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Side by Side Diff: webrtc/modules/audio_coding/test/opus_test.cc

Issue 2750783004: Add mute state field to AudioFrame. (Closed)
Patch Set: don't return from Add() too early Created 3 years, 9 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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256 in_file_mono_.Read10MsData(audio_frame); 256 in_file_mono_.Read10MsData(audio_frame);
257 } else { 257 } else {
258 if (in_file_stereo_.EndOfFile()) { 258 if (in_file_stereo_.EndOfFile()) {
259 break; 259 break;
260 } 260 }
261 in_file_stereo_.Read10MsData(audio_frame); 261 in_file_stereo_.Read10MsData(audio_frame);
262 } 262 }
263 263
264 // If input audio is sampled at 32 kHz, resampling to 48 kHz is required. 264 // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
265 EXPECT_EQ(480, 265 EXPECT_EQ(480,
266 resampler_.Resample10Msec(audio_frame.data_, 266 resampler_.Resample10Msec(audio_frame.data(),
267 audio_frame.sample_rate_hz_, 267 audio_frame.sample_rate_hz_,
268 48000, 268 48000,
269 channels, 269 channels,
270 kBufferSizeSamples - written_samples, 270 kBufferSizeSamples - written_samples,
271 &audio[written_samples])); 271 &audio[written_samples]));
272 written_samples += 480 * channels; 272 written_samples += 480 * channels;
273 273
274 // Sometimes we need to loop over the audio vector to produce the right 274 // Sometimes we need to loop over the audio vector to produce the right
275 // number of packets. 275 // number of packets.
276 size_t loop_encode = (written_samples - read_samples) / 276 size_t loop_encode = (written_samples - read_samples) /
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341 } 341 }
342 342
343 // Run received side of ACM. 343 // Run received side of ACM.
344 bool muted; 344 bool muted;
345 ASSERT_EQ( 345 ASSERT_EQ(
346 0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted)); 346 0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
347 ASSERT_FALSE(muted); 347 ASSERT_FALSE(muted);
348 348
349 // Write output speech to file. 349 // Write output speech to file.
350 out_file_.Write10MsData( 350 out_file_.Write10MsData(
351 audio_frame.data_, 351 audio_frame.data(),
352 audio_frame.samples_per_channel_ * audio_frame.num_channels_); 352 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
353 353
354 // Write stand-alone speech to file. 354 // Write stand-alone speech to file.
355 out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels); 355 out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
356 356
357 if (audio_frame.timestamp_ > start_time_stamp) { 357 if (audio_frame.timestamp_ > start_time_stamp) {
358 // Number of channels should be the same for both stand-alone and 358 // Number of channels should be the same for both stand-alone and
359 // ACM-decoding. 359 // ACM-decoding.
360 EXPECT_EQ(audio_frame.num_channels_, channels); 360 EXPECT_EQ(audio_frame.num_channels_, channels);
361 } 361 }
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382 out_file_.Open(file_name, 48000, "wb"); 382 out_file_.Open(file_name, 48000, "wb");
383 file_stream.str(""); 383 file_stream.str("");
384 file_name = file_stream.str(); 384 file_name = file_stream.str();
385 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" 385 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
386 << test_number << ".pcm"; 386 << test_number << ".pcm";
387 file_name = file_stream.str(); 387 file_name = file_stream.str();
388 out_file_standalone_.Open(file_name, 48000, "wb"); 388 out_file_standalone_.Open(file_name, 48000, "wb");
389 } 389 }
390 390
391 } // namespace webrtc 391 } // namespace webrtc
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