Add missing includes

 * Probably not necessary but a diff when running a step in build.sh

include/talk/app/webrtc/mediastreamobserver.h

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+/*
+ * 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.
+ */
+
+#ifndef TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
+#define TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/sigslot.h"
+
+namespace webrtc {
+
+// Helper class which will listen for changes to a stream and emit the
+// corresponding signals.
+class MediaStreamObserver : public ObserverInterface {
+ public:
+  explicit MediaStreamObserver(MediaStreamInterface* stream);
+  ~MediaStreamObserver();
+
+  const MediaStreamInterface* stream() const { return stream_; }
+
+  void OnChanged() override;
+
+  sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
+      SignalAudioTrackAdded;
+  sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
+      SignalAudioTrackRemoved;
+  sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
+      SignalVideoTrackAdded;
+  sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
+      SignalVideoTrackRemoved;
+
+ private:
+  rtc::scoped_refptr<MediaStreamInterface> stream_;
+  AudioTrackVector cached_audio_tracks_;
+  VideoTrackVector cached_video_tracks_;
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_

include/talk/app/webrtc/remoteaudiotrack.h

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+/*
+ * 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.
+ */
+
+// TODO(tommi): Delete this file when removed from build files in Chromium.

include/webrtc/audio_receive_stream.h

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+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
+#define WEBRTC_AUDIO_RECEIVE_STREAM_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/config.h"
+#include "webrtc/stream.h"
+#include "webrtc/transport.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioDecoder;
+class AudioSinkInterface;
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
+class AudioReceiveStream : public ReceiveStream {
+ public:
+  struct Stats {
+    uint32_t remote_ssrc = 0;
+    int64_t bytes_rcvd = 0;
+    uint32_t packets_rcvd = 0;
+    uint32_t packets_lost = 0;
+    float fraction_lost = 0.0f;
+    std::string codec_name;
+    uint32_t ext_seqnum = 0;
+    uint32_t jitter_ms = 0;
+    uint32_t jitter_buffer_ms = 0;
+    uint32_t jitter_buffer_preferred_ms = 0;
+    uint32_t delay_estimate_ms = 0;
+    int32_t audio_level = -1;
+    float expand_rate = 0.0f;
+    float speech_expand_rate = 0.0f;
+    float secondary_decoded_rate = 0.0f;
+    float accelerate_rate = 0.0f;
+    float preemptive_expand_rate = 0.0f;
+    int32_t decoding_calls_to_silence_generator = 0;
+    int32_t decoding_calls_to_neteq = 0;
+    int32_t decoding_normal = 0;
+    int32_t decoding_plc = 0;
+    int32_t decoding_cng = 0;
+    int32_t decoding_plc_cng = 0;
+    int64_t capture_start_ntp_time_ms = 0;
+  };
+
+  struct Config {
+    std::string ToString() const;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp {
+      std::string ToString() const;
+
+      // Synchronization source (stream identifier) to be received.
+      uint32_t remote_ssrc = 0;
+
+      // Sender SSRC used for sending RTCP (such as receiver reports).
+      uint32_t local_ssrc = 0;
+
+      // RTP header extensions used for the received stream.
+      std::vector<RtpExtension> extensions;
+    } rtp;
+
+    Transport* receive_transport = nullptr;
+    Transport* rtcp_send_transport = nullptr;
+
+    // Underlying VoiceEngine handle, used to map AudioReceiveStream to lower-
+    // level components.
+    // TODO(solenberg): Remove when VoiceEngine channels are created outside
+    // of Call.
+    int voe_channel_id = -1;
+
+    // Identifier for an A/V synchronization group. Empty string to disable.
+    // TODO(pbos): Synchronize streams in a sync group, not just one video
+    // stream to one audio stream. Tracked by issue webrtc:4762.
+    std::string sync_group;
+
+    // Decoders for every payload that we can receive. Call owns the
+    // AudioDecoder instances once the Config is submitted to
+    // Call::CreateReceiveStream().
+    // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
+    std::map<uint8_t, AudioDecoder*> decoder_map;
+
+    // TODO(pbos): Remove config option once combined A/V BWE is always on.
+    bool combined_audio_video_bwe = false;
+  };
+
+  virtual Stats GetStats() const = 0;
+
+  // Sets an audio sink that receives unmixed audio from the receive stream.
+  // Ownership of the sink is passed to the stream and can be used by the
+  // caller to do lifetime management (i.e. when the sink's dtor is called).
+  // Only one sink can be set and passing a null sink, clears an existing one.
+  // NOTE: Audio must still somehow be pulled through AudioTransport for audio
+  // to stream through this sink. In practice, this happens if mixed audio
+  // is being pulled+rendered and/or if audio is being pulled for the purposes
+  // of feeding to the AEC.
+  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_

