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webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
Ivo Creusen c4b391a257 Revert "NetEQ RTP Play: Optionally write output audio file" 
This reverts commit 6330818ec8.

Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.

Original change's description:
> NetEQ RTP Play: Optionally write output audio file
> 
> This CL makes the output audio file optional to more
> quickly run neteq_rtpplay when no audio output is needed.
> The CL also includes necessary adaptations because of pre-existing
> dependencies (e.g., the output audio file name is used to create
> the plotting script file names).
> 
> The command line arguments are retro-compatible - i.e., same behavior
> when specifying the output audio file and the new flag
> --output_files_base_name is not used.
> 
> This CL also includes a test script with which the retro-compatibility
> has been verified.
> 
> Bug: webrtc:10337
> Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27067}

TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org

Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27078}
2019-03-12 14:28:23 +00:00

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/*
* 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 <iostream>
#include <string>
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
#include "rtc_base/flags.h"
#include "system_wrappers/include/field_trial.h"
#include "test/field_trial.h"
namespace {
using TestConfig = webrtc::test::NetEqTestFactory::Config;
WEBRTC_DEFINE_bool(codec_map,
false,
"Prints the mapping between RTP payload type and "
"codec");
WEBRTC_DEFINE_string(
force_fieldtrials,
"",
"Field trials control experimental feature code which can be forced. "
"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
" will assign the group Enable to field trial WebRTC-FooFeature.");
WEBRTC_DEFINE_bool(help, false, "Prints this message");
// Define command line flags.
WEBRTC_DEFINE_int(pcmu,
TestConfig::default_pcmu(),
"RTP payload type for PCM-u");
WEBRTC_DEFINE_int(pcma,
TestConfig::default_pcma(),
"RTP payload type for PCM-a");
WEBRTC_DEFINE_int(ilbc,
TestConfig::default_ilbc(),
"RTP payload type for iLBC");
WEBRTC_DEFINE_int(isac,
TestConfig::default_isac(),
"RTP payload type for iSAC");
WEBRTC_DEFINE_int(isac_swb,
TestConfig::default_isac_swb(),
"RTP payload type for iSAC-swb (32 kHz)");
WEBRTC_DEFINE_int(opus,
TestConfig::default_opus(),
"RTP payload type for Opus");
WEBRTC_DEFINE_int(pcm16b,
TestConfig::default_pcm16b(),
"RTP payload type for PCM16b-nb (8 kHz)");
WEBRTC_DEFINE_int(pcm16b_wb,
TestConfig::default_pcm16b_wb(),
"RTP payload type for PCM16b-wb (16 kHz)");
WEBRTC_DEFINE_int(pcm16b_swb32,
TestConfig::default_pcm16b_swb32(),
"RTP payload type for PCM16b-swb32 (32 kHz)");
WEBRTC_DEFINE_int(pcm16b_swb48,
TestConfig::default_pcm16b_swb48(),
"RTP payload type for PCM16b-swb48 (48 kHz)");
WEBRTC_DEFINE_int(g722,
TestConfig::default_g722(),
"RTP payload type for G.722");
WEBRTC_DEFINE_int(avt,
TestConfig::default_avt(),
"RTP payload type for AVT/DTMF (8 kHz)");
WEBRTC_DEFINE_int(avt_16,
TestConfig::default_avt_16(),
"RTP payload type for AVT/DTMF (16 kHz)");
