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Unified Diff: talk/app/webrtc/mediastreaminterface.h

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/mediastreaminterface.h
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
deleted file mode 100644
index 2d2dc9ac9c5fb06d8aa25ba070f0f7f427cfd1ea..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/mediastreaminterface.h
+++ /dev/null
@@ -1,296 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 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 interfaces for MediaStream, MediaTrack and MediaSource.
-// These interfaces are used for implementing MediaStream and MediaTrack as
-// defined in http://dev.w3.org/2011/webrtc/editor/webrtc.html#stream-api. These
-// interfaces must be used only with PeerConnection. PeerConnectionManager
-// interface provides the factory methods to create MediaStream and MediaTracks.
-
-#ifndef TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
-#define TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
-
-#include <string>
-#include <vector>
-
-#include "webrtc/base/basictypes.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/media/base/videosinkinterface.h"
-
-namespace cricket {
-
-class AudioRenderer;
-class VideoCapturer;
-class VideoRenderer;
-class VideoFrame;
-
-} // namespace cricket
-
-namespace webrtc {
-
-// Generic observer interface.
-class ObserverInterface {
- public:
- virtual void OnChanged() = 0;
-
- protected:
- virtual ~ObserverInterface() {}
-};
-
-class NotifierInterface {
- public:
- virtual void RegisterObserver(ObserverInterface* observer) = 0;
- virtual void UnregisterObserver(ObserverInterface* observer) = 0;
-
- virtual ~NotifierInterface() {}
-};
-
-// Base class for sources. A MediaStreamTrack have an underlying source that
-// provide media. A source can be shared with multiple tracks.
-class MediaSourceInterface : public rtc::RefCountInterface,
- public NotifierInterface {
- public:
- enum SourceState {
- kInitializing,
- kLive,
- kEnded,
- kMuted
- };
-
- virtual SourceState state() const = 0;
-
- virtual bool remote() const = 0;
-
- protected:
- virtual ~MediaSourceInterface() {}
-};
-
-// Information about a track.
-class MediaStreamTrackInterface : public rtc::RefCountInterface,
- public NotifierInterface {
- public:
- enum TrackState {
- kInitializing, // Track is beeing negotiated.
- kLive = 1, // Track alive
- kEnded = 2, // Track have ended
- kFailed = 3, // Track negotiation failed.
- };
-
- static const char kAudioKind[];
- static const char kVideoKind[];
-
- virtual std::string kind() const = 0;
- virtual std::string id() const = 0;
- virtual bool enabled() const = 0;
- virtual TrackState state() const = 0;
- virtual bool set_enabled(bool enable) = 0;
- // These methods should be called by implementation only.
- virtual bool set_state(TrackState new_state) = 0;
-
- protected:
- virtual ~MediaStreamTrackInterface() {}
-};
-
-// Interface for rendering VideoFrames from a VideoTrack
-class VideoRendererInterface
- : public rtc::VideoSinkInterface<cricket::VideoFrame> {
- public:
- // |frame| may have pending rotation. For clients which can't apply rotation,
- // |frame|->GetCopyWithRotationApplied() will return a frame that has the
- // rotation applied.
- virtual void RenderFrame(const cricket::VideoFrame* frame) = 0;
- // Intended to replace RenderFrame.
- void OnFrame(const cricket::VideoFrame& frame) override {
- RenderFrame(&frame);
- }
-
- protected:
- // The destructor is protected to prevent deletion via the interface.
- // This is so that we allow reference counted classes, where the destructor
- // should never be public, to implement the interface.
- virtual ~VideoRendererInterface() {}
-};
-
-class VideoSourceInterface;
-
-class VideoTrackInterface : public MediaStreamTrackInterface {
- public:
- // Register a renderer that will render all frames received on this track.
- virtual void AddRenderer(VideoRendererInterface* renderer) = 0;
- // Deregister a renderer.
- virtual void RemoveRenderer(VideoRendererInterface* renderer) = 0;
-
- virtual VideoSourceInterface* GetSource() const = 0;
-
- // Return the track input sink. I.e., frames sent to this sink are
- // propagated to all renderers registered with the track. The
- // returned sink must not change between calls. Currently, this
- // method is used for remote tracks (VideoRtpReceiver); further
- // refactoring is planned for this path, it's unclear if this method
- // belongs here long term.
-
- // We do this instead of simply implementing the
- // VideoSourceInterface directly, because if we did the latter, we'd
- // need an OnFrame method in VideoTrackProxy, with a thread jump on
- // each call.
