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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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| 641 "SendFrequency Failed, no codec is registered"); | 641 "SendFrequency Failed, no codec is registered"); |
| 642 return -1; | 642 return -1; |
| 643 } | 643 } |
| 644 | 644 |
| 645 return encoder_stack_->SampleRateHz(); | 645 return encoder_stack_->SampleRateHz(); |
| 646 } | 646 } |
| 647 | 647 |
| 648 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) { | 648 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) { |
| 649 rtc::CritScope lock(&acm_crit_sect_); | 649 rtc::CritScope lock(&acm_crit_sect_); |
| 650 if (encoder_stack_) { | 650 if (encoder_stack_) { |
| 651 encoder_stack_->OnReceivedTargetAudioBitrate(bitrate_bps); | 651 encoder_stack_->OnReceivedTargetAudioBitrate(bitrate_bps, |
| 652 rtc::Optional<int64_t>()); |
| 652 } | 653 } |
| 653 } | 654 } |
| 654 | 655 |
| 655 // Register a transport callback which will be called to deliver | 656 // Register a transport callback which will be called to deliver |
| 656 // the encoded buffers. | 657 // the encoded buffers. |
| 657 int AudioCodingModuleImpl::RegisterTransportCallback( | 658 int AudioCodingModuleImpl::RegisterTransportCallback( |
| 658 AudioPacketizationCallback* transport) { | 659 AudioPacketizationCallback* transport) { |
| 659 rtc::CritScope lock(&callback_crit_sect_); | 660 rtc::CritScope lock(&callback_crit_sect_); |
| 660 packetization_callback_ = transport; | 661 packetization_callback_ = transport; |
| 661 return 0; | 662 return 0; |
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| 1375 // Checks the validity of the parameters of the given codec | 1376 // Checks the validity of the parameters of the given codec |
| 1376 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) { | 1377 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) { |
| 1377 bool valid = acm2::RentACodec::IsCodecValid(codec); | 1378 bool valid = acm2::RentACodec::IsCodecValid(codec); |
| 1378 if (!valid) | 1379 if (!valid) |
| 1379 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, | 1380 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, |
| 1380 "Invalid codec setting"); | 1381 "Invalid codec setting"); |
| 1381 return valid; | 1382 return valid; |
| 1382 } | 1383 } |
| 1383 | 1384 |
| 1384 } // namespace webrtc | 1385 } // namespace webrtc |
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