Index: webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc |
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc |
deleted file mode 100644 |
index 857381d250d011ebe78a0f8d350bdae894c3833b..0000000000000000000000000000000000000000 |
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc |
+++ /dev/null |
@@ -1,307 +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 "gflags/gflags.h" |
-#include "testing/gtest/include/gtest/gtest.h" |
-#include "webrtc/base/scoped_ptr.h" |
-#include "webrtc/common_types.h" |
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" |
-#include "webrtc/modules/audio_coding/main/test/Channel.h" |
-#include "webrtc/modules/audio_coding/main/test/PCMFile.h" |
-#include "webrtc/modules/include/module_common_types.h" |
-#include "webrtc/system_wrappers/include/clock.h" |
-#include "webrtc/test/testsupport/fileutils.h" |
- |
-// Codec. |
-DEFINE_string(codec, "opus", "Codec Name"); |
-DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz."); |
-DEFINE_int32(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_int32(input_fs_hz, 32000, "Input sample rate Hz."); |
-DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile"); |
-DEFINE_int32(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_double(loss_rate, 0, "Rate of packet loss < 1"); |
- |
-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(0, sender_clock_)), |
- receive_acm_(AudioCodingModule::Create(0, receiver_clock_)), |
- channel_(new Channel), |
- seq_num_fid_(fopen(FLAGS_seq_num.c_str(), "rt")), |
- send_ts_fid_(fopen(FLAGS_send_ts.c_str(), "rt")), |
- receive_ts_fid_(fopen(FLAGS_receive_ts.c_str(), "rt")), |
- pcm_out_fid_(fopen(FLAGS_output.c_str(), "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(FLAGS_codec.c_str(), &codec, |
- FLAGS_codec_sample_rate_hz, |
- FLAGS_codec_channels)); |
- ASSERT_EQ(0, receive_acm_->InitializeReceiver()); |
- ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec)); |
- ASSERT_EQ(0, receive_acm_->RegisterReceiveCodec(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 (FLAGS_input.size() == 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(FLAGS_input, static_cast<uint16_t>(FLAGS_input_fs_hz), |
- "r", true); // auto-rewind |
- std::cout << "Input file " << FLAGS_input << "at " << FLAGS_input_fs_hz |
- << " Hz in " << ((FLAGS_input_stereo) ? "stereo." : "mono.") |
- << std::endl; |
- pcm_in_fid_.ReadStereo(FLAGS_input_stereo); |
- } |
- |
- ASSERT_TRUE(pcm_out_fid_ != NULL); |
- std::cout << "Output file " << FLAGS_output << " at " << FLAGS_output_fs_hz |
- << " Hz." << std::endl; |
- |
- // Other setups |
- if (FLAGS_loss_rate > 0) |
- loss_threshold_ = RAND_MAX * FLAGS_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; |
- receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz), |
- &frame_); |
- fwrite(frame_.data_, sizeof(frame_.data_[0]), |
- 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 (FLAGS_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_; |
- |
- rtc::scoped_ptr<AudioCodingModule> send_acm_; |
- rtc::scoped_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[]) { |
- google::ParseCommandLineFlags(&argc, &argv, true); |
- webrtc::InsertPacketWithTiming test; |
- test.SetUp(); |
- |
- FILE* delay_log = NULL; |
- if (FLAGS_delay.size() > 0) { |
- delay_log = fopen(FLAGS_delay.c_str(), "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); |
-} |