n", _numFrameTypes[kAudioFrameSpeech]);
+ printf("kAudioFrameCN....... %d\n", _numFrameTypes[kAudioFrameCN]);
+ printf("kVideoFrameKey...... %d\n", _numFrameTypes[kVideoFrameKey]);
+ printf("kVideoFrameDelta.... %d\n", _numFrameTypes[kVideoFrameDelta]);
+}
+
+int32_t VADCallback::InFrameType(FrameType frame_type) {
+ _numFrameTypes[frame_type]++;
+ return 0;
+}
+
+} // namespace webrtc
diff --git a/modules/audio_coding/test/utility.h b/modules/audio_coding/test/utility.h
new file mode 100644
index 0000000..6f17df5
--- /dev/null
+++ b/modules/audio_coding/test/utility.h
@@ -0,0 +1,140 @@
+/*
+ * Copyright (c) 2011 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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#define MODULES_AUDIO_CODING_TEST_UTILITY_H_
+
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+//-----------------------------
+#define CHECK_ERROR(f) \
+ do { \
+ EXPECT_GE(f, 0) << "Error Calling API"; \
+ } while (0)
+
+//-----------------------------
+#define CHECK_PROTECTED(f) \
+ do { \
+ if (f >= 0) { \
+ ADD_FAILURE() << "Error Calling API"; \
+ } else { \
+ printf("An expected error is caught.\n"); \
+ } \
+ } while (0)
+
+//----------------------------
+#define CHECK_ERROR_MT(f) \
+ do { \
+ if (f < 0) { \
+ fprintf(stderr, "Error Calling API in file %s at line %d \n", __FILE__, \
+ __LINE__); \
+ } \
+ } while (0)
+
+//----------------------------
+#define CHECK_PROTECTED_MT(f) \
+ do { \
+ if (f >= 0) { \
+ fprintf(stderr, "Error Calling API in file %s at line %d \n", __FILE__, \
+ __LINE__); \
+ } else { \
+ printf("An expected error is caught.\n"); \
+ } \
+ } while (0)
+
+#define DELETE_POINTER(p) \
+ do { \
+ if (p != NULL) { \
+ delete p; \
+ p = NULL; \
+ } \
+ } while (0)
+
+class ACMTestTimer {
+ public:
+ ACMTestTimer();
+ ~ACMTestTimer();
+
+ void Reset();
+ void Tick10ms();
+ void Tick1ms();
+ void Tick100ms();
+ void Tick1sec();
+ void CurrentTimeHMS(char* currTime);
+ void CurrentTime(unsigned long& h,
+ unsigned char& m,
+ unsigned char& s,
+ unsigned short& ms);
+
+ private:
+ void Adjust();
+
+ unsigned short _msec;
+ unsigned char _sec;
+ unsigned char _min;
+ unsigned long _hour;
+};
+
+// To avoid clashes with CircularBuffer in APM.
+namespace test {
+
+class CircularBuffer {
+ public:
+ CircularBuffer(uint32_t len);
+ ~CircularBuffer();
+
+ void SetArithMean(bool enable);
+ void SetVariance(bool enable);
+
+ void Update(const double newVal);
+ void IsBufferFull();
+
+ int16_t Variance(double& var);
+ int16_t ArithMean(double& mean);
+
+ protected:
+ double* _buff;
+ uint32_t _idx;
+ uint32_t _buffLen;
+
+ bool _buffIsFull;
+ bool _calcAvg;
+ bool _calcVar;
+ double _sum;
+ double _sumSqr;
+};
+
+} // namespace test
+
+int16_t ChooseCodec(CodecInst& codecInst);
+
+void PrintCodecs();
+
+bool FixedPayloadTypeCodec(const char* payloadName);
+
+class VADCallback : public ACMVADCallback {
+ public:
+ VADCallback();
+
+ int32_t InFrameType(FrameType frame_type) override;
+
+ void PrintFrameTypes();
+ void Reset();
+
+ private:
+ uint32_t _numFrameTypes[5];
+};
+
+} // namespace webrtc
+
+#endif // MODULES_AUDIO_CODING_TEST_UTILITY_H_