Revert of Conversational speech tool, simualtor + unit tests (patchset #12 id:220001 of https://codereview.webrtc.org/2790933002/ )
Reason for revert:
Compile Error.
Original issue's description:
> The simulator puts into action the schedule of speech turns encoded in a MultiEndCall instance. The output is a set of audio track pairs. There is one set for each speaker and each set contains one near-end and one far-end audio track. The tracks are directly written into wav files instead of creating them in memory. To speed up the creation of the output wav files, *all* the source audio tracks (i.e., the atomic speech turns) are pre-loaded.
>
> The ConversationalSpeechTest.MultiEndCallSimulator unit test defines a conversational speech sequence and creates two wav files (with pure tones at 440 and 880 Hz) that are used as atomic speech turn tracks.
>
> This CL also patches MultiEndCall in order to allow input audio tracks with same sample rate and single channel only.
>
> BUG=webrtc:7218
>
> Review-Url: https://codereview.webrtc.org/2790933002
> Cr-Commit-Position: refs/heads/master@{#18480}
> Committed: https://chromium.googlesource.com/external/webrtc/+/6b648c4697cede14605fd2b89425866eec5f7c79
TBR=minyue@webrtc.org,alessiob@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2925123003
Cr-Commit-Position: refs/heads/master@{#18481}
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
index af24f8a..b74c05b 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -35,8 +35,6 @@
"config.h",
"multiend_call.cc",
"multiend_call.h",
- "simulator.cc",
- "simulator.h",
"timing.cc",
"timing.h",
"wavreader_abstract_factory.h",
@@ -69,8 +67,5 @@
"../../../../../webrtc/test:test_support",
"//testing/gmock",
"//testing/gtest",
- "//webrtc:webrtc_common",
- "//webrtc/base:rtc_base_approved",
- "//webrtc/test:test_support",
]
}
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc b/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
index 02513c4..6797beb 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
+++ b/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
@@ -40,16 +40,13 @@
#include <cmath>
#include <map>
#include <memory>
-#include <vector>
#include "webrtc/base/logging.h"
-#include "webrtc/base/optional.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/config.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h"
-#include "webrtc/modules/audio_processing/test/conversational_speech/simulator.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h"
#include "webrtc/test/gmock.h"
@@ -86,12 +83,9 @@
constexpr int kDefaultSampleRate = 48000;
const std::map<std::string, const MockWavReaderFactory::Params>
kDefaultMockWavReaderFactoryParamsMap = {
- {"t300", {kDefaultSampleRate, 1u, 14400u}}, // Mono, 0.3 seconds.
- {"t500", {kDefaultSampleRate, 1u, 24000u}}, // Mono, 0.5 seconds.
- {"t1000", {kDefaultSampleRate, 1u, 48000u}}, // Mono, 1.0 seconds.
- {"sr8000", {8000, 1u, 8000u}}, // 8kHz sample rate, mono, 1 second.
- {"sr16000", {16000, 1u, 16000u}}, // 16kHz sample rate, mono, 1 second.
- {"sr16000_stereo", {16000, 2u, 16000u}}, // Like sr16000, but stereo.
+ {"t300", {kDefaultSampleRate, 1u, 14400u}}, // 0.3 seconds.
+ {"t500", {kDefaultSampleRate, 1u, 24000u}}, // 0.5 seconds.
+ {"t1000", {kDefaultSampleRate, 1u, 48000u}}, // 1.0 seconds.
};
const MockWavReaderFactory::Params& kDefaultMockWavReaderFactoryParams =
kDefaultMockWavReaderFactoryParamsMap.at("t500");
@@ -119,57 +113,6 @@
wav_writer.WriteSamples(samples.data(), params.num_samples);
}
-// Parameters to generate audio tracks with CreateSineWavFile.
-struct SineAudioTrackParams {
- MockWavReaderFactory::Params params;
- float frequency;
-};
-
-// Creates a temporary directory in which sine audio tracks are written.
-std::string CreateTemporarySineAudioTracks(
- const std::map<std::string, SineAudioTrackParams>& sine_tracks_params) {
- // Create temporary directory.
- rtc::Pathname temp_directory(OutputPath());
- temp_directory.AppendFolder("TempConversationalSpeechAudioTracks");
- CreateDir(temp_directory.pathname());
-
- // Create sine tracks.
