audio_coding: remove "main" directory
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc
new file mode 100644
index 0000000..a8c137f
--- /dev/null
+++ b/webrtc/modules/audio_coding/test/delay_test.cc
@@ -0,0 +1,265 @@
+/*
+ * 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 <assert.h>
+#include <math.h>
+
+#include <iostream>
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+DEFINE_string(codec, "isac", "Codec Name");
+DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
+DEFINE_int32(num_channels, 1, "Number of Channels.");
+DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
+DEFINE_int32(delay, 0, "Delay in millisecond.");
+DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
+DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
+DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
+
+namespace webrtc {
+
+namespace {
+
+struct CodecSettings {
+ char name[50];
+ int sample_rate_hz;
+ int num_channels;
+};
+
+struct AcmSettings {
+ bool dtx;
+ bool fec;
+};
+
+struct TestSettings {
+ CodecSettings codec;
+ AcmSettings acm;
+ bool packet_loss;
+};
+
+} // namespace
+
+class DelayTest {
+ public:
+ DelayTest()
+ : acm_a_(AudioCodingModule::Create(0)),
+ acm_b_(AudioCodingModule::Create(1)),
+ channel_a2b_(new Channel),
+ test_cntr_(0),
+ encoding_sample_rate_hz_(8000) {}
+
+ ~DelayTest() {
+ if (channel_a2b_ != NULL) {
+ delete channel_a2b_;
+ channel_a2b_ = NULL;
+ }
+ in_file_a_.Close();
+ }
+
+ void Initialize() {
+ test_cntr_ = 0;
+ std::string file_name = webrtc::test::ResourcePath(
+ "audio_coding/testfile32kHz", "pcm");
+ if (FLAGS_input_file.size() > 0)
+ file_name = FLAGS_input_file;
+ in_file_a_.Open(file_name, 32000, "rb");
+ ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
+ "Couldn't initialize receiver.\n";
+ ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
+ "Couldn't initialize receiver.\n";
+
+ if (FLAGS_delay > 0) {
+ ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
+ "Failed to set minimum delay.\n";
+ }
+
+ int num_encoders = acm_a_->NumberOfCodecs();
+ CodecInst my_codec_param;
+ for (int n = 0; n < num_encoders; n++) {
+ EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
+ "Failed to get codec.";
+ if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
+ my_codec_param.channels = 1;
+ else if (my_codec_param.channels > 1)
+ continue;
+ if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
+ my_codec_param.plfreq == 48000)
+ continue;
+ if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
+ continue;
+ ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
+ "Couldn't register receive codec.\n";
+ }
+
+ // Create and connect the channel
+ ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
+ "Couldn't register Transport callback.\n";
+ channel_a2b_->RegisterReceiverACM(acm_b_.get());
+ }
+
+ void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
+ const char* output_prefix) {
+ for (size_t n = 0; n < num_tests; ++n) {
+ ApplyConfig(config[n]);
+ Run(duration_sec, output_prefix);
+ }
+ }
+
+ private:
+ void ApplyConfig(const TestSettings& config) {
+ printf("====================================\n");
+ printf("Test %d \n"
+ "Codec: %s, %d kHz, %d channel(s)\n"
+ "ACM: DTX %s, FEC %s\n"
+ "Channel: %s\n",
+ ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
+ config.codec.num_channels, config.acm.dtx ? "on" : "off",
+ config.acm.fec ? "on" : "off",
+ config.packet_loss ? "with packet-loss" : "no packet-loss");
+ SendCodec(config.codec);
+ ConfigAcm(config.acm);
+ ConfigChannel(config.packet_loss);
+ }
+
+ void SendCodec(const CodecSettings& config) {
+ CodecInst my_codec_param;
+ ASSERT_EQ(0, AudioCodingModule::Codec(
+ config.name, &my_codec_param, config.sample_rate_hz,
+ config.num_channels)) << "Specified codec is not supported.\n";
+
+ encoding_sample_rate_hz_ = my_codec_param.plfreq;
+ ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
+ "Failed to register send-codec.\n";
+ }
+
+ void ConfigAcm(const AcmSettings& config) {
+ ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
+ "Failed to set VAD.\n";
+ ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
+ "Failed to set RED.\n";
+ }
+
+ void ConfigChannel(bool packet_loss) {
+ channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
+ }
+
+ void OpenOutFile(const char* output_id) {
+ std::stringstream file_stream;
+ file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
+ << "Hz" << "_" << FLAGS_delay << "ms.pcm";
+ std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
+ std::string file_name = webrtc::test::OutputPath() + file_stream.str();
+ out_file_b_.Open(file_name.c_str(), 32000, "wb");
+ }
+
+ void Run(int duration_sec, const char* output_prefix) {
+ OpenOutFile(output_prefix);
+ AudioFrame audio_frame;
+ uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
+
+ int num_frames = 0;
+ int in_file_frames = 0;
+ uint32_t playout_ts;
+ uint32_t received_ts;
+ double average_delay = 0;
+ double inst_delay_sec = 0;
+ while (num_frames < (duration_sec * 100)) {
+ if (in_file_a_.EndOfFile()) {
+ in_file_a_.Rewind();
+ }
+
+ // Print delay information every 16 frame
+ if ((num_frames & 0x3F) == 0x3F) {
+ NetworkStatistics statistics;
+ acm_b_->GetNetworkStatistics(&statistics);
+ fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
+ " ts-based average = %6.3f, "
+ "curr buff-lev = %4u opt buff-lev = %4u \n",
+ statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
+ statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
+ average_delay, statistics.currentBufferSize,
+ statistics.preferredBufferSize);
+ fflush (stdout);
+ }
+
+ in_file_a_.Read10MsData(audio_frame);
+ ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
+ ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+ out_file_b_.Write10MsData(
+ audio_frame.data_,
+ audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+ acm_b_->PlayoutTimestamp(&playout_ts);
+ received_ts = channel_a2b_->LastInTimestamp();
+ inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
+ / static_cast<double>(encoding_sample_rate_hz_);
+
+ if (num_frames > 10)
+ average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
+
+ ++num_frames;
+ ++in_file_frames;
+ }
+ out_file_b_.Close();
+ }
+
+ rtc::scoped_ptr<AudioCodingModule> acm_a_;
+ rtc::scoped_ptr<AudioCodingModule> acm_b_;
+
+ Channel* channel_a2b_;
+
+ PCMFile in_file_a_;
+ PCMFile out_file_b_;
+ int test_cntr_;
+ int encoding_sample_rate_hz_;
+};
+
+} // namespace webrtc
+
+int main(int argc, char* argv[]) {
+ google::ParseCommandLineFlags(&argc, &argv, true);
+ webrtc::TestSettings test_setting;
+ strcpy(test_setting.codec.name, FLAGS_codec.c_str());
+
+ if (FLAGS_sample_rate_hz != 8000 &&
+ FLAGS_sample_rate_hz != 16000 &&
+ FLAGS_sample_rate_hz != 32000 &&
+ FLAGS_sample_rate_hz != 48000) {
+ std::cout << "Invalid sampling rate.\n";
+ return 1;
+ }
+ test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
+ if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
+ std::cout << "Only mono and stereo are supported.\n";
+ return 1;
+ }
+ test_setting.codec.num_channels = FLAGS_num_channels;
+ test_setting.acm.dtx = FLAGS_dtx;
+ test_setting.acm.fec = FLAGS_fec;
+ test_setting.packet_loss = FLAGS_packet_loss;
+
+ webrtc::DelayTest delay_test;
+ delay_test.Initialize();
+ delay_test.Perform(&test_setting, 1, 240, "delay_test");
+ return 0;
+}