mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
1f3679884c
Start introducing ArrayView to AudioFrame and code that flows down from there. In this first step: * Add `data_view()` that returns a read-only ArrayView for the audio buffer. When AudioFrame is not initialized however, data_view() will return a nullptr whereas the current data() method never returns nullptr. * Add `mutable_data()` that requires two arguments for properly setting the samples per channel and number of channels that's required for accurately reserving the returned mutable ArrayView. A notable behavior change is that if the requested number of channels is larger than supported or the calculated buffer size is too large, the function will trigger a check. * Add TODOs for following work. Bug: chromium:335805780 Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42202}
99 lines
4.1 KiB
C++
99 lines
4.1 KiB
C++
/*
|
|
* Copyright (c) 2012 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/remix_resample.h"
|
|
|
|
#include "api/audio/audio_frame.h"
|
|
#include "audio/utility/audio_frame_operations.h"
|
|
#include "common_audio/resampler/include/push_resampler.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
namespace voe {
|
|
|
|
void RemixAndResample(const AudioFrame& src_frame,
|
|
PushResampler<int16_t>* resampler,
|
|
AudioFrame* dst_frame) {
|
|
RemixAndResample(src_frame.data(), src_frame.samples_per_channel_,
|
|
src_frame.num_channels_, src_frame.sample_rate_hz_,
|
|
resampler, dst_frame);
|
|
dst_frame->timestamp_ = src_frame.timestamp_;
|
|
dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
|
|
dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
|
|
dst_frame->packet_infos_ = src_frame.packet_infos_;
|
|
}
|
|
|
|
void RemixAndResample(const int16_t* src_data,
|
|
size_t samples_per_channel,
|
|
size_t num_channels,
|
|
int sample_rate_hz,
|
|
PushResampler<int16_t>* resampler,
|
|
AudioFrame* dst_frame) {
|
|
const int16_t* audio_ptr = src_data;
|
|
size_t audio_ptr_num_channels = num_channels;
|
|
int16_t downmixed_audio[AudioFrame::kMaxDataSizeSamples];
|
|
|
|
// Downmix before resampling.
|
|
if (num_channels > dst_frame->num_channels_) {
|
|
RTC_DCHECK(num_channels == 2 || num_channels == 4)
|
|
<< "num_channels: " << num_channels;
|
|
RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
|
|
<< "dst_frame->num_channels_: " << dst_frame->num_channels_;
|
|
|
|
AudioFrameOperations::DownmixChannels(
|
|
src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
|
|
downmixed_audio);
|
|
audio_ptr = downmixed_audio;
|
|
audio_ptr_num_channels = dst_frame->num_channels_;
|
|
}
|
|
|
|
if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
|
|
audio_ptr_num_channels) == -1) {
|
|
RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
|
|
<< sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
|
|
<< dst_frame->sample_rate_hz_
|
|
<< ", audio_ptr_num_channels = " << audio_ptr_num_channels;
|
|
}
|
|
|
|
// TODO(yujo): for muted input frames, don't resample. Either 1) allow
|
|
// resampler to return output length without doing the resample, so we know
|
|
// how much to zero here; or 2) make resampler accept a hint that the input is
|
|
// zeroed.
|
|
const size_t src_length = samples_per_channel * audio_ptr_num_channels;
|
|
// Ensure the `samples_per_channel_` member is set correctly based on the
|
|
// destination sample rate, number of channels and assumed 10ms buffer size.
|
|
// TODO(tommi): Could we rather assume that this has been done by the caller?
|
|
dst_frame->SetSampleRateAndChannelSize(dst_frame->sample_rate_hz_);
|
|
|
|
int out_length = resampler->Resample(
|
|
rtc::ArrayView<const int16_t>(audio_ptr, src_length),
|
|
dst_frame->mutable_data(dst_frame->samples_per_channel_,
|
|
dst_frame->num_channels_));
|
|
if (out_length == -1) {
|
|
RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
|
|
<< ", src_length = " << src_length
|
|
<< ", dst_frame->mutable_data() = "
|
|
<< dst_frame->mutable_data();
|
|
}
|
|
|
|
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
|
|
|
|
// Upmix after resampling.
|
|
if (num_channels == 1 && dst_frame->num_channels_ == 2) {
|
|
// The audio in dst_frame really is mono at this point; MonoToStereo will
|
|
// set this back to stereo.
|
|
dst_frame->num_channels_ = 1;
|
|
AudioFrameOperations::UpmixChannels(2, dst_frame);
|
|
}
|
|
}
|
|
|
|
} // namespace voe
|
|
} // namespace webrtc
|