Conversational Speech tool, MultiEndCall class and unit tests via mocking

MultiEndCall is responsible for analyzing and validating timing information and audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible (and hence valid) is not implemented yet.

MultiEndCall is designed using dependency injection. This allows to use mock objects with which we can quickly simulate different timings and track lengths without needing actual wav files.

BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2761853002
Cr-Commit-Position: refs/heads/master@{#17421}
Committed: https://chromium.googlesource.com/external/webrtc/+/33397438780d5817ddfc0207e17ca27d6ed1a786

[AleBzk, 2017-03-23 14:52:47]

Description was changed from

==========
MultiEndCall class & unit tests for conversational speech tool

The new class is responsible for analyzing timing information and audiotracks
and validating them.
If valid, it provides access to the list of ends (aka speakers) and WavReader
instances (one for each audiotrack).
The class will be used by the main worker of the tool (not written yet) to
create one mixed audiotrack for each end.

BUG=webrtc:7218
==========

to

==========
MultiEndCall class & unit tests for conversational speech tool

The new class is responsible for analyzing timing information and audiotracks
and validating them.
If valid, it provides access to the list of ends (aka speakers) and WavReader
instances (one for each audiotrack).
The class will be used by the main worker of the tool (not written yet) to
create one mixed audiotrack for each end.

BUG=webrtc:7218
==========

[AleBzk, 2017-03-23 14:52:47]

alessiob@webrtc.org changed reviewers:
+ henrik.lundin@webrtc.org

[AleBzk, 2017-03-23 15:05:10]

Description was changed from

==========
MultiEndCall class & unit tests for conversational speech tool

The new class is responsible for analyzing timing information and audiotracks
and validating them.
If valid, it provides access to the list of ends (aka speakers) and WavReader
instances (one for each audiotrack).
The class will be used by the main worker of the tool (not written yet) to
create one mixed audiotrack for each end.

BUG=webrtc:7218
==========

to

==========
MultiEndCall is responsible for analyzing and validating timing information and
audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and
builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible
(and hence valid) is not implemented yet.

MultiEndCall is designed using dependency injection. This allows to use mock
objects with which we can quickly simulate different timings and track lengths
without needing actual wav files.

To facilitate the review, you can start from the comments I added in the files
below (following the order):
- BUILD.gn
- wavreader_abstract_factory.h
- wavreader_interface.h
- multiend_call.h

Finally, note that some files in this CL include TODO comments.
This was intentionally done to keep the size of this CL as compact as possible.

BUG=webrtc:7218
==========

[AleBzk, 2017-03-23 15:07:14]

Read the description of this CL to find a guide on how to review it.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn:46: #
"wavreader_factory.h",
I added these files to (hopefully) help the reviewer having a better picture of
where I want to go and everything fits together.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn:51:
"//webrtc/common_audio",
This dep will be used to borrow WavReader, which is used via the adapter defined
in wavreader_adaptor.h

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc:43:
{"A", "a3", 0},
Note that we have 5 unique track names (a Turn is a triplet speaker_name,
new_audiotrack_file_name, offset). This is used below in the new test.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h:33:
std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
Note the 3rd argument in the constructor; it used for dependency injection.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h:23:
class WavReaderAbstractFactory {
This is the interface of what I inject into MultiEndCall instances.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:21:
class WavReaderInterface {
This is the interface of what is created by a concrete WavReader factory through
the Create() method.
The factory is injected into MultiEndCall instances.

[hlundin-webrtc, 2017-03-24 12:55:15]

On 2017/03/23 15:07:14, AleBzk wrote:
> Read the description of this CL to find a guide on how to review it.
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> File webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
> (right):
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn:46: #
> "wavreader_factory.h",
> I added these files to (hopefully) help the reviewer having a better picture
of
> where I want to go and everything fits together.
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn:51:
> "//webrtc/common_audio",
> This dep will be used to borrow WavReader, which is used via the adapter
defined
> in wavreader_adaptor.h
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> File
>
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
> (right):
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
>
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc:43:
> {"A", "a3", 0},
> Note that we have 5 unique track names (a Turn is a triplet speaker_name,
> new_audiotrack_file_name, offset). This is used below in the new test.
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> File
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
> (right):
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h:33:
> std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
> Note the 3rd argument in the constructor; it used for dependency injection.
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> File
>
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
> (right):
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
>
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h:23:
> class WavReaderAbstractFactory {
> This is the interface of what I inject into MultiEndCall instances.
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
> File
>
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
> (right):
> 
>
https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
>
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:21:
> class WavReaderInterface {
> This is the interface of what is created by a concrete WavReader factory
through
> the Create() method.
> The factory is injected into MultiEndCall instances.

