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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 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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403 } 403 }
404 } 404 }
405 return false; 405 return false;
406 } 406 }
407 407
408 private: 408 private:
409 static const int kMaxNumPacketSize = 6; 409 static const int kMaxNumPacketSize = 6;
410 struct CodecPref { 410 struct CodecPref {
411 const char* name; 411 const char* name;
412 int clockrate; 412 int clockrate;
413 int channels; 413 size_t channels;
414 int payload_type; 414 int payload_type;
415 bool is_multi_rate; 415 bool is_multi_rate;
416 int packet_sizes_ms[kMaxNumPacketSize]; 416 int packet_sizes_ms[kMaxNumPacketSize];
417 }; 417 };
418 // Note: keep the supported packet sizes in ascending order. 418 // Note: keep the supported packet sizes in ascending order.
419 static const CodecPref kCodecPrefs[12]; 419 static const CodecPref kCodecPrefs[12];
420 420
421 static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) { 421 static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
422 int selected_packet_size_ms = codec_pref.packet_sizes_ms[0]; 422 int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
423 for (int packet_size_ms : codec_pref.packet_sizes_ms) { 423 for (int packet_size_ms : codec_pref.packet_sizes_ms) {
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1148 renderer_->SetSink(nullptr); 1148 renderer_->SetSink(nullptr);
1149 renderer_ = nullptr; 1149 renderer_ = nullptr;
1150 } 1150 }
1151 } 1151 }
1152 1152
1153 // AudioRenderer::Sink implementation. 1153 // AudioRenderer::Sink implementation.
1154 // This method is called on the audio thread. 1154 // This method is called on the audio thread.
1155 void OnData(const void* audio_data, 1155 void OnData(const void* audio_data,
1156 int bits_per_sample, 1156 int bits_per_sample,
1157 int sample_rate, 1157 int sample_rate,
1158 int number_of_channels, 1158 size_t number_of_channels,
1159 size_t number_of_frames) override { 1159 size_t number_of_frames) override {
1160 RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread()); 1160 RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
1161 RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread()); 1161 RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
1162 RTC_DCHECK(voe_audio_transport_); 1162 RTC_DCHECK(voe_audio_transport_);
1163 voe_audio_transport_->OnData(config_.voe_channel_id, 1163 voe_audio_transport_->OnData(config_.voe_channel_id,
1164 audio_data, 1164 audio_data,
1165 bits_per_sample, 1165 bits_per_sample,
1166 sample_rate, 1166 sample_rate,
1167 number_of_channels, 1167 number_of_channels,
1168 number_of_frames); 1168 number_of_frames);
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2500 } 2500 }
2501 } else { 2501 } else {
2502 LOG(LS_INFO) << "Stopping playout for channel #" << channel; 2502 LOG(LS_INFO) << "Stopping playout for channel #" << channel;
2503 engine()->voe()->base()->StopPlayout(channel); 2503 engine()->voe()->base()->StopPlayout(channel);
2504 } 2504 }
2505 return true; 2505 return true;
2506 } 2506 }
2507 } // namespace cricket 2507 } // namespace cricket
2508 2508
2509 #endif // HAVE_WEBRTC_VOICE 2509 #endif // HAVE_WEBRTC_VOICE
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