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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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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103 RTC_NOTREACHED(); | 103 RTC_NOTREACHED(); |
104 } | 104 } |
105 | 105 |
106 void AudioTransportProxy::PullRenderData(int bits_per_sample, | 106 void AudioTransportProxy::PullRenderData(int bits_per_sample, |
107 int sample_rate, | 107 int sample_rate, |
108 size_t number_of_channels, | 108 size_t number_of_channels, |
109 size_t number_of_frames, | 109 size_t number_of_frames, |
110 void* audio_data, | 110 void* audio_data, |
111 int64_t* elapsed_time_ms, | 111 int64_t* elapsed_time_ms, |
112 int64_t* ntp_time_ms) { | 112 int64_t* ntp_time_ms) { |
113 RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 16); | 113 RTC_DCHECK_EQ(bits_per_sample, 16); |
114 RTC_DCHECK_GE(number_of_channels, 1); | 114 RTC_DCHECK_GE(number_of_channels, 1); |
115 RTC_DCHECK_LE(number_of_channels, 2); | 115 RTC_DCHECK_LE(number_of_channels, 2); |
116 RTC_DCHECK_GE(static_cast<int>(sample_rate), | 116 RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz); |
117 AudioProcessing::NativeRate::kSampleRate8kHz); | |
118 | 117 |
119 // 100 = 1 second / data duration (10 ms). | 118 // 100 = 1 second / data duration (10 ms). |
120 RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate); | 119 RTC_DCHECK_EQ(number_of_frames * 100, sample_rate); |
121 | 120 |
122 // 8 = bits per byte. | 121 // 8 = bits per byte. |
123 RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels, | 122 RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels, |
124 sizeof(AudioFrame::data_)); | 123 sizeof(AudioFrame::data_)); |
125 mixer_->Mix(number_of_channels, &mixed_frame_); | 124 mixer_->Mix(number_of_channels, &mixed_frame_); |
126 *elapsed_time_ms = mixed_frame_.elapsed_time_ms_; | 125 *elapsed_time_ms = mixed_frame_.elapsed_time_ms_; |
127 *ntp_time_ms = mixed_frame_.ntp_time_ms_; | 126 *ntp_time_ms = mixed_frame_.ntp_time_ms_; |
128 | 127 |
129 const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_, | 128 const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_, |
130 static_cast<int16_t*>(audio_data)); | 129 static_cast<int16_t*>(audio_data)); |
131 RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames); | 130 RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames); |
132 } | 131 } |
133 | 132 |
134 } // namespace webrtc | 133 } // namespace webrtc |
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