Remove the review instructions from the CL description, please.

[AleBzk, 2017-03-24 12:56:15]

Description was changed from

==========
MultiEndCall is responsible for analyzing and validating timing information and
audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and
builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible
(and hence valid) is not implemented yet.

MultiEndCall is designed using dependency injection. This allows to use mock
objects with which we can quickly simulate different timings and track lengths
without needing actual wav files.

To facilitate the review, you can start from the comments I added in the files
below (following the order):
- BUILD.gn
- wavreader_abstract_factory.h
- wavreader_interface.h
- multiend_call.h

Finally, note that some files in this CL include TODO comments.
This was intentionally done to keep the size of this CL as compact as possible.

BUG=webrtc:7218
==========

to

==========
MultiEndCall is responsible for analyzing and validating timing information and
audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and
builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible
(and hence valid) is not implemented yet.

MultiEndCall is designed using dependency injection. This allows to use mock
objects with which we can quickly simulate different timings and track lengths
without needing actual wav files.

BUG=webrtc:7218
==========

[hlundin-webrtc, 2017-03-24 13:44:15]

Nice work.

See some comments inline.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn:46: #
"wavreader_factory.h",
On 2017/03/23 15:07:13, AleBzk wrote:
> I added these files to (hopefully) help the reviewer having a better picture
of
> where I want to go and everything fits together.

Include them in the build, and make stub implementations (returning nullptr, 0,
whatever, and possibly also annotate them with FATAL() from checks.h).

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc:11:
#include <cstdio>
stdio.h

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc:17:
#include
"webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
Order of includes.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:2:
*  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
Can this file be folded into the h-file?

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:13:
#include <stdint.h>
#include "webrtc/typedefs.h"

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:22:
// TODO(alessiob): extract filename, parse to extract sample rate, number of
Since this is a mock, I think you can add extra ctor parameters for the
information you want, and skip any complicated string-parsing for that
information. The ctor of the mock does not have to match the mocked ctor.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:26:
MockWavReader::~MockWavReader() {
= default

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h:14:
#include <stdint.h>
#include "webrtc/typedefs.h" instead of stdint.h and cstddef/stddef.h.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc:2:
*  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
Is this file needed? Can't you just = default the ctor and dtor in the h-file?

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:17:
#include "webrtc/base/array_view.h"
Skip this; already included in header.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:37:
MultiEndCall::~MultiEndCall() {}
= default;

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:48:
void MultiEndCall::FindSpeakerNames() {
This is only going to be called from the constructor, right? If so, I suggest
you skip the local std::set and insert the names directly into speaker_names_.
DCHECK that the vector is empty before.

void MultiEndCall::FindSpeakerNames() {
  RTC_DCHECK(speaker_names_.empty());
  for (const Turn& turn : timing_) {
    speaker_names_.insert(turn.speaker_name);
  }
}

Now, this is short enough to actually do it in the constructor, and skip this
method. Then you can also skip the DCHECK.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:59:
if (it != audiotrack_readers_.end())
This method is also supposed to be called from the ctor only, right? If so,
simply DCHECK that audiotrack_readers_ is empty, and skip this.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:69:
audiotrack_readers_.insert(
I think you can make this somewhat shorter, like
audiotrack_readers_.emplace(turn.audiotrack_file_name, std::move(wavreader));

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h:25:
virtual ~WavReaderAbstractFactory() {}
Use "= default;" instead of "{}".

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.cc:13:
#include <stdint.h>
#include "webrtc/typedefs.h"

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h:16:
#include <stdint.h>
#include "webrtc/typedefs.h"

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc:17:
WavReaderFactory::WavReaderFactory() {}
= default;

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc:19:
WavReaderFactory::~WavReaderFactory() {}
= default;

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:14:
#include <stdint.h>
#include "webrtc/typedefs.h"

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:23:
virtual ~WavReaderInterface() {}
= default;

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:26:
// verifies that the end of the file was reached.
"verifies that the end of the file was reached"
This seems to be an implementation details which does not belong in the
interface. What does it mean to the user if less than requested number of
samples were read?

