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-Original-Commit-Position: refs/heads/master@{#11545}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 15583c19d7d491d53f9bf9c0f6d15a734b0a473d
diff --git a/api/rtpsender.h b/api/rtpsender.h
new file mode 100644
index 0000000..45b765d
--- /dev/null
+++ b/api/rtpsender.h
@@ -0,0 +1,195 @@
+/*
+ * 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.
+ */
+
+// This file contains classes that implement RtpSenderInterface.
+// An RtpSender associates a MediaStreamTrackInterface with an underlying
+// transport (provided by AudioProviderInterface/VideoProviderInterface)
+
+#ifndef WEBRTC_API_RTPSENDER_H_
+#define WEBRTC_API_RTPSENDER_H_
+
+#include <string>
+
+#include "webrtc/api/mediastreamprovider.h"
+#include "webrtc/api/rtpsenderinterface.h"
+#include "webrtc/api/statscollector.h"
+#include "webrtc/base/basictypes.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/media/base/audiorenderer.h"
+
+namespace webrtc {
+
+// LocalAudioSinkAdapter receives data callback as a sink to the local
+// AudioTrack, and passes the data to the sink of AudioRenderer.
+class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
+ public cricket::AudioRenderer {
+ public:
+ LocalAudioSinkAdapter();
+ virtual ~LocalAudioSinkAdapter();
+
+ private:
+ // AudioSinkInterface implementation.
+ void OnData(const void* audio_data,
+ int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames) override;
+
+ // cricket::AudioRenderer implementation.
+ void SetSink(cricket::AudioRenderer::Sink* sink) override;
+
+ cricket::AudioRenderer::Sink* sink_;
+ // Critical section protecting |sink_|.
+ rtc::CriticalSection lock_;
+};
+
+class AudioRtpSender : public ObserverInterface,
+ public rtc::RefCountedObject<RtpSenderInterface> {
+ public:
+ // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
+ // at the appropriate times.
+ AudioRtpSender(AudioTrackInterface* track,
+ const std::string& stream_id,
+ AudioProviderInterface* provider,
+ StatsCollector* stats);
+
+ // Randomly generates stream_id.
+ AudioRtpSender(AudioTrackInterface* track,
+ AudioProviderInterface* provider,
+ StatsCollector* stats);
+
+ // Randomly generates id and stream_id.
+ AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
+
+ virtual ~AudioRtpSender();
+
+ // ObserverInterface implementation
+ void OnChanged() override;
+
+ // RtpSenderInterface implementation
+ bool SetTrack(MediaStreamTrackInterface* track) override;
+ rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+ return track_.get();
+ }
+
+ void SetSsrc(uint32_t ssrc) override;
+
+ uint32_t ssrc() const override { return ssrc_; }
+
+ cricket::MediaType media_type() const override {
+ return cricket::MEDIA_TYPE_AUDIO;
+ }
+
+ std::string id() const override { return id_; }
+
+ void set_stream_id(const std::string& stream_id) override {
+ stream_id_ = stream_id;
+ }
+ std::string stream_id() const override { return stream_id_; }
+
+ void Stop() override;
+
+ private:
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // AudioProviderInterface::SetAudioSend.
+ void SetAudioSend();
+
+ std::string id_;
+ std::string stream_id_;
+ AudioProviderInterface* provider_;
+ StatsCollector* stats_;
+ rtc::scoped_refptr<AudioTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ bool stopped_ = false;
+
+ // Used to pass the data callback from the |track_| to the other end of
+ // cricket::AudioRenderer.
+ rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_;
+};
+
+class VideoRtpSender : public ObserverInterface,
+ public rtc::RefCountedObject<RtpSenderInterface> {
+ public:
+ VideoRtpSender(VideoTrackInterface* track,
+ const std::string& stream_id,
+ VideoProviderInterface* provider);
+
+ // Randomly generates stream_id.
+ VideoRtpSender(VideoTrackInterface* track, VideoProviderInterface* provider);
+
+ // Randomly generates id and stream_id.
+ explicit VideoRtpSender(VideoProviderInterface* provider);
+
+ virtual ~VideoRtpSender();
+
+ // ObserverInterface implementation
+ void OnChanged() override;
+
+ // RtpSenderInterface implementation
+ bool SetTrack(MediaStreamTrackInterface* track) override;
+ rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
+ return track_.get();
+ }
+
+ void SetSsrc(uint32_t ssrc) override;
+
+ uint32_t ssrc() const override { return ssrc_; }
+
+ cricket::MediaType media_type() const override {
+ return cricket::MEDIA_TYPE_VIDEO;
+ }
+
+ std::string id() const override { return id_; }
+
+ void set_stream_id(const std::string& stream_id) override {
+ stream_id_ = stream_id;
+ }
+ std::string stream_id() const override { return stream_id_; }
+
+ void Stop() override;
+
+ private:
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // VideoProviderInterface::SetVideoSend.
+ void SetVideoSend();
+
+ std::string id_;
+ std::string stream_id_;
+ VideoProviderInterface* provider_;
+ rtc::scoped_refptr<VideoTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ bool stopped_ = false;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_RTPSENDER_H_
