Restructure neteq_rtpplay into a library with small executable wrapper.

Most of the code in neteq_rtpplay is moved into a factory class for
NetEqTest. The factory method takes the same argc and argv arguments as
neteq_rtpplay.
This CL also adds a small public API for neteq_test to allow easy
integration into external software.

Bug: webrtc:9667
Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d
Reviewed-on: https://webrtc-review.googlesource.com/96100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24531}
This commit is contained in:
Ivo Creusen
2018-09-03 11:49:27 +02:00
committed by Commit Bot
parent 88c1a9ecbc
commit 55de08e7ef
16 changed files with 987 additions and 554 deletions

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@ -309,6 +309,14 @@ rtc_source_set("libjingle_peerconnection_test_api") {
]
}
rtc_source_set("neteq_simulator_api") {
visibility = [ "*" ]
sources = [
"test/neteq_simulator.cc",
"test/neteq_simulator.h",
]
}
if (rtc_include_tests) {
if (rtc_enable_protobuf) {
rtc_source_set("audioproc_f_api") {
@ -324,6 +332,20 @@ if (rtc_include_tests) {
"../modules/audio_processing:audioproc_f_impl",
]
}
rtc_source_set("neteq_simulator_factory") {
visibility = [ "*" ]
testonly = true
sources = [
"test/neteq_simulator_factory.cc",
"test/neteq_simulator_factory.h",
]
deps = [
":neteq_simulator_api",
"../modules/audio_coding:neteq_test_factory",
"//third_party/abseil-cpp/absl/memory",
]
}
}
rtc_source_set("simulcast_test_fixture_api") {

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@ -0,0 +1,22 @@
/*
* Copyright (c) 2018 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 "api/test/neteq_simulator.h"
namespace webrtc {
namespace test {
NetEqSimulator::SimulationStepResult::SimulationStepResult() = default;
NetEqSimulator::SimulationStepResult::SimulationStepResult(
const NetEqSimulator::SimulationStepResult& other) = default;
NetEqSimulator::SimulationStepResult::~SimulationStepResult() = default;
} // namespace test
} // namespace webrtc

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@ -0,0 +1,63 @@
/*
* Copyright (c) 2018 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.
*/
#ifndef API_TEST_NETEQ_SIMULATOR_H_
#define API_TEST_NETEQ_SIMULATOR_H_
#include <stdint.h>
#include <map>
namespace webrtc {
namespace test {
class NetEqSimulator {
public:
virtual ~NetEqSimulator() = default;
enum class Action { kNormal, kExpand, kAccelerate, kPreemptiveExpand };
// The results of one simulation step.
struct SimulationStepResult {
SimulationStepResult();
SimulationStepResult(const SimulationStepResult& other);
~SimulationStepResult();
bool is_simulation_finished = false;
// The amount of audio produced (in ms) with the actions in this time step.
std::map<Action, int> action_times_ms;
// The amount of wall clock time (in ms) that elapsed since the previous
// event. This is not necessarily equal to the sum of the values in
// action_times_ms.
int64_t simulation_step_ms = 0;
};
struct NetEqState {
// The sum of the packet buffer and sync buffer delay.
int current_delay_ms = 0;
// TODO(ivoc): Expand this struct with more useful metrics.
};
// Runs the simulation until we hit the next GetAudio event. If the simulation
// is finished, is_simulation_finished will be set to true in the returned
// SimulationStepResult.
virtual SimulationStepResult RunToNextGetAudio() = 0;
// Set the next action to be taken by NetEq. This will override any action
// that NetEq would normally decide to take.
virtual void SetNextAction(Action next_operation) = 0;
// Get the current state of NetEq.
virtual NetEqState GetNetEqState() = 0;
};
} // namespace test
} // namespace webrtc
#endif // API_TEST_NETEQ_SIMULATOR_H_

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@ -0,0 +1,31 @@
/*
* Copyright (c) 2018 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 "api/test/neteq_simulator_factory.h"
#include "absl/memory/memory.h"
#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
namespace webrtc {
namespace test {
NetEqSimulatorFactory::NetEqSimulatorFactory()
: factory_(absl::make_unique<NetEqTestFactory>()) {}
NetEqSimulatorFactory::~NetEqSimulatorFactory() = default;
std::unique_ptr<NetEqSimulator> NetEqSimulatorFactory::CreateSimulator(
int argc,
char* argv[]) {
return factory_->InitializeTest(argc, argv);
}
} // namespace test
} // namespace webrtc