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:27:
virtual size_t ReadSamples(size_t num_samples, float* samples) = 0;
The style guide recommends that you create two different names instead of
overloading. E.g.,
ReadFloatSamples
ReadInt16Samples

https://google.github.io/styleguide/cppguide.html#Function_Overloading

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:30:
virtual int sample_rate() const = 0;
Since this is an interface, and there is no implementation to read, you may want
to document these methods better. Also, the underscore_naming is reserved for
accessors and mutators (or functions that are very light-weight, similar to an
accessor or mutator), but you don't know that what implementation of this
interface will look like. It might not even have member variables with these
names.

[AleBzk, 2017-03-27 08:24:14]

Thanks a lot for your comments Henrik!

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn:46: #
"wavreader_factory.h",
On 2017/03/24 13:44:04, hlundin-webrtc wrote:
> On 2017/03/23 15:07:13, AleBzk wrote:
> > I added these files to (hopefully) help the reviewer having a better picture
> of
> > where I want to go and everything fits together.
> 
> Include them in the build, and make stub implementations (returning nullptr,
0,
> whatever, and possibly also annotate them with FATAL() from checks.h).

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc:11:
#include <cstdio>
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> stdio.h

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc:17:
#include
"webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> Order of includes.

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:2:
*  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> Can this file be folded into the h-file?

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:13:
#include <stdint.h>
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> #include "webrtc/typedefs.h"

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:22:
// TODO(alessiob): extract filename, parse to extract sample rate, number of
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> Since this is a mock, I think you can add extra ctor parameters for the
> information you want, and skip any complicated string-parsing for that
> information. The ctor of the mock does not have to match the mocked ctor.

Right. I will delegate its factory (namely, MockWavReaderFactory) to pass
sampling rate, # samples and # channels (parameters) directly to the
MockWavReader ctor. In this way, I don't need to parse strings. However, we
still need additional code to simulate audio tracks with different params.

I will do the following:
- create the mock factory instance
- before injecting the mock factory, the client code sets a map to link
filenames to params
- inject the mock factory
In this way, the factory knows sampling rate, # samples and # channels for each
"fake" audio file.
The MockWavReader objects will then be created by simply receiving the params
from the factory - which uses the aforementioned map and passed filename as key.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc:26:
MockWavReader::~MockWavReader() {
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> = default

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h:14:
#include <stdint.h>
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> #include "webrtc/typedefs.h" instead of stdint.h and cstddef/stddef.h.

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc:2:
*  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> Is this file needed? Can't you just = default the ctor and dtor in the h-file?

I tried but I get the following:
error: [chromium-style] Complex destructor has an inline body.  // same for ctor

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:17:
#include "webrtc/base/array_view.h"
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> Skip this; already included in header.

Same for the includes below except for the first then.
What about the includes above from std lib? They're also included in .h

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:37:
MultiEndCall::~MultiEndCall() {}
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> = default;

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:48:
void MultiEndCall::FindSpeakerNames() {
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> This is only going to be called from the constructor, right? If so, I suggest
> you skip the local std::set and insert the names directly into speaker_names_.
> DCHECK that the vector is empty before.
> 
> void MultiEndCall::FindSpeakerNames() {
>   RTC_DCHECK(speaker_names_.empty());
>   for (const Turn& turn : timing_) {
>     speaker_names_.insert(turn.speaker_name);
>   }
> }
> 
> Now, this is short enough to actually do it in the constructor, and skip this
> method. Then you can also skip the DCHECK.

Note that speaker_names_ is std::vector.
I use speaker_names_set to find the unique names while iterating over the turns.
I then copy the set into the vector so that I can return ArrayView<const
std::string> in the getter speaker_names().
If I'm right, ArrayView doesn't work with std::set.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:59:
if (it != audiotrack_readers_.end())
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> This method is also supposed to be called from the ctor only, right?
Yes

> If so, simply DCHECK that audiotrack_readers_ is empty, and skip this.
Nope. Similarly to MultiEndCall::FindSpeakerNames(), I find the set of unique
audio track files (to avoid loading the same file multiple times).
Line 59 checks whether the audio track file name is already a key of the
collection of WavReaderInterface instances.
If so, it skips to the next turn.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:69:
audiotrack_readers_.insert(
On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> I think you can make this somewhat shorter, like
> audiotrack_readers_.emplace(turn.audiotrack_file_name, std::move(wavreader));

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h:25:
virtual ~WavReaderAbstractFactory() {}
On 2017/03/24 13:44:07, hlundin-webrtc wrote:
> Use "= default;" instead of "{}".

