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Remove CodecInst pt.1

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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}
This commit is contained in:
Fredrik Solenberg
2018-12-03 15:22:16 +01:00
committed by Commit Bot
parent 450b548cad
commit 056f9738bf
24 changed files with 359 additions and 1215 deletions
Download Patch File Download Diff File Expand all files Collapse all files

50
modules/audio_coding/test/PacketLossTest.cc
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@ -30,6 +30,7 @@ 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;
@ -37,7 +38,7 @@ 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);
Receiver::Setup(acm, rtpStream, ss.str(), channels, file_num);
}
bool ReceiverWithPacketLoss::IncomingPacket() {
@ -89,10 +90,11 @@ SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
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 @@ 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) {}
@ -133,40 +133,32 @@ void PacketLossTest::Perform() {
#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
}