| /* |
| * 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 <errno.h> |
| #include <inttypes.h> |
| #include <limits.h> // For ULONG_MAX returned by strtoul. |
| #include <stdio.h> |
| #include <stdlib.h> // For strtoul. |
| |
| #include <algorithm> |
| #include <ios> |
| #include <iostream> |
| #include <memory> |
| #include <numeric> |
| #include <string> |
| |
| #include "gflags/gflags.h" |
| #include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h" |
| #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| // Parses the input string for a valid SSRC (at the start of the string). If a |
| // valid SSRC is found, it is written to the output variable |ssrc|, and true is |
| // returned. Otherwise, false is returned. |
| bool ParseSsrc(const std::string& str, uint32_t* ssrc) { |
| if (str.empty()) |
| return true; |
| int base = 10; |
| // Look for "0x" or "0X" at the start and change base to 16 if found. |
| if ((str.compare(0, 2, "0x") == 0) || (str.compare(0, 2, "0X") == 0)) |
| base = 16; |
| errno = 0; |
| char* end_ptr; |
| unsigned long value = strtoul(str.c_str(), &end_ptr, base); |
| if (value == ULONG_MAX && errno == ERANGE) |
| return false; // Value out of range for unsigned long. |
| if (sizeof(unsigned long) > sizeof(uint32_t) && value > 0xFFFFFFFF) |
| return false; // Value out of range for uint32_t. |
| if (end_ptr - str.c_str() < static_cast<ptrdiff_t>(str.length())) |
| return false; // Part of the string was not parsed. |
| *ssrc = static_cast<uint32_t>(value); |
| return true; |
| } |
| |
| // Flag validators. |
| bool ValidatePayloadType(const char* flagname, int32_t value) { |
| if (value >= 0 && value <= 127) // Value is ok. |
| return true; |
| printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value)); |
| return false; |
| } |
| |
| bool ValidateSsrcValue(const char* flagname, const std::string& str) { |
| uint32_t dummy_ssrc; |
| return ParseSsrc(str, &dummy_ssrc); |
| } |
| |
| static bool ValidateExtensionId(const char* flagname, int32_t value) { |
| if (value > 0 && value <= 255) // Value is ok. |
| return true; |
| printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value)); |
| return false; |
| } |
| |
| // Define command line flags. |
| DEFINE_int32(pcmu, 0, "RTP payload type for PCM-u"); |
| const bool pcmu_dummy = |
| google::RegisterFlagValidator(&FLAGS_pcmu, &ValidatePayloadType); |
| DEFINE_int32(pcma, 8, "RTP payload type for PCM-a"); |
| const bool pcma_dummy = |
| google::RegisterFlagValidator(&FLAGS_pcma, &ValidatePayloadType); |
| DEFINE_int32(ilbc, 102, "RTP payload type for iLBC"); |
| const bool ilbc_dummy = |
| google::RegisterFlagValidator(&FLAGS_ilbc, &ValidatePayloadType); |
| DEFINE_int32(isac, 103, "RTP payload type for iSAC"); |
| const bool isac_dummy = |
| google::RegisterFlagValidator(&FLAGS_isac, &ValidatePayloadType); |
| DEFINE_int32(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)"); |
| const bool isac_swb_dummy = |
| google::RegisterFlagValidator(&FLAGS_isac_swb, &ValidatePayloadType); |
| DEFINE_int32(opus, 111, "RTP payload type for Opus"); |
| const bool opus_dummy = |
| google::RegisterFlagValidator(&FLAGS_opus, &ValidatePayloadType); |
| DEFINE_int32(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)"); |
| const bool pcm16b_dummy = |
| google::RegisterFlagValidator(&FLAGS_pcm16b, &ValidatePayloadType); |
| DEFINE_int32(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)"); |
| const bool pcm16b_wb_dummy = |
| google::RegisterFlagValidator(&FLAGS_pcm16b_wb, &ValidatePayloadType); |
| DEFINE_int32(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)"); |
| const bool pcm16b_swb32_dummy = |
| google::RegisterFlagValidator(&FLAGS_pcm16b_swb32, &ValidatePayloadType); |