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.cc:13:
#include <stdint.h>
On 2017/03/24 13:44:07, hlundin-webrtc wrote:
> #include "webrtc/typedefs.h"

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h:16:
#include <stdint.h>
On 2017/03/24 13:44:07, hlundin-webrtc wrote:
> #include "webrtc/typedefs.h"

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc:17:
WavReaderFactory::WavReaderFactory() {}
On 2017/03/24 13:44:08, hlundin-webrtc wrote:
> = default;

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc:19:
WavReaderFactory::~WavReaderFactory() {}
On 2017/03/24 13:44:09, hlundin-webrtc wrote:
> = default;

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:14:
#include <stdint.h>
On 2017/03/24 13:44:11, hlundin-webrtc wrote:
> #include "webrtc/typedefs.h"

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:23:
virtual ~WavReaderInterface() {}
On 2017/03/24 13:44:13, hlundin-webrtc wrote:
> = default;

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:26:
// verifies that the end of the file was reached.
On 2017/03/24 13:44:11, hlundin-webrtc wrote:
> "verifies that the end of the file was reached"
> This seems to be an implementation details which does not belong in the
> interface. What does it mean to the user if less than requested number of
> samples were read?

Right. I stole the docstring from
https://cs.chromium.org/chromium/src/third_party/webrtc/common_audio/wav_file...
which in fact is for a concrete class.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:27:
virtual size_t ReadSamples(size_t num_samples, float* samples) = 0;
On 2017/03/24 13:44:13, hlundin-webrtc wrote:
> The style guide recommends that you create two different names instead of
> overloading. E.g.,
> ReadFloatSamples
> ReadInt16Samples
> 
> https://google.github.io/styleguide/cppguide.html#Function_Overloading

Done.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:30:
virtual int sample_rate() const = 0;
On 2017/03/24 13:44:10, hlundin-webrtc wrote:
> Since this is an interface, and there is no implementation to read, you may
want
> to document these methods better. Also, the underscore_naming is reserved for
> accessors and mutators (or functions that are very light-weight, similar to an
> accessor or mutator), but you don't know that what implementation of this
> interface will look like. It might not even have member variables with these
> names.

True. But I wanted to provide the same interface as done in
https://cs.chromium.org/chromium/src/third_party/webrtc/common_audio/wav_file....

For now, I keep it as it is. Just let me know if you prefer that I switch to
CapWords.

[hlundin-webrtc, 2017-03-27 09:46:35]

Thanks!

Some more comments.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc:2:
*  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
On 2017/03/27 08:24:13, AleBzk wrote:
> On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> > Is this file needed? Can't you just = default the ctor and dtor in the
h-file?
> 
> I tried but I get the following:
> error: [chromium-style] Complex destructor has an inline body.  // same for
ctor
> 
> 

Acknowledged.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:17:
#include "webrtc/base/array_view.h"
On 2017/03/27 08:24:14, AleBzk wrote:
> On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> > Skip this; already included in header.
> 
> Same for the includes below except for the first then.
> What about the includes above from std lib? They're also included in .h

https://google.github.io/styleguide/cppguide.html#Names_and_Order_of_Includes
"any includes present in the related header do not need to be included again in
the related cc (i.e., foo.cc can rely on foo.h's includes)"

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:48:
void MultiEndCall::FindSpeakerNames() {
On 2017/03/27 08:24:14, AleBzk wrote:
> On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> > This is only going to be called from the constructor, right? If so, I
suggest
> > you skip the local std::set and insert the names directly into
speaker_names_.
> > DCHECK that the vector is empty before.
> > 
> > void MultiEndCall::FindSpeakerNames() {
> >   RTC_DCHECK(speaker_names_.empty());
> >   for (const Turn& turn : timing_) {
> >     speaker_names_.insert(turn.speaker_name);
> >   }
> > }
> > 
> > Now, this is short enough to actually do it in the constructor, and skip
this
> > method. Then you can also skip the DCHECK.
> 
> Note that speaker_names_ is std::vector.
> I use speaker_names_set to find the unique names while iterating over the
turns.
> I then copy the set into the vector so that I can return ArrayView<const
> std::string> in the getter speaker_names().
> If I'm right, ArrayView doesn't work with std::set.

