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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 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/base/audio_splicer.h" | 5 #include "media/base/audio_splicer.h" |
6 | 6 |
7 #include <cstdlib> | 7 #include <cstdlib> |
8 #include <deque> | 8 #include <deque> |
9 | 9 |
10 #include "base/logging.h" | 10 #include "base/logging.h" |
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92 } | 92 } |
93 | 93 |
94 // Transfer all buffers into |output|. Returns false if AddInput() on the | 94 // Transfer all buffers into |output|. Returns false if AddInput() on the |
95 // |output| sanitizer fails for any buffer removed from |this|. | 95 // |output| sanitizer fails for any buffer removed from |this|. |
96 bool DrainInto(AudioStreamSanitizer* output); | 96 bool DrainInto(AudioStreamSanitizer* output); |
97 | 97 |
98 private: | 98 private: |
99 void AddOutputBuffer(const scoped_refptr<AudioBuffer>& buffer); | 99 void AddOutputBuffer(const scoped_refptr<AudioBuffer>& buffer); |
100 | 100 |
101 AudioTimestampHelper output_timestamp_helper_; | 101 AudioTimestampHelper output_timestamp_helper_; |
102 bool received_end_of_stream_; | 102 bool received_end_of_stream_ = false; |
103 | 103 |
104 typedef std::deque<scoped_refptr<AudioBuffer> > BufferQueue; | 104 typedef std::deque<scoped_refptr<AudioBuffer> > BufferQueue; |
105 BufferQueue output_buffers_; | 105 BufferQueue output_buffers_; |
106 | 106 |
107 scoped_refptr<MediaLog> media_log_; | 107 scoped_refptr<MediaLog> media_log_; |
108 | 108 |
109 // To prevent log spam, counts the number of audio gap or overlaps warned in | 109 // To prevent log spam, counts the number of audio gap or overlaps warned in |
110 // logs. | 110 // logs. |
111 int num_warning_logs_; | 111 int num_warning_logs_ = 0; |
112 | 112 |
113 DISALLOW_ASSIGN(AudioStreamSanitizer); | 113 DISALLOW_ASSIGN(AudioStreamSanitizer); |
114 }; | 114 }; |
115 | 115 |
116 AudioStreamSanitizer::AudioStreamSanitizer( | 116 AudioStreamSanitizer::AudioStreamSanitizer( |
117 int samples_per_second, | 117 int samples_per_second, |
118 const scoped_refptr<MediaLog>& media_log) | 118 const scoped_refptr<MediaLog>& media_log) |
119 : output_timestamp_helper_(samples_per_second), | 119 : output_timestamp_helper_(samples_per_second), media_log_(media_log) {} |
120 received_end_of_stream_(false), | |
121 media_log_(media_log), | |
122 num_warning_logs_(0) { | |
123 } | |
124 | 120 |
125 AudioStreamSanitizer::~AudioStreamSanitizer() {} | 121 AudioStreamSanitizer::~AudioStreamSanitizer() {} |
126 | 122 |
127 void AudioStreamSanitizer::Reset() { | 123 void AudioStreamSanitizer::Reset() { |
128 ResetTimestampState(0, kNoTimestamp()); | 124 ResetTimestampState(0, kNoTimestamp()); |
129 } | 125 } |
130 | 126 |
131 void AudioStreamSanitizer::ResetTimestampState(int64 frame_count, | 127 void AudioStreamSanitizer::ResetTimestampState(int64 frame_count, |
132 base::TimeDelta base_timestamp) { | 128 base::TimeDelta base_timestamp) { |
133 output_buffers_.clear(); | 129 output_buffers_.clear(); |
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550 AccurateTrimStart(frames_to_trim, remainder, output_ts_helper); | 546 AccurateTrimStart(frames_to_trim, remainder, output_ts_helper); |
551 CHECK(output_sanitizer_->AddInput(remainder)); | 547 CHECK(output_sanitizer_->AddInput(remainder)); |
552 } | 548 } |
553 | 549 |
554 // Transfer all remaining buffers out and reset once empty. | 550 // Transfer all remaining buffers out and reset once empty. |
555 CHECK(post_splice_sanitizer_->DrainInto(output_sanitizer_.get())); | 551 CHECK(post_splice_sanitizer_->DrainInto(output_sanitizer_.get())); |
556 post_splice_sanitizer_->Reset(); | 552 post_splice_sanitizer_->Reset(); |
557 } | 553 } |
558 | 554 |
559 } // namespace media | 555 } // namespace media |
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