- for (const auto& it : sine_tracks_params) {
- const rtc::Pathname temp_filepath(temp_directory.pathname(), it.first);
- CreateSineWavFile(
- temp_filepath.pathname(), it.second.params, it.second.frequency);
- }
-
- return temp_directory.pathname();
-}
-
-void CheckAudioTrackParams(const WavReaderFactory& wav_reader_factory,
- const std::string& filepath,
- const MockWavReaderFactory::Params& expeted_params) {
- auto wav_reader = wav_reader_factory.Create(filepath);
- EXPECT_EQ(expeted_params.sample_rate, wav_reader->SampleRate());
- EXPECT_EQ(expeted_params.num_channels, wav_reader->NumChannels());
- EXPECT_EQ(expeted_params.num_samples, wav_reader->NumSamples());
-}
-
-void DeleteFolderAndContents(const std::string& dir) {
- if (!DirExists(dir)) { return; }
- rtc::Optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
- EXPECT_TRUE(dir_content);
- for (const auto& path : *dir_content) {
- if (DirExists(path)) {
- DeleteFolderAndContents(path);
- } else if (FileExists(path)) {
- // TODO(alessiob): Wrap with EXPECT_TRUE() once webrtc:7769 bug fixed.
- RemoveFile(path);
- } else {
- FAIL();
- }
- }
- // TODO(alessiob): Wrap with EXPECT_TRUE() once webrtc:7769 bug fixed.
- RemoveDir(dir);
-}
-
} // namespace
using testing::_;
@@ -195,8 +138,8 @@
TEST_F(ConversationalSpeechTest, TimingSaveLoad) {
// Save test timing.
- const std::string temporary_filepath = TempFilename(
- OutputPath(), "TempTimingTestFile");
+ const std::string temporary_filepath = webrtc::test::TempFilename(
+ webrtc::test::OutputPath(), "TempTimingTestFile");
SaveTiming(temporary_filepath, expected_timing);
// Create a std::vector<Turn> instance by loading from file.
@@ -230,52 +173,6 @@
EXPECT_EQ(6u, multiend_call.speaking_turns().size());
}
-TEST_F(ConversationalSpeechTest, MultiEndCallSetupDifferentSampleRates) {
- const std::vector<Turn> timing = {
- {"A", "sr8000", 0},
- {"B", "sr16000", 0},
- };
- auto mock_wavreader_factory = CreateMockWavReaderFactory();
-
- // There are two unique audio tracks to read.
- EXPECT_CALL(*mock_wavreader_factory, Create(testing::_)).Times(2);
-
- MultiEndCall multiend_call(
- timing, audiotracks_path, std::move(mock_wavreader_factory));
- EXPECT_FALSE(multiend_call.valid());
-}
-
-TEST_F(ConversationalSpeechTest, MultiEndCallSetupMultipleChannels) {
- const std::vector<Turn> timing = {
- {"A", "sr16000_stereo", 0},
- {"B", "sr16000_stereo", 0},
- };
- auto mock_wavreader_factory = CreateMockWavReaderFactory();
-
- // There is one unique audio track to read.
- EXPECT_CALL(*mock_wavreader_factory, Create(testing::_)).Times(1);
-
- MultiEndCall multiend_call(
- timing, audiotracks_path, std::move(mock_wavreader_factory));
- EXPECT_FALSE(multiend_call.valid());
-}
-
-TEST_F(ConversationalSpeechTest,
- MultiEndCallSetupDifferentSampleRatesAndMultipleNumChannels) {
- const std::vector<Turn> timing = {
- {"A", "sr8000", 0},
- {"B", "sr16000_stereo", 0},
- };
- auto mock_wavreader_factory = CreateMockWavReaderFactory();
-
- // There are two unique audio tracks to read.