I think you should consider sticking with the set instead of converting to a
vector. This seems like the kind of data that does not benefit from being
represented as a vector. I mean, the actual data in the string objects is most
likely not stored in the continuous array that the vector guarantees.

So, unless the users of the speaker_names_ data will rely on being able to index
the data by numbers (the [] operator), I would recommend to make speaker_names_
itself an std::set, and have speaker_names() return a const ref to it. If you
really rely on the [] indexing, you could consider inserting directly into a
vector, and then run sort and unique on that vector (as in
http://en.cppreference.com/w/cpp/algorithm/unique). But I admit that is not as
elegant.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:69:
audiotrack_readers_.insert(
On 2017/03/27 08:24:14, AleBzk wrote:
> On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> > I think you can make this somewhat shorter, like
> > audiotrack_readers_.emplace(turn.audiotrack_file_name,
std::move(wavreader));
> 
> Done.

Did you try emplace?

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h:25:
virtual ~WavReaderAbstractFactory() {}
On 2017/03/27 08:24:14, AleBzk wrote:
> On 2017/03/24 13:44:07, hlundin-webrtc wrote:
> > Use "= default;" instead of "{}".
> 
> Done.

Not done. Try harder. :)

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:30:
virtual int sample_rate() const = 0;
On 2017/03/27 08:24:14, AleBzk wrote:
> On 2017/03/24 13:44:10, hlundin-webrtc wrote:
> > Since this is an interface, and there is no implementation to read, you may
> want
> > to document these methods better. Also, the underscore_naming is reserved
for
> > accessors and mutators (or functions that are very light-weight, similar to
an
> > accessor or mutator), but you don't know that what implementation of this
> > interface will look like. It might not even have member variables with these
> > names.
> 
> True. But I wanted to provide the same interface as done in
>
https://cs.chromium.org/chromium/src/third_party/webrtc/common_audio/wav_file....
> 
> For now, I keep it as it is. Just let me know if you prefer that I switch to
> CapWords.

I see. This is fine.

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h:43:
int sample_rate_;
Make these consts if you can.

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc:17:
MockWavReaderFactory::MockWavReaderFactory() {}
= default on both.

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:56:
auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
There is no guarantee that audiotrack_file_names are unique?

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:14:
#include <cstddef>
stddef.h
We tend to include the c-version (header.h) instead of the c++ version (cheader)
when both are available.

[AleBzk, 2017-03-27 11:17:06]

Once again thanks!

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:17:
#include "webrtc/base/array_view.h"
On 2017/03/27 09:46:34, hlundin-webrtc wrote:
> On 2017/03/27 08:24:14, AleBzk wrote:
> > On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> > > Skip this; already included in header.
> > 
> > Same for the includes below except for the first then.
> > What about the includes above from std lib? They're also included in .h
> 
> https://google.github.io/styleguide/cppguide.html#Names_and_Order_of_Includes
> "any includes present in the related header do not need to be included again
in
> the related cc (i.e., foo.cc can rely on foo.h's includes)"

Acknowledged.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:69:
audiotrack_readers_.insert(
On 2017/03/27 09:46:34, hlundin-webrtc wrote:
> On 2017/03/27 08:24:14, AleBzk wrote:
> > On 2017/03/24 13:44:05, hlundin-webrtc wrote:
> > > I think you can make this somewhat shorter, like
> > > audiotrack_readers_.emplace(turn.audiotrack_file_name,
> std::move(wavreader));
> > 
> > Done.
> 
> Did you try emplace?

As far as I understand from http://www.cplusplus.com/reference/map/map/emplace/,
emplace() is useful when one can pass the arguments for the constructor.
While this way removes the need to copy/move, it is not applicable to my case
since I need to pull objects from an injected factory.

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
(right):

https://codereview.webrtc.org/2761853002/diff/80001/webrtc/modules/audio_proc...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h:25:
virtual ~WavReaderAbstractFactory() {}
On 2017/03/27 09:46:34, hlundin-webrtc wrote:
> On 2017/03/27 08:24:14, AleBzk wrote:
> > On 2017/03/24 13:44:07, hlundin-webrtc wrote:
> > > Use "= default;" instead of "{}".
> > 
> > Done.
> 
> Not done. Try harder. :)

Best comment ever, thanks :D

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h:43:
int sample_rate_;
On 2017/03/27 09:46:35, hlundin-webrtc wrote:
> Make these consts if you can.

