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Side by Side Diff: media/cast/sender/audio_sender.cc

Issue 560223002: [Cast] Limit frames in flight by duration, and not by number of frames. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Account for faster input than configured max FPS. Created 6 years, 3 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/sender/audio_sender.h" 5 #include "media/cast/sender/audio_sender.h"
6 6
7 #include <algorithm>
8
7 #include "base/bind.h" 9 #include "base/bind.h"
8 #include "base/logging.h" 10 #include "base/logging.h"
9 #include "base/message_loop/message_loop.h" 11 #include "base/message_loop/message_loop.h"
10 #include "media/cast/cast_defines.h" 12 #include "media/cast/cast_defines.h"
11 #include "media/cast/net/cast_transport_config.h" 13 #include "media/cast/net/cast_transport_config.h"
12 #include "media/cast/sender/audio_encoder.h" 14 #include "media/cast/sender/audio_encoder.h"
13 15
14 namespace media { 16 namespace media {
15 namespace cast { 17 namespace cast {
16 namespace { 18 namespace {
(...skipping 37 matching lines...) Expand 10 before | Expand all | Expand 10 after
54 cast_initialization_status_ = audio_encoder_->InitializationResult(); 56 cast_initialization_status_ = audio_encoder_->InitializationResult();
55 } else { 57 } else {
56 NOTREACHED(); // No support for external audio encoding. 58 NOTREACHED(); // No support for external audio encoding.
57 cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED; 59 cast_initialization_status_ = STATUS_AUDIO_UNINITIALIZED;
58 } 60 }
59 61
60 media::cast::CastTransportRtpConfig transport_config; 62 media::cast::CastTransportRtpConfig transport_config;
61 transport_config.ssrc = audio_config.ssrc; 63 transport_config.ssrc = audio_config.ssrc;
62 transport_config.feedback_ssrc = audio_config.incoming_feedback_ssrc; 64 transport_config.feedback_ssrc = audio_config.incoming_feedback_ssrc;
63 transport_config.rtp_payload_type = audio_config.rtp_payload_type; 65 transport_config.rtp_payload_type = audio_config.rtp_payload_type;
64 // TODO(miu): AudioSender needs to be like VideoSender in providing an upper
65 // limit on the number of in-flight frames.
66 transport_config.stored_frames = max_unacked_frames_;
67 transport_config.aes_key = audio_config.aes_key; 66 transport_config.aes_key = audio_config.aes_key;
68 transport_config.aes_iv_mask = audio_config.aes_iv_mask; 67 transport_config.aes_iv_mask = audio_config.aes_iv_mask;
69 68
70 transport_sender->InitializeAudio( 69 transport_sender->InitializeAudio(
71 transport_config, 70 transport_config,
72 base::Bind(&AudioSender::OnReceivedCastFeedback, 71 base::Bind(&AudioSender::OnReceivedCastFeedback,
73 weak_factory_.GetWeakPtr()), 72 weak_factory_.GetWeakPtr()),
74 base::Bind(&AudioSender::OnMeasuredRoundTripTime, 73 base::Bind(&AudioSender::OnMeasuredRoundTripTime,
75 weak_factory_.GetWeakPtr())); 74 weak_factory_.GetWeakPtr()));
76 } 75 }
77 76
78 AudioSender::~AudioSender() {} 77 AudioSender::~AudioSender() {}
79 78
80 void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus, 79 void AudioSender::InsertAudio(scoped_ptr<AudioBus> audio_bus,
81 const base::TimeTicks& recorded_time) { 80 const base::TimeTicks& recorded_time) {
82 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); 81 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
83 if (cast_initialization_status_ != STATUS_AUDIO_INITIALIZED) { 82 if (cast_initialization_status_ != STATUS_AUDIO_INITIALIZED) {
84 NOTREACHED(); 83 NOTREACHED();
85 return; 84 return;
86 } 85 }
87 DCHECK(audio_encoder_.get()) << "Invalid internal state"; 86 DCHECK(audio_encoder_.get()) << "Invalid internal state";
88 87
89 // TODO(miu): An |audio_bus| that represents more duration than a single 88 // Check that enqueuing the samples in |audio_bus| won't cause more frames to
90 // frame's duration can defeat our logic here, causing too much data to become 89 // become in-flight than the system's design limit.
