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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_device/fine_audio_buffer.h"
#include <memory.h>
#include <stdio.h>
#include <algorithm>
#include "modules/audio_device/audio_device_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
int sample_rate,
size_t capacity)
: device_buffer_(device_buffer),
sample_rate_(sample_rate),
samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
playout_buffer_(0, capacity),
record_buffer_(0, capacity) {
RTC_LOG(INFO) << "samples_per_10_ms_:" << samples_per_10_ms_;
}
FineAudioBuffer::~FineAudioBuffer() {}
void FineAudioBuffer::ResetPlayout() {
playout_buffer_.Clear();
}
void FineAudioBuffer::ResetRecord() {
record_buffer_.Clear();
}
void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int8_t> audio_buffer) {
// Ask WebRTC for new data in chunks of 10ms until we have enough to
// fulfill the request. It is possible that the buffer already contains
// enough samples from the last round.
const size_t num_bytes = audio_buffer.size();
while (playout_buffer_.size() < num_bytes) {
// Get 10ms decoded audio from WebRTC.
device_buffer_->RequestPlayoutData(samples_per_10_ms_);
// Append |bytes_per_10_ms_| elements to the end of the buffer.
const size_t bytes_written = playout_buffer_.AppendData(
bytes_per_10_ms_, [&](rtc::ArrayView<int8_t> buf) {
const size_t samples_per_channel =
device_buffer_->GetPlayoutData(buf.data());
// TODO(henrika): this class is only used on mobile devices and is
// currently limited to mono. Modifications are needed for stereo.
return sizeof(int16_t) * samples_per_channel;
});
RTC_DCHECK_EQ(bytes_per_10_ms_, bytes_written);
}
// Provide the requested number of bytes to the consumer.
memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
// Move remaining samples to start of buffer to prepare for next round.
memmove(playout_buffer_.data(), playout_buffer_.data() + num_bytes,
playout_buffer_.size() - num_bytes);
playout_buffer_.SetSize(playout_buffer_.size() - num_bytes);
}
void FineAudioBuffer::DeliverRecordedData(
rtc::ArrayView<const int8_t> audio_buffer,
int playout_delay_ms,
int record_delay_ms) {
// Always append new data and grow the buffer if needed.
record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
// Consume samples from buffer in chunks of 10ms until there is not
// enough data left. The number of remaining bytes in the cache is given by
// the new size of the buffer.
while (record_buffer_.size() >= bytes_per_10_ms_) {
device_buffer_->SetRecordedBuffer(record_buffer_.data(),
samples_per_10_ms_);
device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
device_buffer_->DeliverRecordedData();
memmove(record_buffer_.data(), record_buffer_.data() + bytes_per_10_ms_,
record_buffer_.size() - bytes_per_10_ms_);
record_buffer_.SetSize(record_buffer_.size() - bytes_per_10_ms_);
}
}
} // namespace webrtc
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