| DEFINE_int32(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)"); |
| const bool pcm16b_swb48_dummy = |
| google::RegisterFlagValidator(&FLAGS_pcm16b_swb48, &ValidatePayloadType); |
| DEFINE_int32(g722, 9, "RTP payload type for G.722"); |
| const bool g722_dummy = |
| google::RegisterFlagValidator(&FLAGS_g722, &ValidatePayloadType); |
| DEFINE_int32(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)"); |
| const bool avt_dummy = |
| google::RegisterFlagValidator(&FLAGS_avt, &ValidatePayloadType); |
| DEFINE_int32(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)"); |
| const bool avt_16_dummy = |
| google::RegisterFlagValidator(&FLAGS_avt_16, &ValidatePayloadType); |
| DEFINE_int32(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)"); |
| const bool avt_32_dummy = |
| google::RegisterFlagValidator(&FLAGS_avt_32, &ValidatePayloadType); |
| DEFINE_int32(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)"); |
| const bool avt_48_dummy = |
| google::RegisterFlagValidator(&FLAGS_avt_48, &ValidatePayloadType); |
| DEFINE_int32(red, 117, "RTP payload type for redundant audio (RED)"); |
| const bool red_dummy = |
| google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType); |
| DEFINE_int32(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)"); |
| const bool cn_nb_dummy = |
| google::RegisterFlagValidator(&FLAGS_cn_nb, &ValidatePayloadType); |
| DEFINE_int32(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)"); |
| const bool cn_wb_dummy = |
| google::RegisterFlagValidator(&FLAGS_cn_wb, &ValidatePayloadType); |
| DEFINE_int32(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)"); |
| const bool cn_swb32_dummy = |
| google::RegisterFlagValidator(&FLAGS_cn_swb32, &ValidatePayloadType); |
| DEFINE_int32(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)"); |
| const bool cn_swb48_dummy = |
| google::RegisterFlagValidator(&FLAGS_cn_swb48, &ValidatePayloadType); |
| DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and " |
| "codec"); |
| DEFINE_string(replacement_audio_file, "", |
| "A PCM file that will be used to populate ""dummy"" RTP packets"); |
| DEFINE_string(ssrc, |
| "", |
| "Only use packets with this SSRC (decimal or hex, the latter " |
| "starting with 0x)"); |
| const bool hex_ssrc_dummy = |
| google::RegisterFlagValidator(&FLAGS_ssrc, &ValidateSsrcValue); |
| DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)"); |
| const bool audio_level_dummy = |
| google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId); |
| DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time"); |
| const bool abs_send_time_dummy = |
| google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId); |
| DEFINE_bool(matlabplot, |
| false, |
| "Generates a matlab script for plotting the delay profile"); |
| |
| // Maps a codec type to a printable name string. |
| std::string CodecName(NetEqDecoder codec) { |
| switch (codec) { |
| case NetEqDecoder::kDecoderPCMu: |
| return "PCM-u"; |
| case NetEqDecoder::kDecoderPCMa: |
| return "PCM-a"; |
| case NetEqDecoder::kDecoderILBC: |
| return "iLBC"; |
| case NetEqDecoder::kDecoderISAC: |
| return "iSAC"; |
| case NetEqDecoder::kDecoderISACswb: |
| return "iSAC-swb (32 kHz)"; |
| case NetEqDecoder::kDecoderOpus: |
| return "Opus"; |
| case NetEqDecoder::kDecoderPCM16B: |
| return "PCM16b-nb (8 kHz)"; |
| case NetEqDecoder::kDecoderPCM16Bwb: |
| return "PCM16b-wb (16 kHz)"; |
| case NetEqDecoder::kDecoderPCM16Bswb32kHz: |
| return "PCM16b-swb32 (32 kHz)"; |
| case NetEqDecoder::kDecoderPCM16Bswb48kHz: |
| return "PCM16b-swb48 (48 kHz)"; |
| case NetEqDecoder::kDecoderG722: |
| return "G.722"; |
| case NetEqDecoder::kDecoderRED: |
| return "redundant audio (RED)"; |
| case NetEqDecoder::kDecoderAVT: |
