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Side by Side Diff: webrtc/media/base/mediachannel.h

Issue 1840043005: Don't reconfigure the encoder if the video options aren't changing. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: removing whitespace Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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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181 delay_agnostic_aec == o.delay_agnostic_aec && 181 delay_agnostic_aec == o.delay_agnostic_aec &&
182 experimental_ns == o.experimental_ns && 182 experimental_ns == o.experimental_ns &&
183 adjust_agc_delta == o.adjust_agc_delta && 183 adjust_agc_delta == o.adjust_agc_delta &&
184 tx_agc_target_dbov == o.tx_agc_target_dbov && 184 tx_agc_target_dbov == o.tx_agc_target_dbov &&
185 tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain && 185 tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
186 tx_agc_limiter == o.tx_agc_limiter && 186 tx_agc_limiter == o.tx_agc_limiter &&
187 recording_sample_rate == o.recording_sample_rate && 187 recording_sample_rate == o.recording_sample_rate &&
188 playout_sample_rate == o.playout_sample_rate && 188 playout_sample_rate == o.playout_sample_rate &&
189 combined_audio_video_bwe == o.combined_audio_video_bwe; 189 combined_audio_video_bwe == o.combined_audio_video_bwe;
190 } 190 }
191 bool operator!=(const AudioOptions& o) const { return !(*this == o); }
191 192
192 std::string ToString() const { 193 std::string ToString() const {
193 std::ostringstream ost; 194 std::ostringstream ost;
194 ost << "AudioOptions {"; 195 ost << "AudioOptions {";
195 ost << ToStringIfSet("aec", echo_cancellation); 196 ost << ToStringIfSet("aec", echo_cancellation);
196 ost << ToStringIfSet("agc", auto_gain_control); 197 ost << ToStringIfSet("agc", auto_gain_control);
197 ost << ToStringIfSet("ns", noise_suppression); 198 ost << ToStringIfSet("ns", noise_suppression);
198 ost << ToStringIfSet("hf", highpass_filter); 199 ost << ToStringIfSet("hf", highpass_filter);
199 ost << ToStringIfSet("swap", stereo_swapping); 200 ost << ToStringIfSet("swap", stereo_swapping);
200 ost << ToStringIfSet("audio_jitter_buffer_max_packets", 201 ost << ToStringIfSet("audio_jitter_buffer_max_packets",
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272 SetFrom(&video_noise_reduction, change.video_noise_reduction); 273 SetFrom(&video_noise_reduction, change.video_noise_reduction);
273 SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps); 274 SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
274 SetFrom(&is_screencast, change.is_screencast); 275 SetFrom(&is_screencast, change.is_screencast);
275 } 276 }
276 277
277 bool operator==(const VideoOptions& o) const { 278 bool operator==(const VideoOptions& o) const {
278 return video_noise_reduction == o.video_noise_reduction && 279 return video_noise_reduction == o.video_noise_reduction &&
279 screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps && 280 screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps &&
280 is_screencast == o.is_screencast; 281 is_screencast == o.is_screencast;
281 } 282 }
283 bool operator!=(const VideoOptions& o) const { return !(*this == o); }
282 284
283 std::string ToString() const { 285 std::string ToString() const {
284 std::ostringstream ost; 286 std::ostringstream ost;
285 ost << "VideoOptions {"; 287 ost << "VideoOptions {";
286 ost << ToStringIfSet("noise reduction", video_noise_reduction); 288 ost << ToStringIfSet("noise reduction", video_noise_reduction);
287 ost << ToStringIfSet("screencast min bitrate kbps", 289 ost << ToStringIfSet("screencast min bitrate kbps",
288 screencast_min_bitrate_kbps); 290 screencast_min_bitrate_kbps);
289 ost << ToStringIfSet("is_screencast ", is_screencast); 291 ost << ToStringIfSet("is_screencast ", is_screencast);
290 ost << "}"; 292 ost << "}";
291 return ost.str(); 293 return ost.str();
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1122 // Signal when the media channel is ready to send the stream. Arguments are: 1124 // Signal when the media channel is ready to send the stream. Arguments are:
1123 // writable(bool) 1125 // writable(bool)
1124 sigslot::signal1<bool> SignalReadyToSend; 1126 sigslot::signal1<bool> SignalReadyToSend;
1125 // Signal for notifying that the remote side has closed the DataChannel. 1127 // Signal for notifying that the remote side has closed the DataChannel.
1126 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1128 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1127 }; 1129 };
1128 1130
1129 } // namespace cricket 1131 } // namespace cricket
1130 1132
1131 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1133 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
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