2015-09-25 13:58:30 +02:00
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# Copyright (c) 2015 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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2017-01-24 06:58:22 -08:00
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import("../webrtc.gni")
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2017-03-06 04:01:16 -08:00
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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2015-09-25 13:58:30 +02:00
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2016-09-23 00:38:52 -07:00
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rtc_static_library("audio") {
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2015-09-25 13:58:30 +02:00
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sources = [
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2018-01-17 11:18:31 +01:00
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"audio_level.cc",
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"audio_level.h",
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2015-09-25 13:58:30 +02:00
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"audio_receive_stream.cc",
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"audio_receive_stream.h",
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2015-10-16 14:35:07 -07:00
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"audio_send_stream.cc",
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"audio_send_stream.h",
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2015-11-06 15:34:49 -08:00
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"audio_state.cc",
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"audio_state.h",
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Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
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"audio_transport_impl.cc",
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"audio_transport_impl.h",
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2018-01-17 11:18:31 +01:00
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"channel.cc",
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"channel.h",
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"channel_proxy.cc",
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"channel_proxy.h",
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2015-10-22 10:49:27 +02:00
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"conversion.h",
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2017-11-01 11:06:56 +01:00
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"null_audio_poller.cc",
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"null_audio_poller.h",
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2018-01-17 11:18:31 +01:00
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"remix_resample.cc",
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"remix_resample.h",
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2017-07-19 00:39:19 -07:00
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"time_interval.cc",
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"time_interval.h",
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2018-01-17 11:18:31 +01:00
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"transport_feedback_packet_loss_tracker.cc",
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"transport_feedback_packet_loss_tracker.h",
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2015-09-25 13:58:30 +02:00
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]
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2016-10-16 23:56:12 -07:00
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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2016-09-02 04:10:34 -07:00
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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2015-09-25 13:58:30 +02:00
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}
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deps = [
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"..:webrtc_common",
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2018-01-17 11:18:31 +01:00
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"../api:array_view",
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2016-12-07 08:23:27 -08:00
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"../api:call_api",
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2018-01-17 11:18:31 +01:00
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"../api:libjingle_peerconnection_api",
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"../api:transport_api",
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2018-02-23 13:18:29 +01:00
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"../api/audio:aec3_factory",
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2018-04-12 22:44:09 +02:00
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"../api/audio:audio_frame_api",
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2018-02-16 13:43:49 +01:00
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"../api/audio:audio_mixer_api",
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2017-05-02 06:46:30 -07:00
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"../api/audio_codecs:audio_codecs_api",
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2017-10-10 14:38:11 +02:00
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"../call:bitrate_allocator",
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2016-12-07 04:52:58 -08:00
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"../call:call_interfaces",
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2017-06-01 04:02:35 -07:00
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"../call:rtp_interfaces",
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2016-11-22 06:42:53 -08:00
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"../common_audio",
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2018-02-05 15:50:41 +01:00
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"../common_audio:common_audio_c",
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2018-02-01 11:04:46 -08:00
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"../logging:rtc_event_audio",
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2018-01-17 11:18:31 +01:00
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"../logging:rtc_event_log_api",
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"../modules/audio_coding",
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"../modules/audio_coding:audio_format_conversion",
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"../modules/audio_coding:audio_network_adaptor_config",
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2017-04-27 02:08:52 -07:00
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"../modules/audio_coding:cng",
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2016-11-17 06:28:59 -08:00
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"../modules/audio_device",
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"../modules/audio_processing",
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2017-02-07 07:14:08 -08:00
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"../modules/bitrate_controller:bitrate_controller",
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2016-12-07 08:23:27 -08:00
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"../modules/pacing:pacing",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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2017-12-06 07:51:33 +01:00
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"../modules/rtp_rtcp",
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2018-01-17 11:18:31 +01:00
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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2018-04-03 13:40:05 +02:00
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"../rtc_base:audio_format_to_string",
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2017-12-13 16:05:42 +01:00
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"../rtc_base:checks",
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2018-01-17 11:18:31 +01:00
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"../rtc_base:rate_limiter",
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2017-11-01 11:06:56 +01:00
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"../rtc_base:rtc_base",
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2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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2018-03-07 14:18:56 +01:00
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"../rtc_base:safe_minmax",
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Adding a new string utility class: SimpleStringBuilder.
