Remove deprecated methods from AudioFrame.
Bug: webrtc:6548 Change-Id: I85bdba3acbef3216b3d836515dde10188ff0dbc6 Reviewed-on: https://webrtc-review.googlesource.com/54304 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22824}
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@ -18,7 +18,6 @@ rtc_source_set("audio_frame_api") {
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deps = [
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"../../:typedefs",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:rtc_base_approved",
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]
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}
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@ -13,7 +13,6 @@
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#include "api/audio/audio_frame.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/timeutils.h"
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namespace webrtc {
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@ -43,12 +42,12 @@ void AudioFrame::ResetWithoutMuting() {
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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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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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@ -119,62 +118,6 @@ void AudioFrame::Mute() {
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bool AudioFrame::muted() const { return muted_; }
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AudioFrame& AudioFrame::operator>>=(const int rhs) {
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RTC_CHECK_GT(num_channels_, 0);
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RTC_CHECK_LT(num_channels_, 3);
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if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
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if (muted_) return *this;
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for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
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data_[i] = static_cast<int16_t>(data_[i] >> rhs);
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}
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return *this;
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}
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AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
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// Sanity check
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RTC_CHECK_GT(num_channels_, 0);
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RTC_CHECK_LT(num_channels_, 3);
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if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
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if (num_channels_ != rhs.num_channels_) return *this;
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bool noPrevData = muted_;
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if (samples_per_channel_ != rhs.samples_per_channel_) {
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if (samples_per_channel_ == 0) {
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// special case we have no data to start with
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samples_per_channel_ = rhs.samples_per_channel_;
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noPrevData = true;
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} else {
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return *this;
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}
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}
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if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
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vad_activity_ = kVadActive;
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} else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
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vad_activity_ = kVadUnknown;
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}
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if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
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if (!rhs.muted()) {
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muted_ = false;
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if (noPrevData) {
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memcpy(data_, rhs.data(),
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sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
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} else {
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// IMPROVEMENT this can be done very fast in assembly
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for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
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int32_t wrap_guard =
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static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
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data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
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}
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}
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}
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return *this;
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}
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// static
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const int16_t* AudioFrame::empty_data() {
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static const int16_t kEmptyData[kMaxDataSizeSamples] = {0};
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@ -11,11 +11,7 @@
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#ifndef API_AUDIO_AUDIO_FRAME_H_
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#define API_AUDIO_AUDIO_FRAME_H_
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#include <stdint.h>
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#include <stdlib.h>
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/deprecation.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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@ -68,17 +64,6 @@ class AudioFrame {
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// ResetWithoutMuting() to skip this wasteful zeroing.
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void ResetWithoutMuting();
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// TODO(solenberg): Remove once downstream users of AudioFrame have updated.
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RTC_DEPRECATED
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void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
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size_t samples_per_channel, int sample_rate_hz,
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SpeechType speech_type, VADActivity vad_activity,
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size_t num_channels = 1) {
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RTC_UNUSED(id);
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UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
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speech_type, vad_activity, num_channels);
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}
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void UpdateFrame(uint32_t timestamp, const int16_t* data,
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size_t samples_per_channel, int sample_rate_hz,
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SpeechType speech_type, VADActivity vad_activity,
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@ -108,13 +93,6 @@ class AudioFrame {
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// Frame is muted by default.
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bool muted() const;
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// These methods are deprecated. Use the functions in
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// webrtc/audio/utility instead. These methods will exists for a
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// short period of time until webrtc clients have updated. See
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// webrtc:6548 for details.
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RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
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RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
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// RTP timestamp of the first sample in the AudioFrame.
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uint32_t timestamp_ = 0;
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// Time since the first frame in milliseconds.
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