-
- // TODO(nisse): It has a default implementation so that mock
- // objects, in particular, chrome's MockWebRtcVideoTrack, doesn't
- // need to know about it. Consider removing the implementation (or
- // this comment) after refactoring dust settles.
- virtual rtc::VideoSinkInterface<cricket::VideoFrame>* GetSink() {
- return nullptr;
- };
-
- protected:
- virtual ~VideoTrackInterface() {}
-};
-
-// Interface for receiving audio data from a AudioTrack.
-class AudioTrackSinkInterface {
- public:
- virtual void OnData(const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- size_t number_of_channels,
- size_t number_of_frames) = 0;
-
- protected:
- virtual ~AudioTrackSinkInterface() {}
-};
-
-// AudioSourceInterface is a reference counted source used for AudioTracks.
-// The same source can be used in multiple AudioTracks.
-class AudioSourceInterface : public MediaSourceInterface {
- public:
- class AudioObserver {
- public:
- virtual void OnSetVolume(double volume) = 0;
-
- protected:
- virtual ~AudioObserver() {}
- };
-
- // TODO(xians): Makes all the interface pure virtual after Chrome has their
- // implementations.
- // Sets the volume to the source. |volume| is in the range of [0, 10].
- // TODO(tommi): This method should be on the track and ideally volume should
- // be applied in the track in a way that does not affect clones of the track.
- virtual void SetVolume(double volume) {}
-
- // Registers/unregisters observer to the audio source.
- virtual void RegisterAudioObserver(AudioObserver* observer) {}
- virtual void UnregisterAudioObserver(AudioObserver* observer) {}
-
- // TODO(tommi): Make pure virtual.
- virtual void AddSink(AudioTrackSinkInterface* sink) {}
- virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
-};
-
-// Interface of the audio processor used by the audio track to collect
-// statistics.
-class AudioProcessorInterface : public rtc::RefCountInterface {
- public:
- struct AudioProcessorStats {
- AudioProcessorStats() : typing_noise_detected(false),
- echo_return_loss(0),
- echo_return_loss_enhancement(0),
- echo_delay_median_ms(0),
- aec_quality_min(0.0),
- echo_delay_std_ms(0) {}
- ~AudioProcessorStats() {}
-
- bool typing_noise_detected;
- int echo_return_loss;
- int echo_return_loss_enhancement;
- int echo_delay_median_ms;
- float aec_quality_min;
- int echo_delay_std_ms;
- };
-
- // Get audio processor statistics.
- virtual void GetStats(AudioProcessorStats* stats) = 0;
-
- protected:
- virtual ~AudioProcessorInterface() {}
-};
-
-class AudioTrackInterface : public MediaStreamTrackInterface {
- public:
- // TODO(xians): Figure out if the following interface should be const or not.
- virtual AudioSourceInterface* GetSource() const = 0;
-
- // Add/Remove a sink that will receive the audio data from the track.
- virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
- virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
-
- // Get the signal level from the audio track.
- // Return true on success, otherwise false.
- // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
- // after Chrome has the correct implementation of the interface.
- virtual bool GetSignalLevel(int* level) { return false; }
-
- // Get the audio processor used by the audio track. Return NULL if the track
- // does not have any processor.
- // TODO(xians): Make the interface pure virtual.
- virtual rtc::scoped_refptr<AudioProcessorInterface>
- GetAudioProcessor() { return NULL; }
-
- protected:
- virtual ~AudioTrackInterface() {}
-};
-
-typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
- AudioTrackVector;
-typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
- VideoTrackVector;
-
-class MediaStreamInterface : public rtc::RefCountInterface,
- public NotifierInterface {
- public:
- virtual std::string label() const = 0;
-
- virtual AudioTrackVector GetAudioTracks() = 0;
- virtual VideoTrackVector GetVideoTracks() = 0;
- virtual rtc::scoped_refptr<AudioTrackInterface>
- FindAudioTrack(const std::string& track_id) = 0;
- virtual rtc::scoped_refptr<VideoTrackInterface>
- FindVideoTrack(const std::string& track_id) = 0;
-
- virtual bool AddTrack(AudioTrackInterface* track) = 0;
- virtual bool AddTrack(VideoTrackInterface* track) = 0;
- virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
- virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
-
- protected:
- virtual ~MediaStreamInterface() {}
-};
-
-} // namespace webrtc
-
-#endif // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
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