Reland "Remove CodecInst pt.1"
This is a reland of 056f9738bf7a3d16da45398239656e165c4e0851
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}
Bug: webrtc:7626
Change-Id: I5d6ca0baf6230bfe9bf95c2c25496d2a56812d90
Reviewed-on: https://webrtc-review.googlesource.com/c/112942
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25902}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index fc5a803..34df39f 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1343,8 +1343,6 @@
"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 bb970c1..8cb5baa 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -287,113 +287,6 @@
_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 fd161fb..eb197c6 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -58,14 +58,6 @@
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 2408366..6ca7408 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -14,12 +14,9 @@
#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"
@@ -53,14 +50,12 @@
}
void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int sample_rate, size_t channels) {
- struct CodecInst sendCodec;
- int codecNo;
-
+ std::string in_file_name, int in_sample_rate,
+ int payload_type, SdpAudioFormat format) {
// Open input file
const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
- _pcmFile.Open(file_name, sample_rate, "rb");
- if (channels == 2) {
+ _pcmFile.Open(file_name, in_sample_rate, "rb");
+ if (format.num_channels == 2) {
_pcmFile.ReadStereo(true);
}
// Set test length to 500 ms (50 blocks of 10 ms each).
@@ -68,16 +63,9 @@
// 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(
- sendCodec.pltype, CodecInstToSdp(sendCodec), absl::nullopt));
- _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
+ payload_type, format, absl::nullopt));
+ _packetization = new TestPacketization(rtpStream, format.clockrate_hz);
EXPECT_EQ(0, acm->RegisterTransportCallback(_packetization));
_acm = acm;
@@ -112,30 +100,39 @@
}
void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, size_t channels) {
- struct CodecInst recvCodec = CodecInst();
- int noOfCodecs;
+ std::string out_file_name, size_t channels, int file_num) {
EXPECT_EQ(0, acm->InitializeReceiver());
- 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)));
- }
+ 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"}}}}});
}
int playSampFreq;
std::string file_name;
rtc::StringBuilder file_stream;
- file_stream << webrtc::test::OutputPath() << out_file_name
- << static_cast<int>(codeId) << ".pcm";
+ file_stream << webrtc::test::OutputPath() << out_file_name << file_num
+ << ".pcm";
file_name = file_stream.str();
_rtpStream = rtpStream;
@@ -225,85 +222,47 @@
}
}
-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);
-}
+EncodeDecodeTest::EncodeDecodeTest() = default;
void EncodeDecodeTest::Perform() {
- 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.
+ 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}},
+#ifdef WEBRTC_CODEC_ILBC
+ {102, {"ILBC", 8000, 1}},
+#endif
+ {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())));
- codePars[0] = 0;
- codePars[1] = 0;
- codePars[2] = 0;
+ 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();
- std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
- struct CodecInst sendCodecTmp;
- numCodecs = acm->NumberOfCodecs();
+ 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();
- 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;
- }
+ file_num++;
}
-
- // 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 9132d71..df6ee5f 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -47,13 +47,12 @@
public:
Sender();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string in_file_name, int sample_rate, size_t channels);
+ std::string in_file_name, int in_sample_rate,
+ int payload_type, SdpAudioFormat format);
void Teardown();
void Run();
bool Add10MsData();
- uint8_t codeId;
-
protected:
AudioCodingModule* _acm;
@@ -68,15 +67,12 @@
Receiver();
virtual ~Receiver() {};
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
- std::string out_file_name, size_t channels);
+ std::string out_file_name, size_t channels, int file_num);
void Teardown();
void Run();
virtual bool IncomingPacket();
bool PlayoutData();
- //for auto_test and logging
- uint8_t codeId;
-
private:
PCMFile _pcmFile;
int16_t* _playoutBuffer;
@@ -96,17 +92,8 @@
class EncodeDecodeTest {
public:
- explicit EncodeDecodeTest(int test_mode);
+ EncodeDecodeTest();
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 a1629fd..6f87659 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -30,6 +30,7 @@
