| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/audio/audio_transport_proxy.h" |
| |
| namespace webrtc { |
| |
| AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport, |
| AudioProcessing* apm, |
| AudioMixer* mixer) |
| : voe_audio_transport_(voe_audio_transport) { |
| RTC_DCHECK(voe_audio_transport); |
| RTC_DCHECK(apm); |
| } |
| |
| AudioTransportProxy::~AudioTransportProxy() {} |
| |
| int32_t AudioTransportProxy::RecordedDataIsAvailable( |
| const void* audioSamples, |
| const size_t nSamples, |
| const size_t nBytesPerSample, |
| const size_t nChannels, |
| const uint32_t samplesPerSec, |
| const uint32_t totalDelayMS, |
| const int32_t clockDrift, |
| const uint32_t currentMicLevel, |
| const bool keyPressed, |
| uint32_t& newMicLevel) { |
| // Pass call through to original audio transport instance. |
| return voe_audio_transport_->RecordedDataIsAvailable( |
| audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec, |
| totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel); |
| } |
| |
| int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples, |
| const size_t nBytesPerSample, |
| const size_t nChannels, |
| const uint32_t samplesPerSec, |
| void* audioSamples, |
| size_t& nSamplesOut, |
| int64_t* elapsed_time_ms, |
| int64_t* ntp_time_ms) { |
| RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample); |
| RTC_DCHECK_GE(nChannels, 1u); |
| RTC_DCHECK_LE(nChannels, 2u); |
| RTC_DCHECK_GE( |
| samplesPerSec, |
| static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz)); |
| RTC_DCHECK_EQ(nSamples * 100, samplesPerSec); |
| RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels, |
| sizeof(AudioFrame::data_)); |
| |
| // Pass call through to original audio transport instance. |
| return voe_audio_transport_->NeedMorePlayData( |
| nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples, |
| nSamplesOut, elapsed_time_ms, ntp_time_ms); |
| } |
| |
| void AudioTransportProxy::PushCaptureData(int voe_channel, |
| const void* audio_data, |
| int bits_per_sample, |
| int sample_rate, |
| size_t number_of_channels, |
| size_t number_of_frames) { |
| // This is part of deprecated VoE interface operating on specific |
| // VoE channels. It should not be used. |
| RTC_NOTREACHED(); |
| } |
| |
| void AudioTransportProxy::PullRenderData(int bits_per_sample, |
| int sample_rate, |
| size_t number_of_channels, |
| size_t number_of_frames, |
| void* audio_data, |
| int64_t* elapsed_time_ms, |
| int64_t* ntp_time_ms) { |
| RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 8 * sizeof(int16_t)); |
| RTC_DCHECK_GE(number_of_channels, 1u); |
| RTC_DCHECK_LE(number_of_channels, 2u); |
| RTC_DCHECK_GE(static_cast<int>(sample_rate), |
| AudioProcessing::NativeRate::kSampleRate8kHz); |
| RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate); |
| RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels, |
| sizeof(AudioFrame::data_)); |
| voe_audio_transport_->PullRenderData( |
| bits_per_sample, sample_rate, number_of_channels, number_of_frames, |
| audio_data, elapsed_time_ms, ntp_time_ms); |
| } |
| |
| } // namespace webrtc |