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/*
* 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 "audio/audio_transport_impl.h"
#include <algorithm>
#include <memory>
#include <utility>
#include "audio/remix_resample.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/audio_sender.h"
#include "modules/async_audio_processing/async_audio_processing.h"
#include "modules/audio_processing/include/audio_frame_proxies.h"
#include "rtc_base/checks.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
// We want to process at the lowest sample rate and channel count possible
// without losing information. Choose the lowest native rate at least equal to
// the minimum of input and codec rates, choose lowest channel count, and
// configure the audio frame.
void InitializeCaptureFrame(int input_sample_rate,
int send_sample_rate_hz,
size_t input_num_channels,
size_t send_num_channels,
AudioFrame* audio_frame) {
RTC_DCHECK(audio_frame);
int min_processing_rate_hz = std::min(input_sample_rate, send_sample_rate_hz);
for (int native_rate_hz : AudioProcessing::kNativeSampleRatesHz) {
audio_frame->SetSampleRateAndChannelSize(native_rate_hz);
if (native_rate_hz >= min_processing_rate_hz) {
break;
}
}
audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
}
void ProcessCaptureFrame(uint32_t delay_ms,
bool key_pressed,
bool swap_stereo_channels,
AudioProcessing* audio_processing,
AudioFrame* audio_frame) {
RTC_DCHECK(audio_frame);
if (audio_processing) {
audio_processing->set_stream_delay_ms(delay_ms);
audio_processing->set_stream_key_pressed(key_pressed);
int error = ProcessAudioFrame(audio_processing, audio_frame);
RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
}
if (swap_stereo_channels) {
AudioFrameOperations::SwapStereoChannels(audio_frame);
}
}
// 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,
InterleavedView<int16_t> destination) {
TRACE_EVENT2("webrtc", "Resample", "frame sample rate", frame.sample_rate_hz_,
"destination_sample_rate", destination_sample_rate);
const size_t target_number_of_samples_per_channel =
SampleRateToDefaultChannelSize(destination_sample_rate);
RTC_DCHECK_EQ(NumChannels(destination), frame.num_channels_);
RTC_DCHECK_EQ(SamplesPerChannel(destination),
target_number_of_samples_per_channel);
RTC_CHECK_EQ(destination.data().size(),
frame.num_channels_ * target_number_of_samples_per_channel);
// TODO(yujo): Add special case handling of muted frames.
return resampler->Resample(frame.data_view(), destination);
}
} // namespace
AudioTransportImpl::AudioTransportImpl(
AudioMixer* mixer,
AudioProcessing* audio_processing,
AsyncAudioProcessing::Factory* async_audio_processing_factory)
: audio_processing_(audio_processing),
async_audio_processing_(
async_audio_processing_factory
? async_audio_processing_factory->CreateAsyncAudioProcessing(
[this](std::unique_ptr<AudioFrame> frame) {
this->SendProcessedData(std::move(frame));
})
: nullptr),
mixer_(mixer) {
RTC_DCHECK(mixer);
}
AudioTransportImpl::~AudioTransportImpl() {}
int32_t AudioTransportImpl::RecordedDataIsAvailable(
const void* audio_data,
size_t number_of_frames,
size_t bytes_per_sample,
size_t number_of_channels,
uint32_t sample_rate,
uint32_t audio_delay_milliseconds,
int32_t clock_drift,
uint32_t volume,
bool key_pressed,
uint32_t& new_mic_volume) { // NOLINT: to avoid changing APIs
return RecordedDataIsAvailable(
audio_data, number_of_frames, bytes_per_sample, number_of_channels,
sample_rate, audio_delay_milliseconds, clock_drift, volume, key_pressed,
new_mic_volume, /*estimated_capture_time_ns=*/absl::nullopt);
}
// Not used in Chromium. Process captured audio and distribute to all sending
// streams, and try to do this at the lowest possible sample rate.
int32_t AudioTransportImpl::RecordedDataIsAvailable(
const void* audio_data,
size_t number_of_frames,
size_t bytes_per_sample,
size_t number_of_channels,
uint32_t sample_rate,
uint32_t audio_delay_milliseconds,
int32_t /*clock_drift*/,
uint32_t /*volume*/,
bool key_pressed,
uint32_t& /*new_mic_volume*/,
absl::optional<int64_t>
estimated_capture_time_ns) { // NOLINT: to avoid changing APIs
RTC_DCHECK(audio_data);
RTC_DCHECK_GE(number_of_channels, 1);
RTC_DCHECK_LE(number_of_channels, 2);
RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
// 100 = 1 second / data duration (10 ms).
RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
AudioFrame::kMaxDataSizeBytes);
InterleavedView<const int16_t> source(static_cast<const int16_t*>(audio_data),
number_of_frames, number_of_channels);
int send_sample_rate_hz = 0;
size_t send_num_channels = 0;
bool swap_stereo_channels = false;
{
MutexLock lock(&capture_lock_);
send_sample_rate_hz = send_sample_rate_hz_;
send_num_channels = send_num_channels_;
swap_stereo_channels = swap_stereo_channels_;
}
std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels,
send_num_channels, audio_frame.get());
voe::RemixAndResample(source, sample_rate, &capture_resampler_,
audio_frame.get());
ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
swap_stereo_channels, audio_processing_,
audio_frame.get());
if (estimated_capture_time_ns) {
audio_frame->set_absolute_capture_timestamp_ms(*estimated_capture_time_ns /
1000000);
}
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
if (async_audio_processing_)
async_audio_processing_->Process(std::move(audio_frame));
else
SendProcessedData(std::move(audio_frame));
return 0;
}
void AudioTransportImpl::SendProcessedData(
std::unique_ptr<AudioFrame> audio_frame) {
TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
MutexLock lock(&capture_lock_);
if (audio_senders_.empty())
return;
auto it = audio_senders_.begin();
while (++it != audio_senders_.end()) {
auto audio_frame_copy = std::make_unique<AudioFrame>();
audio_frame_copy->CopyFrom(*audio_frame);
(*it)->SendAudioData(std::move(audio_frame_copy));
}
// Send the original frame to the first stream w/o copying.
(*audio_senders_.begin())->SendAudioData(std::move(audio_frame));
}
// Mix all received streams, feed the result to the AudioProcessing module, then
// resample the result to the requested output rate.
int32_t AudioTransportImpl::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) {
TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
RTC_DCHECK_GE(nChannels, 1);
RTC_DCHECK_LE(nChannels, 2);
RTC_DCHECK_GE(
samplesPerSec,
static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
// 100 = 1 second / data duration (10 ms).
RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
AudioFrame::kMaxDataSizeBytes);
mixer_->Mix(nChannels, &mixed_frame_);
*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
*ntp_time_ms = mixed_frame_.ntp_time_ms_;
if (audio_processing_) {
const auto error =
ProcessReverseAudioFrame(audio_processing_, &mixed_frame_);
RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
}
nSamplesOut =
Resample(mixed_frame_, samplesPerSec, &render_resampler_,
InterleavedView<int16_t>(static_cast<int16_t*>(audioSamples),
nSamples, nChannels));
RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
return 0;
}
// Used by Chromium - same as NeedMorePlayData() but because Chrome has its
// own APM instance, does not call audio_processing_->ProcessReverseStream().
void AudioTransportImpl::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) {
TRACE_EVENT2("webrtc", "AudioTransportImpl::PullRenderData", "sample_rate",
sample_rate, "number_of_frames", number_of_frames);
RTC_DCHECK_EQ(bits_per_sample, 16);
RTC_DCHECK_GE(number_of_channels, 1);
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,
AudioFrame::kMaxDataSizeBytes);
mixer_->Mix(number_of_channels, &mixed_frame_);
*elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
*ntp_time_ms = mixed_frame_.ntp_time_ms_;
int output_samples =
Resample(mixed_frame_, sample_rate, &render_resampler_,
InterleavedView<int16_t>(static_cast<int16_t*>(audio_data),
number_of_frames, number_of_channels));
RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
}
void AudioTransportImpl::UpdateAudioSenders(std::vector<AudioSender*> senders,
int send_sample_rate_hz,
size_t send_num_channels) {
MutexLock lock(&capture_lock_);
audio_senders_ = std::move(senders);
send_sample_rate_hz_ = send_sample_rate_hz;
send_num_channels_ = send_num_channels;
}
void AudioTransportImpl::SetStereoChannelSwapping(bool enable) {
MutexLock lock(&capture_lock_);
swap_stereo_channels_ = enable;
}
} // namespace webrtc