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Issue 1985743002: Propagate muted parameter to VoE::Channel (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Resurrect the PlayoutData10Ms(int, AudioFrame*) method Created 4 years, 7 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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785 } 785 }
786 } 786 }
787 // Verify that the timestamp is updated with expected length 787 // Verify that the timestamp is updated with expected length
788 time_stamp_diff = channel->timestamp_diff(); 788 time_stamp_diff = channel->timestamp_diff();
789 if ((counter_ > 10) && (time_stamp_diff != pack_size_samp_)) { 789 if ((counter_ > 10) && (time_stamp_diff != pack_size_samp_)) {
790 error_count++; 790 error_count++;
791 } 791 }
792 } 792 }
793 793
794 // Run received side of ACM 794 // Run received side of ACM
795 EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame)); 795 bool muted;
796 EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
797 ASSERT_FALSE(muted);
796 798
797 // Write output speech to file 799 // Write output speech to file
798 out_file_.Write10MsData( 800 out_file_.Write10MsData(
799 audio_frame.data_, 801 audio_frame.data_,
800 audio_frame.samples_per_channel_ * audio_frame.num_channels_); 802 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
801 } 803 }
802 804
803 EXPECT_EQ(0, error_count); 805 EXPECT_EQ(0, error_count);
804 806
805 // Check that packet size is in the right range for variable rate codecs, 807 // Check that packet size is in the right range for variable rate codecs,
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835 printf("%s -> ", send_codec->plname); 837 printf("%s -> ", send_codec->plname);
836 } 838 }
837 CodecInst receive_codec; 839 CodecInst receive_codec;
838 acm_b_->ReceiveCodec(&receive_codec); 840 acm_b_->ReceiveCodec(&receive_codec);
839 if (test_mode_ != 0) { 841 if (test_mode_ != 0) {
840 printf("%s\n", receive_codec.plname); 842 printf("%s\n", receive_codec.plname);
841 } 843 }
842 } 844 }
843 845
844 } // namespace webrtc 846 } // namespace webrtc
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