- EXPECT_CALL(*mock_wavreader_factory, Create(testing::_)).Times(2);
-
- MultiEndCall multiend_call(
- timing, audiotracks_path, std::move(mock_wavreader_factory));
- EXPECT_FALSE(multiend_call.valid());
-}
-
TEST_F(ConversationalSpeechTest, MultiEndCallSetupFirstOffsetNegative) {
const std::vector<Turn> timing = {
{"A", "t500", -100},
@@ -628,70 +525,20 @@
const std::size_t num_samples = duration_seconds * sample_rate;
MockWavReaderFactory::Params params = {sample_rate, 1u, num_samples};
CreateSineWavFile(temp_filename.pathname(), params);
+ LOG(LS_VERBOSE) << "wav file @" << sample_rate << " Hz created ("
+ << num_samples << " samples)";
// Load wav file and check if params match.
WavReaderFactory wav_reader_factory;
- MockWavReaderFactory::Params expeted_params = {
- sample_rate, 1u, num_samples};
- CheckAudioTrackParams(
- wav_reader_factory, temp_filename.pathname(), expeted_params);
+ auto wav_reader = wav_reader_factory.Create(temp_filename.pathname());
+ EXPECT_EQ(sample_rate, wav_reader->SampleRate());
+ EXPECT_EQ(1u, wav_reader->NumChannels());
+ EXPECT_EQ(num_samples, wav_reader->NumSamples());
// Clean up.
remove(temp_filename.pathname().c_str());
}
}
-TEST_F(ConversationalSpeechTest, MultiEndCallSimulator) {
- // Simulated call (one character corresponding to 500 ms):
- // A 0*********...........2*********.....
- // B ...........1*********.....3*********
- const std::vector<Turn> expected_timing = {
- {"A", "t5000_440.wav", 0},
- {"B", "t5000_880.wav", 500},
- {"A", "t5000_440.wav", 0},
- {"B", "t5000_880.wav", -2500},
- };
- const std::size_t expected_duration_seconds = 18;
-
- // Create temporary audio track files.
- const int sample_rate = 16000;
- const std::map<std::string, SineAudioTrackParams> sine_tracks_params = {
- {"t5000_440.wav", {{sample_rate, 1u, sample_rate * 5}, 440.0}},
- {"t5000_880.wav", {{sample_rate, 1u, sample_rate * 5}, 880.0}},
- };
- const std::string audiotracks_path = CreateTemporarySineAudioTracks(
- sine_tracks_params);
-
- // Set up the multi-end call.
- auto wavreader_factory = std::unique_ptr<WavReaderFactory>(
- new WavReaderFactory());
- MultiEndCall multiend_call(
- expected_timing, audiotracks_path, std::move(wavreader_factory));
-
- // Simulate the call.
- rtc::Pathname output_path(audiotracks_path);
- output_path.AppendFolder("output");
- CreateDir(output_path.pathname());
- LOG(LS_VERBOSE) << "simulator output path: " << output_path.pathname();
- auto generated_audiotrak_pairs = conversational_speech::Simulate(
- multiend_call, output_path.pathname());
- EXPECT_EQ(2u, generated_audiotrak_pairs->size());
-
- // Check the output.
- WavReaderFactory wav_reader_factory;
- const MockWavReaderFactory::Params expeted_params = {
- sample_rate, 1u, sample_rate * expected_duration_seconds};
- for (const auto& it : *generated_audiotrak_pairs) {
- LOG(LS_VERBOSE) << "checking far/near-end for <" << it.first << ">";
- CheckAudioTrackParams(
- wav_reader_factory, it.second.near_end, expeted_params);
- CheckAudioTrackParams(
- wav_reader_factory, it.second.far_end, expeted_params);
- }
-
- // Clean.
