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Side by Side Diff: webrtc/modules/audio_device/fine_audio_buffer.cc

Issue 1401963002: Adds support for Bluetooth headsets to the iOS audio layer (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: nit Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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108 RTC_CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_); 108 RTC_CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_);
109 RTC_CHECK_EQ(static_cast<size_t>(-bytes_left), playout_cached_bytes_); 109 RTC_CHECK_EQ(static_cast<size_t>(-bytes_left), playout_cached_bytes_);
110 playout_cached_buffer_start_ = 0; 110 playout_cached_buffer_start_ = 0;
111 memcpy(playout_cache_buffer_.get(), cache_ptr, playout_cached_bytes_); 111 memcpy(playout_cache_buffer_.get(), cache_ptr, playout_cached_bytes_);
112 } 112 }
113 113
114 void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer, 114 void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
115 size_t size_in_bytes, 115 size_t size_in_bytes,
116 int playout_delay_ms, 116 int playout_delay_ms,
117 int record_delay_ms) { 117 int record_delay_ms) {
118 RTC_CHECK_EQ(size_in_bytes, desired_frame_size_bytes_);
119 // Check if the temporary buffer can store the incoming buffer. If not, 118 // Check if the temporary buffer can store the incoming buffer. If not,
120 // move the remaining (old) bytes to the beginning of the temporary buffer 119 // move the remaining (old) bytes to the beginning of the temporary buffer
121 // and start adding new samples after the old samples. 120 // and start adding new samples after the old samples.
122 if (record_write_pos_ + size_in_bytes > required_record_buffer_size_bytes_) { 121 if (record_write_pos_ + size_in_bytes > required_record_buffer_size_bytes_) {
123 if (record_cached_bytes_ > 0) { 122 if (record_cached_bytes_ > 0) {
124 memmove(record_cache_buffer_.get(), 123 memmove(record_cache_buffer_.get(),
125 record_cache_buffer_.get() + record_read_pos_, 124 record_cache_buffer_.get() + record_read_pos_,
126 record_cached_bytes_); 125 record_cached_bytes_);
127 } 126 }
128 record_write_pos_ = record_cached_bytes_; 127 record_write_pos_ = record_cached_bytes_;
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141 device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0); 140 device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
142 device_buffer_->DeliverRecordedData(); 141 device_buffer_->DeliverRecordedData();
143 // Read next chunk of 10ms data. 142 // Read next chunk of 10ms data.
144 record_read_pos_ += bytes_per_10_ms_; 143 record_read_pos_ += bytes_per_10_ms_;
145 // Reduce number of cached bytes with the consumed amount. 144 // Reduce number of cached bytes with the consumed amount.
146 record_cached_bytes_ -= bytes_per_10_ms_; 145 record_cached_bytes_ -= bytes_per_10_ms_;
147 } 146 }
148 } 147 }
149 148
150 } // namespace webrtc 149 } // namespace webrtc
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