374aa49e1a
Part of refactoring of ACM, and recent lint-warnings. This CL changes non-const references in the ACM API to pointers. BUG=issue1372 Review URL: https://webrtc-codereview.appspot.com/1103012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3543 4adac7df-926f-26a2-2b94-8c16560cd09d
269 lines
8.1 KiB
C++
269 lines
8.1 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
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#include <math.h>
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#include <cassert>
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#include <iostream>
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#include "gflags/gflags.h"
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#include "gtest/gtest.h"
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#include "testsupport/fileutils.h"
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#include "webrtc/common_types.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
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#include "webrtc/modules/audio_coding/main/test/Channel.h"
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#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
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#include "webrtc/modules/audio_coding/main/test/utility.h"
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#include "webrtc/system_wrappers/interface/event_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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DEFINE_string(codec, "isac", "Codec Name");
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DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
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DEFINE_int32(num_channels, 1, "Number of Channels.");
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DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
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DEFINE_int32(delay, 0, "Delay in millisecond.");
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DEFINE_int32(init_delay, 0, "Initial delay in millisecond.");
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DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
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namespace webrtc {
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namespace {
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struct CodecConfig {
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char name[50];
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int sample_rate_hz;
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int num_channels;
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};
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struct AcmConfig {
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bool dtx;
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bool fec;
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};
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struct Config {
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CodecConfig codec;
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AcmConfig acm;
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bool packet_loss;
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};
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}
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class DelayTest {
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public:
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DelayTest()
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: acm_a_(NULL),
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acm_b_(NULL),
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channel_a2b_(NULL),
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test_cntr_(0),
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encoding_sample_rate_hz_(8000) {
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}
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~DelayTest() {}
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void TearDown() {
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if(acm_a_ != NULL) {
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AudioCodingModule::Destroy(acm_a_);
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acm_a_ = NULL;
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}
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if(acm_b_ != NULL) {
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AudioCodingModule::Destroy(acm_b_);
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acm_b_ = NULL;
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}
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if(channel_a2b_ != NULL) {
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delete channel_a2b_;
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channel_a2b_ = NULL;
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}
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}
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void SetUp() {
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test_cntr_ = 0;
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std::string file_name =
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
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if (FLAGS_input_file.size() > 0)
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file_name = FLAGS_input_file;
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in_file_a_.Open(file_name, 32000, "rb");
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acm_a_ = AudioCodingModule::Create(0);
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acm_b_ = AudioCodingModule::Create(1);
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acm_a_->InitializeReceiver();
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acm_b_->InitializeReceiver();
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if (FLAGS_init_delay > 0) {
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ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay));
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}
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if (FLAGS_delay > 0) {
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ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay));
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}
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WebRtc_UWord8 num_encoders = acm_a_->NumberOfCodecs();
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CodecInst my_codec_param;
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for(int n = 0; n < num_encoders; n++) {
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acm_b_->Codec(n, &my_codec_param);
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if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
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my_codec_param.channels = 1;
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else if (my_codec_param.channels > 1)
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continue;
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if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
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my_codec_param.plfreq == 48000)
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continue;
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if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
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continue;
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acm_b_->RegisterReceiveCodec(my_codec_param);
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}
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// Create and connect the channel
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channel_a2b_ = new Channel;
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acm_a_->RegisterTransportCallback(channel_a2b_);
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channel_a2b_->RegisterReceiverACM(acm_b_);
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}
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void Perform(const Config* config, size_t num_tests, int duration_sec,
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const char* output_prefix) {
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for (size_t n = 0; n < num_tests; ++n) {
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ApplyConfig(config[n]);
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Run(duration_sec, output_prefix);
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}
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}
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private:
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void ApplyConfig(const Config& config) {
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printf("====================================\n");
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printf("Test %d \n"
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"Codec: %s, %d kHz, %d channel(s)\n"
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"ACM: DTX %s, FEC %s\n"
