dd31071e19
All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
98 lines
4.2 KiB
C++
98 lines
4.2 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/audio/audio_transport_proxy.h"
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namespace webrtc {
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AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport,
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AudioProcessing* apm,
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AudioMixer* mixer)
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: voe_audio_transport_(voe_audio_transport) {
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RTC_DCHECK(voe_audio_transport);
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RTC_DCHECK(apm);
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}
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AudioTransportProxy::~AudioTransportProxy() {}
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int32_t AudioTransportProxy::RecordedDataIsAvailable(
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const void* audioSamples,
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const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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const uint32_t totalDelayMS,
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const int32_t clockDrift,
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const uint32_t currentMicLevel,
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const bool keyPressed,
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uint32_t& newMicLevel) {
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// Pass call through to original audio transport instance.
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return voe_audio_transport_->RecordedDataIsAvailable(
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audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
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totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
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}
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int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples,
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const size_t nBytesPerSample,
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const size_t nChannels,
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const uint32_t samplesPerSec,
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void* audioSamples,
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size_t& nSamplesOut,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) {
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RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
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RTC_DCHECK_GE(nChannels, 1u);
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RTC_DCHECK_LE(nChannels, 2u);
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RTC_DCHECK_GE(
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samplesPerSec,
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static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
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RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
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RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
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sizeof(AudioFrame::data_));
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// Pass call through to original audio transport instance.
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return voe_audio_transport_->NeedMorePlayData(
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nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples,
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nSamplesOut, elapsed_time_ms, ntp_time_ms);
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}
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void AudioTransportProxy::PushCaptureData(int voe_channel,
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const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) {
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// This is part of deprecated VoE interface operating on specific
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// VoE channels. It should not be used.
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RTC_NOTREACHED();
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}
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void AudioTransportProxy::PullRenderData(int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames,
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void* audio_data,
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int64_t* elapsed_time_ms,
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int64_t* ntp_time_ms) {
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RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 8 * sizeof(int16_t));
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RTC_DCHECK_GE(number_of_channels, 1u);
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RTC_DCHECK_LE(number_of_channels, 2u);
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RTC_DCHECK_GE(static_cast<int>(sample_rate),
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AudioProcessing::NativeRate::kSampleRate8kHz);
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RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate);
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RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
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sizeof(AudioFrame::data_));
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voe_audio_transport_->PullRenderData(
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bits_per_sample, sample_rate, number_of_channels, number_of_frames,
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audio_data, elapsed_time_ms, ntp_time_ms);
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}
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} // namespace webrtc
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