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/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