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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: Update new usages of AudioFrame::data_ Created 3 years, 6 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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255 in_file_mono_.Read10MsData(audio_frame); 255 in_file_mono_.Read10MsData(audio_frame);
256 } else { 256 } else {
257 if (in_file_stereo_.EndOfFile()) { 257 if (in_file_stereo_.EndOfFile()) {
258 break; 258 break;
259 } 259 }
260 in_file_stereo_.Read10MsData(audio_frame); 260 in_file_stereo_.Read10MsData(audio_frame);
261 } 261 }
262 262
263 // If input audio is sampled at 32 kHz, resampling to 48 kHz is required. 263 // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
264 EXPECT_EQ(480, 264 EXPECT_EQ(480,
265 resampler_.Resample10Msec(audio_frame.data_, 265 resampler_.Resample10Msec(audio_frame.data(),
266 audio_frame.sample_rate_hz_, 266 audio_frame.sample_rate_hz_,
267 48000, 267 48000,
268 channels, 268 channels,
269 kBufferSizeSamples - written_samples, 269 kBufferSizeSamples - written_samples,
270 &audio[written_samples])); 270 &audio[written_samples]));
271 written_samples += 480 * channels; 271 written_samples += 480 * channels;
272 272
273 // Sometimes we need to loop over the audio vector to produce the right 273 // Sometimes we need to loop over the audio vector to produce the right
274 // number of packets. 274 // number of packets.
275 size_t loop_encode = (written_samples - read_samples) / 275 size_t loop_encode = (written_samples - read_samples) /
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340 } 340 }
341 341
342 // Run received side of ACM. 342 // Run received side of ACM.
343 bool muted; 343 bool muted;
344 ASSERT_EQ( 344 ASSERT_EQ(
345 0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted)); 345 0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
346 ASSERT_FALSE(muted); 346 ASSERT_FALSE(muted);
347 347
348 // Write output speech to file. 348 // Write output speech to file.
349 out_file_.Write10MsData( 349 out_file_.Write10MsData(
350 audio_frame.data_, 350 audio_frame.data(),
351 audio_frame.samples_per_channel_ * audio_frame.num_channels_); 351 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
352 352
353 // Write stand-alone speech to file. 353 // Write stand-alone speech to file.
354 out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels); 354 out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
355 355
356 if (audio_frame.timestamp_ > start_time_stamp) { 356 if (audio_frame.timestamp_ > start_time_stamp) {
357 // Number of channels should be the same for both stand-alone and 357 // Number of channels should be the same for both stand-alone and
358 // ACM-decoding. 358 // ACM-decoding.
359 EXPECT_EQ(audio_frame.num_channels_, channels); 359 EXPECT_EQ(audio_frame.num_channels_, channels);
360 } 360 }
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381 out_file_.Open(file_name, 48000, "wb"); 381 out_file_.Open(file_name, 48000, "wb");
382 file_stream.str(""); 382 file_stream.str("");
383 file_name = file_stream.str(); 383 file_name = file_stream.str();
384 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" 384 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
385 << test_number << ".pcm"; 385 << test_number << ".pcm";
386 file_name = file_stream.str(); 386 file_name = file_stream.str();
387 out_file_standalone_.Open(file_name, 48000, "wb"); 387 out_file_standalone_.Open(file_name, 48000, "wb");
388 } 388 }
389 389
390 } // namespace webrtc 390 } // namespace webrtc
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