| /* |
| * 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; |
| 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 int16_t* null_data = new int16_t[kMaxDataSizeSamples](); |
| return &null_data[0]; |
| } |
| |
| } // namespace webrtc |