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Revert "Remove CodecInst pt.1"

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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}
This commit is contained in:
Fredrik Solenberg
2018-12-03 15:50:44 +00:00
committed by Commit Bot
parent 9d54bd8898
commit ec0f45be11
24 changed files with 1215 additions and 359 deletions
Download Patch File Download Diff File Expand all files Collapse all files

50
modules/audio_coding/test/PacketLossTest.cc
View File

@ -30,7 +30,6 @@ void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
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 @@ void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
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 @@ SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
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 @@ PacketLossTest::PacketLossTest(int channels,
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 @@ void PacketLossTest::Perform() {
#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
}