Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions

View File

@ -8,17 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
#include "modules/audio_processing/vad/voice_activity_detector.h"
#include <algorithm>
#include "webrtc/base/checks.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
const size_t kMaxLength = 320;
const int kNumChannels = 1;
const size_t kNumChannels = 1;
const double kDefaultVoiceValue = 1.0;
const double kNeutralProbability = 0.5;
@ -28,8 +27,9 @@ const double kLowProbability = 0.01;
VoiceActivityDetector::VoiceActivityDetector()
: last_voice_probability_(kDefaultVoiceValue),
standalone_vad_(StandaloneVad::Create()) {
}
standalone_vad_(StandaloneVad::Create()) {}
VoiceActivityDetector::~VoiceActivityDetector() = default;
// Because ISAC has a different chunk length, it updates
// |chunkwise_voice_probabilities_| and |chunkwise_rms_| when there is new data.
@ -37,8 +37,7 @@ VoiceActivityDetector::VoiceActivityDetector()
void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
size_t length,
int sample_rate_hz) {
RTC_DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100);
RTC_DCHECK_LE(length, kMaxLength);
RTC_DCHECK_EQ(length, sample_rate_hz / 100);
// Resample to the required rate.
const int16_t* resampled_ptr = audio;
if (sample_rate_hz != kSampleRateHz) {