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Side by Side Diff: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

Issue 1362943004: ACM: Removing runtime APIs related to playout mode (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years, 2 months ago
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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
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652 652
653 int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) { 653 int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
654 if ((time_ms < 0) || (time_ms > 10000)) { 654 if ((time_ms < 0) || (time_ms > 10000)) {
655 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, 655 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
656 "Delay must be in the range of 0-1000 milliseconds."); 656 "Delay must be in the range of 0-1000 milliseconds.");
657 return -1; 657 return -1;
658 } 658 }
659 return receiver_.SetMaximumDelay(time_ms); 659 return receiver_.SetMaximumDelay(time_ms);
660 } 660 }
661 661
662 // Set playout mode for: voice, fax, streaming or off.
663 int AudioCodingModuleImpl::SetPlayoutMode(AudioPlayoutMode mode) {
664 receiver_.SetPlayoutMode(mode);
665 return 0; // TODO(turajs): return value is for backward compatibility.
666 }
667
668 // Get playout mode voice, fax, streaming or off.
669 AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
670 return receiver_.PlayoutMode();
671 }
672
673 // Get 10 milliseconds of raw audio data to play out. 662 // Get 10 milliseconds of raw audio data to play out.
674 // Automatic resample to the requested frequency. 663 // Automatic resample to the requested frequency.
675 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz, 664 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
676 AudioFrame* audio_frame) { 665 AudioFrame* audio_frame) {
677 // GetAudio always returns 10 ms, at the requested sample rate. 666 // GetAudio always returns 10 ms, at the requested sample rate.
678 if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) { 667 if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
679 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, 668 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
680 "PlayoutData failed, RecOut Failed"); 669 "PlayoutData failed, RecOut Failed");
681 return -1; 670 return -1;
682 } 671 }
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1119 *sample_rate_hz = 8000; 1108 *sample_rate_hz = 8000;
1120 *channels = 1; 1109 *channels = 1;
1121 break; 1110 break;
1122 default: 1111 default:
1123 FATAL() << "Codec type " << codec_type << " not supported."; 1112 FATAL() << "Codec type " << codec_type << " not supported.";
1124 } 1113 }
1125 return true; 1114 return true;
1126 } 1115 }
1127 1116
1128 } // namespace webrtc 1117 } // namespace webrtc
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