- EXPECT_NO_FATAL_FAILURE(DeleteFolderAndContents(audiotracks_path));
-}
-
} // namespace test
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
index f83923c..624f981 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
+++ b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
@@ -24,15 +24,36 @@
rtc::ArrayView<const Turn> timing, const std::string& audiotracks_path,
std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory)
: timing_(timing), audiotracks_path_(audiotracks_path),
- wavreader_abstract_factory_(std::move(wavreader_abstract_factory)),
- valid_(false) {
+ wavreader_abstract_factory_(std::move(wavreader_abstract_factory)) {
FindSpeakerNames();
- if (CreateAudioTrackReaders())
- valid_ = CheckTiming();
+ CreateAudioTrackReaders();
+ valid_ = CheckTiming();
}
MultiEndCall::~MultiEndCall() = default;
+const std::set<std::string>& MultiEndCall::speaker_names() const {
+ return speaker_names_;
+}
+
+const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
+ MultiEndCall::audiotrack_readers() const {
+ return audiotrack_readers_;
+}
+
+bool MultiEndCall::valid() const {
+ return valid_;
+}
+
+size_t MultiEndCall::total_duration_samples() const {
+ return total_duration_samples_;
+}
+
+const std::vector<MultiEndCall::SpeakingTurn>& MultiEndCall::speaking_turns()
+ const {
+ return speaking_turns_;
+}
+
void MultiEndCall::FindSpeakerNames() {
RTC_DCHECK(speaker_names_.empty());
for (const Turn& turn : timing_) {
@@ -40,9 +61,8 @@
}
}
-bool MultiEndCall::CreateAudioTrackReaders() {
+void MultiEndCall::CreateAudioTrackReaders() {
RTC_DCHECK(audiotrack_readers_.empty());
- sample_rate_hz_ = 0; // Sample rate will be set when reading the first track.
for (const Turn& turn : timing_) {
auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
if (it != audiotrack_readers_.end())
@@ -55,24 +75,9 @@
// Map the audiotrack file name to a new instance of WavReaderInterface.
std::unique_ptr<WavReaderInterface> wavreader =
wavreader_abstract_factory_->Create(audiotrack_file_path.pathname());
-
- if (sample_rate_hz_ == 0) {
- sample_rate_hz_ = wavreader->SampleRate();
- } else if (sample_rate_hz_ != wavreader->SampleRate()) {
- LOG(LS_ERROR) << "All the audio tracks should have the same sample rate.";
- return false;
- }
-
- if (wavreader->NumChannels() != 1) {
- LOG(LS_ERROR) << "Only mono audio tracks supported.";
- return false;
- }
-
audiotrack_readers_.emplace(
turn.audiotrack_file_name, std::move(wavreader));
}
-
- return true;
}
bool MultiEndCall::CheckTiming() {
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
index 1daeea0..dd03a07 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
+++ b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
@@ -50,23 +50,19 @@
std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
~MultiEndCall();
- const std::set<std::string>& speaker_names() const { return speaker_names_; }
+ const std::set<std::string>& speaker_names() const;
const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
- audiotrack_readers() const { return audiotrack_readers_; }
- bool valid() const { return valid_; }
- int sample_rate() const { return sample_rate_hz_; }
- size_t total_duration_samples() const { return total_duration_samples_; }
- const std::vector<SpeakingTurn>& speaking_turns() const {
- return speaking_turns_; }
+ audiotrack_readers() const;
+ bool valid() const;
+ size_t total_duration_samples() const;
+ const std::vector<SpeakingTurn>& speaking_turns() const;
private:
// Finds unique speaker names.
void FindSpeakerNames();
- // Creates one WavReader instance for each unique audiotrack. It returns false
- // if the audio tracks do not have the same sample rate or if they are not
- // mono.
- bool CreateAudioTrackReaders();
+ // Creates one WavReader instance for each unique audiotrack.
+ void CreateAudioTrackReaders();
// Validates the speaking turns timing information. Accepts cross-talk, but
// only up to 2 speakers. Rejects unordered turns and self cross-talk.
@@ -79,7 +75,6 @@
std::map<std::string, std::unique_ptr<WavReaderInterface>>
audiotrack_readers_;
bool valid_;
- int sample_rate_hz_;
size_t total_duration_samples_;
std::vector<SpeakingTurn> speaking_turns_;
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc b/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
deleted file mode 100644
index 705b1df..0000000
--- a/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
+++ /dev/null
@@ -1,221 +0,0 @@
-/*
- * Copyright (c) 2017 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 "webrtc/modules/audio_processing/test/conversational_speech/simulator.h"
-
-#include <set>
-#include <utility>
-#include <vector>
-
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/pathutils.h"
-#include "webrtc/base/ptr_util.h"
-#include "webrtc/common_audio/wav_file.h"
-#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
-
-namespace webrtc {
-namespace test {
-namespace {
-
-using conversational_speech::MultiEndCall;
-using conversational_speech::SpeakerOutputFilePaths;
-using conversational_speech::WavReaderInterface;
-
-// Combines output path and speaker names to define the output file paths for
-// the near-end and far=end audio tracks.
