2013-07-10 00:45:36 +00:00
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/*
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* libjingle
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2015-01-20 21:36:13 +00:00
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* Copyright 2013 Google Inc.
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2013-07-10 00:45:36 +00:00
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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2016-02-10 10:53:12 +01:00
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#include "webrtc/api/localaudiosource.h"
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2013-07-10 00:45:36 +00:00
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#include <vector>
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2016-02-10 10:53:12 +01:00
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#include "webrtc/api/mediaconstraintsinterface.h"
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Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
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#include "webrtc/media/base/mediaengine.h"
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2013-07-10 00:45:36 +00:00
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using webrtc::MediaConstraintsInterface;
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using webrtc::MediaSourceInterface;
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namespace webrtc {
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namespace {
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// Convert constraints to audio options. Return false if constraints are
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// invalid.
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2014-08-13 13:51:58 +00:00
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void FromConstraints(const MediaConstraintsInterface::Constraints& constraints,
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2013-07-10 00:45:36 +00:00
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cricket::AudioOptions* options) {
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// This design relies on the fact that all the audio constraints are actually
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// "options", i.e. boolean-valued and always satisfiable. If the constraints
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// are extended to include non-boolean values or actual format constraints,
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// a different algorithm will be required.
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2015-06-09 16:03:13 +02:00
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struct {
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const char* name;
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2015-11-10 22:34:18 +01:00
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rtc::Optional<bool>& value;
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2015-06-09 16:03:13 +02:00
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} key_to_value[] = {
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2015-06-15 09:14:03 +02:00
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{MediaConstraintsInterface::kGoogEchoCancellation,
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2015-06-09 16:03:13 +02:00
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options->echo_cancellation},
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{MediaConstraintsInterface::kExtendedFilterEchoCancellation,
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options->extended_filter_aec},
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{MediaConstraintsInterface::kDAEchoCancellation,
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options->delay_agnostic_aec},
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{MediaConstraintsInterface::kAutoGainControl, options->auto_gain_control},
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{MediaConstraintsInterface::kExperimentalAutoGainControl,
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options->experimental_agc},
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{MediaConstraintsInterface::kNoiseSuppression,
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options->noise_suppression},
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{MediaConstraintsInterface::kExperimentalNoiseSuppression,
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options->experimental_ns},
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{MediaConstraintsInterface::kHighpassFilter, options->highpass_filter},
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{MediaConstraintsInterface::kTypingNoiseDetection,
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options->typing_detection},
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{MediaConstraintsInterface::kAudioMirroring, options->stereo_swapping},
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{MediaConstraintsInterface::kAecDump, options->aec_dump}
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};
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for (const auto& constraint : constraints) {
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2013-07-10 00:45:36 +00:00
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bool value = false;
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2015-06-09 16:03:13 +02:00
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if (!rtc::FromString(constraint.value, &value))
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2013-07-10 00:45:36 +00:00
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continue;
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2015-06-09 16:03:13 +02:00
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for (auto& entry : key_to_value) {
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if (constraint.key.compare(entry.name) == 0)
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2015-11-10 22:34:18 +01:00
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entry.value = rtc::Optional<bool>(value);
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2015-06-09 16:03:13 +02:00
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}
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2013-07-10 00:45:36 +00:00
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}
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}
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} // namespace
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2014-07-29 17:36:52 +00:00
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rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
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2013-10-25 21:18:33 +00:00
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const PeerConnectionFactoryInterface::Options& options,
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2013-07-10 00:45:36 +00:00
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const MediaConstraintsInterface* constraints) {
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2014-07-29 17:36:52 +00:00
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rtc::scoped_refptr<LocalAudioSource> source(
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new rtc::RefCountedObject<LocalAudioSource>());
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2013-10-25 21:18:33 +00:00
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source->Initialize(options, constraints);
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2013-07-10 00:45:36 +00:00
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return source;
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}
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void LocalAudioSource::Initialize(
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2013-10-25 21:18:33 +00:00
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const PeerConnectionFactoryInterface::Options& options,
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2013-07-10 00:45:36 +00:00
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const MediaConstraintsInterface* constraints) {
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if (!constraints)
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return;
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// Apply optional constraints first, they will be overwritten by mandatory
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// constraints.
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FromConstraints(constraints->GetOptional(), &options_);
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2014-08-13 13:51:58 +00:00
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cricket::AudioOptions mandatory_options;
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FromConstraints(constraints->GetMandatory(), &mandatory_options);
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options_.SetAll(mandatory_options);
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2013-07-10 00:45:36 +00:00
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source_state_ = kLive;
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}
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} // namespace webrtc
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