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- Directly include api/audio/audio_frame.h everywhere AudioFrame is used. - This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM. - audio_frame.h still included from module_common_types.h for backwards compatibility with clients. Bug: webrtc:9139, webrtc:7504 Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897 Reviewed-on: https://webrtc-review.googlesource.com/62464 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22845}
739 lines
29 KiB
C++
739 lines
29 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 <errno.h>
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#include <inttypes.h>
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#include <limits.h> // For ULONG_MAX returned by strtoul.
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#include <stdio.h>
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#include <stdlib.h> // For strtoul.
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#include <string.h>
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#include <algorithm>
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#include <ios>
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#include <iostream>
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#include <memory>
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#include <numeric>
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#include <string>
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#include "modules/audio_coding/neteq/include/neteq.h"
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#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
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#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
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#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
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#include "modules/audio_coding/neteq/tools/neteq_test.h"
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#include "modules/audio_coding/neteq/tools/output_audio_file.h"
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#include "modules/audio_coding/neteq/tools/output_wav_file.h"
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/flags.h"
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#include "test/testsupport/fileutils.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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namespace test {
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namespace {
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// Parses the input string for a valid SSRC (at the start of the string). If a
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// valid SSRC is found, it is written to the output variable |ssrc|, and true is
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// returned. Otherwise, false is returned.
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bool ParseSsrc(const std::string& str, uint32_t* ssrc) {
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if (str.empty())
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return true;
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int base = 10;
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// Look for "0x" or "0X" at the start and change base to 16 if found.
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if ((str.compare(0, 2, "0x") == 0) || (str.compare(0, 2, "0X") == 0))
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base = 16;
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errno = 0;
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char* end_ptr;
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unsigned long value = strtoul(str.c_str(), &end_ptr, base);
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if (value == ULONG_MAX && errno == ERANGE)
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return false; // Value out of range for unsigned long.
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if (sizeof(unsigned long) > sizeof(uint32_t) && value > 0xFFFFFFFF)
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return false; // Value out of range for uint32_t.
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if (end_ptr - str.c_str() < static_cast<ptrdiff_t>(str.length()))
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return false; // Part of the string was not parsed.
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*ssrc = static_cast<uint32_t>(value);
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return true;
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}
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// Flag validators.
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bool ValidatePayloadType(int value) {
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if (value >= 0 && value <= 127) // Value is ok.
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return true;
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printf("Payload type must be between 0 and 127, not %d\n",
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static_cast<int>(value));
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return false;
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}
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bool ValidateSsrcValue(const std::string& str) {
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uint32_t dummy_ssrc;
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if (ParseSsrc(str, &dummy_ssrc)) // Value is ok.
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return true;
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printf("Invalid SSRC: %s\n", str.c_str());
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return false;
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}
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static bool ValidateExtensionId(int value) {
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if (value > 0 && value <= 255) // Value is ok.
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return true;
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printf("Extension ID must be between 1 and 255, not %d\n",
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static_cast<int>(value));
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return false;
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}
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// Define command line flags.
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DEFINE_int(pcmu, 0, "RTP payload type for PCM-u");
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DEFINE_int(pcma, 8, "RTP payload type for PCM-a");
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DEFINE_int(ilbc, 102, "RTP payload type for iLBC");
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DEFINE_int(isac, 103, "RTP payload type for iSAC");
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DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
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DEFINE_int(opus, 111, "RTP payload type for Opus");
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DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
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DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
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DEFINE_int(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
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DEFINE_int(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
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DEFINE_int(g722, 9, "RTP payload type for G.722");
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DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
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DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
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DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
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DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
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DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)");
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DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
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DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
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DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
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DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
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DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and "
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"codec");
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DEFINE_string(replacement_audio_file, "",
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"A PCM file that will be used to populate ""dummy"" RTP packets");
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DEFINE_string(ssrc,
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"",
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"Only use packets with this SSRC (decimal or hex, the latter "
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"starting with 0x)");
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DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
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DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
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DEFINE_int(transport_seq_no, 5, "Extension ID for transport sequence number");
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DEFINE_bool(matlabplot,
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false,
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"Generates a matlab script for plotting the delay profile");
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DEFINE_bool(pythonplot,
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false,
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"Generates a python script for plotting the delay profile");
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DEFINE_bool(help, false, "Prints this message");
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DEFINE_bool(concealment_events, false, "Prints concealment events");
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// Maps a codec type to a printable name string.