Done.

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc:17:
MockWavReaderFactory::MockWavReaderFactory() {}
On 2017/03/27 09:46:35, hlundin-webrtc wrote:
> = default on both.

Done.

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:56:
auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
On 2017/03/27 09:46:35, hlundin-webrtc wrote:
> There is no guarantee that audiotrack_file_names are unique?

Nope. This allows more flexibility.

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h:14:
#include <cstddef>
On 2017/03/27 09:46:35, hlundin-webrtc wrote:
> stddef.h
> We tend to include the c-version (header.h) instead of the c++ version
(cheader)
> when both are available.

Done.

[hlundin-webrtc, 2017-03-28 11:36:03]

Well done! LGTM.

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
File webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
(right):

https://codereview.webrtc.org/2761853002/diff/100001/webrtc/modules/audio_pro...
webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc:56:
auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
On 2017/03/27 11:17:06, AleBzk wrote:
> On 2017/03/27 09:46:35, hlundin-webrtc wrote:
> > There is no guarantee that audiotrack_file_names are unique?
> 
> Nope. This allows more flexibility.

Acknowledged.

[AleBzk, 2017-03-28 11:37:13]

The CQ bit was checked by alessiob@webrtc.org

[commit-bot: I haz the power, 2017-03-28 11:37:19]

CQ is trying da patch. Follow status at
 
https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/...

[commit-bot: I haz the power, 2017-03-28 11:39:08]

The CQ bit was unchecked by commit-bot@chromium.org

[commit-bot: I haz the power, 2017-03-28 11:39:09]

Try jobs failed on following builders:
  ios64_sim_ios9_dbg on master.tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/ios64_sim_ios9_dbg/buil...)
  ios_arm64_dbg on master.tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/ios_arm64_dbg/builds/18995)
  ios_rel on master.tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/ios_rel/builds/23095)
  mac_asan on master.tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/mac_asan/builds/23383)
  mac_compile_dbg on master.tryserver.webrtc (JOB_FAILED,
http://build.chromium.org/p/tryserver.webrtc/builders/mac_compile_dbg/builds/...)

[AleBzk, 2017-03-28 12:02:40]

The CQ bit was checked by alessiob@webrtc.org

[AleBzk, 2017-03-28 12:02:40]

The patchset sent to the CQ was uploaded after l-g-t-m from
henrik.lundin@webrtc.org
Link to the patchset: https://codereview.webrtc.org/2761853002/#ps160001 (title:
"BUILD deps fixed")

[commit-bot: I haz the power, 2017-03-28 12:02:44]

CQ is trying da patch. Follow status at
 
https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/...

[commit-bot: I haz the power, 2017-03-28 12:40:00]

CQ is committing da patch.

Bot data: {"patchset_id": 160001, "attempt_start_ts": 1490702560021610,
"parent_rev": "83862e3c14a39065cb87e86a61536355f1c9284f", "commit_rev":
"33397438780d5817ddfc0207e17ca27d6ed1a786"}

[commit-bot: I haz the power, 2017-03-28 12:40:03]

Description was changed from

==========
MultiEndCall is responsible for analyzing and validating timing information and
audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and
builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible
(and hence valid) is not implemented yet.

MultiEndCall is designed using dependency injection. This allows to use mock
objects with which we can quickly simulate different timings and track lengths
without needing actual wav files.

BUG=webrtc:7218
==========

to

==========
MultiEndCall is responsible for analyzing and validating timing information and
audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and
builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible
(and hence valid) is not implemented yet.

MultiEndCall is designed using dependency injection. This allows to use mock
objects with which we can quickly simulate different timings and track lengths
without needing actual wav files.

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2761853002
Cr-Commit-Position: refs/heads/master@{#17421}
Committed:
https://chromium.googlesource.com/external/webrtc/+/33397438780d5817ddfc0207e...
==========

[commit-bot: I haz the power, 2017-03-28 12:40:04]

Committed patchset #9 (id:160001) as
https://chromium.googlesource.com/external/webrtc/+/33397438780d5817ddfc0207e...