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@ -0,0 +1,37 @@
/*
* Copyright (c) 2018 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.
*/
#ifndef API_TEST_NETEQ_SIMULATOR_FACTORY_H_
#define API_TEST_NETEQ_SIMULATOR_FACTORY_H_
#include <memory>
#include "api/test/neteq_simulator.h"
namespace webrtc {
namespace test {
class NetEqTestFactory;
class NetEqSimulatorFactory {
public:
NetEqSimulatorFactory();
~NetEqSimulatorFactory();
// This function takes the same arguments as the neteq_rtpplay utility.
std::unique_ptr<NetEqSimulator> CreateSimulator(int argc, char* argv[]);
private:
std::unique_ptr<NetEqTestFactory> factory_;
};
} // namespace test
} // namespace webrtc
#endif // API_TEST_NETEQ_SIMULATOR_FACTORY_H_

View File

@ -1089,6 +1089,7 @@ rtc_source_set("neteq_tools_minimal") {
":neteq",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../api:neteq_simulator_api",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
@ -1164,6 +1165,8 @@ rtc_source_set("neteq_tools") {
"neteq/tools/neteq_replacement_input.h",
"neteq/tools/neteq_stats_getter.cc",
"neteq/tools/neteq_stats_getter.h",
"neteq/tools/neteq_stats_plotter.cc",
"neteq/tools/neteq_stats_plotter.h",
]
if (!build_with_chromium && is_clang) {
@ -1505,8 +1508,9 @@ if (rtc_include_tests) {
proto_out_dir = "modules/audio_coding/neteq"
}
rtc_test("neteq_rtpplay") {
rtc_source_set("neteq_test_factory") {
testonly = true
visibility += webrtc_default_visibility
defines = []
deps = [
"../../rtc_base:checks",
@ -1514,7 +1518,8 @@ if (rtc_include_tests) {
"../../test:fileutils",
]
sources = [
"neteq/tools/neteq_rtpplay.cc",
"neteq/tools/neteq_test_factory.cc",
"neteq/tools/neteq_test_factory.h",
]
if (!build_with_chromium && is_clang) {
@ -1529,6 +1534,19 @@ if (rtc_include_tests) {
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_test("neteq_rtpplay") {
testonly = true
defines = []
deps = [
":neteq_test_factory",
":neteq_test_tools",
]
sources = [
"neteq/tools/neteq_rtpplay.cc",
]
}
}

View File

@ -21,6 +21,7 @@
#include "api/audio_codecs/audio_decoder.h"
#include "api/rtp_headers.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/neteq/defines.h"
#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/scoped_ref_ptr.h"
@ -129,9 +130,14 @@ class NetEq {
// If muted state is enabled (through Config::enable_muted_state), |muted|
// may be set to true after a prolonged expand period. When this happens, the
// |data_| in |audio_frame| is not written, but should be interpreted as being
// all zeros.
// all zeros. For testing purposes, an override can be supplied in the
// |action_override| argument, which will cause NetEq to take this action
// next, instead of the action it would normally choose.
// Returns kOK on success, or kFail in case of an error.
virtual int GetAudio(AudioFrame* audio_frame, bool* muted) = 0;
virtual int GetAudio(
AudioFrame* audio_frame,
bool* muted,
absl::optional<Operations> action_override = absl::nullopt) = 0;
// Replaces the current set of decoders with the given one.
virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;

View File

@ -199,10 +199,12 @@ void SetAudioFrameActivityAndType(bool vad_enabled,
}
} // namespace
int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted) {
int NetEqImpl::GetAudio(AudioFrame* audio_frame,
bool* muted,
absl::optional<Operations> action_override) {
TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
rtc::CritScope lock(&crit_sect_);
if (GetAudioInternal(audio_frame, muted) != 0) {
if (GetAudioInternal(audio_frame, muted, action_override) != 0) {
return kFail;
}
RTC_DCHECK_EQ(
@ -798,7 +800,9 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
return 0;
}
int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
absl::optional<Operations> action_override) {
PacketList packet_list;
DtmfEvent dtmf_event;
Operations operation;
@ -831,9 +835,8 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
*muted = true;
return 0;
}
int return_value =
GetDecision(&operation, &packet_list, &dtmf_event, &play_dtmf);
int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
&play_dtmf, action_override);
if (return_value != 0) {
last_mode_ = kModeError;
return return_value;
@ -1021,7 +1024,8 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
int NetEqImpl::GetDecision(Operations* operation,
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf) {
bool* play_dtmf,
absl::optional<Operations> action_override) {
// Initialize output variables.
*play_dtmf = false;
*operation = kUndefined;
@ -1093,6 +1097,10 @@ int NetEqImpl::GetDecision(Operations* operation,
*sync_buffer_, *expand_, decoder_frame_length_, packet, last_mode_,
*play_dtmf, generated_noise_samples, &reset_decoder_);
if (action_override) {
// Use the provided action instead of the decision NetEq decided on.
*operation = *action_override;
}
// Check if we already have enough samples in the |sync_buffer_|. If so,
// change decision to normal, unless the decision was merge, accelerate, or
// preemptive expand.

