mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
c9aaf11985
ChannelReceive is now owning and interfacing with NetEq directly. A new ResamplerHelper is added to acm_resampler.cc/.h, to do the audio resampling that was previously done inside AcmReceiver. AcmReceiver still remains, since it is used in other places for now. Bug: webrtc:14867 Change-Id: If3eb6415e06b9b5e729d393713f3fccb31b0570f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361820 Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42974}
122 lines
4.4 KiB
C++
122 lines
4.4 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 "modules/audio_coding/acm2/acm_resampler.h"
|
|
|
|
#include <string.h>
|
|
|
|
#include "api/audio/audio_frame.h"
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
namespace acm2 {
|
|
|
|
ACMResampler::ACMResampler() {}
|
|
|
|
ACMResampler::~ACMResampler() {}
|
|
|
|
int ACMResampler::Resample10Msec(const int16_t* in_audio,
|
|
int in_freq_hz,
|
|
int out_freq_hz,
|
|
size_t num_audio_channels,
|
|
size_t out_capacity_samples,
|
|
int16_t* out_audio) {
|
|
InterleavedView<const int16_t> src(
|
|
in_audio, SampleRateToDefaultChannelSize(in_freq_hz), num_audio_channels);
|
|
InterleavedView<int16_t> dst(out_audio,
|
|
SampleRateToDefaultChannelSize(out_freq_hz),
|
|
num_audio_channels);
|
|
RTC_DCHECK_GE(out_capacity_samples, dst.size());
|
|
if (in_freq_hz == out_freq_hz) {
|
|
if (out_capacity_samples < src.data().size()) {
|
|
RTC_DCHECK_NOTREACHED();
|
|
return -1;
|
|
}
|
|
CopySamples(dst, src);
|
|
RTC_DCHECK_EQ(dst.samples_per_channel(), src.samples_per_channel());
|
|
return static_cast<int>(dst.samples_per_channel());
|
|
}
|
|
|
|
int out_length = resampler_.Resample(src, dst);
|
|
if (out_length == -1) {
|
|
RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << src.data().size()
|
|
<< ", " << out_audio << ", " << out_capacity_samples
|
|
<< ") failed.";
|
|
return -1;
|
|
}
|
|
RTC_DCHECK_EQ(out_length, dst.size());
|
|
RTC_DCHECK_EQ(out_length / num_audio_channels, dst.samples_per_channel());
|
|
return static_cast<int>(dst.samples_per_channel());
|
|
}
|
|
|
|
ResamplerHelper::ResamplerHelper() {
|
|
ClearSamples(last_audio_buffer_);
|
|
}
|
|
|
|
bool ResamplerHelper::MaybeResample(int desired_sample_rate_hz,
|
|
AudioFrame* audio_frame) {
|
|
const int current_sample_rate_hz = audio_frame->sample_rate_hz_;
|
|
RTC_DCHECK_NE(current_sample_rate_hz, 0);
|
|
|
|
// Update if resampling is required.
|
|
const bool need_resampling =
|
|
(desired_sample_rate_hz != -1) &&
|
|
(current_sample_rate_hz != desired_sample_rate_hz);
|
|
|
|
if (need_resampling && !resampled_last_output_frame_) {
|
|
// Prime the resampler with the last frame.
|
|
int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
|
|
int samples_per_channel_int = resampler_.Resample10Msec(
|
|
last_audio_buffer_.data(), current_sample_rate_hz,
|
|
desired_sample_rate_hz, audio_frame->num_channels_,
|
|
AudioFrame::kMaxDataSizeSamples, temp_output);
|
|
if (samples_per_channel_int < 0) {
|
|
RTC_LOG(LS_ERROR) << "AcmReceiver::GetAudio - "
|
|
"Resampling last_audio_buffer_ failed.";
|
|
return false;
|
|
}
|
|
}
|
|
|
|
// TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output
|
|
// rate from NetEq changes.
|
|
if (need_resampling) {
|
|
// TODO(yujo): handle this more efficiently for muted frames.
|
|
int samples_per_channel_int = resampler_.Resample10Msec(
|
|
audio_frame->data(), current_sample_rate_hz, desired_sample_rate_hz,
|
|
audio_frame->num_channels_, AudioFrame::kMaxDataSizeSamples,
|
|
audio_frame->mutable_data());
|
|
if (samples_per_channel_int < 0) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
|
|
return false;
|
|
}
|
|
audio_frame->samples_per_channel_ =
|
|
static_cast<size_t>(samples_per_channel_int);
|
|
audio_frame->sample_rate_hz_ = desired_sample_rate_hz;
|
|
RTC_DCHECK_EQ(
|
|
audio_frame->sample_rate_hz_,
|
|
rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
|
|
resampled_last_output_frame_ = true;
|
|
} else {
|
|
resampled_last_output_frame_ = false;
|
|
// We might end up here ONLY if codec is changed.
|
|
}
|
|
|
|
// Store current audio in `last_audio_buffer_` for next time.
|
|
// TODO: b/335805780 - Use CopySamples().
|
|
memcpy(last_audio_buffer_.data(), audio_frame->data(),
|
|
sizeof(int16_t) * audio_frame->samples_per_channel_ *
|
|
audio_frame->num_channels_);
|
|
|
|
return true;
|
|
}
|
|
|
|
} // namespace acm2
|
|
} // namespace webrtc
|