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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) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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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206 << static_cast<int>(header->payloadType) 206 << static_cast<int>(header->payloadType)
207 << " Failed to insert packet"; 207 << " Failed to insert packet";
208 return -1; 208 return -1;
209 } 209 }
210 return 0; 210 return 0;
211 } 211 }
212 212
213 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) { 213 int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
214 enum NetEqOutputType type; 214 enum NetEqOutputType type;
215 size_t samples_per_channel; 215 size_t samples_per_channel;
216 int num_channels; 216 size_t num_channels;
217 217
218 // Accessing members, take the lock. 218 // Accessing members, take the lock.
219 CriticalSectionScoped lock(crit_sect_.get()); 219 CriticalSectionScoped lock(crit_sect_.get());
220 220
221 // Always write the output to |audio_buffer_| first. 221 // Always write the output to |audio_buffer_| first.
222 if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples, 222 if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
223 audio_buffer_.get(), 223 audio_buffer_.get(),
224 &samples_per_channel, 224 &samples_per_channel,
225 &num_channels, 225 &num_channels,
226 &type) != NetEq::kOK) { 226 &type) != NetEq::kOK) {
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294 } else { 294 } else {
295 // Remain 0 until we have a valid |playout_timestamp|. 295 // Remain 0 until we have a valid |playout_timestamp|.
296 audio_frame->timestamp_ = 0; 296 audio_frame->timestamp_ = 0;
297 } 297 }
298 298
299 return 0; 299 return 0;
300 } 300 }
301 301
302 int32_t AcmReceiver::AddCodec(int acm_codec_id, 302 int32_t AcmReceiver::AddCodec(int acm_codec_id,
303 uint8_t payload_type, 303 uint8_t payload_type,
304 int channels, 304 size_t channels,
305 int sample_rate_hz, 305 int sample_rate_hz,
306 AudioDecoder* audio_decoder, 306 AudioDecoder* audio_decoder,
307 const std::string& name) { 307 const std::string& name) {
308 const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder { 308 const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
309 if (acm_codec_id == -1) 309 if (acm_codec_id == -1)
310 return NetEqDecoder::kDecoderArbitrary; // External decoder. 310 return NetEqDecoder::kDecoderArbitrary; // External decoder.
311 const rtc::Optional<RentACodec::CodecId> cid = 311 const rtc::Optional<RentACodec::CodecId> cid =
312 RentACodec::CodecIdFromIndex(acm_codec_id); 312 RentACodec::CodecIdFromIndex(acm_codec_id);
313 RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id; 313 RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
314 const rtc::Optional<NetEqDecoder> ned = 314 const rtc::Optional<NetEqDecoder> ned =
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532 532
533 void AcmReceiver::GetDecodingCallStatistics( 533 void AcmReceiver::GetDecodingCallStatistics(
534 AudioDecodingCallStats* stats) const { 534 AudioDecodingCallStats* stats) const {
535 CriticalSectionScoped lock(crit_sect_.get()); 535 CriticalSectionScoped lock(crit_sect_.get());
536 *stats = call_stats_.GetDecodingStatistics(); 536 *stats = call_stats_.GetDecodingStatistics();
537 } 537 }
538 538
539 } // namespace acm2 539 } // namespace acm2
540 540
541 } // namespace webrtc 541 } // namespace webrtc
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