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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}
59 lines
1.8 KiB
C++
59 lines
1.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
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#define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include "api/audio/audio_frame.h"
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#include "common_audio/resampler/include/push_resampler.h"
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namespace webrtc {
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namespace acm2 {
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class ACMResampler {
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public:
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ACMResampler();
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~ACMResampler();
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// TODO: b/335805780 - Change to accept InterleavedView<>.
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int Resample10Msec(const int16_t* in_audio,
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int in_freq_hz,
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int out_freq_hz,
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size_t num_audio_channels,
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size_t out_capacity_samples,
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int16_t* out_audio);
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private:
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PushResampler<int16_t> resampler_;
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};
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// Helper class to perform resampling if needed, meant to be used after
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// receiving the audio_frame from NetEq. Provides reasonably glitch free
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// transitions between different output sample rates from NetEq.
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class ResamplerHelper {
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public:
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ResamplerHelper();
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// Resamples audio_frame if it is not already in desired_sample_rate_hz.
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bool MaybeResample(int desired_sample_rate_hz, AudioFrame* audio_frame);
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private:
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ACMResampler resampler_;
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bool resampled_last_output_frame_ = true;
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std::array<int16_t, AudioFrame::kMaxDataSizeSamples> last_audio_buffer_;
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
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