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std::string CodecName(NetEqDecoder codec) {
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switch (codec) {
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case NetEqDecoder::kDecoderPCMu:
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return "PCM-u";
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case NetEqDecoder::kDecoderPCMa:
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return "PCM-a";
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case NetEqDecoder::kDecoderILBC:
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return "iLBC";
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case NetEqDecoder::kDecoderISAC:
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return "iSAC";
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case NetEqDecoder::kDecoderISACswb:
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return "iSAC-swb (32 kHz)";
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case NetEqDecoder::kDecoderOpus:
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return "Opus";
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case NetEqDecoder::kDecoderPCM16B:
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return "PCM16b-nb (8 kHz)";
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case NetEqDecoder::kDecoderPCM16Bwb:
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return "PCM16b-wb (16 kHz)";
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case NetEqDecoder::kDecoderPCM16Bswb32kHz:
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return "PCM16b-swb32 (32 kHz)";
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case NetEqDecoder::kDecoderPCM16Bswb48kHz:
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return "PCM16b-swb48 (48 kHz)";
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case NetEqDecoder::kDecoderG722:
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return "G.722";
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case NetEqDecoder::kDecoderRED:
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return "redundant audio (RED)";
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case NetEqDecoder::kDecoderAVT:
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return "AVT/DTMF (8 kHz)";
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case NetEqDecoder::kDecoderAVT16kHz:
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return "AVT/DTMF (16 kHz)";
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case NetEqDecoder::kDecoderAVT32kHz:
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return "AVT/DTMF (32 kHz)";
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case NetEqDecoder::kDecoderAVT48kHz:
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return "AVT/DTMF (48 kHz)";
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case NetEqDecoder::kDecoderCNGnb:
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return "comfort noise (8 kHz)";
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case NetEqDecoder::kDecoderCNGwb:
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return "comfort noise (16 kHz)";
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case NetEqDecoder::kDecoderCNGswb32kHz:
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return "comfort noise (32 kHz)";
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case NetEqDecoder::kDecoderCNGswb48kHz:
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return "comfort noise (48 kHz)";
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default:
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FATAL();
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return "undefined";
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}
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}
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void PrintCodecMappingEntry(NetEqDecoder codec, int flag) {
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std::cout << CodecName(codec) << ": " << flag << std::endl;
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}
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void PrintCodecMapping() {
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PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAG_pcmu);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAG_pcma);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAG_ilbc);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAG_isac);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAG_isac_swb);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAG_opus);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAG_pcm16b);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAG_pcm16b_wb);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb32kHz,
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FLAG_pcm16b_swb32);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb48kHz,
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FLAG_pcm16b_swb48);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAG_g722);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAG_avt);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAG_avt_16);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAG_avt_32);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAG_avt_48);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAG_red);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAG_cn_nb);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAG_cn_wb);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAG_cn_swb32);
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PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAG_cn_swb48);
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}
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rtc::Optional<int> CodecSampleRate(uint8_t payload_type) {
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if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
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payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
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payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
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return 8000;
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if (payload_type == FLAG_isac || payload_type == FLAG_pcm16b_wb ||
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payload_type == FLAG_g722 || payload_type == FLAG_cn_wb ||
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payload_type == FLAG_avt_16)
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return 16000;
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if (payload_type == FLAG_isac_swb || payload_type == FLAG_pcm16b_swb32 ||
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payload_type == FLAG_cn_swb32 || payload_type == FLAG_avt_32)
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return 32000;
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if (payload_type == FLAG_opus || payload_type == FLAG_pcm16b_swb48 ||
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payload_type == FLAG_cn_swb48 || payload_type == FLAG_avt_48)
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return 48000;
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if (payload_type == FLAG_red)
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return 0;
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return rtc::nullopt;
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}
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// Class to let through only the packets with a given SSRC. Should be used as an
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// outer layer on another NetEqInput object.
