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

Issue 1316523002: Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Fix compile Created 4 years, 11 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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317 const AudioFrame* ptr_frame; 317 const AudioFrame* ptr_frame;
318 // Perform a resampling, also down-mix if it is required and can be 318 // Perform a resampling, also down-mix if it is required and can be
319 // performed before resampling (a down mix prior to resampling will take 319 // performed before resampling (a down mix prior to resampling will take
320 // place if both primary and secondary encoders are mono and input is in 320 // place if both primary and secondary encoders are mono and input is in
321 // stereo). 321 // stereo).
322 if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) { 322 if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) {
323 return -1; 323 return -1;
324 } 324 }
325 325
326 // Check whether we need an up-mix or down-mix? 326 // Check whether we need an up-mix or down-mix?
327 const int current_num_channels = 327 const size_t current_num_channels =
328 rent_a_codec_.GetEncoderStack()->NumChannels(); 328 rent_a_codec_.GetEncoderStack()->NumChannels();
329 const bool same_num_channels = 329 const bool same_num_channels =
330 ptr_frame->num_channels_ == current_num_channels; 330 ptr_frame->num_channels_ == current_num_channels;
331 331
332 if (!same_num_channels) { 332 if (!same_num_channels) {
333 if (ptr_frame->num_channels_ == 1) { 333 if (ptr_frame->num_channels_ == 1) {
334 if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0) 334 if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
335 return -1; 335 return -1;
336 } else { 336 } else {
337 if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0) 337 if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
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582 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_, 582 WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
583 "PlayoutFrequency()"); 583 "PlayoutFrequency()");
584 return receiver_.last_output_sample_rate_hz(); 584 return receiver_.last_output_sample_rate_hz();
585 } 585 }
586 586
587 // Register possible receive codecs, can be called multiple times, 587 // Register possible receive codecs, can be called multiple times,
588 // for codecs, CNG (NB, WB and SWB), DTMF, RED. 588 // for codecs, CNG (NB, WB and SWB), DTMF, RED.
589 int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) { 589 int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
590 CriticalSectionScoped lock(acm_crit_sect_.get()); 590 CriticalSectionScoped lock(acm_crit_sect_.get());
591 RTC_DCHECK(receiver_initialized_); 591 RTC_DCHECK(receiver_initialized_);
592 if (codec.channels > 2 || codec.channels < 0) { 592 if (codec.channels > 2) {
593 LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels; 593 LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
594 return -1; 594 return -1;
595 } 595 }
596 596
597 auto codec_id = 597 auto codec_id =
598 RentACodec::CodecIdByParams(codec.plname, codec.plfreq, codec.channels); 598 RentACodec::CodecIdByParams(codec.plname, codec.plfreq, codec.channels);
599 if (!codec_id) { 599 if (!codec_id) {
600 LOG_F(LS_ERROR) << "Wrong codec params to be registered as receive codec"; 600 LOG_F(LS_ERROR) << "Wrong codec params to be registered as receive codec";
601 return -1; 601 return -1;
602 } 602 }
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819 return receiver_.LeastRequiredDelayMs(); 819 return receiver_.LeastRequiredDelayMs();
820 } 820 }
821 821
822 void AudioCodingModuleImpl::GetDecodingCallStatistics( 822 void AudioCodingModuleImpl::GetDecodingCallStatistics(
823 AudioDecodingCallStats* call_stats) const { 823 AudioDecodingCallStats* call_stats) const {
824 receiver_.GetDecodingCallStatistics(call_stats); 824 receiver_.GetDecodingCallStatistics(call_stats);
825 } 825 }
826 826
827 } // namespace acm2 827 } // namespace acm2
828 } // namespace webrtc 828 } // namespace webrtc
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