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

Issue 2262203002: Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 3 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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152 // Maximum playout delay. 152 // Maximum playout delay.
153 int SetMaximumPlayoutDelay(int time_ms) override; 153 int SetMaximumPlayoutDelay(int time_ms) override;
154 154
155 // Smallest latency NetEq will maintain. 155 // Smallest latency NetEq will maintain.
156 int LeastRequiredDelayMs() const override; 156 int LeastRequiredDelayMs() const override;
157 157
158 RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override; 158 RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;
159 159
160 rtc::Optional<uint32_t> PlayoutTimestamp() override; 160 rtc::Optional<uint32_t> PlayoutTimestamp() override;
161 161
162 int FilteredCurrentDelayMs() const override;
163
162 // Get 10 milliseconds of raw audio data to play out, and 164 // Get 10 milliseconds of raw audio data to play out, and
163 // automatic resample to the requested frequency if > 0. 165 // automatic resample to the requested frequency if > 0.
164 int PlayoutData10Ms(int desired_freq_hz, 166 int PlayoutData10Ms(int desired_freq_hz,
165 AudioFrame* audio_frame, 167 AudioFrame* audio_frame,
166 bool* muted) override; 168 bool* muted) override;
167 int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override; 169 int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
168 170
169 ///////////////////////////////////////// 171 /////////////////////////////////////////
170 // Statistics 172 // Statistics
171 // 173 //
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1218 if (!ts) 1220 if (!ts)
1219 return -1; 1221 return -1;
1220 *timestamp = *ts; 1222 *timestamp = *ts;
1221 return 0; 1223 return 0;
1222 } 1224 }
1223 1225
1224 rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() { 1226 rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
1225 return receiver_.GetPlayoutTimestamp(); 1227 return receiver_.GetPlayoutTimestamp();
1226 } 1228 }
1227 1229
1230 int AudioCodingModuleImpl::FilteredCurrentDelayMs() const {
1231 return receiver_.FilteredCurrentDelayMs();
1232 }
1233
1228 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const { 1234 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
1229 if (!encoder_stack_) { 1235 if (!encoder_stack_) {
1230 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_, 1236 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
1231 "%s failed: No send codec is registered.", caller_name); 1237 "%s failed: No send codec is registered.", caller_name);
1232 return false; 1238 return false;
1233 } 1239 }
1234 return true; 1240 return true;
1235 } 1241 }
1236 1242
1237 int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) { 1243 int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
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1341 // Checks the validity of the parameters of the given codec 1347 // Checks the validity of the parameters of the given codec
1342 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) { 1348 bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
1343 bool valid = acm2::RentACodec::IsCodecValid(codec); 1349 bool valid = acm2::RentACodec::IsCodecValid(codec);
1344 if (!valid) 1350 if (!valid)
1345 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, 1351 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
1346 "Invalid codec setting"); 1352 "Invalid codec setting");
1347 return valid; 1353 return valid;
1348 } 1354 }
1349 1355
1350 } // namespace webrtc 1356 } // namespace webrtc
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