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class FilterSsrcInput : public NetEqInput {
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public:
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FilterSsrcInput(std::unique_ptr<NetEqInput> source, uint32_t ssrc)
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: source_(std::move(source)), ssrc_(ssrc) {
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FindNextWithCorrectSsrc();
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RTC_CHECK(source_->NextHeader()) << "Found no packet with SSRC = 0x"
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<< std::hex << ssrc_;
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}
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// All methods but PopPacket() simply relay to the |source_| object.
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rtc::Optional<int64_t> NextPacketTime() const override {
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return source_->NextPacketTime();
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}
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rtc::Optional<int64_t> NextOutputEventTime() const override {
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return source_->NextOutputEventTime();
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}
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// Returns the next packet, and throws away upcoming packets that do not match
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// the desired SSRC.
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std::unique_ptr<PacketData> PopPacket() override {
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std::unique_ptr<PacketData> packet_to_return = source_->PopPacket();
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RTC_DCHECK(!packet_to_return || packet_to_return->header.ssrc == ssrc_);
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// Pre-fetch the next packet with correct SSRC. Hence, |source_| will always
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// be have a valid packet (or empty if no more packets are available) when
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// this method returns.
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FindNextWithCorrectSsrc();
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return packet_to_return;
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}
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void AdvanceOutputEvent() override { source_->AdvanceOutputEvent(); }
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bool ended() const override { return source_->ended(); }
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rtc::Optional<RTPHeader> NextHeader() const override {
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return source_->NextHeader();
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}
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private:
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void FindNextWithCorrectSsrc() {
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while (source_->NextHeader() && source_->NextHeader()->ssrc != ssrc_) {
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source_->PopPacket();
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}
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}
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std::unique_ptr<NetEqInput> source_;
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uint32_t ssrc_;
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};
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// A callback class which prints whenver the inserted packet stream changes
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// the SSRC.
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class SsrcSwitchDetector : public NetEqPostInsertPacket {
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public:
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// Takes a pointer to another callback object, which will be invoked after
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// this object finishes. This does not transfer ownership, and null is a
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// valid value.
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explicit SsrcSwitchDetector(NetEqPostInsertPacket* other_callback)
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: other_callback_(other_callback) {}
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void AfterInsertPacket(const NetEqInput::PacketData& packet,
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NetEq* neteq) override {
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if (last_ssrc_ && packet.header.ssrc != *last_ssrc_) {
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std::cout << "Changing streams from 0x" << std::hex << *last_ssrc_
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<< " to 0x" << std::hex << packet.header.ssrc
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<< std::dec << " (payload type "
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<< static_cast<int>(packet.header.payloadType) << ")"
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<< std::endl;
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}
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last_ssrc_ = packet.header.ssrc;
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if (other_callback_) {
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other_callback_->AfterInsertPacket(packet, neteq);
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}
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}
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private:
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NetEqPostInsertPacket* other_callback_;
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rtc::Optional<uint32_t> last_ssrc_;
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};
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class StatsGetter : public NetEqGetAudioCallback {
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public:
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// This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
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// values stored in double precision.
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struct Stats {
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double current_buffer_size_ms = 0.0;
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double preferred_buffer_size_ms = 0.0;
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double jitter_peaks_found = 0.0;
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double packet_loss_rate = 0.0;
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double expand_rate = 0.0;
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double speech_expand_rate = 0.0;
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double preemptive_rate = 0.0;
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double accelerate_rate = 0.0;
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double secondary_decoded_rate = 0.0;
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double secondary_discarded_rate = 0.0;
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double clockdrift_ppm = 0.0;
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double added_zero_samples = 0.0;
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double mean_waiting_time_ms = 0.0;
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double median_waiting_time_ms = 0.0;
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double min_waiting_time_ms = 0.0;
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double max_waiting_time_ms = 0.0;
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};
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struct ConcealmentEvent {
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uint64_t duration_ms;
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size_t concealment_event_number;
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int64_t time_from_previous_event_end_ms;
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friend std::ostream& operator<<(std::ostream& stream,
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const ConcealmentEvent& concealment_event) {
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stream << "ConcealmentEvent duration_ms:" << concealment_event.duration_ms
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<< " event_number:" << concealment_event.concealment_event_number
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<< " time_from_previous_event_end_ms:"
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<< concealment_event.time_from_previous_event_end_ms << "\n";
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return stream;
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}
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};
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// Takes a pointer to another callback object, which will be invoked after
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// this object finishes. This does not transfer ownership, and null is a
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// valid value.