WEBRTC_DEFINE_int(avt_32,
TestConfig::default_avt_32(),
"RTP payload type for AVT/DTMF (32 kHz)");
WEBRTC_DEFINE_int(avt_48,
TestConfig::default_avt_48(),
"RTP payload type for AVT/DTMF (48 kHz)");
WEBRTC_DEFINE_int(red,
TestConfig::default_red(),
"RTP payload type for redundant audio (RED)");
WEBRTC_DEFINE_int(cn_nb,
TestConfig::default_cn_nb(),
"RTP payload type for comfort noise (8 kHz)");
WEBRTC_DEFINE_int(cn_wb,
TestConfig::default_cn_wb(),
"RTP payload type for comfort noise (16 kHz)");
WEBRTC_DEFINE_int(cn_swb32,
TestConfig::default_cn_swb32(),
"RTP payload type for comfort noise (32 kHz)");
WEBRTC_DEFINE_int(cn_swb48,
TestConfig::default_cn_swb48(),
"RTP payload type for comfort noise (48 kHz)");
WEBRTC_DEFINE_string(replacement_audio_file,
"",
"A PCM file that will be used to populate dummy"
" RTP packets");
WEBRTC_DEFINE_string(
ssrc,
"",
"Only use packets with this SSRC (decimal or hex, the latter "
"starting with 0x)");
WEBRTC_DEFINE_int(audio_level,
TestConfig::default_audio_level(),
"Extension ID for audio level (RFC 6464)");
WEBRTC_DEFINE_int(abs_send_time,
TestConfig::default_abs_send_time(),
"Extension ID for absolute sender time");
WEBRTC_DEFINE_int(transport_seq_no,
TestConfig::default_transport_seq_no(),
"Extension ID for transport sequence number");
WEBRTC_DEFINE_int(video_content_type,
TestConfig::default_video_content_type(),
"Extension ID for video content type");
WEBRTC_DEFINE_int(video_timing,
TestConfig::default_video_timing(),
"Extension ID for video timing");
WEBRTC_DEFINE_bool(matlabplot,
false,
"Generates a matlab script for plotting the delay profile");
WEBRTC_DEFINE_bool(pythonplot,
false,
"Generates a python script for plotting the delay profile");
WEBRTC_DEFINE_bool(textlog,
false,
"Generates a text log describing the simulation on a "
"step-by-step basis.");
WEBRTC_DEFINE_bool(concealment_events, false, "Prints concealment events");
WEBRTC_DEFINE_int(max_nr_packets_in_buffer,
TestConfig::default_max_nr_packets_in_buffer(),
"Maximum allowed number of packets in the buffer");
WEBRTC_DEFINE_bool(enable_fast_accelerate,
false,
"Enables jitter buffer fast accelerate");
// 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); // NOLINT
if (value == ULONG_MAX && errno == ERANGE)
return false; // Value out of range for unsigned long.
if (sizeof(unsigned long) > sizeof(uint32_t) && value > 0xFFFFFFFF) // NOLINT
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;
}
static bool ValidateExtensionId(int value) {
if (value > 0 && value <= 255) // Value is ok.
return true;
printf("Extension ID must be between 1 and 255, not %d\n",
static_cast<int>(value));
return false;
}
// Flag validators.
bool ValidatePayloadType(int value) {
if (value >= 0 && value <= 127) // Value is ok.
return true;
printf("Payload type must be between 0 and 127, not %d\n",
static_cast<int>(value));
return false;
}
bool ValidateSsrcValue(const std::string& str) {
uint32_t dummy_ssrc;
if (ParseSsrc(str, &dummy_ssrc)) // Value is ok.
return true;
printf("Invalid SSRC: %s\n", str.c_str());
return false;
}
void PrintCodecMappingEntry(const char* codec, int flag) {
std::cout << codec << ": " << flag << std::endl;
}
void PrintCodecMapping() {
