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/*
* 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/time_utils.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;
channel_layout_ = CHANNEL_LAYOUT_NONE;
speech_type_ = kUndefined;
vad_activity_ = kVadUnknown;
profile_timestamp_ms_ = 0;
packet_infos_ = RtpPacketInfos();
absolute_capture_timestamp_ms_ = absl::nullopt;
}
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;
channel_layout_ = GuessChannelLayout(num_channels);
if (channel_layout_ != CHANNEL_LAYOUT_UNSUPPORTED) {
RTC_DCHECK_EQ(num_channels, ChannelLayoutToChannelCount(channel_layout_));
}
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_;
packet_infos_ = src.packet_infos_;
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_;
channel_layout_ = src.channel_layout_;
absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms();
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 int16_t* null_data = new int16_t[kMaxDataSizeSamples]();
return &null_data[0];
}
} // namespace webrtc
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