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Side by Side Diff: media/cast/transport/transport_audio_sender.cc

Issue 252923007: Cast: Fix two video freezing problems (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: mac fix Created 6 years, 7 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 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 #include "media/cast/transport/transport_audio_sender.h" 5 #include "media/cast/transport/transport_audio_sender.h"
6 6
7 #include "base/bind.h" 7 #include "base/bind.h"
8 #include "base/logging.h" 8 #include "base/logging.h"
9 #include "base/message_loop/message_loop.h" 9 #include "base/message_loop/message_loop.h"
10 #include "media/cast/transport/rtp_sender/rtp_sender.h" 10 #include "media/cast/transport/rtp_sender/rtp_sender.h"
11 11
12 namespace media { 12 namespace media {
13 namespace cast { 13 namespace cast {
14 namespace transport { 14 namespace transport {
15 15
16 TransportAudioSender::TransportAudioSender( 16 TransportAudioSender::TransportAudioSender(
17 const CastTransportAudioConfig& config, 17 const CastTransportAudioConfig& config,
18 base::TickClock* clock, 18 base::TickClock* clock,
19 const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner, 19 const scoped_refptr<base::SingleThreadTaskRunner>& transport_task_runner,
20 PacedSender* const paced_packet_sender) 20 PacedSender* const paced_packet_sender)
21 : rtp_sender_(clock, transport_task_runner, paced_packet_sender), 21 : rtp_sender_(clock, transport_task_runner, paced_packet_sender),
22 encryptor_() { 22 encryptor_() {
23 rtp_sender_.InitializeAudio(config); 23 initialized_ = rtp_sender_.InitializeAudio(config) &&
24 initialized_ = 24 encryptor_.Initialize(config.rtp.config.aes_key,
25 encryptor_.Initialize(config.base.aes_key, config.base.aes_iv_mask); 25 config.rtp.config.aes_iv_mask);
26 } 26 }
27 27
28 TransportAudioSender::~TransportAudioSender() {} 28 TransportAudioSender::~TransportAudioSender() {}
29 29
30 void TransportAudioSender::InsertCodedAudioFrame( 30 void TransportAudioSender::InsertCodedAudioFrame(
31 const EncodedAudioFrame* audio_frame, 31 const EncodedAudioFrame* audio_frame,
32 const base::TimeTicks& recorded_time) { 32 const base::TimeTicks& recorded_time) {
33 if (!initialized_) {
34 return;
35 }
33 if (encryptor_.initialized()) { 36 if (encryptor_.initialized()) {
34 EncodedAudioFrame encrypted_frame; 37 EncodedAudioFrame encrypted_frame;
35 if (!EncryptAudioFrame(*audio_frame, &encrypted_frame)) { 38 if (!EncryptAudioFrame(*audio_frame, &encrypted_frame)) {
36 return; 39 return;
37 } 40 }
38 rtp_sender_.IncomingEncodedAudioFrame(&encrypted_frame, recorded_time); 41 rtp_sender_.IncomingEncodedAudioFrame(&encrypted_frame, recorded_time);
39 } else { 42 } else {
40 rtp_sender_.IncomingEncodedAudioFrame(audio_frame, recorded_time); 43 rtp_sender_.IncomingEncodedAudioFrame(audio_frame, recorded_time);
41 } 44 }
42 } 45 }
43 46
44 bool TransportAudioSender::EncryptAudioFrame( 47 bool TransportAudioSender::EncryptAudioFrame(
45 const EncodedAudioFrame& audio_frame, 48 const EncodedAudioFrame& audio_frame,
46 EncodedAudioFrame* encrypted_frame) { 49 EncodedAudioFrame* encrypted_frame) {
50 if (!initialized_) {
51 return false;
52 }
47 if (!encryptor_.Encrypt( 53 if (!encryptor_.Encrypt(
48 audio_frame.frame_id, audio_frame.data, &encrypted_frame->data)) 54 audio_frame.frame_id, audio_frame.data, &encrypted_frame->data))
49 return false; 55 return false;
50 56
51 encrypted_frame->codec = audio_frame.codec; 57 encrypted_frame->codec = audio_frame.codec;
52 encrypted_frame->frame_id = audio_frame.frame_id; 58 encrypted_frame->frame_id = audio_frame.frame_id;
53 encrypted_frame->rtp_timestamp = audio_frame.rtp_timestamp; 59 encrypted_frame->rtp_timestamp = audio_frame.rtp_timestamp;
54 return true; 60 return true;
55 } 61 }
56 62
57 void TransportAudioSender::ResendPackets( 63 void TransportAudioSender::ResendPackets(
58 const MissingFramesAndPacketsMap& missing_frames_and_packets) { 64 const MissingFramesAndPacketsMap& missing_frames_and_packets) {
65 if (!initialized_) {
66 return;
67 }
59 rtp_sender_.ResendPackets(missing_frames_and_packets); 68 rtp_sender_.ResendPackets(missing_frames_and_packets);
60 } 69 }
61 70
62 void TransportAudioSender::SubscribeAudioRtpStatsCallback( 71 void TransportAudioSender::SubscribeAudioRtpStatsCallback(
63 const CastTransportRtpStatistics& callback) { 72 const CastTransportRtpStatistics& callback) {
64 rtp_sender_.SubscribeRtpStatsCallback(callback); 73 rtp_sender_.SubscribeRtpStatsCallback(callback);
65 } 74 }
66 75
67 } // namespace transport 76 } // namespace transport
68 } // namespace cast 77 } // namespace cast
69 } // namespace media 78 } // namespace media
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