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Remove executable insert_packet_with_timing

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It appears to have been created in mid-2013, and hasn't been changed
since except to keep the compiler happy when surrounding code changed.
It crashes when I try to run it without arguments, and no one
remembers how to use it.

Bug: webrtc:8396
Change-Id: I2eae36cf468f28c5bf05c85e6a3aaeebc48a1ffc
Reviewed-on: https://webrtc-review.googlesource.com/83581
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23618}
This commit is contained in:
Karl Wiberg
2018-06-14 11:46:53 +02:00
committed by Commit Bot
parent 0f173bde9e
commit 8fbe4f10e2
2 changed files with 0 additions and 370 deletions
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36
modules/audio_coding/BUILD.gn
View File

@ -1358,7 +1358,6 @@ if (rtc_include_tests) {
":g711_test",
":g722_test",
":ilbc_test",
":insert_packet_with_timing",
":isac_api_test",
":isac_fix_test",
":isac_switch_samprate_test",
@ -1552,41 +1551,6 @@ if (rtc_include_tests) {
]
} # delay_test
rtc_executable("insert_packet_with_timing") {
testonly = true
sources = [
"test/Channel.cc",
"test/Channel.h",
"test/PCMFile.cc",
"test/PCMFile.h",
"test/insert_packet_with_timing.cc",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":audio_coding",
":audio_format_conversion",
"..:module_api",
"../..:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:fileutils",
"../../test:test_support",
"../rtp_rtcp",
"//testing/gtest",
]
} # insert_packet_with_timing
audio_decoder_unittests_resources =
[ "../../resources/audio_coding/testfile32kHz.pcm" ]

