| /* |
| * 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 "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 "modules/include/module_common_types.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 { |
| |
| class InsertPacketWithTiming { |
| public: |
| InsertPacketWithTiming() |
| : sender_clock_(new SimulatedClock(0)), |
| receiver_clock_(new SimulatedClock(0)), |
| send_acm_(AudioCodingModule::Create(sender_clock_)), |
| receive_acm_(AudioCodingModule::Create(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, ¤t_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); |
| } |