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explicit StatsGetter(NetEqGetAudioCallback* other_callback)
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: other_callback_(other_callback) {}
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void BeforeGetAudio(NetEq* neteq) override {
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if (other_callback_) {
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other_callback_->BeforeGetAudio(neteq);
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}
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}
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void AfterGetAudio(int64_t time_now_ms,
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const AudioFrame& audio_frame,
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bool muted,
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NetEq* neteq) override {
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if (++counter_ >= 100) {
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counter_ = 0;
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NetEqNetworkStatistics stats;
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RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0);
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stats_.push_back(stats);
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}
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const auto lifetime_stat = neteq->GetLifetimeStatistics();
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if (current_concealment_event_ != lifetime_stat.concealment_events &&
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voice_concealed_samples_until_last_event_ <
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lifetime_stat.voice_concealed_samples) {
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if (last_event_end_time_ms_ > 0) {
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// Do not account for the first event to avoid start of the call
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// skewing.
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ConcealmentEvent concealment_event;
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uint64_t last_event_voice_concealed_samples =
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lifetime_stat.voice_concealed_samples -
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voice_concealed_samples_until_last_event_;
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RTC_CHECK_GT(last_event_voice_concealed_samples, 0);
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concealment_event.duration_ms = last_event_voice_concealed_samples /
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(audio_frame.sample_rate_hz_ / 1000);
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concealment_event.concealment_event_number = current_concealment_event_;
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concealment_event.time_from_previous_event_end_ms =
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time_now_ms - last_event_end_time_ms_;
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concealment_events_.emplace_back(concealment_event);
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voice_concealed_samples_until_last_event_ =
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lifetime_stat.voice_concealed_samples;
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}
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last_event_end_time_ms_ = time_now_ms;
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voice_concealed_samples_until_last_event_ =
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lifetime_stat.voice_concealed_samples;
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current_concealment_event_ = lifetime_stat.concealment_events;
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}
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if (other_callback_) {
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other_callback_->AfterGetAudio(time_now_ms, audio_frame, muted, neteq);
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}
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}
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double AverageSpeechExpandRate() const {
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double sum_speech_expand =
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std::accumulate(stats_.begin(), stats_.end(), double{0.0},
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[](double a, NetEqNetworkStatistics b) {
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return a + static_cast<double>(b.speech_expand_rate);
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});
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return sum_speech_expand / 16384.0 / stats_.size();
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}
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const std::vector<ConcealmentEvent>& concealment_events() {
|
|
// Do not account for the last concealment event to avoid potential end
|
|
// call skewing.
|
|
return concealment_events_;
|
|
}
|
|
|
|
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 =
|
|
std::min(a.min_waiting_time_ms,
|
|
static_cast<double>(b.min_waiting_time_ms));
|
|
a.max_waiting_time_ms =
|
|
std::max(a.max_waiting_time_ms,
|
|
static_cast<double>(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();
|
|
|
|
return sum_stats;
|
|
}
|
|
|
|
private:
|
|
NetEqGetAudioCallback* other_callback_;
|
|
size_t counter_ = 0;
|
|
std::vector<NetEqNetworkStatistics> stats_;
|
|
size_t current_concealment_event_ = 1;
|
|
uint64_t voice_concealed_samples_until_last_event_ = 0;
|
|
std::vector<ConcealmentEvent> concealment_events_;
|
|
int64_t last_event_end_time_ms_ = 0;
|
|
};
|
|
|
|
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 + " --help for usage.\n"
|
|
"Example usage:\n" + program_name +
|
|
" input.rtp output.{pcm, wav}\n";
|
|
if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
|
|
return 1;
|
|
}
|
|
if (FLAG_help) {
|
|
std::cout << usage;
|
|
rtc::FlagList::Print(nullptr, false);
|
|
return 0;
|
|
}
|
|
|
|
if (FLAG_codec_map) {
|
|
PrintCodecMapping();
|
|
}
|
|
|
|
if (argc != 3) {
|
|
if (FLAG_codec_map) {
|
|
// We have already printed the codec map. Just end the program.
|
|
return 0;
|
|
}
|
|
// Print usage information.