| return "AVT/DTMF (8 kHz)"; |
| case NetEqDecoder::kDecoderAVT16kHz: |
| return "AVT/DTMF (16 kHz)"; |
| case NetEqDecoder::kDecoderAVT32kHz: |
| return "AVT/DTMF (32 kHz)"; |
| case NetEqDecoder::kDecoderAVT48kHz: |
| return "AVT/DTMF (48 kHz)"; |
| case NetEqDecoder::kDecoderCNGnb: |
| return "comfort noise (8 kHz)"; |
| case NetEqDecoder::kDecoderCNGwb: |
| return "comfort noise (16 kHz)"; |
| case NetEqDecoder::kDecoderCNGswb32kHz: |
| return "comfort noise (32 kHz)"; |
| case NetEqDecoder::kDecoderCNGswb48kHz: |
| return "comfort noise (48 kHz)"; |
| default: |
| FATAL(); |
| return "undefined"; |
| } |
| } |
| |
| void PrintCodecMappingEntry(NetEqDecoder codec, google::int32 flag) { |
| std::cout << CodecName(codec) << ": " << flag << std::endl; |
| } |
| |
| void PrintCodecMapping() { |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAGS_pcmu); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAGS_pcma); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAGS_ilbc); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAGS_isac); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAGS_isac_swb); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAGS_opus); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAGS_pcm16b); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAGS_pcm16b_wb); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb32kHz, |
| FLAGS_pcm16b_swb32); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb48kHz, |
| FLAGS_pcm16b_swb48); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAGS_g722); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAGS_avt); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAGS_avt_16); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAGS_avt_32); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAGS_avt_48); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAGS_red); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAGS_cn_nb); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAGS_cn_wb); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAGS_cn_swb32); |
| PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAGS_cn_swb48); |
| } |
| |
| rtc::Optional<int> CodecSampleRate(uint8_t payload_type) { |
| if (payload_type == FLAGS_pcmu || payload_type == FLAGS_pcma || |
| payload_type == FLAGS_ilbc || payload_type == FLAGS_pcm16b || |
| payload_type == FLAGS_cn_nb || payload_type == FLAGS_avt) |
| return rtc::Optional<int>(8000); |
| if (payload_type == FLAGS_isac || payload_type == FLAGS_pcm16b_wb || |
| payload_type == FLAGS_g722 || payload_type == FLAGS_cn_wb || |
| payload_type == FLAGS_avt_16) |
| return rtc::Optional<int>(16000); |
| if (payload_type == FLAGS_isac_swb || payload_type == FLAGS_pcm16b_swb32 || |
| payload_type == FLAGS_cn_swb32 || payload_type == FLAGS_avt_32) |
| return rtc::Optional<int>(32000); |
| if (payload_type == FLAGS_opus || payload_type == FLAGS_pcm16b_swb48 || |
| payload_type == FLAGS_cn_swb48 || payload_type == FLAGS_avt_48) |
| return rtc::Optional<int>(48000); |
| if (payload_type == FLAGS_red) |
| return rtc::Optional<int>(0); |
| return rtc::Optional<int>(); |
| } |
| |
| // Class to let through only the packets with a given SSRC. Should be used as an |
| // outer layer on another NetEqInput object. |
| class FilterSsrcInput : public NetEqInput { |
| public: |
| FilterSsrcInput(std::unique_ptr<NetEqInput> source, uint32_t ssrc) |
| : source_(std::move(source)), ssrc_(ssrc) { |
| FindNextWithCorrectSsrc(); |
| RTC_CHECK(source_->NextHeader()) << "Found no packet with SSRC = 0x" |
| << std::hex << ssrc_; |
| } |
| |
| // All methods but PopPacket() simply relay to the |source_| object. |
| rtc::Optional<int64_t> NextPacketTime() const override { |