This is a fairly minimalistic string building class that
can be used instead of stringstream, which is discouraged
but tempting to use due to its convenient interface and
familiarity for anyone using our logging macros.
As a starter, I'm changing the string building code in
ReceiveStatisticsProxy and SendStatisticsProxy from using
stringstream and using SimpleStringBuilder instead.
In the case of SimpleStringBuilder, there's a single allocation,
it's done on the stack (fast), and minimal code is required for
each concatenation. The developer is responsible for ensuring
that the buffer size is adequate but the class won't overflow
the buffer. In dcheck-enabled builds, a check will go off if
we run out of buffer space.
As part of using SimpleStringBuilder for a small part of
rtc::LogMessage, a few more changes were made:
- SimpleStringBuilder is used for formatting errors instead of ostringstream.
- A new 'noop' state has been introduced for log messages that will be dropped.
- Use a static (singleton) noop ostream object for noop logging messages
instead of building up an actual ostringstream object that will be dropped.
- Add a LogMessageForTest class for better state inspection/testing.
- Fix benign bug in LogTest.Perf, change the test to not use File IO and
always enable it.
- Ensure that minimal work is done for noop messages.
- Remove dependency on rtc::Thread.
- Add tests for the extra_ field, correctly parsed paths and noop handling.
Bug: webrtc:8529, webrtc:4364, webrtc:8933
Change-Id: Ifa258c135135945e4560d9e24315f7d96f784acb
Reviewed-on: https://webrtc-review.googlesource.com/55520
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22203}
2018-02-27 13:51:08 +01:00
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"../rtc_base:stringutils",
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2015-09-25 13:58:30 +02:00
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"../system_wrappers",
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2017-12-07 20:54:55 +01:00
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"../system_wrappers:field_trial_api",
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2018-01-17 11:18:31 +01:00
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"../system_wrappers:metrics_api",
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Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
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"utility:audio_frame_operations",
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Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
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"//third_party/abseil-cpp/absl/memory",
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2018-06-15 12:28:07 +02:00
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"//third_party/abseil-cpp/absl/types:optional",
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2015-09-25 13:58:30 +02:00
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]
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}
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2016-06-14 12:52:54 +02:00
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if (rtc_include_tests) {
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2017-09-14 14:46:47 +02:00
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rtc_source_set("audio_end_to_end_test") {
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testonly = true
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sources = [
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"test/audio_end_to_end_test.cc",
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"test/audio_end_to_end_test.h",
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]
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deps = [
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":audio",
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2018-08-17 14:26:54 +02:00
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"../api:simulated_network_api",
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2018-08-20 13:30:39 +02:00
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"../call:fake_network",
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"../call:simulated_network",
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2017-09-14 14:46:47 +02:00
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"../system_wrappers:system_wrappers",
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"../test:test_common",
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"../test:test_support",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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2016-09-02 04:10:34 -07:00
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rtc_source_set("audio_tests") {
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2016-06-14 12:52:54 +02:00
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testonly = true
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2016-12-07 08:23:27 -08:00
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2016-06-14 12:52:54 +02:00
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sources = [
