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/*
* Copyright (c) 2014 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 "absl/flags/flag.h"
#include "modules/audio_coding/codecs/opus/opus_inst.h"
#include "modules/audio_coding/codecs/opus/opus_interface.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
ABSL_FLAG(int,
complexity,
10,
"Complexity: 0 ~ 10 -- defined as in Opus"
"specification.");
ABSL_FLAG(int, maxplaybackrate, 48000, "Maximum playback rate (Hz).");
ABSL_FLAG(int, application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
ABSL_FLAG(int, reported_loss_rate, 10, "Reported percentile of packet loss.");
ABSL_FLAG(bool, fec, false, "Enable FEC for encoding (-nofec to disable).");
ABSL_FLAG(bool, dtx, false, "Enable DTX for encoding (-nodtx to disable).");
ABSL_FLAG(int, sub_packets, 1, "Number of sub packets to repacketize.");
using ::testing::InitGoogleTest;
namespace webrtc {
namespace test {
namespace {
static const int kOpusBlockDurationMs = 20;
static const int kOpusSamplingKhz = 48;
} // namespace
class NetEqOpusQualityTest : public NetEqQualityTest {
protected:
NetEqOpusQualityTest();
void SetUp() override;
void TearDown() override;
int EncodeBlock(int16_t* in_data,
size_t block_size_samples,
rtc::Buffer* payload,
size_t max_bytes) override;
private:
WebRtcOpusEncInst* opus_encoder_;
OpusRepacketizer* repacketizer_;
size_t sub_block_size_samples_;
int bit_rate_kbps_;
bool fec_;
bool dtx_;
int complexity_;
int maxplaybackrate_;
int target_loss_rate_;
int sub_packets_;
int application_;
};
NetEqOpusQualityTest::NetEqOpusQualityTest()
: NetEqQualityTest(kOpusBlockDurationMs * absl::GetFlag(FLAGS_sub_packets),
kOpusSamplingKhz,
kOpusSamplingKhz,
SdpAudioFormat("opus", 48000, 2)),
opus_encoder_(NULL),
repacketizer_(NULL),
sub_block_size_samples_(
static_cast<size_t>(kOpusBlockDurationMs * kOpusSamplingKhz)),
bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)),
fec_(absl::GetFlag(FLAGS_fec)),
dtx_(absl::GetFlag(FLAGS_dtx)),
complexity_(absl::GetFlag(FLAGS_complexity)),
maxplaybackrate_(absl::GetFlag(FLAGS_maxplaybackrate)),
target_loss_rate_(absl::GetFlag(FLAGS_reported_loss_rate)),
sub_packets_(absl::GetFlag(FLAGS_sub_packets)) {
// Flag validation
RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 6 &&
absl::GetFlag(FLAGS_bit_rate_kbps) <= 510)
<< "Invalid bit rate, should be between 6 and 510 kbps.";
RTC_CHECK(absl::GetFlag(FLAGS_complexity) >= -1 &&
absl::GetFlag(FLAGS_complexity) <= 10)
<< "Invalid complexity setting, should be between 0 and 10.";
RTC_CHECK(absl::GetFlag(FLAGS_application) == 0 ||
absl::GetFlag(FLAGS_application) == 1)
<< "Invalid application mode, should be 0 or 1.";
RTC_CHECK(absl::GetFlag(FLAGS_reported_loss_rate) >= 0 &&
absl::GetFlag(FLAGS_reported_loss_rate) <= 100)
<< "Invalid packet loss percentile, should be between 0 and 100.";
RTC_CHECK(absl::GetFlag(FLAGS_sub_packets) >= 1 &&
absl::GetFlag(FLAGS_sub_packets) <= 3)
<< "Invalid number of sub packets, should be between 1 and 3.";
// Redefine decoder type if input is stereo.
if (channels_ > 1) {
audio_format_ = SdpAudioFormat("opus", 48000, 2,
SdpAudioFormat::Parameters{{"stereo", "1"}});
}
application_ = absl::GetFlag(FLAGS_application);
}
void NetEqOpusQualityTest::SetUp() {
// Create encoder memory.
WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_, 48000);
ASSERT_TRUE(opus_encoder_);
// Create repacketizer.
repacketizer_ = opus_repacketizer_create();
ASSERT_TRUE(repacketizer_);
// Set bitrate.
EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_kbps_ * 1000));
if (fec_) {
EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
}
if (dtx_) {
EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
}
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity_));
EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, maxplaybackrate_));
EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, target_loss_rate_));
NetEqQualityTest::SetUp();
}
void NetEqOpusQualityTest::TearDown() {
// Free memory.
EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
opus_repacketizer_destroy(repacketizer_);
NetEqQualityTest::TearDown();
}
int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
size_t block_size_samples,
rtc::Buffer* payload,
size_t max_bytes) {
EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
int16_t* pointer = in_data;
int value;
opus_repacketizer_init(repacketizer_);
for (int idx = 0; idx < sub_packets_; idx++) {
payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
max_bytes, payload.data());
Log() << "Encoded a frame with Opus mode "
<< (value == 0 ? 0 : payload[0] >> 3) << std::endl;
return (value >= 0) ? static_cast<size_t>(value) : 0;
});
if (OPUS_OK !=
opus_repacketizer_cat(repacketizer_, payload->data(), value)) {
opus_repacketizer_init(repacketizer_);
// If the repacketization fails, we discard this frame.
return 0;
}
pointer += sub_block_size_samples_ * channels_;
}
value = opus_repacketizer_out(repacketizer_, payload->data(),
static_cast<opus_int32>(max_bytes));
EXPECT_GE(value, 0);
return value;
}
TEST_F(NetEqOpusQualityTest, Test) {
Simulate();
}
} // namespace test
} // namespace webrtc
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