golanggohlsrtmpwebrtcmedia-serverobs-studiortcprtmp-proxyrtmp-serverrtprtsprtsp-proxyrtsp-relayrtsp-serversrtstreamingwebrtc-proxy
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70 lines
1.6 KiB
70 lines
1.6 KiB
package message |
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import ( |
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"fmt" |
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"time" |
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"github.com/notedit/rtmp/format/flv/flvio" |
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"github.com/aler9/rtsp-simple-server/internal/rtmp/chunk" |
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"github.com/aler9/rtsp-simple-server/internal/rtmp/rawmessage" |
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) |
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const ( |
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// MsgAudioChunkStreamID is the chunk stream ID that is usually used to send MsgAudio{} |
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MsgAudioChunkStreamID = 6 |
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) |
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// MsgAudio is an audio message. |
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type MsgAudio struct { |
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ChunkStreamID byte |
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DTS time.Duration |
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MessageStreamID uint32 |
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Rate uint8 |
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Depth uint8 |
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Channels uint8 |
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AACType uint8 |
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Payload []byte |
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} |
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// Unmarshal implements Message. |
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func (m *MsgAudio) Unmarshal(raw *rawmessage.Message) error { |
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m.ChunkStreamID = raw.ChunkStreamID |
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m.DTS = raw.Timestamp |
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m.MessageStreamID = raw.MessageStreamID |
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if len(raw.Body) < 2 { |
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return fmt.Errorf("invalid body size") |
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} |
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codec := raw.Body[0] >> 4 |
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if codec != flvio.SOUND_AAC { |
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return fmt.Errorf("unsupported audio codec: %d", codec) |
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} |
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m.Rate = (raw.Body[0] >> 2) & 0x03 |
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m.Depth = (raw.Body[0] >> 1) & 0x01 |
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m.Channels = raw.Body[0] & 0x01 |
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m.AACType = raw.Body[1] |
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m.Payload = raw.Body[2:] |
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return nil |
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} |
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// Marshal implements Message. |
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func (m MsgAudio) Marshal() (*rawmessage.Message, error) { |
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body := make([]byte, 2+len(m.Payload)) |
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body[0] = flvio.SOUND_AAC<<4 | m.Rate<<2 | m.Depth<<1 | m.Channels |
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body[1] = m.AACType |
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copy(body[2:], m.Payload) |
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return &rawmessage.Message{ |
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ChunkStreamID: m.ChunkStreamID, |
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Timestamp: m.DTS, |
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Type: chunk.MessageTypeAudio, |
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MessageStreamID: m.MessageStreamID, |
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Body: body, |
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}, nil |
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}
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