RTPStream* rtpStream,
std::string out_file_name,
int channels,
+ int file_num,
int loss_rate,
int burst_length) {
loss_rate_ = loss_rate;
@@ -37,7 +38,7 @@
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);
+ Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num);
}
bool ReceiverWithPacketLoss::IncomingPacket() {
@@ -89,10 +90,11 @@
void SenderWithFEC::Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string in_file_name,
- int sample_rate,
- int channels,
+ int payload_type,
+ SdpAudioFormat format,
int expected_loss_rate) {
- Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
+ Sender::Setup(acm, rtpStream, in_file_name, format.clockrate_hz, payload_type,
+ format);
EXPECT_TRUE(SetFEC(true));
EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
}
@@ -123,8 +125,6 @@
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) {}
@@ -133,40 +133,32 @@
#ifndef WEBRTC_CODEC_OPUS
return;
#else
- 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::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
+ SdpAudioFormat send_format = SdpAudioFormat("opus", 48000, 2);
+ if (channels_ == 2) {
+ send_format.parameters = {{"stereo", "1"}};
+ }
+
std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
"packet_loss_test");
-
- // Encode to file
rtpFile.Open(fileName.c_str(), "wb+");
rtpFile.WriteHeader();
-
- sender_->codeId = codec_id;
-
- sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
+ SenderWithFEC sender;
+ sender.Setup(acm.get(), &rtpFile, in_file_name_, 120, send_format,
expected_loss_rate_);
- if (acm->SendCodec()) {
- sender_->Run();
- }
- sender_->Teardown();
+ sender.Run();
+ sender.Teardown();
rtpFile.Close();
- // Decode to file
rtpFile.Open(fileName.c_str(), "rb");
rtpFile.ReadHeader();
-
- receiver_->codeId = codec_id;
-
- receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
+ ReceiverWithPacketLoss receiver;
+ receiver.Setup(acm.get(), &rtpFile, "packetLoss_out", channels_, 15,
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 6018301..b26f6ec 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -11,7 +11,6 @@
#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"
@@ -24,6 +23,7 @@
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 sample_rate,
- int channels,
+ int payload_type,
+ SdpAudioFormat format,
int expected_loss_rate);
bool SetPacketLossRate(int expected_loss_rate);
bool SetFEC(bool enable_fec);
@@ -65,8 +65,6 @@
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 0099b2a..aad80e8 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -14,13 +14,11 @@
#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/audio_coding/test/utility.h"
+#include "modules/include/module_common_types.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/strings/string_builder.h"
@@ -35,6 +33,11 @@
// 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();
}
@@ -101,7 +104,7 @@
payload_size_ = 0;
}
-TestAllCodecs::TestAllCodecs(int test_mode)
+TestAllCodecs::TestAllCodecs()
: acm_a_(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
acm_b_(AudioCodingModule::Create(
@@ -110,8 +113,6 @@
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() {
@@ -126,23 +127,28 @@
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();
- 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));
- }
+ 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}}});
// Create and connect the channel
channel_a_to_b_ = new TestPack;
@@ -151,9 +157,6 @@
// 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";
@@ -171,9 +174,6 @@
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";
@@ -188,9 +188,6 @@
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";
@@ -205,9 +202,6 @@
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);
@@ -220,9 +214,6 @@
Run(channel_a_to_b_);
outfile_b_.Close();
#endif
- if (test_mode_ != 0) {
- printf("===============================================================\n");
- }
test_count_++;
OpenOutFile(test_count_);
char codec_l16[] = "L16";
@@ -235,9 +226,7 @@
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);
@@ -249,9 +238,7 @@
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);
@@ -259,9 +246,7 @@
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";
@@ -277,9 +262,7 @@
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_);
@@ -295,9 +278,6 @@
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";
@@ -317,24 +297,6 @@
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
@@ -354,21 +316,21 @@
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.
