| Index: media/cast/transport/cast_transport_config.h
|
| diff --git a/media/cast/transport/cast_transport_config.h b/media/cast/transport/cast_transport_config.h
|
| deleted file mode 100644
|
| index 4525a5bd4c2db2e5ad1db03681996e107b98b516..0000000000000000000000000000000000000000
|
| --- a/media/cast/transport/cast_transport_config.h
|
| +++ /dev/null
|
| @@ -1,202 +0,0 @@
|
| -// Copyright 2014 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_TRANSPORT_CAST_TRANSPORT_CONFIG_H_
|
| -#define MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_CONFIG_H_
|
| -
|
| -#include <string>
|
| -#include <vector>
|
| -
|
| -#include "base/basictypes.h"
|
| -#include "base/callback.h"
|
| -#include "base/memory/linked_ptr.h"
|
| -#include "base/memory/ref_counted.h"
|
| -#include "base/stl_util.h"
|
| -#include "media/cast/transport/cast_transport_defines.h"
|
| -#include "net/base/ip_endpoint.h"
|
| -
|
| -namespace media {
|
| -namespace cast {
|
| -namespace transport {
|
| -
|
| -enum RtcpMode {
|
| - kRtcpCompound, // Compound RTCP mode is described by RFC 4585.
|
| - kRtcpReducedSize, // Reduced-size RTCP mode is described by RFC 5506.
|
| -};
|
| -
|
| -enum Codec {
|
| - CODEC_UNKNOWN,
|
| - CODEC_AUDIO_OPUS,
|
| - CODEC_AUDIO_PCM16,
|
| - CODEC_VIDEO_FAKE,
|
| - CODEC_VIDEO_VP8,
|
| - CODEC_VIDEO_H264,
|
| - CODEC_LAST = CODEC_VIDEO_H264
|
| -};
|
| -
|
| -struct CastTransportRtpConfig {
|
| - CastTransportRtpConfig();
|
| - ~CastTransportRtpConfig();
|
| -
|
| - // Identifier refering to this sender.
|
| - uint32 ssrc;
|
| -
|
| - // RTP payload type enum: Specifies the type/encoding of frame data.
|
| - int rtp_payload_type;
|
| -
|
| - // The number of most-recent frames that must be stored in the transport
|
| - // layer, to facilitate re-transmissions.
|
| - int stored_frames;
|
| -
|
| - // The AES crypto key and initialization vector. Each of these strings
|
| - // contains the data in binary form, of size kAesKeySize. If they are empty
|
| - // strings, crypto is not being used.
|
| - std::string aes_key;
|
| - std::string aes_iv_mask;
|
| -};
|
| -
|
| -// A combination of metadata and data for one encoded frame. This can contain
|
| -// audio data or video data or other.
|
| -struct EncodedFrame {
|
| - enum Dependency {
|
| - // "null" value, used to indicate whether |dependency| has been set.
|
| - UNKNOWN_DEPENDENCY,
|
| -
|
| - // Not decodable without the reference frame indicated by
|
| - // |referenced_frame_id|.
|
| - DEPENDENT,
|
| -
|
| - // Independently decodable.
|
| - INDEPENDENT,
|
| -
|
| - // Independently decodable, and no future frames will depend on any frames
|
| - // before this one.
|
| - KEY,
|
| -
|
| - DEPENDENCY_LAST = KEY
|
| - };
|
| -
|
| - EncodedFrame();
|
| - ~EncodedFrame();
|
| -
|
| - // Convenience accessors to data as an array of uint8 elements.
|
| - const uint8* bytes() const {
|
| - return reinterpret_cast<uint8*>(string_as_array(
|
| - const_cast<std::string*>(&data)));
|
| - }
|
| - uint8* mutable_bytes() {
|
| - return reinterpret_cast<uint8*>(string_as_array(&data));
|
| - }
|
| -
|
| - // Copies all data members except |data| to |dest|.
|
| - // Does not modify |dest->data|.
|
| - void CopyMetadataTo(EncodedFrame* dest) const;
|
| -
|
| - // This frame's dependency relationship with respect to other frames.
|
| - Dependency dependency;
|
| -
|
| - // The label associated with this frame. Implies an ordering relative to
|
| - // other frames in the same stream.
|
| - uint32 frame_id;
|
| -
|
| - // The label associated with the frame upon which this frame depends. If
|
| - // this frame does not require any other frame in order to become decodable
|
| - // (e.g., key frames), |referenced_frame_id| must equal |frame_id|.