View File

@ -131,7 +131,10 @@ class NetEqImpl : public webrtc::NetEq {
void InsertEmptyPacket(const RTPHeader& rtp_header) override;
int GetAudio(AudioFrame* audio_frame, bool* muted) override;
int GetAudio(
AudioFrame* audio_frame,
bool* muted,
absl::optional<Operations> action_override = absl::nullopt) override;
void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
@ -230,7 +233,9 @@ class NetEqImpl : public webrtc::NetEq {
// Delivers 10 ms of audio data. The data is written to |audio_frame|.
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
int GetAudioInternal(AudioFrame* audio_frame,
bool* muted,
absl::optional<Operations> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Provides a decision to the GetAudioInternal method. The decision what to
@ -241,7 +246,9 @@ class NetEqImpl : public webrtc::NetEq {
int GetDecision(Operations* operation,
PacketList* packet_list,
DtmfEvent* dtmf_event,
bool* play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
bool* play_dtmf,
absl::optional<Operations> action_override)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Decodes the speech packets in |packet_list|, and writes the results to
// |decoded_buffer|, which is allocated to hold |decoded_buffer_length|

View File

@ -8,544 +8,15 @@
* 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 <iostream>
#include <memory>
#include <string>
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "test/field_trial.h"
#include "test/testsupport/fileutils.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(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;
}
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;
}
// Define command line flags.
DEFINE_int(pcmu, 0, "RTP payload type for PCM-u");
DEFINE_int(pcma, 8, "RTP payload type for PCM-a");
DEFINE_int(ilbc, 102, "RTP payload type for iLBC");
DEFINE_int(isac, 103, "RTP payload type for iSAC");
DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
DEFINE_int(opus, 111, "RTP payload type for Opus");
DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
DEFINE_int(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
DEFINE_int(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
DEFINE_int(g722, 9, "RTP payload type for G.722");
DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)");
DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
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)");
DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
DEFINE_int(transport_seq_no, 5, "Extension ID for transport sequence number");
DEFINE_int(video_content_type, 7, "Extension ID for video content type");
DEFINE_int(video_timing, 8, "Extension ID for video timing");
DEFINE_bool(matlabplot,
false,
"Generates a matlab script for plotting the delay profile");
DEFINE_bool(pythonplot,
false,
"Generates a python script for plotting the delay profile");
DEFINE_bool(help, false, "Prints this message");
DEFINE_bool(concealment_events, false, "Prints concealment events");
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.");
// 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, int flag) {
std::cout << CodecName(codec) << ": " << flag << std::endl;
}
void PrintCodecMapping() {
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAG_pcmu);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAG_pcma);
PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAG_ilbc);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAG_isac);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAG_isac_swb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAG_opus);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAG_pcm16b);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAG_pcm16b_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb32kHz,
FLAG_pcm16b_swb32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb48kHz,
FLAG_pcm16b_swb48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAG_g722);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAG_avt);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAG_avt_16);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAG_avt_32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAG_avt_48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAG_red);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAG_cn_nb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAG_cn_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAG_cn_swb32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAG_cn_swb48);
}
absl::optional<int> CodecSampleRate(uint8_t payload_type) {
if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
return 8000;
if (payload_type == FLAG_isac || payload_type == FLAG_pcm16b_wb ||
payload_type == FLAG_g722 || payload_type == FLAG_cn_wb ||
payload_type == FLAG_avt_16)
return 16000;
if (payload_type == FLAG_isac_swb || payload_type == FLAG_pcm16b_swb32 ||
payload_type == FLAG_cn_swb32 || payload_type == FLAG_avt_32)
return 32000;
if (payload_type == FLAG_opus || payload_type == FLAG_pcm16b_swb48 ||
payload_type == FLAG_cn_swb48 || payload_type == FLAG_avt_48)
return 48000;
if (payload_type == FLAG_red)
return 0;
return absl::nullopt;
}
// 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_ = packet.header.ssrc;
if (other_callback_) {
other_callback_->AfterInsertPacket(packet, neteq);
}
}
private:
NetEqPostInsertPacket* other_callback_;
absl::optional<uint32_t> last_ssrc_;
};
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;
}
ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
ScopedFieldTrials field_trials(FLAG_force_fieldtrials);
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));
// 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},
{FLAG_video_content_type, kRtpExtensionVideoContentType},
{FLAG_video_timing, kRtpExtensionVideoTiming}};
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.";
static_cast<NetEqPacketSourceInput*>(input.get())->SelectSsrc(ssrc);
}
// Check the sample rate.
absl::optional<int> sample_rate_hz;
std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
while (absl::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")},
#ifdef WEBRTC_CODEC_ILBC
{FLAG_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
#endif
{FLAG_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
#if !defined(WEBRTC_ANDROID)
{FLAG_isac_swb, std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
#endif
#ifdef WEBRTC_CODEC_OPUS
{FLAG_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
#endif
{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;
NetEqStatsGetter stats_getter(std::move(delay_analyzer));
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;
stats_getter.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;
stats_getter.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:" << std::endl;
for (auto concealment_event : stats_getter.concealment_events())
std::cout << concealment_event.ToString() << std::endl;
std::cout << " end of concealment_events_ms" << std::endl;
}
return 0;
}
} // namespace
} // namespace test
} // namespace webrtc
#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
int main(int argc, char* argv[]) {
return webrtc::test::RunTest(argc, argv);
webrtc::test::NetEqTestFactory factory;
std::unique_ptr<webrtc::test::NetEqTest> test =
factory.InitializeTest(argc, argv);
test->Run();
return 0;
}