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"audio_receive_stream_unittest.cc",
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2017-09-15 09:56:08 -07:00
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"audio_send_stream_tests.cc",
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2016-06-14 12:52:54 +02:00
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"audio_send_stream_unittest.cc",
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"audio_state_unittest.cc",
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2018-01-17 11:18:31 +01:00
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"mock_voe_channel_proxy.h",
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"remix_resample_unittest.cc",
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2018-08-07 08:53:41 +02:00
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"test/audio_stats_test.cc",
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2017-07-19 00:39:19 -07:00
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"time_interval_unittest.cc",
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2018-01-17 11:18:31 +01:00
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"transport_feedback_packet_loss_tracker_unittest.cc",
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2016-06-14 12:52:54 +02:00
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]
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deps = [
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":audio",
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2017-09-14 14:46:47 +02:00
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":audio_end_to_end_test",
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2016-11-22 06:42:53 -08:00
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"../api:mock_audio_mixer",
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2018-04-12 22:44:09 +02:00
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"../api/audio:audio_frame_api",
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2018-05-08 14:52:22 +02:00
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"../api/units:time_delta",
|
Remove voe::TransmitMixer
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
2017-12-15 16:42:15 +01:00
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"../call:mock_call_interfaces",
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2018-02-22 14:49:02 +01:00
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"../call:mock_rtp_interfaces",
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2017-11-13 17:04:05 +01:00
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"../call:rtp_interfaces",
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2017-06-21 01:05:22 -07:00
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"../call:rtp_receiver",
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2018-01-17 11:18:31 +01:00
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"../common_audio",
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2018-01-03 09:08:20 +01:00
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"../logging:mocks",
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2016-11-17 06:28:59 -08:00
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"../modules/audio_device:mock_audio_device",
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2016-11-22 06:42:53 -08:00
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"../modules/audio_mixer:audio_mixer_impl",
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2017-11-24 17:29:59 +01:00
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"../modules/audio_processing:audio_processing_statistics",
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2017-12-19 16:44:45 +01:00
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"../modules/audio_processing:mocks",
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2018-01-02 14:20:17 +01:00
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"../modules/bitrate_controller:mocks",
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2016-12-07 08:23:27 -08:00
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"../modules/pacing:pacing",
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2017-11-13 17:04:05 +01:00
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"../modules/rtp_rtcp:mock_rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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2018-01-17 11:18:31 +01:00
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"../rtc_base:checks",
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2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_base_approved",
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2017-11-13 17:04:05 +01:00
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"../rtc_base:rtc_base_tests_utils",
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2017-07-19 10:40:47 -07:00
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"../rtc_base:rtc_task_queue",
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2017-12-13 16:05:42 +01:00
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"../rtc_base:safe_compare",
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2017-11-13 17:04:05 +01:00