334
modules/audio_coding/test/insert_packet_with_timing.cc
View File

@ -1,334 +0,0 @@
/*
* 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 <stdio.h>
#include <string.h>
#include <memory>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/audio_format_conversion.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/Channel.h"
#include "modules/audio_coding/test/PCMFile.h"
#include "rtc_base/flags.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
// Codec.
DEFINE_string(codec, "opus", "Codec Name");
DEFINE_int(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
DEFINE_int(codec_channels, 1, "Number of channels of the codec.");
// PCM input/output.
DEFINE_string(input, "", "Input PCM file at 16 kHz.");
DEFINE_bool(input_stereo, false, "Input is stereo.");
DEFINE_int(input_fs_hz, 32000, "Input sample rate Hz.");
DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile");
DEFINE_int(output_fs_hz, 32000, "Output sample rate Hz");
// Timing files
DEFINE_string(seq_num, "seq_num", "Sequence number file.");
DEFINE_string(send_ts, "send_timestamp", "Send timestamp file.");
DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
// Delay logging
DEFINE_string(delay, "", "Log for delay.");
// Other setups
DEFINE_bool(verbose, false, "Verbosity.");
DEFINE_float(loss_rate, 0, "Rate of packet loss < 1");
DEFINE_bool(help, false, "Prints this message.");
const int32_t kAudioPlayedOut = 0x00000001;
const int32_t kPacketPushedIn = 0x00000001 << 1;
const int kPlayoutPeriodMs = 10;
namespace webrtc {
namespace {
AudioCodingModule::Config AcmConfig(Clock* clock) {
AudioCodingModule::Config config;
config.clock = clock;
config.decoder_factory = CreateBuiltinAudioDecoderFactory();
return config;
}
} // namespace
class InsertPacketWithTiming {
public:
InsertPacketWithTiming()
: sender_clock_(new SimulatedClock(0)),
receiver_clock_(new SimulatedClock(0)),
send_acm_(AudioCodingModule::Create(AcmConfig(sender_clock_))),
receive_acm_(AudioCodingModule::Create(AcmConfig(receiver_clock_))),
channel_(new Channel),
seq_num_fid_(fopen(FLAG_seq_num, "rt")),
send_ts_fid_(fopen(FLAG_send_ts, "rt")),
receive_ts_fid_(fopen(FLAG_receive_ts, "rt")),
pcm_out_fid_(fopen(FLAG_output, "wb")),
samples_in_1ms_(48),
num_10ms_in_codec_frame_(2), // Typical 20 ms frames.
time_to_insert_packet_ms_(3), // An arbitrary offset on pushing packet.
next_receive_ts_(0),
time_to_playout_audio_ms_(kPlayoutPeriodMs),
loss_threshold_(0),
playout_timing_fid_(fopen("playout_timing.txt", "wt")) {}
void SetUp() {
ASSERT_TRUE(sender_clock_ != NULL);
ASSERT_TRUE(receiver_clock_ != NULL);
ASSERT_TRUE(send_acm_.get() != NULL);
ASSERT_TRUE(receive_acm_.get() != NULL);
ASSERT_TRUE(channel_ != NULL);
ASSERT_TRUE(seq_num_fid_ != NULL);
ASSERT_TRUE(send_ts_fid_ != NULL);
ASSERT_TRUE(receive_ts_fid_ != NULL);
ASSERT_TRUE(playout_timing_fid_ != NULL);
next_receive_ts_ = ReceiveTimestamp();
CodecInst codec;
ASSERT_EQ(0, AudioCodingModule::Codec(FLAG_codec, &codec,
FLAG_codec_sample_rate_hz,
FLAG_codec_channels));
ASSERT_EQ(0, receive_acm_->InitializeReceiver());
ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
ASSERT_EQ(true, receive_acm_->RegisterReceiveCodec(codec.pltype,
CodecInstToSdp(codec)));
// Set codec-dependent parameters.
samples_in_1ms_ = codec.plfreq / 1000;
num_10ms_in_codec_frame_ = codec.pacsize / (codec.plfreq / 100);
channel_->RegisterReceiverACM(receive_acm_.get());
send_acm_->RegisterTransportCallback(channel_);
if (strlen(FLAG_input) == 0) {
std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
"pcm");
pcm_in_fid_.Open(file_name, 32000, "r", true); // auto-rewind
std::cout << "Input file " << file_name << " 32 kHz mono." << std::endl;
} else {
pcm_in_fid_.Open(FLAG_input, static_cast<uint16_t>(FLAG_input_fs_hz),
"r", true); // auto-rewind
std::cout << "Input file " << FLAG_input << "at " << FLAG_input_fs_hz
<< " Hz in " << ((FLAG_input_stereo) ? "stereo." : "mono.")
<< std::endl;
pcm_in_fid_.ReadStereo(FLAG_input_stereo);
}
ASSERT_TRUE(pcm_out_fid_ != NULL);
std::cout << "Output file " << FLAG_output << " at " << FLAG_output_fs_hz
<< " Hz." << std::endl;
// Other setups
if (FLAG_loss_rate > 0)
loss_threshold_ = RAND_MAX * FLAG_loss_rate;
else
loss_threshold_ = 0;
}
void TickOneMillisecond(uint32_t* action) {
// One millisecond passed.
time_to_insert_packet_ms_--;
