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Now using rtc::Buffer in FineAudioBuffer

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BUG=b/35589717

Review-Url: https://codereview.webrtc.org/2706923006
Cr-Commit-Position: refs/heads/master@{#16793}
This commit is contained in:
henrika 2017-02-23 02:44:55 -08:00 committed by Commit bot
parent 02f994b4e9
commit f166e1bcab
2 changed files with 14 additions and 47 deletions
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51
webrtc/modules/audio_device/fine_audio_buffer.cc
View File

@ -29,17 +29,8 @@ FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
playout_cached_buffer_start_(0),
playout_cached_bytes_(0),
// Allocate extra space on the recording side to reduce the number of
// memmove() calls.
required_record_buffer_size_bytes_(
5 * (desired_frame_size_bytes + bytes_per_10_ms_)),
record_cached_bytes_(0),
record_read_pos_(0),
record_write_pos_(0) {
playout_cached_bytes_(0) {
playout_cache_buffer_.reset(new int8_t[bytes_per_10_ms_]);
record_cache_buffer_.reset(new int8_t[required_record_buffer_size_bytes_]);
memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
}
FineAudioBuffer::~FineAudioBuffer() {}
@ -58,10 +49,7 @@ void FineAudioBuffer::ResetPlayout() {
}
void FineAudioBuffer::ResetRecord() {
record_cached_bytes_ = 0;
record_read_pos_ = 0;
record_write_pos_ = 0;
memset(record_cache_buffer_.get(), 0, required_record_buffer_size_bytes_);
record_buffer_.Clear();
}
void FineAudioBuffer::GetPlayoutData(int8_t* buffer) {
@ -115,34 +103,19 @@ void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
size_t size_in_bytes,
int playout_delay_ms,
int record_delay_ms) {
// Check if the temporary buffer can store the incoming buffer. If not,
// move the remaining (old) bytes to the beginning of the temporary buffer
// and start adding new samples after the old samples.
if (record_write_pos_ + size_in_bytes > required_record_buffer_size_bytes_) {
if (record_cached_bytes_ > 0) {
memmove(record_cache_buffer_.get(),
record_cache_buffer_.get() + record_read_pos_,
record_cached_bytes_);
}
record_write_pos_ = record_cached_bytes_;
record_read_pos_ = 0;
}
// Add recorded samples to a temporary buffer.
memcpy(record_cache_buffer_.get() + record_write_pos_, buffer, size_in_bytes);
record_write_pos_ += size_in_bytes;
record_cached_bytes_ += size_in_bytes;
// Consume samples in temporary buffer in chunks of 10ms until there is not
// Always append new data and grow the buffer if needed.
record_buffer_.AppendData(buffer, size_in_bytes);
// 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
// |record_cached_bytes_| after this while loop is done.
while (record_cached_bytes_ >= bytes_per_10_ms_) {
device_buffer_->SetRecordedBuffer(
record_cache_buffer_.get() + record_read_pos_, samples_per_10_ms_);
// 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();
// Read next chunk of 10ms data.
record_read_pos_ += bytes_per_10_ms_;
// Reduce number of cached bytes with the consumed amount.
record_cached_bytes_ -= bytes_per_10_ms_;
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_);
}
}

10
webrtc/modules/audio_device/fine_audio_buffer.h
View File

@ -13,6 +13,7 @@
#include <memory>
#include "webrtc/base/buffer.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -94,14 +95,7 @@ class FineAudioBuffer {
size_t playout_cached_bytes_;
// Storage for input samples that are about to be delivered to the WebRTC
// ADB or remains from the last successful delivery of a 10ms audio buffer.
std::unique_ptr<int8_t[]> record_cache_buffer_;
// Required (max) size in bytes of the |record_cache_buffer_|.
const size_t required_record_buffer_size_bytes_;
// Number of bytes in input (contains recorded samples) cache.
size_t record_cached_bytes_;
// Read and write pointers used in the buffering scheme on the recording side.
size_t record_read_pos_;
size_t record_write_pos_;
rtc::BufferT<int8_t> record_buffer_;
};
} // namespace webrtc