| return source_->NextPacketTime(); |
| } |
| rtc::Optional<int64_t> NextOutputEventTime() const override { |
| return source_->NextOutputEventTime(); |
| } |
| |
| // Returns the next packet, and throws away upcoming packets that do not match |
| // the desired SSRC. |
| std::unique_ptr<PacketData> PopPacket() override { |
| std::unique_ptr<PacketData> packet_to_return = source_->PopPacket(); |
| RTC_DCHECK(!packet_to_return || packet_to_return->header.ssrc == ssrc_); |
| // Pre-fetch the next packet with correct SSRC. Hence, |source_| will always |
| // be have a valid packet (or empty if no more packets are available) when |
| // this method returns. |
| FindNextWithCorrectSsrc(); |
| return packet_to_return; |
| } |
| |
| void AdvanceOutputEvent() override { source_->AdvanceOutputEvent(); } |
| |
| bool ended() const override { return source_->ended(); } |
| |
| rtc::Optional<RTPHeader> NextHeader() const override { |
| return source_->NextHeader(); |
| } |
| |
| private: |
| void FindNextWithCorrectSsrc() { |
| while (source_->NextHeader() && source_->NextHeader()->ssrc != ssrc_) { |
| source_->PopPacket(); |
| } |
| } |
| |
| std::unique_ptr<NetEqInput> source_; |
| uint32_t ssrc_; |
| }; |
| |
| // A callback class which prints whenver the inserted packet stream changes |
| // the SSRC. |
| class SsrcSwitchDetector : public NetEqPostInsertPacket { |
| public: |
| // Takes a pointer to another callback object, which will be invoked after |
| // this object finishes. This does not transfer ownership, and null is a |
| // valid value. |
| explicit SsrcSwitchDetector(NetEqPostInsertPacket* other_callback) |
| : other_callback_(other_callback) {} |
| |
| void AfterInsertPacket(const NetEqInput::PacketData& packet, |
| NetEq* neteq) override { |
| if (last_ssrc_ && packet.header.ssrc != *last_ssrc_) { |
| std::cout << "Changing streams from 0x" << std::hex << *last_ssrc_ |
| << " to 0x" << std::hex << packet.header.ssrc |
| << std::dec << " (payload type " |
| << static_cast<int>(packet.header.payloadType) << ")" |
| << std::endl; |
| } |
| last_ssrc_ = rtc::Optional<uint32_t>(packet.header.ssrc); |
| if (other_callback_) { |
| other_callback_->AfterInsertPacket(packet, neteq); |
| } |
| } |
| |
| private: |
| NetEqPostInsertPacket* other_callback_; |
| rtc::Optional<uint32_t> last_ssrc_; |
| }; |
| |
| class StatsGetter : public NetEqGetAudioCallback { |
| public: |
| // This struct is a replica of webrtc::NetEqNetworkStatistics, but with all |
| // values stored in double precision. |
| struct Stats { |
| double current_buffer_size_ms = 0.0; |
| double preferred_buffer_size_ms = 0.0; |
| double jitter_peaks_found = 0.0; |
| double packet_loss_rate = 0.0; |
| double expand_rate = 0.0; |
| double speech_expand_rate = 0.0; |
| double preemptive_rate = 0.0; |
| double accelerate_rate = 0.0; |
| double secondary_decoded_rate = 0.0; |
| double secondary_discarded_rate = 0.0; |
| double clockdrift_ppm = 0.0; |
| double added_zero_samples = 0.0; |
| double mean_waiting_time_ms = 0.0; |
| double median_waiting_time_ms = 0.0; |
| double min_waiting_time_ms = 0.0; |
| double max_waiting_time_ms = 0.0; |
| }; |
| |
| // Takes a pointer to another callback object, which will be invoked after |
| // this object finishes. This does not transfer ownership, and null is a |
| // valid value. |
| explicit StatsGetter(NetEqGetAudioCallback* other_callback) |
| : other_callback_(other_callback) {} |
| |
| void BeforeGetAudio(NetEq* neteq) override { |
| if (other_callback_) { |
| other_callback_->BeforeGetAudio(neteq); |
| } |
| } |
| |
| void AfterGetAudio(int64_t time_now_ms, |
| const AudioFrame& audio_frame, |
| bool muted, |
| NetEq* neteq) override { |
| if (++counter_ >= 100) { |
| counter_ = 0; |
