Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(546)

Side by Side Diff: webrtc/modules/audio_processing/gain_control_impl.cc

Issue 1571013002: Remove additional channel constraints when Beamforming is enabled in AudioProcessing (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebasing Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 417 matching lines...) Expand 10 before | Expand all | Expand 10 after
428 config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_); 428 config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
429 config.compressionGaindB = 429 config.compressionGaindB =
430 static_cast<int16_t>(compression_gain_db_); 430 static_cast<int16_t>(compression_gain_db_);
431 config.limiterEnable = limiter_enabled_; 431 config.limiterEnable = limiter_enabled_;
432 432
433 return WebRtcAgc_set_config(static_cast<Handle*>(handle), config); 433 return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
434 } 434 }
435 435
436 int GainControlImpl::num_handles_required() const { 436 int GainControlImpl::num_handles_required() const {
437 // Not locked as it only relies on APM public API which is threadsafe. 437 // Not locked as it only relies on APM public API which is threadsafe.
438 return apm_->num_output_channels(); 438 return apm_->num_proc_channels();
439 } 439 }
440 440
441 int GainControlImpl::GetHandleError(void* handle) const { 441 int GainControlImpl::GetHandleError(void* handle) const {
442 // The AGC has no get_error() function. 442 // The AGC has no get_error() function.
443 // (Despite listing errors in its interface...) 443 // (Despite listing errors in its interface...)
444 assert(handle != NULL); 444 assert(handle != NULL);
445 return AudioProcessing::kUnspecifiedError; 445 return AudioProcessing::kUnspecifiedError;
446 } 446 }
447 } // namespace webrtc 447 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/audio_processing/echo_cancellation_impl.cc ('k') | webrtc/modules/audio_processing/include/audio_processing.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698