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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 "webrtc/modules/audio_coding/main/acm2/acm_receive_test.h"
#include <assert.h>
#include <stdio.h>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
namespace webrtc {
namespace test {
AcmReceiveTest::AcmReceiveTest(PacketSource* packet_source,
AudioSink* audio_sink,
int output_freq_hz,
NumOutputChannels exptected_output_channels)
: clock_(0),
packet_source_(packet_source),
audio_sink_(audio_sink),
output_freq_hz_(output_freq_hz),
exptected_output_channels_(exptected_output_channels) {
webrtc::AudioCoding::Config config;
config.clock = &clock_;
config.playout_frequency_hz = output_freq_hz_;
acm_.reset(webrtc::AudioCoding::Create(config));
}
void AcmReceiveTest::RegisterDefaultCodecs() {
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kOpus, 120));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
#ifndef WEBRTC_ANDROID
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 105));
#endif
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 107));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 108));
ASSERT_TRUE(
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 109));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B_2ch, 111));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb_2ch, 112));
ASSERT_TRUE(
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch, 113));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU_2ch, 110));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA_2ch, 118));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722_2ch, 119));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 127));
}
void AcmReceiveTest::RegisterNetEqTestCodecs() {
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISAC, 103));
#ifndef WEBRTC_ANDROID
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACSWB, 104));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kISACFB, 124));
#endif
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16B, 93));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bwb, 94));
ASSERT_TRUE(
acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCM16Bswb32kHz, 95));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMU, 0));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kPCMA, 8));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kILBC, 102));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kG722, 9));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNNB, 13));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNWB, 98));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kCNSWB, 99));
ASSERT_TRUE(acm_->RegisterReceiveCodec(acm2::ACMCodecDB::kRED, 117));
}
void AcmReceiveTest::Run() {
for (scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
packet.reset(packet_source_->NextPacket())) {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
AudioFrame output_frame;
EXPECT_TRUE(acm_->Get10MsAudio(&output_frame));
EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
const int samples_per_block = 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);
}
// Insert packet after converting from RTPHeader to WebRtcRTPHeader.
WebRtcRTPHeader header;
header.header = packet->header();
header.frameType = kAudioFrameSpeech;
memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
EXPECT_TRUE(acm_->InsertPacket(packet->payload(),
packet->payload_length_bytes(),
header))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(header.header.payloadType) << std::endl
<< " TS = " << header.header.timestamp << std::endl
<< " SN = " << header.header.sequenceNumber;
}
}
} // namespace test
} // namespace webrtc
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