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| 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 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 |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #include "webrtc/audio/audio_transport_proxy.h" |
| 12 |
| 13 namespace webrtc { |
| 14 |
| 15 AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport, |
| 16 AudioProcessing* apm, |
| 17 AudioMixer* mixer) |
| 18 : voe_audio_transport_(voe_audio_transport) { |
| 19 RTC_DCHECK(voe_audio_transport); |
| 20 RTC_DCHECK(apm); |
| 21 } |
| 22 |
| 23 AudioTransportProxy::~AudioTransportProxy() {} |
| 24 |
| 25 int32_t AudioTransportProxy::RecordedDataIsAvailable( |
| 26 const void* audioSamples, |
| 27 const size_t nSamples, |
| 28 const size_t nBytesPerSample, |
| 29 const size_t nChannels, |
| 30 const uint32_t samplesPerSec, |
| 31 const uint32_t totalDelayMS, |
| 32 const int32_t clockDrift, |
| 33 const uint32_t currentMicLevel, |
| 34 const bool keyPressed, |
| 35 uint32_t& newMicLevel) { |
| 36 // Pass call through to original audio transport instance. |
| 37 return voe_audio_transport_->RecordedDataIsAvailable( |
| 38 audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec, |
| 39 totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel); |
| 40 } |
| 41 |
| 42 int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples, |
| 43 const size_t nBytesPerSample, |
| 44 const size_t nChannels, |
| 45 const uint32_t samplesPerSec, |
| 46 void* audioSamples, |
| 47 size_t& nSamplesOut, |
| 48 int64_t* elapsed_time_ms, |
| 49 int64_t* ntp_time_ms) { |
| 50 RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample); |
| 51 RTC_DCHECK_GE(nChannels, 1u); |
| 52 RTC_DCHECK_LE(nChannels, 2u); |
| 53 RTC_DCHECK_GE( |
| 54 samplesPerSec, |
| 55 static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz)); |
| 56 RTC_DCHECK_EQ(nSamples * 100, samplesPerSec); |
| 57 RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels, |
| 58 sizeof(AudioFrame::data_)); |
| 59 |
| 60 // Pass call through to original audio transport instance. |
| 61 return voe_audio_transport_->NeedMorePlayData( |
| 62 nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples, |
| 63 nSamplesOut, elapsed_time_ms, ntp_time_ms); |
| 64 } |
| 65 |
| 66 void AudioTransportProxy::PushCaptureData(int voe_channel, |
| 67 const void* audio_data, |
| 68 int bits_per_sample, |
| 69 int sample_rate, |
| 70 size_t number_of_channels, |
| 71 size_t number_of_frames) { |
| 72 // This is part of deprecated VoE interface operating on specific |
| 73 // VoE channels. It should not be used. |
| 74 RTC_NOTREACHED(); |
| 75 } |
| 76 |
| 77 void AudioTransportProxy::PullRenderData(int bits_per_sample, |
| 78 int sample_rate, |
| 79 size_t number_of_channels, |
| 80 size_t number_of_frames, |
| 81 void* audio_data, |
| 82 int64_t* elapsed_time_ms, |
| 83 int64_t* ntp_time_ms) { |
| 84 RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 8 * sizeof(int16_t)); |
| 85 RTC_DCHECK_GE(number_of_channels, 1u); |
| 86 RTC_DCHECK_LE(number_of_channels, 2u); |
| 87 RTC_DCHECK_GE(static_cast<int>(sample_rate), |
| 88 AudioProcessing::NativeRate::kSampleRate8kHz); |
| 89 RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate); |
| 90 RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels, |
| 91 sizeof(AudioFrame::data_)); |
| 92 voe_audio_transport_->PullRenderData( |
| 93 bits_per_sample, sample_rate, number_of_channels, number_of_frames, |
| 94 audio_data, elapsed_time_ms, ntp_time_ms); |
| 95 } |
| 96 |
| 97 } // namespace webrtc |
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