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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
92 lines
3.6 KiB
C++
92 lines
3.6 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_device/fine_audio_buffer.h"
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#include <memory.h>
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#include <stdio.h>
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#include <algorithm>
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#include "modules/audio_device/audio_device_buffer.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
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int sample_rate,
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size_t capacity)
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: device_buffer_(device_buffer),
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sample_rate_(sample_rate),
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samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
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bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
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playout_buffer_(0, capacity),
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record_buffer_(0, capacity) {
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LOG(INFO) << "samples_per_10_ms_:" << samples_per_10_ms_;
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}
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FineAudioBuffer::~FineAudioBuffer() {}
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void FineAudioBuffer::ResetPlayout() {
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playout_buffer_.Clear();
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}
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void FineAudioBuffer::ResetRecord() {
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record_buffer_.Clear();
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}
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void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int8_t> audio_buffer) {
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// Ask WebRTC for new data in chunks of 10ms until we have enough to
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// fulfill the request. It is possible that the buffer already contains
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// enough samples from the last round.
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const size_t num_bytes = audio_buffer.size();
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while (playout_buffer_.size() < num_bytes) {
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// Get 10ms decoded audio from WebRTC.
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device_buffer_->RequestPlayoutData(samples_per_10_ms_);
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// Append |bytes_per_10_ms_| elements to the end of the buffer.
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const size_t bytes_written = playout_buffer_.AppendData(
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bytes_per_10_ms_, [&](rtc::ArrayView<int8_t> buf) {
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const size_t samples_per_channel =
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device_buffer_->GetPlayoutData(buf.data());
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// TODO(henrika): this class is only used on mobile devices and is
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// currently limited to mono. Modifications are needed for stereo.
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return sizeof(int16_t) * samples_per_channel;
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});
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RTC_DCHECK_EQ(bytes_per_10_ms_, bytes_written);
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}
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// Provide the requested number of bytes to the consumer.
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memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
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// Move remaining samples to start of buffer to prepare for next round.
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memmove(playout_buffer_.data(), playout_buffer_.data() + num_bytes,
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playout_buffer_.size() - num_bytes);
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playout_buffer_.SetSize(playout_buffer_.size() - num_bytes);
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}
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void FineAudioBuffer::DeliverRecordedData(
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rtc::ArrayView<const int8_t> audio_buffer,
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int playout_delay_ms,
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int record_delay_ms) {
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// Always append new data and grow the buffer if needed.
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record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
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// Consume samples from buffer in chunks of 10ms until there is not
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// enough data left. The number of remaining bytes in the cache is given by
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// the new size of the buffer.
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while (record_buffer_.size() >= bytes_per_10_ms_) {
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device_buffer_->SetRecordedBuffer(record_buffer_.data(),
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samples_per_10_ms_);
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device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
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device_buffer_->DeliverRecordedData();
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memmove(record_buffer_.data(), record_buffer_.data() + bytes_per_10_ms_,
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record_buffer_.size() - bytes_per_10_ms_);
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record_buffer_.SetSize(record_buffer_.size() - bytes_per_10_ms_);
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}
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}
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} // namespace webrtc
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