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Side by Side Diff: media/cast/audio_receiver/audio_decoder.h

Issue 214273003: [Cast] Remove AudioDecoder's dependency on WebRTC, and refactor/clean-up AudioReceiver. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: One moar Windows compile fix. Created 6 years, 9 months ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_ 5 #ifndef MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
6 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_ 6 #define MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
7 7
8 #include "base/callback.h" 8 #include "base/callback.h"
9 #include "base/synchronization/lock.h" 9 #include "base/memory/ref_counted.h"
10 #include "media/base/audio_bus.h"
10 #include "media/cast/cast_config.h" 11 #include "media/cast/cast_config.h"
11 #include "media/cast/cast_environment.h" 12 #include "media/cast/cast_environment.h"
12 #include "media/cast/framer/cast_message_builder.h" 13 #include "media/cast/transport/cast_transport_config.h"
13 #include "media/cast/framer/frame_id_map.h"
14 #include "media/cast/rtp_receiver/rtp_receiver_defines.h"
15
16 namespace webrtc {
17 class AudioCodingModule;
18 }
19 14
20 namespace media { 15 namespace media {
21 namespace cast { 16 namespace cast {
22 17
23 typedef std::map<uint32, uint32> FrameIdRtpTimestampMap;
24
25 // Thread safe class.
26 class AudioDecoder { 18 class AudioDecoder {
27 public: 19 public:
28 AudioDecoder(scoped_refptr<CastEnvironment> cast_environment, 20 // Callback passed to DecodeFrame, to deliver decoded audio data from the
29 const AudioReceiverConfig& audio_config, 21 // decoder. The number of samples in |audio_bus| may vary, and |audio_bus|
30 RtpPayloadFeedback* incoming_payload_feedback); 22 // can be NULL when errors occur. |is_continuous| is normally true, but will
23 // be false if the decoder has detected a frame skip since the last decode
24 // operation; and the client should take steps to smooth audio discontinuities
25 // in this case.
26 typedef base::Callback<void(scoped_ptr<AudioBus> audio_bus,
27 bool is_continuous)> DecodeFrameCallback;
28
29 AudioDecoder(const scoped_refptr<CastEnvironment>& cast_environment,
30 const AudioReceiverConfig& audio_config);
31 virtual ~AudioDecoder(); 31 virtual ~AudioDecoder();
32 32
33 // Extract a raw audio frame from the decoder. 33 // Returns STATUS_AUDIO_INITIALIZED if the decoder was successfully
34 // Set the number of desired 10ms blocks and frequency. 34 // constructed from the given AudioReceiverConfig. If this method returns any
35 // Should be called from the cast audio decoder thread; however that is not 35 // other value, calls to DecodeFrame() will not succeed.
36 // required. 36 CastInitializationStatus InitializationResult() const;
37 bool GetRawAudioFrame(int number_of_10ms_blocks,
38 int desired_frequency,
39 PcmAudioFrame* audio_frame,
40 uint32* rtp_timestamp);
41 37
42 // Insert an RTP packet to the decoder. 38 // Decode the payload in |encoded_frame| asynchronously. |callback| will be
43 // Should be called from the main cast thread; however that is not required. 39 // invoked on the CastEnvironment::MAIN thread with the result.
44 void IncomingParsedRtpPacket(const uint8* payload_data, 40 //
45 size_t payload_size, 41 // In the normal case, |encoded_frame->frame_id| will be
46 const RtpCastHeader& rtp_header); 42 // monotonically-increasing by 1 for each successive call to this method.
47 43 // When it is not, the decoder will assume one or more frames have been
48 bool TimeToSendNextCastMessage(base::TimeTicks* time_to_send); 44 // dropped (e.g., due to packet loss), and will perform recovery actions.
49 void SendCastMessage(); 45 void DecodeFrame(scoped_ptr<transport::EncodedAudioFrame> encoded_frame,
46 const DecodeFrameCallback& callback);
50 47
51 private: 48 private:
52 scoped_refptr<CastEnvironment> cast_environment_; 49 class ImplBase;
50 class OpusImpl;
51 class Pcm16Impl;
53 52
54 // The webrtc AudioCodingModule is thread safe. 53 const scoped_refptr<CastEnvironment> cast_environment_;
55 scoped_ptr<webrtc::AudioCodingModule> audio_decoder_; 54 scoped_refptr<ImplBase> impl_;
56
57 FrameIdMap frame_id_map_;
58 CastMessageBuilder cast_message_builder_;
59
60 base::Lock lock_;
61 bool have_received_packets_;
62 FrameIdRtpTimestampMap frame_id_rtp_timestamp_map_;
63 uint32 last_played_out_timestamp_;
64 55
65 DISALLOW_COPY_AND_ASSIGN(AudioDecoder); 56 DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
66 }; 57 };
67 58
68 } // namespace cast 59 } // namespace cast
69 } // namespace media 60 } // namespace media
70 61
71 #endif // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_ 62 #endif // MEDIA_CAST_AUDIO_RECEIVER_AUDIO_DECODER_H_
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