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/*
* Copyright (c) 2013 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/interface/audio_coding_module.h"
#include <assert.h>
#include <math.h>
#include <iostream>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
namespace {
double FrameRms(AudioFrame& frame) {
int samples = frame.num_channels_ * frame.samples_per_channel_;
double rms = 0;
for (int n = 0; n < samples; ++n)
rms += frame.data_[n] * frame.data_[n];
rms /= samples;
rms = sqrt(rms);
return rms;
}
}
class InitialPlayoutDelayTest : public ::testing::Test {
protected:
InitialPlayoutDelayTest()
: acm_a_(AudioCodingModule::Create(0)),
acm_b_(AudioCodingModule::Create(1)),
channel_a2b_(NULL) {}
~InitialPlayoutDelayTest() {
if (channel_a2b_ != NULL) {
delete channel_a2b_;
channel_a2b_ = NULL;
}
}
void SetUp() {
ASSERT_TRUE(acm_a_.get() != NULL);
ASSERT_TRUE(acm_b_.get() != NULL);
EXPECT_EQ(0, acm_b_->InitializeReceiver());
EXPECT_EQ(0, acm_a_->InitializeReceiver());
// Register all L16 codecs in receiver.
CodecInst codec;
const int kFsHz[3] = { 8000, 16000, 32000 };
const int kChannels[2] = { 1, 2 };
for (int n = 0; n < 3; ++n) {
for (int k = 0; k < 2; ++k) {
AudioCodingModule::Codec("L16", &codec, kFsHz[n], kChannels[k]);
acm_b_->RegisterReceiveCodec(codec);
}
}
// Create and connect the channel
channel_a2b_ = new Channel;
acm_a_->RegisterTransportCallback(channel_a2b_);
channel_a2b_->RegisterReceiverACM(acm_b_.get());
}
void NbMono() {
CodecInst codec;
AudioCodingModule::Codec("L16", &codec, 8000, 1);
codec.pacsize = codec.plfreq * 30 / 1000; // 30 ms packets.
Run(codec, 1000);
}
void WbMono() {
CodecInst codec;
AudioCodingModule::Codec("L16", &codec, 16000, 1);
codec.pacsize = codec.plfreq * 30 / 1000; // 30 ms packets.
Run(codec, 1000);
}
void SwbMono() {
CodecInst codec;
AudioCodingModule::Codec("L16", &codec, 32000, 1);
codec.pacsize = codec.plfreq * 10 / 1000; // 10 ms packets.
Run(codec, 400); // Memory constraints limit the buffer at <500 ms.
}
void NbStereo() {
CodecInst codec;
AudioCodingModule::Codec("L16", &codec, 8000, 2);
codec.pacsize = codec.plfreq * 30 / 1000; // 30 ms packets.
Run(codec, 1000);
}
void WbStereo() {
CodecInst codec;
AudioCodingModule::Codec("L16", &codec, 16000, 2);
codec.pacsize = codec.plfreq * 30 / 1000; // 30 ms packets.
Run(codec, 1000);
}
void SwbStereo() {
CodecInst codec;
AudioCodingModule::Codec("L16", &codec, 32000, 2);
codec.pacsize = codec.plfreq * 10 / 1000; // 10 ms packets.
Run(codec, 400); // Memory constraints limit the buffer at <500 ms.
}
private:
void Run(CodecInst codec, int initial_delay_ms) {
AudioFrame in_audio_frame;
AudioFrame out_audio_frame;
int num_frames = 0;
const int kAmp = 10000;
in_audio_frame.sample_rate_hz_ = codec.plfreq;
in_audio_frame.num_channels_ = codec.channels;
in_audio_frame.samples_per_channel_ = codec.plfreq / 100; // 10 ms.
int samples = in_audio_frame.num_channels_ *
in_audio_frame.samples_per_channel_;
for (int n = 0; n < samples; ++n) {
in_audio_frame.data_[n] = kAmp;
}
uint32_t timestamp = 0;
double rms = 0;
ASSERT_EQ(0, acm_a_->RegisterSendCodec(codec));
acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
while (rms < kAmp / 2) {
in_audio_frame.timestamp_ = timestamp;
timestamp += in_audio_frame.samples_per_channel_;
ASSERT_EQ(0, acm_a_->Add10MsData(in_audio_frame));
ASSERT_LE(0, acm_a_->Process());
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
rms = FrameRms(out_audio_frame);
++num_frames;
}
ASSERT_GE(num_frames * 10, initial_delay_ms);
ASSERT_LE(num_frames * 10, initial_delay_ms + 100);
}
scoped_ptr<AudioCodingModule> acm_a_;
scoped_ptr<AudioCodingModule> acm_b_;
Channel* channel_a2b_;
};
TEST_F(InitialPlayoutDelayTest, NbMono) { NbMono(); }
TEST_F(InitialPlayoutDelayTest, WbMono) { WbMono(); }
TEST_F(InitialPlayoutDelayTest, SwbMono) { SwbMono(); }
TEST_F(InitialPlayoutDelayTest, NbStereo) { NbStereo(); }
TEST_F(InitialPlayoutDelayTest, WbStereo) { WbStereo(); }
TEST_F(InitialPlayoutDelayTest, SwbStereo) { SwbStereo(); }
} // namespace webrtc
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