- if (!strcmp(codec_name, "G722")) {
+ int clockrate_hz = sampling_freq_hz;
+ size_t num_channels = 1;
+ if (absl::EqualsIgnoreCase(codec_name, "G722")) {
packet_size_samples_ = packet_size / 2;
- } else if (!strcmp(codec_name, "ISAC") && (rate == -1)) {
+ clockrate_hz = sampling_freq_hz / 2;
+ } else if (absl::EqualsIgnoreCase(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;
}
@@ -402,16 +364,9 @@
}
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 = CodecInstToSdp(my_codec_param);
+ SdpAudioFormat format = { codec_name, clockrate_hz, num_channels };
format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
my_acm->SetEncoder(
@@ -485,11 +440,4 @@
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 669a118..3125efe 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/test/Channel.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
namespace webrtc {
@@ -48,7 +48,7 @@
class TestAllCodecs {
public:
- explicit TestAllCodecs(int test_mode);
+ TestAllCodecs();
~TestAllCodecs();
void Perform();
@@ -67,9 +67,7 @@
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 8bb3971..ac51666 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -10,8 +10,9 @@
#include "modules/audio_coding/test/TestRedFec.h"
-#include <assert.h>
+#include <utility>
+#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"
@@ -24,12 +25,11 @@
#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 {
@@ -173,6 +173,7 @@
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;
@@ -181,22 +182,22 @@
config.payload_type = cn_payload_type;
config.vad_mode = vad_mode.value();
encoder = CreateComfortNoiseEncoder(std::move(config));
- EXPECT_EQ(true,
- other_acm->RegisterReceiveCodec(
- cn_payload_type, {"CN", codec_format.clockrate_hz, 1}));
+ receive_codecs.emplace(
+ std::make_pair(cn_payload_type,
+ SdpAudioFormat("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));
- EXPECT_EQ(true,
- other_acm->RegisterReceiveCodec(
- red_payload_type, {"red", codec_format.clockrate_hz, 1}));
+ receive_codecs.emplace(
+ std::make_pair(red_payload_type,
+ SdpAudioFormat("red", codec_format.clockrate_hz, 1)));
}
}
acm->SetEncoder(std::move(encoder));
- EXPECT_EQ(true, other_acm->RegisterReceiveCodec(payload_type, codec_format));
+ other_acm->SetReceiveCodecs(receive_codecs);
}
void TestRedFec::Run() {
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index bf8e189..dd4be6a 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -15,9 +15,8 @@
#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/audio_coding/test/utility.h"
+#include "modules/include/module_common_types.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
@@ -32,7 +31,6 @@
last_in_timestamp_(0),
total_bytes_(0),
payload_size_(0),
- codec_mode_(kNotSet),
lost_packet_(false) {}
TestPackStereo::~TestPackStereo() {}
@@ -98,7 +96,7 @@
lost_packet_ = lost;
}
-TestStereo::TestStereo(int test_mode)
+TestStereo::TestStereo()
: acm_a_(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
acm_b_(AudioCodingModule::Create(
@@ -108,8 +106,6 @@
pack_size_samp_(0),
pack_size_bytes_(0),
counter_(0) {
- // test_mode = 0 for silent test (auto test)
- test_mode_ = test_mode;
}
TestStereo::~TestStereo() {
@@ -142,27 +138,25 @@
EXPECT_EQ(0, acm_a_->InitializeReceiver());
EXPECT_EQ(0, acm_b_->InitializeReceiver());
- // 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)));
- }
+ 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}}});
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
@@ -171,9 +165,6 @@
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.
@@ -183,11 +174,6 @@
// 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_);
@@ -206,11 +192,6 @@
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_);
@@ -225,11 +206,6 @@
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);
@@ -242,11 +218,6 @@
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);
@@ -255,11 +226,6 @@
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;
@@ -278,13 +244,8 @@
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();
- 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";
@@ -303,11 +264,6 @@
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;
@@ -340,11 +296,6 @@
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_);
@@ -352,43 +303,25 @@
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_);
@@ -399,12 +332,6 @@
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);
@@ -426,54 +353,30 @@
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);
@@ -483,26 +386,11 @@
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-mono\n");
- }
test_cntr_++;
OpenOutFile(test_cntr_);
// Encode and decode in mono.
RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels);
- 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;
- }
- }
+ acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
Run(channel_a2b_, audio_channels, codec_channels);
// Encode in stereo, decode in mono.
@@ -516,65 +404,22 @@
// 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_);
- 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)));
+ acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}}});
Run(channel_a2b_, audio_channels, 2);
out_file_.Close();
// 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_);
- }
- opus_codec_param.channels = 1;
- EXPECT_EQ(true,
- acm_b_->RegisterReceiveCodec(opus_codec_param.pltype,
- CodecInstToSdp(opus_codec_param)));
+ acm_b_->SetReceiveCodecs({{120, {"OPUS", 48000, 2}}});
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_;
@@ -594,12 +439,6 @@
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;
@@ -719,7 +558,7 @@
}
}
- // Run received side of ACM
+ // Run receive side of ACM
bool muted;
EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
ASSERT_FALSE(muted);
@@ -758,17 +597,4 @@
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 0d80631..da10bf1 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/test/Channel.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/PCMFile.h"
#define PCMA_AND_PCMU
@@ -58,7 +58,7 @@
class TestStereo {
public:
- explicit TestStereo(int test_mode);
+ TestStereo();
~TestStereo();
void Perform();
@@ -79,9 +79,6 @@
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 09c69f9..8e16280 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 {
@@ -94,8 +94,9 @@
channel_->SetIsStereo(encoder->NumChannels() > 1);
acm_send_->SetEncoder(std::move(encoder));
- EXPECT_EQ(true,
- acm_receive_->RegisterReceiveCodec(payload_type, codec_format));
+ std::map<int, SdpAudioFormat> receive_codecs = {{payload_type, codec_format}};
+ acm_receive_->SetReceiveCodecs(receive_codecs);
+
return added_comfort_noise;
}
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index 68b2c1e..f2358e7 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -16,7 +16,6 @@
#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 85926f1..750656f 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -25,12 +25,8 @@
#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(ACM_TEST_MODE).Perform();
+ webrtc::TestAllCodecs().Perform();
}
#if defined(WEBRTC_ANDROID)
@@ -38,7 +34,7 @@
#else
TEST(AudioCodingModuleTest, TestEncodeDecode) {
#endif
- webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
+ webrtc::EncodeDecodeTest().Perform();
}
TEST(AudioCodingModuleTest, TestRedFec) {
@@ -50,7 +46,7 @@
#else
TEST(AudioCodingModuleTest, TestIsac) {
#endif
- webrtc::ISACTest(ACM_TEST_MODE).Perform();
+ webrtc::ISACTest().Perform();
}
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
@@ -70,7 +66,7 @@
#else
TEST(AudioCodingModuleTest, TestStereo) {
#endif
- webrtc::TestStereo(ACM_TEST_MODE).Perform();
+ webrtc::TestStereo().Perform();
}
TEST(AudioCodingModuleTest, TestWebRtcVadDtx) {
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index 2079a94..3af114e 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -16,15 +16,9 @@
#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"
@@ -65,25 +59,26 @@
const int payload_type1,
const SdpAudioFormat& format2,
const int payload_type2) {
+
//--- Set A codecs
_acmA->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
- EXPECT_EQ(true, _acmA->RegisterReceiveCodec(payload_type2, format2));
+ _acmA->SetReceiveCodecs({{payload_type2, format2}});
//--- Set ref-A codecs
_acmRefA->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type1, format1, absl::nullopt));
- EXPECT_EQ(true, _acmRefA->RegisterReceiveCodec(payload_type2, format2));
+ _acmRefA->SetReceiveCodecs({{payload_type2, format2}});
//--- Set B codecs
_acmB->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
- EXPECT_EQ(true, _acmB->RegisterReceiveCodec(payload_type1, format1));
+ _acmB->SetReceiveCodecs({{payload_type1, format1}});