PrintCodecMappingEntry("PCM-u", FLAG_pcmu);
PrintCodecMappingEntry("PCM-a", FLAG_pcma);
PrintCodecMappingEntry("iLBC", FLAG_ilbc);
PrintCodecMappingEntry("iSAC", FLAG_isac);
PrintCodecMappingEntry("iSAC-swb (32 kHz)", FLAG_isac_swb);
PrintCodecMappingEntry("Opus", FLAG_opus);
PrintCodecMappingEntry("PCM16b-nb (8 kHz)", FLAG_pcm16b);
PrintCodecMappingEntry("PCM16b-wb (16 kHz)", FLAG_pcm16b_wb);
PrintCodecMappingEntry("PCM16b-swb32 (32 kHz)", FLAG_pcm16b_swb32);
PrintCodecMappingEntry("PCM16b-swb48 (48 kHz)", FLAG_pcm16b_swb48);
PrintCodecMappingEntry("G.722", FLAG_g722);
PrintCodecMappingEntry("AVT/DTMF (8 kHz)", FLAG_avt);
PrintCodecMappingEntry("AVT/DTMF (16 kHz)", FLAG_avt_16);
PrintCodecMappingEntry("AVT/DTMF (32 kHz)", FLAG_avt_32);
PrintCodecMappingEntry("AVT/DTMF (48 kHz)", FLAG_avt_48);
PrintCodecMappingEntry("redundant audio (RED)", FLAG_red);
PrintCodecMappingEntry("comfort noise (8 kHz)", FLAG_cn_nb);
PrintCodecMappingEntry("comfort noise (16 kHz)", FLAG_cn_wb);
PrintCodecMappingEntry("comfort noise (32 kHz)", FLAG_cn_swb32);
PrintCodecMappingEntry("comfort noise (48 kHz)", FLAG_cn_swb48);
}
} // namespace
int main(int argc, char* argv[]) {
webrtc::test::NetEqTestFactory factory;
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)) {
exit(1);
}
if (FLAG_help) {
std::cout << usage;
rtc::FlagList::Print(nullptr, false);
exit(0);
}
if (FLAG_codec_map) {
PrintCodecMapping();
exit(0);
}
if (argc != 3) {
// Print usage information.
std::cout << usage;
exit(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));
RTC_CHECK(ValidateExtensionId(FLAG_video_content_type));
RTC_CHECK(ValidateExtensionId(FLAG_video_timing));
webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
webrtc::test::NetEqTestFactory::Config config;
config.pcmu = FLAG_pcmu;
config.pcma = FLAG_pcma;
config.ilbc = FLAG_ilbc;
config.isac = FLAG_isac;
config.isac_swb = FLAG_isac_swb;
config.opus = FLAG_opus;
config.pcm16b = FLAG_pcm16b;
config.pcm16b_wb = FLAG_pcm16b_wb;
config.pcm16b_swb32 = FLAG_pcm16b_swb32;
config.pcm16b_swb48 = FLAG_pcm16b_swb48;
config.g722 = FLAG_g722;
config.avt = FLAG_avt;
config.avt_16 = FLAG_avt_16;
config.avt_32 = FLAG_avt_32;
config.avt_48 = FLAG_avt_48;
config.red = FLAG_red;
config.cn_nb = FLAG_cn_nb;
config.cn_wb = FLAG_cn_wb;
config.cn_swb32 = FLAG_cn_swb32;
config.cn_swb48 = FLAG_cn_swb48;
config.replacement_audio_file = FLAG_replacement_audio_file;
config.audio_level = FLAG_audio_level;
config.abs_send_time = FLAG_abs_send_time;
config.transport_seq_no = FLAG_transport_seq_no;
config.video_content_type = FLAG_video_content_type;
config.video_timing = FLAG_video_timing;
config.matlabplot = FLAG_matlabplot;
config.pythonplot = FLAG_pythonplot;
config.textlog = FLAG_textlog;
config.concealment_events = FLAG_concealment_events;
config.max_nr_packets_in_buffer = FLAG_max_nr_packets_in_buffer;
config.enable_fast_accelerate = FLAG_enable_fast_accelerate;
// 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.";
config.ssrc_filter = absl::make_optional(ssrc);
}
std::unique_ptr<webrtc::test::NetEqTest> test =
factory.InitializeTest(argv[1], argv[2], config);
test->Run();
return 0;
}
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