| NetEqNetworkStatistics stats; |
| RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0); |
| stats_.push_back(stats); |
| } |
| if (other_callback_) { |
| other_callback_->BeforeGetAudio(neteq); |
| } |
| } |
| |
| double AverageSpeechExpandRate() const { |
| double sum_speech_expand = |
| std::accumulate(stats_.begin(), stats_.end(), double{0.0}, |
| [](double a, NetEqNetworkStatistics b) { |
| return a + static_cast<double>(b.speech_expand_rate); |
| }); |
| return sum_speech_expand / 16384.0 / stats_.size(); |
| } |
| |
| Stats AverageStats() const { |
| Stats sum_stats = std::accumulate( |
| stats_.begin(), stats_.end(), Stats(), |
| [](Stats a, NetEqNetworkStatistics b) { |
| a.current_buffer_size_ms += b.current_buffer_size_ms; |
| a.preferred_buffer_size_ms += b.preferred_buffer_size_ms; |
| a.jitter_peaks_found += b.jitter_peaks_found; |
| a.packet_loss_rate += b.packet_loss_rate / 16384.0; |
| a.expand_rate += b.expand_rate / 16384.0; |
| a.speech_expand_rate += b.speech_expand_rate / 16384.0; |
| a.preemptive_rate += b.preemptive_rate / 16384.0; |
| a.accelerate_rate += b.accelerate_rate / 16384.0; |
| a.secondary_decoded_rate += b.secondary_decoded_rate / 16384.0; |
| a.secondary_discarded_rate += b.secondary_discarded_rate / 16384.0; |
| a.clockdrift_ppm += b.clockdrift_ppm; |
| a.added_zero_samples += b.added_zero_samples; |
| a.mean_waiting_time_ms += b.mean_waiting_time_ms; |
| a.median_waiting_time_ms += b.median_waiting_time_ms; |
| a.min_waiting_time_ms += b.min_waiting_time_ms; |
| a.max_waiting_time_ms += b.max_waiting_time_ms; |
| return a; |
| }); |
| |
| sum_stats.current_buffer_size_ms /= stats_.size(); |
| sum_stats.preferred_buffer_size_ms /= stats_.size(); |
| sum_stats.jitter_peaks_found /= stats_.size(); |
| sum_stats.packet_loss_rate /= stats_.size(); |
| sum_stats.expand_rate /= stats_.size(); |
| sum_stats.speech_expand_rate /= stats_.size(); |
| sum_stats.preemptive_rate /= stats_.size(); |
| sum_stats.accelerate_rate /= stats_.size(); |
| sum_stats.secondary_decoded_rate /= stats_.size(); |
| sum_stats.secondary_discarded_rate /= stats_.size(); |
| sum_stats.clockdrift_ppm /= stats_.size(); |
| sum_stats.added_zero_samples /= stats_.size(); |
| sum_stats.mean_waiting_time_ms /= stats_.size(); |
| sum_stats.median_waiting_time_ms /= stats_.size(); |
| sum_stats.min_waiting_time_ms /= stats_.size(); |
| sum_stats.max_waiting_time_ms /= stats_.size(); |
| |
| return sum_stats; |
| } |
| |
| private: |
| NetEqGetAudioCallback* other_callback_; |
| size_t counter_ = 0; |
| std::vector<NetEqNetworkStatistics> stats_; |
| }; |
| |
| int RunTest(int argc, char* argv[]) { |
| std::string program_name = argv[0]; |
| std::string usage = "Tool for decoding an RTP dump file using NetEq.\n" |
| "Run " + program_name + " --helpshort for usage.\n" |
| "Example usage:\n" + program_name + |
| " input.rtp output.{pcm, wav}\n"; |
| google::SetUsageMessage(usage); |
| google::ParseCommandLineFlags(&argc, &argv, true); |
| |
| if (FLAGS_codec_map) { |
| PrintCodecMapping(); |
| } |
| |
| if (argc != 3) { |
| if (FLAGS_codec_map) { |
| // We have already printed the codec map. Just end the program. |
| return 0; |
| } |
| // Print usage information. |
| std::cout << google::ProgramUsage(); |
| return 0; |
| } |
| |
| // Gather RTP header extensions in a map. |
| NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = { |
| {FLAGS_audio_level, kRtpExtensionAudioLevel}, |
| {FLAGS_abs_send_time, kRtpExtensionAbsoluteSendTime}}; |
| |
| const std::string input_file_name = argv[1]; |
| std::unique_ptr<NetEqInput> input; |
| if (RtpFileSource::ValidRtpDump(input_file_name) || |
| RtpFileSource::ValidPcap(input_file_name)) { |
