Bump to WebRTC M120 release
Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone. We're continuing to carry iSAC even though it's gone upstream, but maybe we'll want to drop that soon.
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@ -11,8 +11,6 @@
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#include "api/audio/audio_frame.h"
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#include <string.h>
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#include <algorithm>
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#include <utility>
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#include "rtc_base/checks.h"
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#include "rtc_base/time_utils.h"
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@ -24,35 +22,13 @@ AudioFrame::AudioFrame() {
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static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
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}
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void swap(AudioFrame& a, AudioFrame& b) {
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using std::swap;
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swap(a.timestamp_, b.timestamp_);
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swap(a.elapsed_time_ms_, b.elapsed_time_ms_);
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swap(a.ntp_time_ms_, b.ntp_time_ms_);
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swap(a.samples_per_channel_, b.samples_per_channel_);
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swap(a.sample_rate_hz_, b.sample_rate_hz_);
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swap(a.num_channels_, b.num_channels_);
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swap(a.channel_layout_, b.channel_layout_);
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swap(a.speech_type_, b.speech_type_);
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swap(a.vad_activity_, b.vad_activity_);
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swap(a.profile_timestamp_ms_, b.profile_timestamp_ms_);
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swap(a.packet_infos_, b.packet_infos_);
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const size_t length_a = a.samples_per_channel_ * a.num_channels_;
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const size_t length_b = b.samples_per_channel_ * b.num_channels_;
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RTC_DCHECK_LE(length_a, AudioFrame::kMaxDataSizeSamples);
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RTC_DCHECK_LE(length_b, AudioFrame::kMaxDataSizeSamples);
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std::swap_ranges(a.data_, a.data_ + std::max(length_a, length_b), b.data_);
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swap(a.muted_, b.muted_);
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swap(a.absolute_capture_timestamp_ms_, b.absolute_capture_timestamp_ms_);
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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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// 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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