| Index: media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer_config.h
|
| diff --git a/media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer_config.h b/media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer_config.h
|
| deleted file mode 100644
|
| index 1a2549e66b2e78571eb00845f949e476058fc499..0000000000000000000000000000000000000000
|
| --- a/media/cast/net/rtp_sender/rtp_packetizer/rtp_packetizer_config.h
|
| +++ /dev/null
|
| @@ -1,39 +0,0 @@
|
| -// 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 CAST_NET_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_CONFIG_H_
|
| -#define CAST_NET_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_CONFIG_H_
|
| -
|
| -#include "media/cast/cast_config.h"
|
| -#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
|
| -
|
| -namespace media {
|
| -namespace cast {
|
| -
|
| -struct RtpPacketizerConfig {
|
| - RtpPacketizerConfig();
|
| -
|
| - // General.
|
| - bool audio;
|
| - int payload_type;
|
| - uint16 max_payload_length;
|
| - uint16 sequence_number;
|
| - uint32 rtp_timestamp;
|
| - int frequency;
|
| -
|
| - // SSRC.
|
| - unsigned int ssrc;
|
| -
|
| - // Video.
|
| - VideoCodec video_codec;
|
| -
|
| - // Audio.
|
| - uint8 channels;
|
| - AudioCodec audio_codec;
|
| -};
|
| -
|
| -} // namespace cast
|
| -} // namespace media
|
| -
|
| -#endif // CAST_NET_RTP_SENDER_RTP_PACKETIZER_RTP_PACKETIZER_CONFIG_H_
|
|
|