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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 "modules/audio_coding/acm2/acm_receive_test.h"
#include <stdio.h>
#include <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/environment/environment_factory.h"
#include "api/neteq/default_neteq_factory.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "test/gtest.h"
namespace webrtc {
namespace test {
AcmReceiveTestOldApi::AcmReceiveTestOldApi(
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels,
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: clock_(0),
neteq_(DefaultNetEqFactory().Create(CreateEnvironment(&clock_),
NetEq::Config(),
std::move(decoder_factory))),
packet_source_(packet_source),
audio_sink_(audio_sink),
output_freq_hz_(output_freq_hz),
exptected_output_channels_(exptected_output_channels) {}
AcmReceiveTestOldApi::~AcmReceiveTestOldApi() = default;
void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
neteq_->SetCodecs({{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{107, {"L16", 8000, 1}},
{108, {"L16", 16000, 1}},
{109, {"L16", 32000, 1}},
{111, {"L16", 8000, 2}},
{112, {"L16", 16000, 2}},
{113, {"L16", 32000, 2}},
{0, {"PCMU", 8000, 1}},
{110, {"PCMU", 8000, 2}},
{8, {"PCMA", 8000, 1}},
{118, {"PCMA", 8000, 2}},
{102, {"ILBC", 8000, 1}},
{9, {"G722", 8000, 1}},
{119, {"G722", 8000, 2}},
{120, {"OPUS", 48000, 2, {{"stereo", "1"}}}},
{13, {"CN", 8000, 1}},
{98, {"CN", 16000, 1}},
{99, {"CN", 32000, 1}}});
}
// Remaps payload types from ACM's default to those used in the resource file
// neteq_universal_new.rtp.
void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
neteq_->SetCodecs({{103, {"ISAC", 16000, 1}},
{104, {"ISAC", 32000, 1}},
{93, {"L16", 8000, 1}},
{94, {"L16", 16000, 1}},
{95, {"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}}});
}
void AcmReceiveTestOldApi::Run() {
for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
packet = packet_source_->NextPacket()) {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_frame, &muted));
EXPECT_TRUE(
resampler_helper_.MaybeResample(output_freq_hz_, &output_frame));
ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
ASSERT_FALSE(muted);
const size_t samples_per_block =
static_cast<size_t>(output_freq_hz_ * 10 / 1000);
EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
if (exptected_output_channels_ != kArbitraryChannels) {
if (output_frame.speech_type_ == webrtc::AudioFrame::kPLC) {
// Don't check number of channels for PLC output, since each test run
// usually starts with a short period of mono PLC before decoding the
// first packet.
} else {
EXPECT_EQ(exptected_output_channels_, output_frame.num_channels_);
}
}
ASSERT_TRUE(audio_sink_->WriteAudioFrame(output_frame));
clock_.AdvanceTimeMilliseconds(10);
AfterGetAudio();
}
EXPECT_EQ(0, neteq_->InsertPacket(
packet->header(),
rtc::ArrayView<const uint8_t>(
packet->payload(), packet->payload_length_bytes()),
clock_.CurrentTime()))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(packet->header().payloadType)
<< std::endl
<< " TS = " << packet->header().timestamp << std::endl
<< " SN = " << packet->header().sequenceNumber;
}
}
AcmReceiveTestToggleOutputFreqOldApi::AcmReceiveTestToggleOutputFreqOldApi(
PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz_1,
int output_freq_hz_2,
int toggle_period_ms,
NumOutputChannels exptected_output_channels)
: AcmReceiveTestOldApi(packet_source,
audio_sink,
output_freq_hz_1,
exptected_output_channels,
CreateBuiltinAudioDecoderFactory()),
output_freq_hz_1_(output_freq_hz_1),
output_freq_hz_2_(output_freq_hz_2),
toggle_period_ms_(toggle_period_ms),
last_toggle_time_ms_(clock_.TimeInMilliseconds()) {}
void AcmReceiveTestToggleOutputFreqOldApi::AfterGetAudio() {
if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
? output_freq_hz_2_
: output_freq_hz_1_;
last_toggle_time_ms_ = clock_.TimeInMilliseconds();
}
}
} // namespace test
} // namespace webrtc