webrtc_m130/api/test/neteq_simulator_factory.cc
Alessio Bazzica 5ad789ceff Reland "NetEQ RTP Play: Optionally write output audio file"
This reverts commit c4b391a257ebf85448e58e73a96eb267635b6d6a.

Reason for revert: issue fixed

Original change's description:
> Revert "NetEQ RTP Play: Optionally write output audio file"
>
> This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f.
>
> Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots.
>
> Original change's description:
> > NetEQ RTP Play: Optionally write output audio file
> >
> > This CL makes the output audio file optional to more
> > quickly run neteq_rtpplay when no audio output is needed.
> > The CL also includes necessary adaptations because of pre-existing
> > dependencies (e.g., the output audio file name is used to create
> > the plotting script file names).
> >
> > The command line arguments are retro-compatible - i.e., same behavior
> > when specifying the output audio file and the new flag
> > --output_files_base_name is not used.
> >
> > This CL also includes a test script with which the retro-compatibility
> > has been verified.
> >
> > Bug: webrtc:10337
> > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27067}
>
> TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org
>
> Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10337
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27078}

TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org

Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539
Bug: webrtc:10337
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27106}
2019-03-13 15:33:29 +00:00

58 lines
1.9 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/test/neteq_simulator_factory.h"
#include <string>
#include "absl/memory/memory.h"
#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
#include "rtc_base/checks.h"
#include "rtc_base/flags.h"
namespace {
WEBRTC_DEFINE_string(replacement_audio_file,
"",
"A PCM file that will be used to populate dummy"
" RTP packets");
WEBRTC_DEFINE_int(max_nr_packets_in_buffer,
50,
"Maximum allowed number of packets in the buffer");
} // namespace
namespace webrtc {
namespace test {
NetEqSimulatorFactory::NetEqSimulatorFactory()
: factory_(absl::make_unique<NetEqTestFactory>()) {}
NetEqSimulatorFactory::~NetEqSimulatorFactory() = default;
std::unique_ptr<NetEqSimulator> NetEqSimulatorFactory::CreateSimulator(
int argc,
char* argv[]) {
RTC_CHECK(!rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true))
<< "Error while parsing command-line flags";
RTC_CHECK_EQ(argc, 3) << "Wrong number of input arguments. Expected 3, got "
<< argc;
// TODO(ivoc) Stop (ab)using command-line flags in this function.
const std::string output_audio_filename(argv[2]);
NetEqTestFactory::Config config;
config.replacement_audio_file = FLAG_replacement_audio_file;
config.max_nr_packets_in_buffer = FLAG_max_nr_packets_in_buffer;
config.output_audio_filename = output_audio_filename;
return factory_->InitializeTest(argv[1], config);
}
} // namespace test
} // namespace webrtc