-std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>>
- InitSpeakerOutputFilePaths(const std::set<std::string>& speaker_names,
- const std::string& output_path) {
- // Create map.
- auto speaker_output_file_paths_map = rtc::MakeUnique<
- std::map<std::string, SpeakerOutputFilePaths>>();
-
- // Add near-end and far-end output paths into the map.
- for (const auto& speaker_name : speaker_names) {
- const rtc::Pathname near_end_path(
- output_path, "s_" + speaker_name + "-near_end.wav");
- LOG(LS_VERBOSE) << "The near-end audio track will be created in "
- << near_end_path.pathname() << ".";
-
- const rtc::Pathname far_end_path(
- output_path, "s_" + speaker_name + "-far_end.wav");
- LOG(LS_VERBOSE) << "The far-end audio track will be created in "
- << far_end_path.pathname() << ".";
-
- // Add to map.
- speaker_output_file_paths_map->emplace(
- std::piecewise_construct,
- std::forward_as_tuple(speaker_name),
- std::forward_as_tuple(near_end_path.pathname(),
- far_end_path.pathname()));
- }
-
- return speaker_output_file_paths_map;
-}
-
-// Class that provides one WavWriter for the near-end and one for the far-end
-// output track of a speaker.
-class SpeakerWavWriters {
- public:
- SpeakerWavWriters(
- const SpeakerOutputFilePaths& output_file_paths, int sample_rate)
- : near_end_wav_writer_(output_file_paths.near_end, sample_rate, 1u),
- far_end_wav_writer_(output_file_paths.far_end, sample_rate, 1u) {}
- WavWriter* near_end_wav_writer() {
- return &near_end_wav_writer_;
- }
- WavWriter* far_end_wav_writer() {
- return &far_end_wav_writer_;
- }
- private:
- WavWriter near_end_wav_writer_;
- WavWriter far_end_wav_writer_;
-};
-
-// Initializes one WavWriter instance for each speaker and both the near-end and
-// far-end output tracks.
-std::unique_ptr<std::map<std::string, SpeakerWavWriters>>
- InitSpeakersWavWriters(const std::map<std::string, SpeakerOutputFilePaths>&
- speaker_output_file_paths, int sample_rate) {
- // Create map.
- auto speaker_wav_writers_map = rtc::MakeUnique<
- std::map<std::string, SpeakerWavWriters>>();
-
- // Add SpeakerWavWriters instance into the map.
- for (auto it = speaker_output_file_paths.begin();
- it != speaker_output_file_paths.end(); ++it) {
- speaker_wav_writers_map->emplace(
- std::piecewise_construct,
- std::forward_as_tuple(it->first),
- std::forward_as_tuple(it->second, sample_rate));
- }
-
- return speaker_wav_writers_map;
-}
-
-// Reads all the samples for each audio track.
-std::unique_ptr<std::map<std::string, std::vector<int16_t>>> PreloadAudioTracks(
- const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
- audiotrack_readers) {
- // Create map.
- auto audiotracks_map = rtc::MakeUnique<
- std::map<std::string, std::vector<int16_t>>>();
-
- // Add audio track vectors.
- for (auto it = audiotrack_readers.begin(); it != audiotrack_readers.end();
- ++it) {
- // Add map entry.
- audiotracks_map->emplace(
- std::piecewise_construct,
- std::forward_as_tuple(it->first),
- std::forward_as_tuple(it->second->NumSamples()));
-
- // Read samples.
- it->second->ReadInt16Samples(audiotracks_map->at(it->first));
- }
-
- return audiotracks_map;
-}
-
-// Writes all the values in |source_samples| via |wav_writer|. If the number of
-// previously written samples in |wav_writer| is less than |interval_begin|, it
-// adds zeros as left padding. The padding corresponds to intervals during which
-// a speaker is not active.
-void PadLeftWriteChunk(rtc::ArrayView<const int16_t> source_samples,
- size_t interval_begin, WavWriter* wav_writer) {
- // Add left padding.
- RTC_CHECK(wav_writer);
- RTC_CHECK_GE(interval_begin, wav_writer->num_samples());
- size_t padding_size = interval_begin - wav_writer->num_samples();
- if (padding_size != 0) {
- const std::vector<int16_t> padding(padding_size, 0);
- wav_writer->WriteSamples(padding.data(), padding_size);
- }
-
- // Write source samples.
