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Unified Diff: talk/app/webrtc/rtpreceiver.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/rtpreceiver.cc
diff --git a/talk/app/webrtc/rtpreceiver.cc b/talk/app/webrtc/rtpreceiver.cc
deleted file mode 100644
index 34efe5c415c703f0b3982e528c40e057317c4c47..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/rtpreceiver.cc
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * libjingle
- * Copyright 2015 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/rtpreceiver.h"
-
-#include "talk/app/webrtc/videosourceinterface.h"
-
-namespace webrtc {
-
-AudioRtpReceiver::AudioRtpReceiver(AudioTrackInterface* track,
- uint32_t ssrc,
- AudioProviderInterface* provider)
- : id_(track->id()),
- track_(track),
- ssrc_(ssrc),
- provider_(provider),
- cached_track_enabled_(track->enabled()) {
- RTC_DCHECK(track_->GetSource()->remote());
- track_->RegisterObserver(this);
- track_->GetSource()->RegisterAudioObserver(this);
- Reconfigure();
-}
-
-AudioRtpReceiver::~AudioRtpReceiver() {
- track_->GetSource()->UnregisterAudioObserver(this);
- track_->UnregisterObserver(this);
- Stop();
-}
-
-void AudioRtpReceiver::OnChanged() {
- if (cached_track_enabled_ != track_->enabled()) {
- cached_track_enabled_ = track_->enabled();
- Reconfigure();
- }
-}
-
-void AudioRtpReceiver::OnSetVolume(double volume) {
- // When the track is disabled, the volume of the source, which is the
- // corresponding WebRtc Voice Engine channel will be 0. So we do not allow
- // setting the volume to the source when the track is disabled.
- if (provider_ && track_->enabled())
- provider_->SetAudioPlayoutVolume(ssrc_, volume);
-}
-
-void AudioRtpReceiver::Stop() {
- // TODO(deadbeef): Need to do more here to fully stop receiving packets.
- if (!provider_) {
- return;
- }
- provider_->SetAudioPlayout(ssrc_, false);
- provider_ = nullptr;
-}
-
-void AudioRtpReceiver::Reconfigure() {
- if (!provider_) {
- return;
- }
- provider_->SetAudioPlayout(ssrc_, track_->enabled());
-}
-
-VideoRtpReceiver::VideoRtpReceiver(VideoTrackInterface* track,
- uint32_t ssrc,
- VideoProviderInterface* provider)
- : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
- provider_->SetVideoPlayout(ssrc_, true, track_->GetSink());
-}
-
-VideoRtpReceiver::~VideoRtpReceiver() {
- // Since cricket::VideoRenderer is not reference counted,
- // we need to remove it from the provider before we are deleted.
- Stop();
-}
-
-void VideoRtpReceiver::Stop() {
- // TODO(deadbeef): Need to do more here to fully stop receiving packets.
- if (!provider_) {
- return;
- }
- provider_->SetVideoPlayout(ssrc_, false, nullptr);
- provider_ = nullptr;
-}
-
-} // namespace webrtc
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