| input.reset(new NetEqRtpDumpInput(input_file_name, rtp_ext_map)); |
| } else { |
| input.reset(new NetEqEventLogInput(input_file_name, rtp_ext_map)); |
| } |
| |
| std::cout << "Input file: " << input_file_name << std::endl; |
| RTC_CHECK(input) << "Cannot open input file"; |
| RTC_CHECK(!input->ended()) << "Input file is empty"; |
| |
| // Check if an SSRC value was provided. |
| if (!FLAGS_ssrc.empty()) { |
| uint32_t ssrc; |
| RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc)) << "Flag verification has failed."; |
| input.reset(new FilterSsrcInput(std::move(input), ssrc)); |
| } |
| |
| // Check the sample rate. |
| rtc::Optional<int> sample_rate_hz; |
| std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc; |
| while (input->NextHeader()) { |
| rtc::Optional<RTPHeader> first_rtp_header = input->NextHeader(); |
| RTC_DCHECK(first_rtp_header); |
| sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType); |
| if (sample_rate_hz) { |
| std::cout << "Found valid packet with payload type " |
| << static_cast<int>(first_rtp_header->payloadType) |
| << " and SSRC 0x" << std::hex << first_rtp_header->ssrc |
| << std::dec << std::endl; |
| break; |
| } |
| // Discard this packet and move to the next. Keep track of discarded payload |
| // types and SSRCs. |
| discarded_pt_and_ssrc.emplace(first_rtp_header->payloadType, |
| first_rtp_header->ssrc); |
| input->PopPacket(); |
| } |
| if (!discarded_pt_and_ssrc.empty()) { |
| std::cout << "Discarded initial packets with the following payload types " |
| "and SSRCs:" |
| << std::endl; |
| for (const auto& d : discarded_pt_and_ssrc) { |
| std::cout << "PT " << d.first << "; SSRC 0x" << std::hex |
| << static_cast<int>(d.second) << std::dec << std::endl; |
| } |
| } |
| if (!sample_rate_hz) { |
| std::cout << "Cannot find any packets with known payload types" |
| << std::endl; |
| RTC_NOTREACHED(); |
| } |
| |
| // Open the output file now that we know the sample rate. (Rate is only needed |
| // for wav files.) |
| const std::string output_file_name = argv[2]; |
| std::unique_ptr<AudioSink> output; |
| if (output_file_name.size() >= 4 && |
| output_file_name.substr(output_file_name.size() - 4) == ".wav") { |
| // Open a wav file. |
| output.reset(new OutputWavFile(output_file_name, *sample_rate_hz)); |
| } else { |
| // Open a pcm file. |
| output.reset(new OutputAudioFile(output_file_name)); |
| } |
| |
| std::cout << "Output file: " << output_file_name << std::endl; |
| |
| NetEqTest::DecoderMap codecs = { |
| {FLAGS_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")}, |
| {FLAGS_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")}, |
| {FLAGS_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")}, |
| {FLAGS_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")}, |
| {FLAGS_isac_swb, |
| std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")}, |
| {FLAGS_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")}, |
| {FLAGS_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")}, |
| {FLAGS_pcm16b_wb, |
| std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")}, |
| {FLAGS_pcm16b_swb32, |
| std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")}, |
| {FLAGS_pcm16b_swb48, |
| std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")}, |
| {FLAGS_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")}, |
| {FLAGS_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")}, |
| {FLAGS_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")}, |
| {FLAGS_avt_32, |
| std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")}, |
| {FLAGS_avt_48, |
| std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")}, |
| {FLAGS_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")}, |
| {FLAGS_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")}, |
