mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
131 lines
3.5 KiB
C++
131 lines
3.5 KiB
C++
/*
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* Copyright (c) 2018 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 "api/audio/audio_frame.h"
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#include <string.h>
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#include "rtc_base/checks.h"
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#include "rtc_base/timeutils.h"
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namespace webrtc {
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AudioFrame::AudioFrame() {
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// Visual Studio doesn't like this in the class definition.
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static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
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}
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void AudioFrame::Reset() {
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ResetWithoutMuting();
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muted_ = true;
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}
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void AudioFrame::ResetWithoutMuting() {
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// TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
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// to an invalid value, or add a new member to indicate invalidity.
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timestamp_ = 0;
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elapsed_time_ms_ = -1;
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ntp_time_ms_ = -1;
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samples_per_channel_ = 0;
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sample_rate_hz_ = 0;
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num_channels_ = 0;
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speech_type_ = kUndefined;
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vad_activity_ = kVadUnknown;
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profile_timestamp_ms_ = 0;
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}
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void AudioFrame::UpdateFrame(uint32_t timestamp,
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const int16_t* data,
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size_t samples_per_channel,
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int sample_rate_hz,
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SpeechType speech_type,
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VADActivity vad_activity,
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size_t num_channels) {
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timestamp_ = timestamp;
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samples_per_channel_ = samples_per_channel;
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sample_rate_hz_ = sample_rate_hz;
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speech_type_ = speech_type;
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vad_activity_ = vad_activity;
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num_channels_ = num_channels;
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const size_t length = samples_per_channel * num_channels;
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RTC_CHECK_LE(length, kMaxDataSizeSamples);
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if (data != nullptr) {
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memcpy(data_, data, sizeof(int16_t) * length);
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muted_ = false;
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} else {
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muted_ = true;
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}
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}
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void AudioFrame::CopyFrom(const AudioFrame& src) {
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if (this == &src)
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return;
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timestamp_ = src.timestamp_;
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elapsed_time_ms_ = src.elapsed_time_ms_;
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ntp_time_ms_ = src.ntp_time_ms_;
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muted_ = src.muted();
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samples_per_channel_ = src.samples_per_channel_;
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sample_rate_hz_ = src.sample_rate_hz_;
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speech_type_ = src.speech_type_;
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vad_activity_ = src.vad_activity_;
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num_channels_ = src.num_channels_;
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const size_t length = samples_per_channel_ * num_channels_;
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RTC_CHECK_LE(length, kMaxDataSizeSamples);
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if (!src.muted()) {
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memcpy(data_, src.data(), sizeof(int16_t) * length);
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muted_ = false;
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}
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}
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void AudioFrame::UpdateProfileTimeStamp() {
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profile_timestamp_ms_ = rtc::TimeMillis();
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}
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int64_t AudioFrame::ElapsedProfileTimeMs() const {
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if (profile_timestamp_ms_ == 0) {
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// Profiling has not been activated.
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return -1;
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}
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return rtc::TimeSince(profile_timestamp_ms_);
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}
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const int16_t* AudioFrame::data() const {
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return muted_ ? empty_data() : data_;
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}
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// TODO(henrik.lundin) Can we skip zeroing the buffer?
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// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
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int16_t* AudioFrame::mutable_data() {
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if (muted_) {
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memset(data_, 0, kMaxDataSizeBytes);
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muted_ = false;
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}
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return data_;
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}
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void AudioFrame::Mute() {
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muted_ = true;
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}
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bool AudioFrame::muted() const {
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return muted_;
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}
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// static
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const int16_t* AudioFrame::empty_data() {
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static int16_t* null_data = new int16_t[kMaxDataSizeSamples]();
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return &null_data[0];
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}
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} // namespace webrtc
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