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"Channel: %s\n",
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++test_cntr_,
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config.codec.name, config.codec.sample_rate_hz,
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config.codec.num_channels, config.acm.dtx ? "on" : "off",
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config.acm.fec ? "on" : "off",
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config.packet_loss ? "with packet-loss" : "no packet-loss");
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SendCodec(config.codec);
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ConfigAcm(config.acm);
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ConfigChannel(config.packet_loss);
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}
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void SendCodec(const CodecConfig& config) {
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CodecInst my_codec_param;
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ASSERT_EQ(0, AudioCodingModule::Codec(config.name, &my_codec_param,
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config.sample_rate_hz,
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config.num_channels));
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encoding_sample_rate_hz_ = my_codec_param.plfreq;
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ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param));
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}
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void ConfigAcm(const AcmConfig& config) {
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ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr));
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ASSERT_EQ(0, acm_a_->SetFECStatus(config.fec));
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}
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void ConfigChannel(bool packet_loss) {
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channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
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}
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void OpenOutFile(const char* output_id) {
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std::stringstream file_stream;
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file_stream << "delay_test_" << FLAGS_codec << "_"
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<< FLAGS_sample_rate_hz << "Hz" << "_"
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<< FLAGS_init_delay << "ms_"
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<< FLAGS_delay << "ms.pcm";
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std::cout << "Output file: " << file_stream.str() << std::endl <<std::endl;
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std::string file_name = webrtc::test::OutputPath() + file_stream.str();
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out_file_b_.Open(file_name.c_str(), 32000, "wb");
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}
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void Run(int duration_sec, const char* output_prefix) {
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OpenOutFile(output_prefix);
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AudioFrame audio_frame;
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uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
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int num_frames = 0;
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int in_file_frames = 0;
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uint32_t playout_ts;
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uint32_t received_ts;
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double average_delay = 0;
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double inst_delay_sec = 0;
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while(num_frames < (duration_sec * 100)) {
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if (in_file_a_.EndOfFile()) {
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in_file_a_.Rewind();
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}
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// Print delay information every 16 frame
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if ((num_frames & 0x3F) == 0x3F) {
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ACMNetworkStatistics statistics;
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acm_b_->NetworkStatistics(&statistics);
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fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
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" ts-based average = %6.3f, "
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"curr buff-lev = %4u opt buff-lev = %4u \n",
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statistics.minWaitingTimeMs,
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statistics.maxWaitingTimeMs,
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statistics.meanWaitingTimeMs,
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statistics.medianWaitingTimeMs,
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average_delay,
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statistics.currentBufferSize,
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statistics.preferredBufferSize);
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fflush(stdout);
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}
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in_file_a_.Read10MsData(audio_frame);
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ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
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ASSERT_LE(0, acm_a_->Process());
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ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
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out_file_b_.Write10MsData(audio_frame.data_,
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audio_frame.samples_per_channel_ *
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audio_frame.num_channels_);
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acm_b_->PlayoutTimestamp(&playout_ts);
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received_ts = channel_a2b_->LastInTimestamp();
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inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts) /
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static_cast<double>(encoding_sample_rate_hz_);
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if (num_frames > 10)
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average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
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++num_frames;
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++in_file_frames;
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}
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out_file_b_.Close();
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}
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AudioCodingModule* acm_a_;
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AudioCodingModule* acm_b_;
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Channel* channel_a2b_;
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PCMFile in_file_a_;
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PCMFile out_file_b_;
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int test_cntr_;
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int encoding_sample_rate_hz_;
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};
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} // namespace webrtc
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int main(int argc, char* argv[]) {
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google::ParseCommandLineFlags(&argc, &argv, true);
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webrtc::Config config;
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strcpy(config.codec.name, FLAGS_codec.c_str());
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config.codec.sample_rate_hz = FLAGS_sample_rate_hz;
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config.codec.num_channels = FLAGS_num_channels;
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config.acm.dtx = FLAGS_dtx;
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config.acm.fec = false;
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config.packet_loss = false;
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webrtc::DelayTest delay_test;
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delay_test.SetUp();
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delay_test.Perform(&config, 1, 240, "delay_test");
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delay_test.TearDown();
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}
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