Index: webrtc/modules/audio_processing/residual_echo_detector.cc |
diff --git a/webrtc/modules/audio_processing/residual_echo_detector.cc b/webrtc/modules/audio_processing/residual_echo_detector.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..3e230f101ccf3ee50baf37cc656c418280d1ac7b |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/residual_echo_detector.cc |
@@ -0,0 +1,85 @@ |
+/* |
hlundin-webrtc
2016/10/13 11:45:20
The order of the methods in this file does not mat
ivoc
2016/10/13 13:46:15
Fixed now.
|
+ * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#include "webrtc/modules/audio_processing/residual_echo_detector.h" |
+#include "webrtc/modules/audio_processing/audio_buffer.h" |
+#include "webrtc/modules/audio_processing/echo_detector/echo_detector.h" |
+ |
+namespace webrtc { |
+ |
+ResidualEchoDetector::ResidualEchoDetector(rtc::CriticalSection* crit_render, |
+ rtc::CriticalSection* crit_capture) |
+ : crit_render_(crit_render), crit_capture_(crit_capture) { |
+ RTC_DCHECK(crit_render); |
+ RTC_DCHECK(crit_capture); |
+} |
+ |
+ResidualEchoDetector::~ResidualEchoDetector() {} |
+ |
+void ResidualEchoDetector::AnalyzeRenderAudio(const AudioBuffer* audio) { |
+ rtc::CritScope cs_render(crit_render_); |
+ |
+ RTC_DCHECK_GE(160u, audio->num_frames_per_band()); |
+ |
+ render_queue_buffer_.clear(); |
+ |
+ // Buffer the samples in the render queue. |
+ render_queue_buffer_.insert(render_queue_buffer_.end(), |
+ audio->split_bands_const_f(0)[kBand0To8kHz], |
+ (audio->split_bands_const_f(0)[kBand0To8kHz] + |
+ audio->num_frames_per_band())); |
+ |
+ // Insert the samples into the queue. |
+ if (!render_signal_queue_->Insert(&render_queue_buffer_)) { |
+ // The data queue is full and needs to be emptied. |
+ ReadQueuedRenderData(); |
+ |
+ // Retry the insert (should always work). |
+ RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true); |
hlundin-webrtc
2016/10/13 11:45:20
The argument of DCHECK is only evaluated in debug
ivoc
2016/10/13 13:46:16
Good point, since I copied this from the AEC, it s
hlundin-webrtc
2016/10/14 06:59:18
Oh. Yes, please do!
ivoc
2016/10/14 09:53:18
Seems like Per beat me to it :)
|
+ } |
+} |
+ |
+float ResidualEchoDetector::get_echo_likelihood() { |
+ return 0.0f; |
+} |
+ |
+// Read chunks of data that were received and queued on the render side from |
+// a queue. All the data chunks are buffered into the farend signal of the AEC. |
+void ResidualEchoDetector::ReadQueuedRenderData() { |
+ rtc::CritScope cs_capture(crit_capture_); |
+ |
+ while (render_signal_queue_->Remove(&capture_queue_buffer_)) { |
hlundin-webrtc
2016/10/13 11:45:20
Is capture_queue_buffer_ only a container that con
ivoc
2016/10/13 13:46:16
I think this is indeed done for performance reason
hlundin-webrtc
2016/10/14 06:59:18
Acknowledged.
|
+ const size_t num_frames_per_band = capture_queue_buffer_.size(); |
+ detector_->BufferFarend(capture_queue_buffer_.data(), num_frames_per_band); |
+ } |
+} |
+ |
+void ResidualEchoDetector::AnalyzeCaptureAudio(AudioBuffer* audio) { |
+ rtc::CritScope cs_capture(crit_capture_); |
+ |
+ RTC_DCHECK_GE(160u, audio->num_frames_per_band()); |
+ |
+ detector_->Process(audio->split_bands_const_f(0)[kBand0To8kHz], |
+ audio->num_frames_per_band()); |
+} |
+ |
+void ResidualEchoDetector::Initialize(int sample_rate_hz) { |
+ rtc::CritScope cs_render(crit_render_); |
+ rtc::CritScope cs_capture(crit_capture_); |
+ |
+ RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || |
+ sample_rate_hz == AudioProcessing::kSampleRate16kHz || |
+ sample_rate_hz == AudioProcessing::kSampleRate32kHz || |
+ sample_rate_hz == AudioProcessing::kSampleRate48kHz); |
+ |
+ detector_->Initialize(sample_rate_hz); |
+} |
+ |
+} // namespace webrtc |