7ce3091d8a
Commit bbf21a3fd617abb37567a86e65f8ba18b8d64eb2 ("Remove dependencies on
modules:module_api from AudioProcessing") causes the build to fail with
libstdc++ due to several files using memcpy(3) or memset(3) while relying on
string.h being included implicitly by other headers.
Bug: webrtc:9139
Change-Id: Ib73284962f8694d8bed0551968265bfd13cab967
Reviewed-on: https://webrtc-review.googlesource.com/70180
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#22895}
129 lines
3.5 KiB
C++
129 lines
3.5 KiB
C++
/*
|
|
* Copyright (c) 2018 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 "api/audio/audio_frame.h"
|
|
|
|
#include <string.h>
|
|
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/timeutils.h"
|
|
|
|
namespace webrtc {
|
|
|
|
AudioFrame::AudioFrame() {
|
|
// Visual Studio doesn't like this in the class definition.
|
|
static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
|
|
}
|
|
|
|
void AudioFrame::Reset() {
|
|
ResetWithoutMuting();
|
|
muted_ = true;
|
|
}
|
|
|
|
void AudioFrame::ResetWithoutMuting() {
|
|
// TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
|
|
// to an invalid value, or add a new member to indicate invalidity.
|
|
timestamp_ = 0;
|
|
elapsed_time_ms_ = -1;
|
|
ntp_time_ms_ = -1;
|
|
samples_per_channel_ = 0;
|
|
sample_rate_hz_ = 0;
|
|
num_channels_ = 0;
|
|
speech_type_ = kUndefined;
|
|
vad_activity_ = kVadUnknown;
|
|
profile_timestamp_ms_ = 0;
|
|
}
|
|
|
|
void AudioFrame::UpdateFrame(uint32_t timestamp,
|
|
const int16_t* data,
|
|
size_t samples_per_channel,
|
|
int sample_rate_hz,
|
|
SpeechType speech_type,
|
|
VADActivity vad_activity,
|
|
size_t num_channels) {
|
|
timestamp_ = timestamp;
|
|
samples_per_channel_ = samples_per_channel;
|
|
sample_rate_hz_ = sample_rate_hz;
|
|
speech_type_ = speech_type;
|
|
vad_activity_ = vad_activity;
|
|
num_channels_ = num_channels;
|
|
|
|
const size_t length = samples_per_channel * num_channels;
|
|
RTC_CHECK_LE(length, kMaxDataSizeSamples);
|
|
if (data != nullptr) {
|
|
memcpy(data_, data, sizeof(int16_t) * length);
|
|
muted_ = false;
|
|
} else {
|
|
muted_ = true;
|
|
}
|
|
}
|
|
|
|
void AudioFrame::CopyFrom(const AudioFrame& src) {
|
|
if (this == &src) return;
|
|
|
|
timestamp_ = src.timestamp_;
|
|
elapsed_time_ms_ = src.elapsed_time_ms_;
|
|
ntp_time_ms_ = src.ntp_time_ms_;
|
|
muted_ = src.muted();
|
|
samples_per_channel_ = src.samples_per_channel_;
|
|
sample_rate_hz_ = src.sample_rate_hz_;
|
|
speech_type_ = src.speech_type_;
|
|
vad_activity_ = src.vad_activity_;
|
|
num_channels_ = src.num_channels_;
|
|
|
|
const size_t length = samples_per_channel_ * num_channels_;
|
|
RTC_CHECK_LE(length, kMaxDataSizeSamples);
|
|
if (!src.muted()) {
|
|
memcpy(data_, src.data(), sizeof(int16_t) * length);
|
|
muted_ = false;
|
|
}
|
|
}
|
|
|
|
void AudioFrame::UpdateProfileTimeStamp() {
|
|
profile_timestamp_ms_ = rtc::TimeMillis();
|
|
}
|
|
|
|
int64_t AudioFrame::ElapsedProfileTimeMs() const {
|
|
if (profile_timestamp_ms_ == 0) {
|
|
// Profiling has not been activated.
|
|
return -1;
|
|
}
|
|
return rtc::TimeSince(profile_timestamp_ms_);
|
|
}
|
|
|
|
const int16_t* AudioFrame::data() const {
|
|
return muted_ ? empty_data() : data_;
|
|
}
|
|
|
|
// TODO(henrik.lundin) Can we skip zeroing the buffer?
|
|
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
|
|
int16_t* AudioFrame::mutable_data() {
|
|
if (muted_) {
|
|
memset(data_, 0, kMaxDataSizeBytes);
|
|
muted_ = false;
|
|
}
|
|
return data_;
|
|
}
|
|
|
|
void AudioFrame::Mute() {
|
|
muted_ = true;
|
|
}
|
|
|
|
bool AudioFrame::muted() const { return muted_; }
|
|
|
|
// static
|
|
const int16_t* AudioFrame::empty_data() {
|
|
static const int16_t kEmptyData[kMaxDataSizeSamples] = {0};
|
|
static_assert(sizeof(kEmptyData) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
|
|
return kEmptyData;
|
|
}
|
|
|
|
} // namespace webrtc
|