time_to_playout_audio_ms_--;
sender_clock_->AdvanceTimeMilliseconds(1);
receiver_clock_->AdvanceTimeMilliseconds(1);
// Reset action.
*action = 0;
// Is it time to pull audio?
if (time_to_playout_audio_ms_ == 0) {
time_to_playout_audio_ms_ = kPlayoutPeriodMs;
bool muted;
receive_acm_->PlayoutData10Ms(static_cast<int>(FLAG_output_fs_hz),
&frame_, &muted);
ASSERT_FALSE(muted);
fwrite(frame_.data(), sizeof(*frame_.data()),
frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
*action |= kAudioPlayedOut;
}
// Is it time to push in next packet?
if (time_to_insert_packet_ms_ <= .5) {
*action |= kPacketPushedIn;
// Update time-to-insert packet.
uint32_t t = next_receive_ts_;
next_receive_ts_ = ReceiveTimestamp();
time_to_insert_packet_ms_ += static_cast<float>(next_receive_ts_ - t) /
samples_in_1ms_;
// Push in just enough audio.
for (int n = 0; n < num_10ms_in_codec_frame_; n++) {
pcm_in_fid_.Read10MsData(frame_);
EXPECT_GE(send_acm_->Add10MsData(frame_), 0);
}
// Set the parameters for the packet to be pushed in receiver ACM right
// now.
uint32_t ts = SendTimestamp();
int seq_num = SequenceNumber();
bool lost = false;
channel_->set_send_timestamp(ts);
channel_->set_sequence_number(seq_num);
if (loss_threshold_ > 0 && rand() < loss_threshold_) {
channel_->set_num_packets_to_drop(1);
lost = true;
}
if (FLAG_verbose) {
if (!lost) {
std::cout << "\nInserting packet number " << seq_num
<< " timestamp " << ts << std::endl;
} else {
std::cout << "\nLost packet number " << seq_num
<< " timestamp " << ts << std::endl;
}
}
}
}
void TearDown() {
delete channel_;
fclose(seq_num_fid_);
fclose(send_ts_fid_);
fclose(receive_ts_fid_);
fclose(pcm_out_fid_);
pcm_in_fid_.Close();
}
~InsertPacketWithTiming() {
delete sender_clock_;
delete receiver_clock_;
}
// Are there more info to simulate.
bool HasPackets() {
if (feof(seq_num_fid_) || feof(send_ts_fid_) || feof(receive_ts_fid_))
return false;
return true;
}
// Jitter buffer delay.
void Delay(int* optimal_delay, int* current_delay) {
NetworkStatistics statistics;
receive_acm_->GetNetworkStatistics(&statistics);
*optimal_delay = statistics.preferredBufferSize;
*current_delay = statistics.currentBufferSize;
}
private:
uint32_t SendTimestamp() {
uint32_t t;
EXPECT_EQ(1, fscanf(send_ts_fid_, "%u\n", &t));
return t;
}
uint32_t ReceiveTimestamp() {
uint32_t t;
EXPECT_EQ(1, fscanf(receive_ts_fid_, "%u\n", &t));
return t;
}
int SequenceNumber() {
int n;
EXPECT_EQ(1, fscanf(seq_num_fid_, "%d\n", &n));
return n;
}
// This class just creates these pointers, not deleting them. They are deleted
// by the associated ACM.
SimulatedClock* sender_clock_;
SimulatedClock* receiver_clock_;
std::unique_ptr<AudioCodingModule> send_acm_;
std::unique_ptr<AudioCodingModule> receive_acm_;
Channel* channel_;
FILE* seq_num_fid_; // Input (text), one sequence number per line.
FILE* send_ts_fid_; // Input (text), one send timestamp per line.
FILE* receive_ts_fid_; // Input (text), one receive timestamp per line.
FILE* pcm_out_fid_; // Output PCM16.
PCMFile pcm_in_fid_; // Input PCM16.
int samples_in_1ms_;
// TODO(turajs): this can be computed from the send timestamp, but there is
// some complication to account for lost and reordered packets.
int num_10ms_in_codec_frame_;
float time_to_insert_packet_ms_;
uint32_t next_receive_ts_;
uint32_t time_to_playout_audio_ms_;
AudioFrame frame_;
double loss_threshold_;
// Output (text), sequence number, playout timestamp, time (ms) of playout,
// per line.
FILE* playout_timing_fid_;
};
} // webrtc
int main(int argc, char* argv[]) {
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
return 1;
}
if (FLAG_help) {
rtc::FlagList::Print(nullptr, false);
return 0;
}
webrtc::InsertPacketWithTiming test;
test.SetUp();
FILE* delay_log = NULL;
if (strlen(FLAG_delay) > 0) {
delay_log = fopen(FLAG_delay, "wt");
if (delay_log == NULL) {
std::cout << "Cannot open the file to log delay values." << std::endl;
exit(1);
}
}
uint32_t action_taken;
int optimal_delay_ms;
int current_delay_ms;
while (test.HasPackets()) {
test.TickOneMillisecond(&action_taken);
if (action_taken != 0) {
test.Delay(&optimal_delay_ms, &current_delay_ms);
if (delay_log != NULL) {
fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
}
}
}
std::cout << std::endl;
test.TearDown();
if (delay_log != NULL)
fclose(delay_log);
return 0;
}