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Unified Diff: talk/app/webrtc/localaudiosource.cc

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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Index: talk/app/webrtc/localaudiosource.cc
diff --git a/talk/app/webrtc/localaudiosource.cc b/talk/app/webrtc/localaudiosource.cc
deleted file mode 100644
index 9789fe6f543a3cab6a8a36f4d26a916b230d27b0..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/localaudiosource.cc
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/localaudiosource.h"
-
-#include <vector>
-
-#include "talk/app/webrtc/mediaconstraintsinterface.h"
-#include "webrtc/media/base/mediaengine.h"
-
-using webrtc::MediaConstraintsInterface;
-using webrtc::MediaSourceInterface;
-
-namespace webrtc {
-
-namespace {
-
-// Convert constraints to audio options. Return false if constraints are
-// invalid.
-void FromConstraints(const MediaConstraintsInterface::Constraints& constraints,
- cricket::AudioOptions* options) {
- // This design relies on the fact that all the audio constraints are actually
- // "options", i.e. boolean-valued and always satisfiable. If the constraints
- // are extended to include non-boolean values or actual format constraints,
- // a different algorithm will be required.
- struct {
- const char* name;
- rtc::Optional<bool>& value;
- } key_to_value[] = {
- {MediaConstraintsInterface::kGoogEchoCancellation,
- options->echo_cancellation},
- {MediaConstraintsInterface::kExtendedFilterEchoCancellation,
- options->extended_filter_aec},
- {MediaConstraintsInterface::kDAEchoCancellation,
- options->delay_agnostic_aec},
- {MediaConstraintsInterface::kAutoGainControl, options->auto_gain_control},
- {MediaConstraintsInterface::kExperimentalAutoGainControl,
- options->experimental_agc},
- {MediaConstraintsInterface::kNoiseSuppression,
- options->noise_suppression},
- {MediaConstraintsInterface::kExperimentalNoiseSuppression,
- options->experimental_ns},
- {MediaConstraintsInterface::kHighpassFilter, options->highpass_filter},
- {MediaConstraintsInterface::kTypingNoiseDetection,
- options->typing_detection},
- {MediaConstraintsInterface::kAudioMirroring, options->stereo_swapping},
- {MediaConstraintsInterface::kAecDump, options->aec_dump}
- };
-
- for (const auto& constraint : constraints) {
- bool value = false;
- if (!rtc::FromString(constraint.value, &value))
- continue;
-
- for (auto& entry : key_to_value) {
- if (constraint.key.compare(entry.name) == 0)
- entry.value = rtc::Optional<bool>(value);
- }
- }
-}
-
-} // namespace
-
-rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
- const PeerConnectionFactoryInterface::Options& options,
- const MediaConstraintsInterface* constraints) {
- rtc::scoped_refptr<LocalAudioSource> source(
- new rtc::RefCountedObject<LocalAudioSource>());
- source->Initialize(options, constraints);
- return source;
-}
-
-void LocalAudioSource::Initialize(
- const PeerConnectionFactoryInterface::Options& options,
- const MediaConstraintsInterface* constraints) {
- if (!constraints)
- return;
-
- // Apply optional constraints first, they will be overwritten by mandatory
- // constraints.
- FromConstraints(constraints->GetOptional(), &options_);
-
- cricket::AudioOptions mandatory_options;
- FromConstraints(constraints->GetMandatory(), &mandatory_options);
- options_.SetAll(mandatory_options);
- source_state_ = kLive;
-}
-
-} // namespace webrtc
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