Move talk/app/webrtc to webrtc/api
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.
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
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc 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/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
diff --git a/webrtc/api/rtpreceiver.cc b/webrtc/api/rtpreceiver.cc
new file mode 100644
index 0000000..11d074a
--- /dev/null
+++ b/webrtc/api/rtpreceiver.cc
@@ -0,0 +1,107 @@
+/*
+ * 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 "webrtc/api/rtpreceiver.h"
+
+#include "webrtc/api/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