blob: 73b009154035ad61481396d91fa05671f5a5e9c9 [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 {
namespace {
// Resample audio in |frame| to given sample rate preserving the
// channel count and place the result in |destination|.
int Resample(const AudioFrame& frame,
const int destination_sample_rate,
PushResampler<int16_t>* resampler,
int16_t* destination) {
const int number_of_channels = static_cast<int>(frame.num_channels_);
const int target_number_of_samples_per_channel =
destination_sample_rate / 100;
resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
// TODO(yujo): make resampler take an AudioFrame, and add special case
// handling of muted frames.
return resampler->Resample(, frame.samples_per_channel_ * number_of_channels,
destination, number_of_channels * target_number_of_samples_per_channel);
} // namespace
AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport,
AudioProcessing* audio_processing,
AudioMixer* mixer)
: voe_audio_transport_(voe_audio_transport),
mixer_(mixer) {
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) { // NOLINT: to avoid changing APIs
// 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, 1);
RTC_DCHECK_LE(nChannels, 2);
// 100 = 1 second / data duration (10 ms).
RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
mixer_->Mix(nChannels, &mixed_frame_);
*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
*ntp_time_ms = mixed_frame_.ntp_time_ms_;
const auto error = audio_processing_->ProcessReverseStream(&mixed_frame_);
RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
nSamplesOut = Resample(mixed_frame_, samplesPerSec, &resampler_,
RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
return 0;
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.
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(bits_per_sample, 16);
RTC_DCHECK_GE(number_of_channels, 1);
RTC_DCHECK_LE(number_of_channels, 2);
RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
// 100 = 1 second / data duration (10 ms).
RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
// 8 = bits per byte.
RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
mixer_->Mix(number_of_channels, &mixed_frame_);
*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
*ntp_time_ms = mixed_frame_.ntp_time_ms_;
const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_,
RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
} // namespace webrtc