blob: 2f5f13aea2d710f98329573fd26cecc5fac3434b (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
|
// 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_AUDIO_RECEIVER_AUDIO_DECODER_H_
#define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
#include "base/callback.h"
#include "base/memory/ref_counted.h"
#include "media/cast/cast_config.h"
#include "media/cast/cast_thread.h"
#include "media/cast/rtp_common/rtp_defines.h"
namespace webrtc {
class AudioCodingModule;
}
namespace media {
namespace cast {
// Thread safe class.
// It should be called from the main cast thread; however that is not required.
class AudioDecoder : public base::RefCountedThreadSafe<AudioDecoder> {
public:
explicit AudioDecoder(scoped_refptr<CastThread> cast_thread,
const AudioReceiverConfig& audio_config);
virtual ~AudioDecoder();
// Extract a raw audio frame from the decoder.
// Set the number of desired 10ms blocks and frequency.
bool GetRawAudioFrame(int number_of_10ms_blocks,
int desired_frequency,
PcmAudioFrame* audio_frame,
uint32* rtp_timestamp);
// Insert an RTP packet to the decoder.
void IncomingParsedRtpPacket(const uint8* payload_data,
int payload_size,
const RtpCastHeader& rtp_header);
private:
// Can't use scoped_ptr due to protected constructor within webrtc.
webrtc::AudioCodingModule* audio_decoder_;
bool have_received_packets_;
scoped_refptr<CastThread> cast_thread_;
DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
};
} // namespace cast
} // namespace media
#endif // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
|