| {FLAGS_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")}, |
| {FLAGS_cn_swb32, |
| std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")}, |
| {FLAGS_cn_swb48, |
| std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48")}}; |
| |
| // Check if a replacement audio file was provided. |
| std::unique_ptr<AudioDecoder> replacement_decoder; |
| NetEqTest::ExtDecoderMap ext_codecs; |
| if (!FLAGS_replacement_audio_file.empty()) { |
| // Find largest unused payload type. |
| int replacement_pt = 127; |
| while (!(codecs.find(replacement_pt) == codecs.end() && |
| ext_codecs.find(replacement_pt) == ext_codecs.end())) { |
| --replacement_pt; |
| RTC_CHECK_GE(replacement_pt, 0); |
| } |
| |
| auto std_set_int32_to_uint8 = [](const std::set<int32_t>& a) { |
| std::set<uint8_t> b; |
| for (auto& x : a) { |
| b.insert(static_cast<uint8_t>(x)); |
| } |
| return b; |
| }; |
| |
| std::set<uint8_t> cn_types = std_set_int32_to_uint8( |
| {FLAGS_cn_nb, FLAGS_cn_wb, FLAGS_cn_swb32, FLAGS_cn_swb48}); |
| std::set<uint8_t> forbidden_types = |
| std_set_int32_to_uint8({FLAGS_g722, FLAGS_red, FLAGS_avt, |
| FLAGS_avt_16, FLAGS_avt_32, FLAGS_avt_48}); |
| input.reset(new NetEqReplacementInput(std::move(input), replacement_pt, |
| cn_types, forbidden_types)); |
| |
| replacement_decoder.reset(new FakeDecodeFromFile( |
| std::unique_ptr<InputAudioFile>( |
| new InputAudioFile(FLAGS_replacement_audio_file)), |
| 48000, false)); |
| NetEqTest::ExternalDecoderInfo ext_dec_info = { |
| replacement_decoder.get(), NetEqDecoder::kDecoderArbitrary, |
| "replacement codec"}; |
| ext_codecs[replacement_pt] = ext_dec_info; |
| } |
| |
| NetEqTest::Callbacks callbacks; |
| std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer; |
| if (FLAGS_matlabplot) { |
| delay_analyzer.reset(new NetEqDelayAnalyzer); |
| } |
| |
| SsrcSwitchDetector ssrc_switch_detector(delay_analyzer.get()); |
| callbacks.post_insert_packet = &ssrc_switch_detector; |
| StatsGetter stats_getter(delay_analyzer.get()); |
| callbacks.get_audio_callback = &stats_getter; |
| NetEq::Config config; |
| config.sample_rate_hz = *sample_rate_hz; |
| NetEqTest test(config, codecs, ext_codecs, std::move(input), |
| std::move(output), callbacks); |
| |
| int64_t test_duration_ms = test.Run(); |
| |
| if (FLAGS_matlabplot) { |
| std::cout << "Creating Matlab plot script " << output_file_name + ".m" |
| << std::endl; |
| delay_analyzer->CreateMatlabScript(output_file_name + ".m"); |
| } |
| |
| printf("Simulation statistics:\n"); |
| printf(" output duration: %" PRId64 " ms\n", test_duration_ms); |
| auto stats = stats_getter.AverageStats(); |
| printf(" packet_loss_rate: %f %%\n", 100.0 * stats.packet_loss_rate); |
| printf(" expand_rate: %f %%\n", 100.0 * stats.expand_rate); |
| printf(" speech_expand_rate: %f %%\n", 100.0 * stats.speech_expand_rate); |
| printf(" preemptive_rate: %f %%\n", 100.0 * stats.preemptive_rate); |
| printf(" accelerate_rate: %f %%\n", 100.0 * stats.accelerate_rate); |
| printf(" secondary_decoded_rate: %f %%\n", |
| 100.0 * stats.secondary_decoded_rate); |
| printf(" secondary_discarded_rate: %f %%\n", |
| 100.0 * stats.secondary_discarded_rate); |
| printf(" clockdrift_ppm: %f ppm\n", stats.clockdrift_ppm); |
| printf(" mean_waiting_time_ms: %f ms\n", stats.mean_waiting_time_ms); |
| printf(" median_waiting_time_ms: %f ms\n", stats.median_waiting_time_ms); |
| printf(" min_waiting_time_ms: %f ms\n", stats.min_waiting_time_ms); |
| printf(" max_waiting_time_ms: %f ms\n", stats.max_waiting_time_ms); |
| |
| return 0; |
| } |
| |
| } // namespace |
| } // namespace test |
| } // namespace webrtc |
| |
| int main(int argc, char* argv[]) { |
| webrtc::test::RunTest(argc, argv); |
| } |