- wav_writer->WriteSamples(source_samples.data(), source_samples.size());
-}
-
-// Appends zeros via |wav_writer|. The number of zeros is always non-negative
-// and equal to the difference between the previously written samples and
-// |pad_samples|.
-void PadRightWrite(WavWriter* wav_writer, size_t pad_samples) {
- RTC_CHECK(wav_writer);
- RTC_CHECK_GE(pad_samples, wav_writer->num_samples());
- size_t padding_size = pad_samples - wav_writer->num_samples();
- if (padding_size != 0) {
- const std::vector<int16_t> padding(padding_size, 0);
- wav_writer->WriteSamples(padding.data(), padding_size);
- }
-}
-
-} // namespace
-
-namespace conversational_speech {
-
-std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>> Simulate(
- const MultiEndCall& multiend_call, const std::string& output_path) {
- // Set output file paths and initialize wav writers.
- const auto& speaker_names = multiend_call.speaker_names();
- auto speaker_output_file_paths = InitSpeakerOutputFilePaths(
- speaker_names, output_path);
- auto speakers_wav_writers = InitSpeakersWavWriters(
- *speaker_output_file_paths, multiend_call.sample_rate());
-
- // Preload all the input audio tracks.
- const auto& audiotrack_readers = multiend_call.audiotrack_readers();
- auto audiotracks = PreloadAudioTracks(audiotrack_readers);
-
- // TODO(alessiob): When speaker_names.size() == 2, near-end and far-end
- // across the 2 speakers are symmetric; hence, the code below could be
- // replaced by only creating the near-end or the far-end. However, this would
- // require to split the unit tests and document the behavior in README.md.
- // In practice, it should not be an issue since the files are not expected to
- // be signinificant.
-
- // Write near-end and far-end output tracks.
- for (const auto& speaking_turn : multiend_call.speaking_turns()) {
- const std::string& active_speaker_name = speaking_turn.speaker_name;
- auto source_audiotrack = audiotracks->at(
- speaking_turn.audiotrack_file_name);
-
- // Write active speaker's chunk to active speaker's near-end.
- PadLeftWriteChunk(source_audiotrack, speaking_turn.begin,
- speakers_wav_writers->at(
- active_speaker_name).near_end_wav_writer());
-
- // Write active speaker's chunk to other participants' far-ends.
- for (const std::string& speaker_name : speaker_names) {
- if (speaker_name == active_speaker_name)
- continue;
- PadLeftWriteChunk(source_audiotrack, speaking_turn.begin,
- speakers_wav_writers->at(
- speaker_name).far_end_wav_writer());
- }
- }
-
- // Finalize all the output tracks with right padding.
- // This is required to make all the output tracks duration equal.
- size_t duration_samples = multiend_call.total_duration_samples();
- for (const std::string& speaker_name : speaker_names) {
- PadRightWrite(speakers_wav_writers->at(speaker_name).near_end_wav_writer(),
- duration_samples);
- PadRightWrite(speakers_wav_writers->at(speaker_name).far_end_wav_writer(),
- duration_samples);
- }
-
- return speaker_output_file_paths;
-}
-
-} // namespace conversational_speech
-} // namespace test
-} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/simulator.h b/webrtc/modules/audio_processing/test/conversational_speech/simulator.h
deleted file mode 100644
index 6224162..0000000
--- a/webrtc/modules/audio_processing/test/conversational_speech/simulator.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (c) 2017 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_SIMULATOR_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_SIMULATOR_H_
-
-#include <map>
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h"
-
-namespace webrtc {
-namespace test {
-namespace conversational_speech {
-
-struct SpeakerOutputFilePaths {
- SpeakerOutputFilePaths(const std::string& new_near_end,
- const std::string& new_far_end)
- : near_end(new_near_end),
- far_end(new_far_end) {}
- // Paths to the near-end and far-end audio track files.
- const std::string near_end;
- const std::string far_end;
-};
-
-// Generates the near-end and far-end audio track pairs for each speaker.
-std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>>
- Simulate(const MultiEndCall& multiend_call, const std::string& output_path);
-
-} // namespace conversational_speech
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_SIMULATOR_H_