//--- Set ref-B codecs
_acmRefB->SetEncoder(
encoder_factory->MakeAudioEncoder(payload_type2, format2, absl::nullopt));
- EXPECT_EQ(true, _acmRefB->RegisterReceiveCodec(payload_type1, format1));
+ _acmRefB->SetReceiveCodecs({{payload_type1, format1}});
uint16_t frequencyHz;
@@ -184,14 +179,13 @@
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)) {
- EXPECT_EQ(true, _acmA->RegisterReceiveCodec(payload_type2, format2));
+ _acmA->SetReceiveCodecs({{payload_type2, format2}});
}
}
}
diff --git a/modules/audio_coding/test/TwoWayCommunication.h b/modules/audio_coding/test/TwoWayCommunication.h
index a679732..7d0cdb9 100644
--- a/modules/audio_coding/test/TwoWayCommunication.h
+++ b/modules/audio_coding/test/TwoWayCommunication.h
@@ -18,7 +18,6 @@
#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 c332fe0..339d419 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -14,20 +14,9 @@
#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"
@@ -43,15 +32,10 @@
namespace {
-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;
-}
+constexpr int kISAC16kPayloadType = 103;
+constexpr int kISAC32kPayloadType = 104;
+const SdpAudioFormat kISAC16kFormat = { "ISAC", 16000, 1 };
+const SdpAudioFormat kISAC32kFormat = { "ISAC", 32000, 1 };
AudioEncoderIsacFloat::Config TweakConfig(
AudioEncoderIsacFloat::Config config,
@@ -77,43 +61,96 @@
} // namespace
-ISACTest::ISACTest(int testMode)
+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()
: _acmA(AudioCodingModule::Create(
AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
_acmB(AudioCodingModule::Create(
- AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
- _testMode(testMode) {}
+ AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {}
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.
- 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)));
+ std::map<int, SdpAudioFormat> receive_codecs =
+ {{kISAC16kPayloadType, kISAC16kFormat},
+ {kISAC32kPayloadType, kISAC32kFormat}};
+ _acmA->SetReceiveCodecs(receive_codecs);
+ _acmB->SetReceiveCodecs(receive_codecs);
//--- Set A-to-B channel
_channel_A2B.reset(new Channel);
@@ -128,10 +165,14 @@
file_name_swb_ =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
- _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
- _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
+ _acmB->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
+ _acmA->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
_inFileA.Open(file_name_swb_, 32000, "rb");
// Set test length to 500 ms (50 blocks of 10 ms each).
@@ -146,9 +187,9 @@
while (!_inFileA.EndOfFile()) {
Run10ms();
}
- CodecInst receiveCodec;
- EXPECT_EQ(0, _acmA->ReceiveCodec(&receiveCodec));
- EXPECT_EQ(0, _acmB->ReceiveCodec(&receiveCodec));
+
+ EXPECT_TRUE(_acmA->ReceiveFormat());
+ EXPECT_TRUE(_acmB->ReceiveFormat());
_inFileA.Close();
_outFileA.Close();
@@ -170,45 +211,13 @@
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++;
- if (_testMode == 0) {
- SwitchingSamplingRate(testNr, 4);
- } else {
- SwitchingSamplingRate(testNr, 80);
- }
+ SwitchingSamplingRate(testNr, 4);
}
void ISACTest::Run10ms() {
@@ -245,12 +254,16 @@
_outFileB.Open(file_name_out, 32000, "wb");
// Side A is sending super-wideband, and side B is sending wideband.
- _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- TweakConfig(MakeConfig(_paramISAC32kHz), swbISACConfig),
- _paramISAC32kHz.pltype));
- _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- TweakConfig(MakeConfig(_paramISAC16kHz), wbISACConfig),
- _paramISAC16kHz.pltype));
+ _acmA->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ swbISACConfig),
+ kISAC32kPayloadType));
+ _acmB->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ wbISACConfig),
+ kISAC16kPayloadType));
bool adaptiveMode = false;
if ((swbISACConfig.currentRateBitPerSec == -1) ||
@@ -262,30 +275,10 @@
_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();
@@ -316,10 +309,14 @@
// Start with side A sending super-wideband and side B seding wideband.
