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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio/audio_frame.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/audio/audio_frame.cc')
-rw-r--r-- | third_party/libwebrtc/api/audio/audio_frame.cc | 140 |
1 files changed, 140 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio/audio_frame.cc b/third_party/libwebrtc/api/audio/audio_frame.cc new file mode 100644 index 0000000000..3e12006386 --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_frame.cc @@ -0,0 +1,140 @@ +/* + * 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 |