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@ -0,0 +1,81 @@
/*
* Copyright (c) 2018 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 "modules/audio_coding/neteq/tools/neteq_stats_plotter.h"
#include <inttypes.h>
#include <stdio.h>
#include <utility>
namespace webrtc {
namespace test {
NetEqStatsPlotter::NetEqStatsPlotter(bool make_matlab_plot,
bool make_python_plot,
bool show_concealment_events,
std::string base_file_name)
: make_matlab_plot_(make_matlab_plot),
make_python_plot_(make_python_plot),
show_concealment_events_(show_concealment_events),
base_file_name_(base_file_name) {
std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer;
if (make_matlab_plot || make_python_plot) {
delay_analyzer.reset(new NetEqDelayAnalyzer);
}
stats_getter_.reset(new NetEqStatsGetter(std::move(delay_analyzer)));
}
void NetEqStatsPlotter::SimulationEnded(int64_t simulation_time_ms) {
if (make_matlab_plot_) {
auto matlab_script_name = base_file_name_;
std::replace(matlab_script_name.begin(), matlab_script_name.end(), '.',
'_');
printf("Creating Matlab plot script %s.m\n", matlab_script_name.c_str());
stats_getter_->delay_analyzer()->CreateMatlabScript(matlab_script_name +
".m");
}
if (make_python_plot_) {
auto python_script_name = base_file_name_;
std::replace(python_script_name.begin(), python_script_name.end(), '.',
'_');
printf("Creating Python plot script %s.py\n", python_script_name.c_str());
stats_getter_->delay_analyzer()->CreatePythonScript(python_script_name +
".py");
}
printf("Simulation statistics:\n");
printf(" output duration: %" PRId64 " ms\n", simulation_time_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 (show_concealment_events_) {
printf(" concealment_events_ms:\n");
for (auto concealment_event : stats_getter_->concealment_events())
printf("%s\n", concealment_event.ToString().c_str());
printf(" end of concealment_events_ms\n");
}
}
} // namespace test
} // namespace webrtc

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@ -0,0 +1,46 @@
/*
* Copyright (c) 2018 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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
#include <memory>
#include <string>
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
namespace webrtc {
namespace test {
class NetEqStatsPlotter : public NetEqSimulationEndedCallback {
public:
NetEqStatsPlotter(bool make_matlab_plot,
bool make_python_plot,
bool show_concealment_events,
std::string base_file_name);
void SimulationEnded(int64_t simulation_time_ms) override;
NetEqStatsGetter* stats_getter() { return stats_getter_.get(); }
private:
std::unique_ptr<NetEqStatsGetter> stats_getter_;
const bool make_matlab_plot_;
const bool make_python_plot_;
const bool show_concealment_events_;
const std::string base_file_name_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_

