blob: ed72200379d1d07fab29e04482b761ea732cb6e2 [file] [log] [blame]
/*
* 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