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"../system_wrappers:system_wrappers",
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"../test:audio_codec_mocks",
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"../test:rtp_test_utils",
|
2016-11-17 06:48:48 -08:00
|
|
|
"../test:test_common",
|
2016-12-05 01:46:09 -08:00
|
|
|
"../test:test_support",
|
2017-01-05 06:03:24 -08:00
|
|
|
"utility:utility_tests",
|
2016-06-14 12:52:54 +02:00
|
|
|
"//testing/gtest",
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
2016-06-14 12:52:54 +02:00
|
|
|
]
|
2017-03-20 02:06:18 -07:00
|
|
|
|
2016-10-16 23:56:12 -07:00
|
|
|
if (!build_with_chromium && is_clang) {
|
|
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
2016-09-02 04:10:34 -07:00
|
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
2016-06-14 12:52:54 +02:00
|
|
|
}
|
|
|
|
|
}
|
2017-03-06 04:01:16 -08:00
|
|
|
|
|
|
|
|
if (rtc_enable_protobuf) {
|
2017-03-30 04:01:30 -07:00
|
|
|
rtc_test("low_bandwidth_audio_test") {
|
2017-03-06 04:01:16 -08:00
|
|
|
testonly = true
|
|
|
|
|
|
|
|
|
|
sources = [
|
|
|
|
|
"test/low_bandwidth_audio_test.cc",
|
|
|
|
|
]
|
|
|
|
|
|
2017-03-23 03:40:03 -07:00
|
|
|
deps = [
|
2017-09-14 14:46:47 +02:00
|
|
|
":audio_end_to_end_test",
|
2018-08-17 14:26:54 +02:00
|
|
|
"../api:simulated_network_api",
|
2017-03-23 03:40:03 -07:00
|
|
|
"../common_audio",
|
2017-08-29 05:51:57 -07:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-03-23 03:40:03 -07:00
|
|
|
"../system_wrappers",
|
2018-03-15 15:05:39 +01:00
|
|
|
"../test:fileutils",
|
2017-03-23 03:40:03 -07:00
|
|
|
"../test:test_common",
|
|
|
|
|
"../test:test_main",
|
2017-07-17 01:41:41 -07:00
|
|
|
"//testing/gtest",
|
2017-03-23 03:40:03 -07:00
|
|
|
]
|
2017-03-06 04:01:16 -08:00
|
|
|
if (is_android) {
|
2017-03-23 03:40:03 -07:00
|
|
|
deps += [ "//testing/android/native_test:native_test_native_code" ]
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
data = [
|
2017-09-15 06:47:31 +02:00
|
|
|
"../resources/voice_engine/audio_tiny16.wav",
|
2018-03-19 15:58:08 +01:00
|
|
|
"../resources/voice_engine/audio_tiny48.wav",
|
2018-03-18 22:31:06 +01:00
|
|
|
]
|
|
|
|
|
|
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
|
|
|
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
group("low_bandwidth_audio_perf_test") {
|
|
|
|
|
testonly = true
|
|
|
|
|
|
|
|
|
|
deps = [
|
|
|
|
|
":low_bandwidth_audio_test",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
data = [
|
2017-09-28 16:14:37 +02:00
|
|
|
"test/low_bandwidth_audio_test.py",
|
2018-03-18 22:31:06 +01:00
|
|
|
"../resources/voice_engine/audio_tiny16.wav",
|
|
|
|
|
"../resources/voice_engine/audio_tiny48.wav",
|
2017-03-23 03:40:03 -07:00
|
|
|
]
|
2018-03-19 11:33:03 +01:00
|
|
|
if (is_win) {
|
|
|
|
|
data += [ "${root_out_dir}/low_bandwidth_audio_test.exe" ]
|
|
|
|
|
} else {
|
|
|
|
|
data += [ "${root_out_dir}/low_bandwidth_audio_test" ]
|
|
|
|
|
}
|
|
|
|
|
|
2018-01-15 17:37:04 +01:00
|
|
|
if (is_linux || is_android) {
|
2017-09-28 16:14:37 +02:00
|
|
|
data += [
|
|
|
|
|
"../tools_webrtc/audio_quality/linux/PolqaOem64",
|
|
|
|
|
"../tools_webrtc/audio_quality/linux/pesq",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
if (is_win) {
|
|
|
|
|
data += [
|
|
|
|
|
"../tools_webrtc/audio_quality/win/PolqaOem64.dll",
|
|
|
|
|
"../tools_webrtc/audio_quality/win/PolqaOem64.exe",
|
|
|
|
|
"../tools_webrtc/audio_quality/win/pesq.exe",
|
|
|
|
|
"../tools_webrtc/audio_quality/win/vcomp120.dll",
|
|
|
|
|
]
|
|
|
|
|
}
|
|
|
|
|
if (is_mac) {
|
|
|
|
|
data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
|
|
|
|
|
}
|
2018-03-15 13:52:41 +00:00
|
|
|
|
2018-03-18 22:31:06 +01:00
|
|
|
write_runtime_deps = "${root_out_dir}/${target_name}.runtime_deps"
|
2017-03-06 04:01:16 -08:00
|
|
|
}
|
|
|
|
|
}
|
2017-07-17 01:41:41 -07:00
|
|
|
|
|
|
|
|
rtc_source_set("audio_perf_tests") {
|
|
|
|
|
testonly = true
|
|
|
|
|
|
|
|
|
|
sources = [
|
|
|
|
|
"test/audio_bwe_integration_test.cc",
|
|
|
|
|
"test/audio_bwe_integration_test.h",
|
|
|
|
|
]
|
|
|
|
|
deps = [
|
2018-08-17 14:26:54 +02:00
|
|
|
"../api:simulated_network_api",
|
2018-08-20 13:30:39 +02:00
|
|
|
"../call:fake_network",
|
|
|
|
|
"../call:simulated_network",
|
2017-07-17 01:41:41 -07:00
|
|
|
"../common_audio",
|
2017-07-19 10:40:47 -07:00
|
|
|
"../rtc_base:rtc_base_approved",
|
2017-07-17 01:41:41 -07:00
|
|
|
"../system_wrappers",
|
|
|
|
|
"../test:field_trial",
|
2018-03-15 15:05:39 +01:00
|
|
|
"../test:fileutils",
|
2017-08-22 04:02:52 -07:00
|
|
|
"../test:single_threaded_task_queue",
|
2017-07-17 01:41:41 -07:00
|
|
|
"../test:test_common",
|
|
|
|
|
"../test:test_main",
|
|
|
|
|
"//testing/gtest",
|
Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
|
|
|
"//third_party/abseil-cpp/absl/memory",
|
2017-07-17 01:41:41 -07:00
|
|
|
]
|
|
|
|
|
|
|
|
|
|
data = [
|
|
|
|
|
"//resources/voice_engine/audio_dtx16.wav",
|
|
|
|
|
]
|
|
|
|
|
|
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
|
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
|
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
|
|
|
}
|
|
|
|
|
}
|
2016-06-14 12:52:54 +02:00
|
|
|
}
|