|
|
std::cout << usage;
|
|
return 0;
|
|
}
|
|
RTC_CHECK(ValidatePayloadType(FLAG_pcmu));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_pcma));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_ilbc));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_isac));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_isac_swb));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_opus));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_wb));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_swb32));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_pcm16b_swb48));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_g722));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_avt));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_avt_16));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_avt_32));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_avt_48));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_red));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_cn_nb));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_cn_wb));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_cn_swb32));
|
|
RTC_CHECK(ValidatePayloadType(FLAG_cn_swb48));
|
|
RTC_CHECK(ValidateSsrcValue(FLAG_ssrc));
|
|
RTC_CHECK(ValidateExtensionId(FLAG_audio_level));
|
|
RTC_CHECK(ValidateExtensionId(FLAG_abs_send_time));
|
|
RTC_CHECK(ValidateExtensionId(FLAG_transport_seq_no));
|
|
|
|
// Gather RTP header extensions in a map.
|
|
NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
|
|
{FLAG_audio_level, kRtpExtensionAudioLevel},
|
|
{FLAG_abs_send_time, kRtpExtensionAbsoluteSendTime},
|
|
{FLAG_transport_seq_no, kRtpExtensionTransportSequenceNumber}};
|
|
|
|
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 (strlen(FLAG_ssrc) > 0) {
|
|
uint32_t ssrc;
|
|
RTC_CHECK(ParseSsrc(FLAG_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 = {
|
|
{FLAG_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
|
|
{FLAG_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
|
|
{FLAG_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
|
|
{FLAG_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
|
|
{FLAG_isac_swb,
|
|
std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
|
|
{FLAG_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
|
|
{FLAG_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
|
|
{FLAG_pcm16b_wb,
|
|
std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")},
|
|
{FLAG_pcm16b_swb32,
|
|
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")},
|
|
{FLAG_pcm16b_swb48,
|
|
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")},
|
|
{FLAG_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
|
|
{FLAG_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
|
|
{FLAG_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
|
|
{FLAG_avt_32,
|
|
std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")},
|
|
{FLAG_avt_48,
|
|
std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")},
|
|
{FLAG_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
|
|
{FLAG_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
|
|
{FLAG_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
|
|
{FLAG_cn_swb32,
|
|
std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")},
|
|
{FLAG_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 (strlen(FLAG_replacement_audio_file) > 0) {
|
|
// 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(
|
|
{FLAG_cn_nb, FLAG_cn_wb, FLAG_cn_swb32, FLAG_cn_swb48});
|
|
std::set<uint8_t> forbidden_types =
|
|
std_set_int32_to_uint8({FLAG_g722, FLAG_red, FLAG_avt,
|
|
FLAG_avt_16, FLAG_avt_32, FLAG_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(FLAG_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 (FLAG_matlabplot || FLAG_pythonplot) {
|
|
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 (FLAG_matlabplot) {
|
|
auto matlab_script_name = output_file_name;
|
|
std::replace(matlab_script_name.begin(), matlab_script_name.end(), '.',
|
|
'_');
|
|
std::cout << "Creating Matlab plot script " << matlab_script_name + ".m"
|
|
<< std::endl;
|
|
delay_analyzer->CreateMatlabScript(matlab_script_name + ".m");
|
|
}
|
|
if (FLAG_pythonplot) {
|
|
auto python_script_name = output_file_name;
|
|
std::replace(python_script_name.begin(), python_script_name.end(), '.',
|
|
'_');
|
|
std::cout << "Creating Python plot script " << python_script_name + ".py"
|
|
<< std::endl;
|
|
delay_analyzer->CreatePythonScript(python_script_name + ".py");
|
|
}
|
|
|
|
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);
|
|
printf(" current_buffer_size_ms: %f ms\n", stats.current_buffer_size_ms);
|
|
printf(" preferred_buffer_size_ms: %f ms\n", stats.preferred_buffer_size_ms);
|
|
if (FLAG_concealment_events) {
|
|
std::cout << " concealment_events_ms:"
|
|
<< "\n";
|
|
for (auto concealment_event : stats_getter.concealment_events())
|
|
std::cout << concealment_event;
|
|
std::cout << " end of concealment_events_ms\n";
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
} // namespace
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
|
|
int main(int argc, char* argv[]) {
|
|
return webrtc::test::RunTest(argc, argv);
|
|
}
|