1 files changed, 218 insertions, 0 deletions
diff --git a/chromium/media/cast/cast_config.h b/chromium/media/cast/cast_config.h
new file mode 100644
index 00000000000..988924aab45
--- /dev/null
+++ b/chromium/media/cast/cast_config.h
@@ -0,0 +1,218 @@
+// Copyright 2013 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef MEDIA_CAST_CAST_CONFIG_H_
+#define MEDIA_CAST_CAST_CONFIG_H_
+
+#include <string>
+#include <vector>
+
+#include "base/basictypes.h"
+#include "base/callback.h"
+#include "base/memory/ref_counted.h"
+#include "media/cast/cast_defines.h"
+
+namespace media {
+namespace cast {
+
+enum RtcpMode {
+  kRtcpCompound,  // Compound RTCP mode is described by RFC 4585.
+  kRtcpReducedSize,  // Reduced-size RTCP mode is described by RFC 5506.
+};
+
+enum VideoCodec {
+  kVp8,
+  kH264,
+  kExternalVideo,
+};
+
+enum AudioCodec {
+  kOpus,
+  kPcm16,
+  kExternalAudio,
+};
+
+struct AudioSenderConfig {
+  AudioSenderConfig();
+
+  uint32 sender_ssrc;
+  uint32 incoming_feedback_ssrc;
+
+  int rtcp_interval;
+  std::string rtcp_c_name;
+  RtcpMode rtcp_mode;
+
+  int rtp_history_ms;  // The time RTP packets are stored for retransmissions.
+  int rtp_max_delay_ms;
+  int rtp_payload_type;
+
+  bool use_external_encoder;
+  int frequency;
+  int channels;
+  int bitrate;
+  AudioCodec codec;
+};
+
+struct VideoSenderConfig {
+  VideoSenderConfig();
+
+  uint32 sender_ssrc;
+  uint32 incoming_feedback_ssrc;
+
+  int rtcp_interval;
+  std::string rtcp_c_name;
+  RtcpMode rtcp_mode;
+
+  int rtp_history_ms;  // The time RTP packets are stored for retransmissions.
+  int rtp_max_delay_ms;
+  int rtp_payload_type;
+
+  bool use_external_encoder;
+  int width;  // Incoming frames will be scaled to this size.
+  int height;
+
+  float congestion_control_back_off;
+  int max_bitrate;
+  int min_bitrate;
+  int start_bitrate;
+  int max_qp;
+  int min_qp;
+  int max_frame_rate;
+  int max_number_of_video_buffers_used;  // Max value depend on codec.
+  VideoCodec codec;
+  int number_of_cores;
+};
+
+struct AudioReceiverConfig {
+  AudioReceiverConfig();
+
+  uint32 feedback_ssrc;
+  uint32 incoming_ssrc;
+
+  int rtcp_interval;
+  std::string rtcp_c_name;
+  RtcpMode rtcp_mode;
+
+  // The time the receiver is prepared to wait for retransmissions.
+  int rtp_max_delay_ms;
+  int rtp_payload_type;
+
+  bool use_external_decoder;
+  int frequency;
+  int channels;
+  AudioCodec codec;
+};
+
+struct VideoReceiverConfig {
+  VideoReceiverConfig();
+
+  uint32 feedback_ssrc;
+  uint32 incoming_ssrc;
+
+  int rtcp_interval;
+  std::string rtcp_c_name;
+  RtcpMode rtcp_mode;
+
+  // The time the receiver is prepared to wait for retransmissions.
+  int rtp_max_delay_ms;
+  int rtp_payload_type;
+
+  bool use_external_decoder;
+  int max_frame_rate;
+
+  // Some HW decoders can not run faster than the frame rate, preventing it
+  // from catching up after a glitch.
+  bool decoder_faster_than_max_frame_rate;
+  VideoCodec codec;
+};
+
+struct I420VideoPlane {
+  int stride;
+  int length;
+  uint8* data;
+};
+
+struct I420VideoFrame {
+  int width;
+  int height;
+  I420VideoPlane y_plane;
+  I420VideoPlane u_plane;
+  I420VideoPlane v_plane;
+};
+
+struct EncodedVideoFrame {
+  EncodedVideoFrame();
+  ~EncodedVideoFrame();
+
+  VideoCodec codec;
+  bool key_frame;
+  uint8 frame_id;
+  uint8 last_referenced_frame_id;
+  std::vector<uint8> data;
+};
+
+struct PcmAudioFrame {
+  PcmAudioFrame();
+  ~PcmAudioFrame();
+
+  int channels;  // Samples in interleaved stereo format. L0, R0, L1 ,R1 ,...
+  int frequency;
+  std::vector<int16> samples;
+};
+
+struct EncodedAudioFrame {
+  EncodedAudioFrame();
+  ~EncodedAudioFrame();
+
+  AudioCodec codec;
+  uint8 frame_id;  // Needed to release the frame. Not used send side.
+  int samples;  // Needed send side to advance the RTP timestamp.
+                // Not used receive side.
+  std::vector<uint8> data;
+};
+
+class PacketSender {
+ public:
+  // All packets to be sent to the network will be delivered via this function.
+  virtual bool SendPacket(const uint8* packet, int length) = 0;
+
+  virtual ~PacketSender() {}
+};
+
+class PacketReceiver : public base::RefCountedThreadSafe<PacketReceiver> {
+ public:
+  // All packets received from the network should be delivered via this
+  // function.
+  virtual void ReceivedPacket(const uint8* packet, int length,
+                              const base::Closure callback) = 0;
+
+  virtual ~PacketReceiver() {}
+};
+
+class VideoEncoderController {
+ public:
+  // Inform the encoder about the new target bit rate.
+  virtual void SetBitRate(int new_bit_rate) = 0;
+
+  // Inform the encoder to not encode the next frame.
+  // Note: this setting is sticky and should last until called with false.
+  virtual void SkipNextFrame(bool skip_next_frame) = 0;
+
+  // Inform the encoder to encode the next frame as a key frame.
+  virtual void GenerateKeyFrame() = 0;
+
+  // Inform the encoder to only reference frames older or equal to frame_id;
+  virtual void LatestFrameIdToReference(uint8 frame_id) = 0;
+
+  // Query the codec about how many frames it has skipped due to slow ACK.
+  virtual int NumberOfSkippedFrames() const = 0;
+
+ protected:
+  virtual ~VideoEncoderController() {}
+};
+
+}  // namespace cast
+}  // namespace media
+
+#endif  // MEDIA_CAST_CAST_CONFIG_H_