// Toggle sending wideband/super-wideband in this test.
- _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
- _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
+ _acmA->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
+ _acmB->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
int numSendCodecChanged = 0;
_myTimer.Reset();
@@ -328,21 +325,23 @@
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(
- MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
+ _acmA->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
} else {
// Switch side A to send wideband.
_inFileA.Close();
_inFileA.Open(file_name_swb_, 32000, "rb");
- _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
+ _acmA->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
}
numSendCodecChanged++;
}
@@ -352,14 +351,18 @@
// Switch side B to send super-wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
- _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- MakeConfig(_paramISAC32kHz), _paramISAC32kHz.pltype));
+ _acmB->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat),
+ kISAC32kPayloadType));
} else {
// Switch side B to send wideband.
_inFileB.Close();
_inFileB.Open(file_name_swb_, 32000, "rb");
- _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder(
- MakeConfig(_paramISAC16kHz), _paramISAC16kHz.pltype));
+ _acmB->SetEncoder(
+ AudioEncoderIsacFloat::MakeAudioEncoder(
+ *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat),
+ kISAC16kPayloadType));
}
numSendCodecChanged++;
}
diff --git a/modules/audio_coding/test/iSACTest.h b/modules/audio_coding/test/iSACTest.h
index 0b140b6..e000476 100644
--- a/modules/audio_coding/test/iSACTest.h
+++ b/modules/audio_coding/test/iSACTest.h
@@ -15,14 +15,9 @@
#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 {
@@ -37,12 +32,37 @@
class ISACTest {
public:
- explicit ISACTest(int testMode);
+ ISACTest();
~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();
@@ -65,15 +85,9 @@
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 954b183..1e24e5d 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -13,12 +13,9 @@
#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"
@@ -89,13 +86,10 @@
EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
// Register Opus stereo as receiving codec.
- 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)));
+ constexpr int kOpusPayloadType = 120;
+ const SdpAudioFormat kOpusFormatStereo("opus", 48000, 2, {{"stereo", "1"}});
+ payload_type_ = kOpusPayloadType;
+ acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatStereo}});
// Create and connect the channel.
channel_a2b_ = new TestPackStereo;
@@ -159,10 +153,8 @@
OpenOutFile(test_cntr);
// Register Opus mono as receiving codec.
- opus_codec_param.channels = 1;
- EXPECT_EQ(true,
- acm_receiver_->RegisterReceiveCodec(
- opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
+ const SdpAudioFormat kOpusFormatMono("opus", 48000, 2);
+ acm_receiver_->SetReceiveCodecs({{kOpusPayloadType, kOpusFormatMono}});
// 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 019e96b..c69f922 100644
--- a/modules/audio_coding/test/opus_test.h
+++ b/modules/audio_coding/test/opus_test.h
@@ -17,7 +17,6 @@
#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 071a6d8..8d82b6e 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -12,10 +12,8 @@
#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"
@@ -35,8 +33,9 @@
ASSERT_EQ(0, acm_->InitializeReceiver());
constexpr int pltype = 108;
- ASSERT_EQ(true,
- acm_->RegisterReceiveCodec(pltype, {"L16", kSampleRateHz, 1}));
+ std::map<int, SdpAudioFormat> receive_codecs =
+ {{pltype, {"L16", kSampleRateHz, 1}}};
+ acm_->SetReceiveCodecs(receive_codecs);
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
deleted file mode 100644
index 53f8077..0000000
--- a/modules/audio_coding/test/utility.cc
+++ /dev/null
@@ -1,299 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 6f17df5..0000000
--- a/modules/audio_coding/test/utility.h
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * 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_
