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/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