91 // enqueued. This will be addressed in a soon-upcoming change. 90 const int count_unacked_frames = GetUnackedFrameCount();
92 if (ShouldDropNextFrame(recorded_time)) { 91 const int64 samples_unacked =
93 VLOG(1) << "Dropping frame due to too many frames currently in-flight."; 92 count_unacked_frames * audio_encoder_->GetSamplesPerFrame();
93 const int64 samples_would_be_in_flight =
94 samples_unacked + samples_in_encoder_ + audio_bus->frames();
95 const int frames_would_be_in_flight =
96 samples_would_be_in_flight / audio_encoder_->GetSamplesPerFrame();
97 if (frames_would_be_in_flight > kMaxUnackedFrames) {
98 VLOG(1) << "Dropping audio: Too many frames would be in-flight.";
99 return;
100 }
101
102 // Check that enqueuing the samples in |audio_bus| won't exceed the allowed
103 // in-flight media duration.
104 const int64 max_samples_in_flight =
105 TimeDeltaToRtpDelta(GetAllowedInFlightMediaDuration(), rtp_timebase());
106 VLOG(2) << "Audio samples in-flight: "
107 << samples_unacked << " unacked + "
108 << samples_in_encoder_ << " in encoder + "
109 << audio_bus->frames() << " additional would be "
110 << (max_samples_in_flight > 0 ?
111 100 * samples_would_be_in_flight / max_samples_in_flight :
112 kint64max) << "% of allowed in-flight.";
113 if (samples_would_be_in_flight > max_samples_in_flight) {
114 VLOG(1) << "Dropping audio: Too long an audio duration would be in-flight.";
94 return; 115 return;
95 } 116 }
96 117
97 samples_in_encoder_ += audio_bus->frames(); 118 samples_in_encoder_ += audio_bus->frames();
98
99 audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time); 119 audio_encoder_->InsertAudio(audio_bus.Pass(), recorded_time);
100 } 120 }
101 121
102 int AudioSender::GetNumberOfFramesInEncoder() const {
103 // Note: It's possible for a partial frame to be in the encoder, but returning
104 // the floor() is good enough for the "design limit" check in FrameSender.
105 return samples_in_encoder_ / audio_encoder_->GetSamplesPerFrame();
106 }
107
108 void AudioSender::OnAck(uint32 frame_id) { 122 void AudioSender::OnAck(uint32 frame_id) {
109 } 123 }
110 124
111 void AudioSender::OnEncodedAudioFrame( 125 void AudioSender::OnEncodedAudioFrame(
112 int encoder_bitrate, 126 int encoder_bitrate,
113 scoped_ptr<EncodedFrame> encoded_frame, 127 scoped_ptr<EncodedFrame> encoded_frame,
114 int samples_skipped) { 128 int samples_skipped) {
115 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN)); 129 DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
116 130
117 samples_in_encoder_ -= audio_encoder_->GetSamplesPerFrame() + samples_skipped; 131 samples_in_encoder_ -= audio_encoder_->GetSamplesPerFrame() + samples_skipped;
118 DCHECK_GE(samples_in_encoder_, 0); 132 DCHECK_GE(samples_in_encoder_, 0);
119 133
120 SendEncodedFrame(encoder_bitrate, encoded_frame.Pass()); 134 SendEncodedFrame(encoder_bitrate, encoded_frame.Pass());
121 } 135 }
122 136
123 } // namespace cast 137 } // namespace cast
124 } // namespace media 138 } // namespace media
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