View File

@ -16,6 +16,26 @@
namespace webrtc {
namespace test {
namespace {
absl::optional<Operations> ActionToOperations(
absl::optional<NetEqSimulator::Action> a) {
if (!a) {
return absl::nullopt;
}
switch (*a) {
case NetEqSimulator::Action::kAccelerate:
return absl::make_optional(kAccelerate);
case NetEqSimulator::Action::kExpand:
return absl::make_optional(kExpand);
case NetEqSimulator::Action::kNormal:
return absl::make_optional(kNormal);
case NetEqSimulator::Action::kPreemptiveExpand:
return absl::make_optional(kPreemptiveExpand);
}
}
} // namespace
void DefaultNetEqTestErrorCallback::OnInsertPacketError(
const NetEqInput::PacketData& packet) {
@ -49,6 +69,20 @@ NetEqTest::NetEqTest(const NetEq::Config& config,
NetEqTest::~NetEqTest() = default;
int64_t NetEqTest::Run() {
int64_t simulation_time = 0;
SimulationStepResult step_result;
do {
step_result = RunToNextGetAudio();
simulation_time += step_result.simulation_step_ms;
} while (!step_result.is_simulation_finished);
if (callbacks_.simulation_ended_callback) {
callbacks_.simulation_ended_callback->SimulationEnded(simulation_time);
}
return simulation_time;
}
NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
SimulationStepResult result;
const int64_t start_time_ms = *input_->NextEventTime();
int64_t time_now_ms = start_time_ms;
@ -81,7 +115,9 @@ int64_t NetEqTest::Run() {
}
AudioFrame out_frame;
bool muted;
int error = neteq_->GetAudio(&out_frame, &muted);
int error = neteq_->GetAudio(&out_frame, &muted,
ActionToOperations(next_action_));
next_action_ = absl::nullopt;
RTC_CHECK(!muted) << "The code does not handle enable_muted_state";
if (error != NetEq::kOK) {
if (callbacks_.error_callback) {
@ -102,9 +138,26 @@ int64_t NetEqTest::Run() {
}
input_->AdvanceOutputEvent();
result.simulation_step_ms = time_now_ms - start_time_ms;
// TODO(ivoc): Set the result.<action>_ms values correctly.
result.is_simulation_finished = input_->ended();
return result;
}
}
return time_now_ms - start_time_ms;
result.simulation_step_ms = time_now_ms - start_time_ms;
result.is_simulation_finished = true;
return result;
}
void NetEqTest::SetNextAction(NetEqTest::Action next_operation) {
next_action_ = absl::optional<Action>(next_operation);
}
NetEqTest::NetEqState NetEqTest::GetNetEqState() {
NetEqState state;
const auto network_stats = SimulationStats();
state.current_delay_ms = network_stats.current_buffer_size_ms;
return state;
}
NetEqNetworkStatistics NetEqTest::SimulationStats() {

View File

@ -16,6 +16,8 @@
#include <string>
#include <utility>
#include "absl/types/optional.h"
#include "api/test/neteq_simulator.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
@ -52,10 +54,16 @@ class NetEqGetAudioCallback {
NetEq* neteq) = 0;
};
class NetEqSimulationEndedCallback {
public:
virtual ~NetEqSimulationEndedCallback() = default;
virtual void SimulationEnded(int64_t simulation_time_ms) = 0;
};
// Class that provides an input--output test for NetEq. The input (both packets
// and output events) is provided by a NetEqInput object, while the output is
// directed to an AudioSink object.
class NetEqTest {
class NetEqTest : public NetEqSimulator {
public:
using DecoderMap = std::map<int, std::pair<NetEqDecoder, std::string> >;
@ -71,6 +79,7 @@ class NetEqTest {
NetEqTestErrorCallback* error_callback = nullptr;
NetEqPostInsertPacket* post_insert_packet = nullptr;
NetEqGetAudioCallback* get_audio_callback = nullptr;
NetEqSimulationEndedCallback* simulation_ended_callback = nullptr;
};
// Sets up the test with given configuration, codec mappings, input, ouput,
@ -82,10 +91,17 @@ class NetEqTest {
std::unique_ptr<AudioSink> output,
Callbacks callbacks);
~NetEqTest();
~NetEqTest() override;
// Runs the test. Returns the duration of the produced audio in ms.
int64_t Run();
// Runs the simulation until we hit the next GetAudio event. If the simulation
// is finished, is_simulation_finished will be set to true in the returned
// SimulationStepResult.
SimulationStepResult RunToNextGetAudio() override;
void SetNextAction(Action next_operation) override;
NetEqState GetNetEqState() override;
// Returns the statistics from NetEq.
NetEqNetworkStatistics SimulationStats();
@ -96,7 +112,7 @@ class NetEqTest {
private:
void RegisterDecoders(const DecoderMap& codecs);
void RegisterExternalDecoders(const ExtDecoderMap& codecs);
absl::optional<Action> next_action_;
std::unique_ptr<NetEq> neteq_;
std::unique_ptr<NetEqInput> input_;
std::unique_ptr<AudioSink> output_;

