Alex Loiko f475e3aa0e Change levels of different speech signal in tool.
The conversational_speech_generator tool now adjusts the level of
different speech segments.

Implementation:
The Turn and MultiEndCall::SpeakingTurn structs have an extra 'gain'
member.  It's read and parsed in timing.cc and put in a Turn
struct. It's put in a SpeakingTurn struct in multiend_call.cc and read
and applied to the signal in simulator.cc

Bug: webrtc:7494
Change-Id: I9b82a896eb616c8b5ef14d41dfdfd085ef1d3fbb
Reviewed-on: https://webrtc-review.googlesource.com/26280
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21714}
2018-01-22 14:19:28 +00:00

51 lines
1.5 KiB
C++

/*
* 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 MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_
#include <string>
#include <vector>
#include "api/array_view.h"
namespace webrtc {
namespace test {
namespace conversational_speech {
struct Turn{
Turn(std::string new_speaker_name,
std::string new_audiotrack_file_name,
int new_offset,
int gain)
: speaker_name(new_speaker_name),
audiotrack_file_name(new_audiotrack_file_name),
offset(new_offset),
gain(gain) {}
bool operator==(const Turn &b) const;
std::string speaker_name;
std::string audiotrack_file_name;
int offset;
int gain;
};
// Loads a list of turns from a file.
std::vector<Turn> LoadTiming(const std::string& timing_filepath);
// Writes a list of turns into a file.
void SaveTiming(const std::string& timing_filepath,
rtc::ArrayView<const Turn> timing);
} // namespace conversational_speech
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_TIMING_H_