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}diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn
index f3bca7c..5558ab1 100644
--- a/api/audio/BUILD.gn
+++ b/api/audio/BUILD.gn
@@ -18,7 +18,6 @@
deps = [
"../../:typedefs",
"../../rtc_base:checks",
- "../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
]
}
diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index 108a523..864188d 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -13,7 +13,6 @@
#include "api/audio/audio_frame.h"
#include "rtc_base/checks.h"
-#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/timeutils.h"
namespace webrtc {
@@ -43,12 +42,12 @@
}
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) {
+ 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;
@@ -119,62 +118,6 @@
bool AudioFrame::muted() const { return muted_; }
-AudioFrame& AudioFrame::operator>>=(const int rhs) {
- RTC_CHECK_GT(num_channels_, 0);
- RTC_CHECK_LT(num_channels_, 3);
- if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
- if (muted_) return *this;
-
- for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
- data_[i] = static_cast<int16_t>(data_[i] >> rhs);
- }
- return *this;
-}
-
-AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
- // Sanity check
- RTC_CHECK_GT(num_channels_, 0);
- RTC_CHECK_LT(num_channels_, 3);
- if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
- if (num_channels_ != rhs.num_channels_) return *this;
-
- bool noPrevData = muted_;
- if (samples_per_channel_ != rhs.samples_per_channel_) {
- if (samples_per_channel_ == 0) {
- // special case we have no data to start with
- samples_per_channel_ = rhs.samples_per_channel_;
- noPrevData = true;
- } else {
- return *this;
- }
- }
-
- if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
- vad_activity_ = kVadActive;
- } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
- vad_activity_ = kVadUnknown;
- }
-
- if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
-
- if (!rhs.muted()) {
- muted_ = false;
- if (noPrevData) {
- memcpy(data_, rhs.data(),
- sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
- } else {
- // IMPROVEMENT this can be done very fast in assembly
- for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
- int32_t wrap_guard =
- static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
- data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
- }
- }
- }
-
- return *this;
-}
-
// static
const int16_t* AudioFrame::empty_data() {
static const int16_t kEmptyData[kMaxDataSizeSamples] = {0};
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index 5cb2019..d69f607 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -11,11 +11,7 @@
#ifndef API_AUDIO_AUDIO_FRAME_H_
#define API_AUDIO_AUDIO_FRAME_H_
-#include <stdint.h>
-#include <stdlib.h>
-
#include "rtc_base/constructormagic.h"
-#include "rtc_base/deprecation.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@@ -68,17 +64,6 @@
// ResetWithoutMuting() to skip this wasteful zeroing.
void ResetWithoutMuting();
- // TODO(solenberg): Remove once downstream users of AudioFrame have updated.
- RTC_DEPRECATED
- void UpdateFrame(int id, 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 = 1) {
- RTC_UNUSED(id);
- UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
- speech_type, vad_activity, num_channels);
- }
-
void UpdateFrame(uint32_t timestamp, const int16_t* data,
size_t samples_per_channel, int sample_rate_hz,
SpeechType speech_type, VADActivity vad_activity,
@@ -108,13 +93,6 @@
// Frame is muted by default.
bool muted() const;
- // These methods are deprecated. Use the functions in
- // webrtc/audio/utility instead. These methods will exists for a
- // short period of time until webrtc clients have updated. See
- // webrtc:6548 for details.
- RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
- RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
-
// RTP timestamp of the first sample in the AudioFrame.
uint32_t timestamp_ = 0;
// Time since the first frame in milliseconds.
