2014-03-21 12:07:40 +00:00
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/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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2017-09-15 06:47:31 +02:00
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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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2014-03-21 12:07:40 +00:00
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2015-05-22 11:22:11 +02:00
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#include <fstream>
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2016-02-14 09:28:33 -08:00
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#include <memory>
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2016-09-30 22:29:43 -07:00
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2017-09-15 13:58:09 +02:00
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#include "common_types.h" // NOLINT(build/include)
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2017-09-15 06:47:31 +02:00
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#include "modules/audio_coding/neteq/include/neteq.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "modules/audio_coding/neteq/tools/rtp_generator.h"
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#include "modules/include/module_common_types.h"
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#include "rtc_base/flags.h"
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#include "test/gtest.h"
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2017-09-15 13:58:09 +02:00
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#include "typedefs.h" // NOLINT(build/include)
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2014-03-21 12:07:40 +00:00
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namespace webrtc {
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namespace test {
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2014-06-25 12:17:41 +00:00
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class LossModel {
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public:
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virtual ~LossModel() {};
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virtual bool Lost() = 0;
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};
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class NoLoss : public LossModel {
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public:
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2015-03-04 12:58:35 +00:00
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bool Lost() override;
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2014-06-25 12:17:41 +00:00
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};
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class UniformLoss : public LossModel {
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public:
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2014-07-22 09:55:51 +00:00
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UniformLoss(double loss_rate);
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2015-03-04 12:58:35 +00:00
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bool Lost() override;
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2014-06-25 12:17:41 +00:00
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void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
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2014-07-22 09:55:51 +00:00
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2014-06-25 12:17:41 +00:00
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private:
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double loss_rate_;
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};
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class GilbertElliotLoss : public LossModel {
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public:
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GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
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2016-08-10 02:11:30 -07:00
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~GilbertElliotLoss() override;
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2015-03-04 12:58:35 +00:00
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bool Lost() override;
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2014-07-22 09:55:51 +00:00
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2014-06-25 12:17:41 +00:00
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private:
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// Prob. of losing current packet, when previous packet is lost.
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double prob_trans_11_;
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// Prob. of losing current packet, when previous packet is not lost.
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double prob_trans_01_;
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bool lost_last_;
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2016-02-14 09:28:33 -08:00
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std::unique_ptr<UniformLoss> uniform_loss_model_;
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2014-06-25 12:17:41 +00:00
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};
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2014-03-21 12:07:40 +00:00
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class NetEqQualityTest : public ::testing::Test {
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protected:
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NetEqQualityTest(int block_duration_ms,
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int in_sampling_khz,
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int out_sampling_khz,
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2015-10-29 06:20:28 -07:00
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NetEqDecoder decoder_type);
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2016-08-10 02:11:30 -07:00
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~NetEqQualityTest() override;
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2015-05-22 11:22:11 +02:00
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2015-03-04 12:58:35 +00:00
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void SetUp() override;
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2014-03-21 12:07:40 +00:00
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// EncodeBlock(...) does the following:
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// 1. encodes a block of audio, saved in |in_data| and has a length of
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// |block_size_samples| (samples per channel),
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// 2. save the bit stream to |payload| of |max_bytes| bytes in size,
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// 3. returns the length of the payload (in bytes),
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
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2016-03-01 00:41:31 -08:00
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rtc::Buffer* payload, size_t max_bytes) = 0;
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2014-03-21 12:07:40 +00:00
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2014-06-25 12:17:41 +00:00
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// PacketLost(...) determines weather a packet sent at an indicated time gets
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2014-03-21 12:07:40 +00:00
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// lost or not.
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2014-06-25 12:17:41 +00:00
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bool PacketLost();
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2014-03-21 12:07:40 +00:00
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// DecodeBlock() decodes a block of audio using the payload stored in
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// |payload_| with the length of |payload_size_bytes_| (bytes). The decoded
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// audio is to be stored in |out_data_|.
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int DecodeBlock();
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// Transmit() uses |rtp_generator_| to generate a packet and passes it to
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// |neteq_|.
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int Transmit();
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2015-05-12 12:09:59 +02:00
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// Runs encoding / transmitting / decoding.
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void Simulate();
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2014-03-21 12:07:40 +00:00
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2015-05-22 11:22:11 +02:00
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// Write to log file. Usage Log() << ...
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std::ofstream& Log();
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2015-10-29 06:20:28 -07:00
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NetEqDecoder decoder_type_;
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Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1316523002 .
Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-12 16:26:35 -08:00
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const size_t channels_;
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2015-05-22 11:22:11 +02:00
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2014-03-21 12:07:40 +00:00
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private:
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int decoded_time_ms_;
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int decodable_time_ms_;
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double drift_factor_;
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2014-06-25 12:17:41 +00:00
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int packet_loss_rate_;
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2014-03-21 12:07:40 +00:00
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const int block_duration_ms_;
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const int in_sampling_khz_;
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const int out_sampling_khz_;
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// Number of samples per channel in a frame.
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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const size_t in_size_samples_;
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2014-03-21 12:07:40 +00:00
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Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
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size_t payload_size_bytes_;
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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size_t max_payload_bytes_;
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2014-03-21 12:07:40 +00:00
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2016-02-14 09:28:33 -08:00
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std::unique_ptr<InputAudioFile> in_file_;
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std::unique_ptr<AudioSink> output_;
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2015-05-22 11:22:11 +02:00
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std::ofstream log_file_;
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2014-03-21 12:07:40 +00:00
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2016-02-14 09:28:33 -08:00
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std::unique_ptr<RtpGenerator> rtp_generator_;
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std::unique_ptr<NetEq> neteq_;
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std::unique_ptr<LossModel> loss_model_;
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2014-03-21 12:07:40 +00:00
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2016-02-14 09:28:33 -08:00
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std::unique_ptr<int16_t[]> in_data_;
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2016-03-01 00:41:31 -08:00
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rtc::Buffer payload_;
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2016-03-04 10:34:21 -08:00
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AudioFrame out_frame_;
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2017-04-24 09:14:32 -07:00
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RTPHeader rtp_header_;
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2014-06-25 12:17:41 +00:00
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Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
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size_t total_payload_size_bytes_;
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2014-03-21 12:07:40 +00:00
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};
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} // namespace test
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} // namespace webrtc
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2017-09-15 06:47:31 +02:00
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
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