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@ -0,0 +1,509 @@
/*
* Copyright (c) 2018 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 "modules/audio_coding/neteq/tools/neteq_test_factory.h"
#include <errno.h>
#include <limits.h> // For ULONG_MAX returned by strtoul.
#include <stdio.h>
#include <stdlib.h> // For strtoul.
#include <iostream>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
#include "modules/audio_coding/neteq/tools/neteq_stats_plotter.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
#include "test/field_trial.h"
#include "test/testsupport/fileutils.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); // 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;
}
// 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;
}
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;
}
// Define command line flags.
DEFINE_int(pcmu, 0, "RTP payload type for PCM-u");
DEFINE_int(pcma, 8, "RTP payload type for PCM-a");
DEFINE_int(ilbc, 102, "RTP payload type for iLBC");
DEFINE_int(isac, 103, "RTP payload type for iSAC");
DEFINE_int(isac_swb, 104, "RTP payload type for iSAC-swb (32 kHz)");
DEFINE_int(opus, 111, "RTP payload type for Opus");
DEFINE_int(pcm16b, 93, "RTP payload type for PCM16b-nb (8 kHz)");
DEFINE_int(pcm16b_wb, 94, "RTP payload type for PCM16b-wb (16 kHz)");
DEFINE_int(pcm16b_swb32, 95, "RTP payload type for PCM16b-swb32 (32 kHz)");
DEFINE_int(pcm16b_swb48, 96, "RTP payload type for PCM16b-swb48 (48 kHz)");
DEFINE_int(g722, 9, "RTP payload type for G.722");
DEFINE_int(avt, 106, "RTP payload type for AVT/DTMF (8 kHz)");
DEFINE_int(avt_16, 114, "RTP payload type for AVT/DTMF (16 kHz)");
DEFINE_int(avt_32, 115, "RTP payload type for AVT/DTMF (32 kHz)");
DEFINE_int(avt_48, 116, "RTP payload type for AVT/DTMF (48 kHz)");
DEFINE_int(red, 117, "RTP payload type for redundant audio (RED)");
DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
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)");
DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
DEFINE_int(transport_seq_no, 5, "Extension ID for transport sequence number");
DEFINE_int(video_content_type, 7, "Extension ID for video content type");
DEFINE_int(video_timing, 8, "Extension ID for video timing");
DEFINE_bool(matlabplot,
false,
"Generates a matlab script for plotting the delay profile");
DEFINE_bool(pythonplot,
false,
"Generates a python script for plotting the delay profile");
DEFINE_bool(help, false, "Prints this message");
DEFINE_bool(concealment_events, false, "Prints concealment events");
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.");
// 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, int flag) {
std::cout << CodecName(codec) << ": " << flag << std::endl;
}
void PrintCodecMapping() {
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMu, FLAG_pcmu);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCMa, FLAG_pcma);
PrintCodecMappingEntry(NetEqDecoder::kDecoderILBC, FLAG_ilbc);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISAC, FLAG_isac);
PrintCodecMappingEntry(NetEqDecoder::kDecoderISACswb, FLAG_isac_swb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderOpus, FLAG_opus);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16B, FLAG_pcm16b);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bwb, FLAG_pcm16b_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb32kHz,
FLAG_pcm16b_swb32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderPCM16Bswb48kHz,
FLAG_pcm16b_swb48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderG722, FLAG_g722);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT, FLAG_avt);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT16kHz, FLAG_avt_16);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT32kHz, FLAG_avt_32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderAVT48kHz, FLAG_avt_48);
PrintCodecMappingEntry(NetEqDecoder::kDecoderRED, FLAG_red);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGnb, FLAG_cn_nb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGwb, FLAG_cn_wb);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb32kHz, FLAG_cn_swb32);
PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAG_cn_swb48);
}
absl::optional<int> CodecSampleRate(uint8_t payload_type) {
if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
return 8000;
if (payload_type == FLAG_isac || payload_type == FLAG_pcm16b_wb ||
payload_type == FLAG_g722 || payload_type == FLAG_cn_wb ||
payload_type == FLAG_avt_16)
return 16000;
if (payload_type == FLAG_isac_swb || payload_type == FLAG_pcm16b_swb32 ||
payload_type == FLAG_cn_swb32 || payload_type == FLAG_avt_32)
return 32000;
if (payload_type == FLAG_opus || payload_type == FLAG_pcm16b_swb48 ||
payload_type == FLAG_cn_swb48 || payload_type == FLAG_avt_48)
return 48000;
if (payload_type == FLAG_red)
return 0;
return absl::nullopt;
}
} // namespace
// 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_ = packet.header.ssrc;
if (other_callback_) {
other_callback_->AfterInsertPacket(packet, neteq);
}
}
private:
NetEqPostInsertPacket* other_callback_;
absl::optional<uint32_t> last_ssrc_;
};
NetEqTestFactory::NetEqTestFactory() = default;
NetEqTestFactory::~NetEqTestFactory() = default;
std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(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)) {
exit(1);
}
if (FLAG_help) {
std::cout << usage;
rtc::FlagList::Print(nullptr, false);
exit(0);
}
if (FLAG_codec_map) {
PrintCodecMapping();
}
if (argc != 3) {
if (FLAG_codec_map) {
// We have already printed the codec map. Just end the program.
exit(0);
}
// Print usage information.
std::cout << usage;
exit(0);
}
ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
ScopedFieldTrials field_trials(FLAG_force_fieldtrials);
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));
// 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},
{FLAG_video_content_type, kRtpExtensionVideoContentType},
{FLAG_video_timing, kRtpExtensionVideoTiming}};
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.";
static_cast<NetEqPacketSourceInput*>(input.get())->SelectSsrc(ssrc);
}
// Check the sample rate.
absl::optional<int> sample_rate_hz;
std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
while (absl::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")},
#ifdef WEBRTC_CODEC_ILBC
{FLAG_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
#endif
{FLAG_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
#if !defined(WEBRTC_ANDROID)
{FLAG_isac_swb, std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
#endif
#ifdef WEBRTC_CODEC_OPUS
{FLAG_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
#endif
{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.
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;
stats_plotter_.reset(new NetEqStatsPlotter(FLAG_matlabplot, FLAG_pythonplot,
FLAG_concealment_events,
output_file_name));
ssrc_switch_detector_.reset(
new SsrcSwitchDetector(stats_plotter_->stats_getter()->delay_analyzer()));
callbacks.post_insert_packet = ssrc_switch_detector_.get();
callbacks.get_audio_callback = stats_plotter_->stats_getter();
callbacks.simulation_ended_callback = stats_plotter_.get();
NetEq::Config config;
config.sample_rate_hz = *sample_rate_hz;
return absl::make_unique<NetEqTest>(config, codecs, ext_codecs_,
std::move(input), std::move(output),
callbacks);
}
} // namespace test
} // namespace webrtc

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/*
* Copyright (c) 2018 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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
#include <memory>
#include "modules/audio_coding/neteq/tools/neteq_test.h"
namespace webrtc {
namespace test {
class SsrcSwitchDetector;
class NetEqStatsGetter;
class NetEqStatsPlotter;
// Note that the NetEqTestFactory needs to be alive when the NetEqTest object is
// used for a simulation.
class NetEqTestFactory {
public:
NetEqTestFactory();
~NetEqTestFactory();
std::unique_ptr<NetEqTest> InitializeTest(int argc, char* argv[]);
private:
std::unique_ptr<AudioDecoder> replacement_decoder_;
NetEqTest::ExtDecoderMap ext_codecs_;
std::unique_ptr<SsrcSwitchDetector> ssrc_switch_detector_;
std::unique_ptr<NetEqStatsPlotter> stats_plotter_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_