include/webrtc/audio_send_stream.h

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+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_SEND_STREAM_H_
+#define WEBRTC_AUDIO_SEND_STREAM_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/stream.h"
+#include "webrtc/transport.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
+class AudioSendStream : public SendStream {
+ public:
+  struct Stats {
+    // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
+    uint32_t local_ssrc = 0;
+    int64_t bytes_sent = 0;
+    int32_t packets_sent = 0;
+    int32_t packets_lost = -1;
+    float fraction_lost = -1.0f;
+    std::string codec_name;
+    int32_t ext_seqnum = -1;
+    int32_t jitter_ms = -1;
+    int64_t rtt_ms = -1;
+    int32_t audio_level = -1;
+    float aec_quality_min = -1.0f;
+    int32_t echo_delay_median_ms = -1;
+    int32_t echo_delay_std_ms = -1;
+    int32_t echo_return_loss = -100;
+    int32_t echo_return_loss_enhancement = -100;
+    bool typing_noise_detected = false;
+  };
+
+  struct Config {
+    Config() = delete;
+    explicit Config(Transport* send_transport)
+        : send_transport(send_transport) {}
+
+    std::string ToString() const;
+
+    // Receive-stream specific RTP settings.
+    struct Rtp {
+      std::string ToString() const;
+
+      // Sender SSRC.
+      uint32_t ssrc = 0;
+
+      // RTP header extensions used for the sent stream.
+      std::vector<RtpExtension> extensions;
+
+      // RTCP CNAME, see RFC 3550.
+      std::string c_name;
+    } rtp;
+
+    // Transport for outgoing packets. The transport is expected to exist for
+    // the entire life of the AudioSendStream and is owned by the API client.
+    Transport* send_transport = nullptr;
+
+    // Underlying VoiceEngine handle, used to map AudioSendStream to lower-level
+    // components.
+    // TODO(solenberg): Remove when VoiceEngine channels are created outside
+    // of Call.
+    int voe_channel_id = -1;
+
+    // Ownership of the encoder object is transferred to Call when the config is
+    // passed to Call::CreateAudioSendStream().
+    // TODO(solenberg): Implement, once we configure codecs through the new API.
+    // rtc::scoped_ptr<AudioEncoder> encoder;
+    int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
+    int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
+  };
+
+  // TODO(solenberg): Make payload_type a config property instead.
+  virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
+                                  uint32_t duration_ms) = 0;
+  virtual Stats GetStats() const = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_SEND_STREAM_H_

include/webrtc/frame_callback.h

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+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_FRAME_CALLBACK_H_
+#define WEBRTC_FRAME_CALLBACK_H_
+
+#include <stddef.h>
+
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+class VideoFrame;
+
+struct EncodedFrame {
+ public:
+  EncodedFrame() : data_(NULL), length_(0), frame_type_(kEmptyFrame) {}
+  EncodedFrame(const uint8_t* data, size_t length, FrameType frame_type)
+    : data_(data), length_(length), frame_type_(frame_type) {}
+
+  const uint8_t* data_;
+  const size_t length_;
+  const FrameType frame_type_;
+};
+
+class I420FrameCallback {
+ public:
+  // This function is called with a I420 frame allowing the user to modify the
+  // frame content.
+  virtual void FrameCallback(VideoFrame* video_frame) = 0;
+
+ protected:
+  virtual ~I420FrameCallback() {}
+};
+
+class EncodedFrameObserver {
+ public:
+  virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) = 0;
+
+ protected:
+  virtual ~EncodedFrameObserver() {}
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_FRAME_CALLBACK_H_