|
| - uint32 referenced_frame_id;
|
| -
|
| - // The stream timestamp, on the timeline of the signal data. For example, RTP
|
| - // timestamps for audio are usually defined as the total number of audio
|
| - // samples encoded in all prior frames. A playback system uses this value to
|
| - // detect gaps in the stream, and otherwise stretch the signal to match
|
| - // playout targets.
|
| - uint32 rtp_timestamp;
|
| -
|
| - // The common reference clock timestamp for this frame. This value originates
|
| - // from a sender and is used to provide lip synchronization between streams in
|
| - // a receiver. Thus, in the sender context, this is set to the time at which
|
| - // the frame was captured/recorded. In the receiver context, this is set to
|
| - // the target playout time. Over a sequence of frames, this time value is
|
| - // expected to drift with respect to the elapsed time implied by the RTP
|
| - // timestamps; and it may not necessarily increment with precise regularity.
|
| - base::TimeTicks reference_time;
|
| -
|
| - // The encoded signal data.
|
| - std::string data;
|
| -};
|
| -
|
| -typedef std::vector<uint8> Packet;
|
| -typedef scoped_refptr<base::RefCountedData<Packet> > PacketRef;
|
| -typedef std::vector<PacketRef> PacketList;
|
| -
|
| -typedef base::Callback<void(scoped_ptr<Packet> packet)> PacketReceiverCallback;
|
| -
|
| -class PacketSender {
|
| - public:
|
| - // Send a packet to the network. Returns false if the network is blocked
|
| - // and we should wait for |cb| to be called. It is not allowed to called
|
| - // SendPacket again until |cb| has been called. Any other errors that
|
| - // occur will be reported through side channels, in such cases, this function
|
| - // will return true indicating that the channel is not blocked.
|
| - virtual bool SendPacket(PacketRef packet, const base::Closure& cb) = 0;
|
| - virtual ~PacketSender() {}
|
| -};
|
| -
|
| -struct RtcpSenderInfo {
|
| - RtcpSenderInfo();
|
| - ~RtcpSenderInfo();
|
| - // First three members are used for lipsync.
|
| - // First two members are used for rtt.
|
| - uint32 ntp_seconds;
|
| - uint32 ntp_fraction;
|
| - uint32 rtp_timestamp;
|
| - uint32 send_packet_count;
|
| - size_t send_octet_count;
|
| -};
|
| -
|
| -struct RtcpReportBlock {
|
| - RtcpReportBlock();
|
| - ~RtcpReportBlock();
|
| - uint32 remote_ssrc; // SSRC of sender of this report.
|
| - uint32 media_ssrc; // SSRC of the RTP packet sender.
|
| - uint8 fraction_lost;
|
| - uint32 cumulative_lost; // 24 bits valid.
|
| - uint32 extended_high_sequence_number;
|
| - uint32 jitter;
|
| - uint32 last_sr;
|
| - uint32 delay_since_last_sr;
|
| -};
|
| -
|
| -struct RtcpDlrrReportBlock {
|
| - RtcpDlrrReportBlock();
|
| - ~RtcpDlrrReportBlock();
|
| - uint32 last_rr;
|
| - uint32 delay_since_last_rr;
|
| -};
|
| -
|
| -// This is only needed because IPC messages don't support more than
|
| -// 5 arguments.
|
| -struct SendRtcpFromRtpSenderData {
|
| - SendRtcpFromRtpSenderData();
|
| - ~SendRtcpFromRtpSenderData();
|
| - uint32 packet_type_flags;
|
| - uint32 sending_ssrc;
|
| - std::string c_name;
|
| - uint32 ntp_seconds;
|
| - uint32 ntp_fraction;
|
| - uint32 rtp_timestamp;
|
| -};
|
| -
|
| -inline bool operator==(RtcpSenderInfo lhs, RtcpSenderInfo rhs) {
|
| - return lhs.ntp_seconds == rhs.ntp_seconds &&
|
| - lhs.ntp_fraction == rhs.ntp_fraction &&
|
| - lhs.rtp_timestamp == rhs.rtp_timestamp &&
|
| - lhs.send_packet_count == rhs.send_packet_count &&
|
| - lhs.send_octet_count == rhs.send_octet_count;
|
| -}
|
| -
|
| -} // namespace transport
|
| -} // namespace cast
|
| -} // namespace media
|
| -
|
| -#endif // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_CONFIG_H_
|
|
|