2250b05778
Two new classes are added to WebRTC from Chrome: ChannelMixer and ChannelMixingMatrix but they are not yet utilized in the audio path for WebRTC. The idea is to utilize these new classes when adding support for multi- channel encoding/decoding in WebRTC/Chrome. Adds support for a new enumerator call webrtc::ChannelLayout and some helper methods which maps between channel layout and number of channels. These parts are also copied from Chrome. Minor (cosmetic) changes are also done on the AudioFrame to prepare for upcoming work. Bug: webrtc:10783 Change-Id: I6cd7a13a3bc1c8bbfa19bc974c7a011d22d19197 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141674 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28482}
139 lines
3.8 KiB
C++
139 lines
3.8 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/time_utils.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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channel_layout_ = CHANNEL_LAYOUT_NONE;
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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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packet_infos_ = RtpPacketInfos();
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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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channel_layout_ = GuessChannelLayout(num_channels);
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if (channel_layout_ != CHANNEL_LAYOUT_UNSUPPORTED) {
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RTC_DCHECK_EQ(num_channels, ChannelLayoutToChannelCount(channel_layout_));
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}
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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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packet_infos_ = src.packet_infos_;
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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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channel_layout_ = src.channel_layout_;
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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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