118 lines
4.4 KiB
C++
118 lines
4.4 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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#include "webrtc/voice_engine/utility.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/utility/include/audio_frame_operations.h"
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#include "webrtc/voice_engine/voice_engine_defines.h"
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namespace webrtc {
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namespace voe {
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void RemixAndResample(const AudioFrame& src_frame,
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PushResampler<int16_t>* resampler,
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AudioFrame* dst_frame) {
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RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
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src_frame.num_channels_, src_frame.sample_rate_hz_,
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resampler, dst_frame);
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dst_frame->timestamp_ = src_frame.timestamp_;
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dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
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dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
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}
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void RemixAndResample(const int16_t* src_data,
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size_t samples_per_channel,
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size_t num_channels,
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int sample_rate_hz,
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PushResampler<int16_t>* resampler,
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AudioFrame* dst_frame) {
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const int16_t* audio_ptr = src_data;
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size_t audio_ptr_num_channels = num_channels;
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int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
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// Downmix before resampling.
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if (num_channels == 2 && dst_frame->num_channels_ == 1) {
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AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
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mono_audio);
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audio_ptr = mono_audio;
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audio_ptr_num_channels = 1;
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}
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if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
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audio_ptr_num_channels) == -1) {
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FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
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<< ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
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<< ", audio_ptr_num_channels = " << audio_ptr_num_channels;
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}
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const size_t src_length = samples_per_channel * audio_ptr_num_channels;
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int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
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AudioFrame::kMaxDataSizeSamples);
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if (out_length == -1) {
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FATAL() << "Resample failed: audio_ptr = " << audio_ptr
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<< ", src_length = " << src_length
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<< ", dst_frame->data_ = " << dst_frame->data_;
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}
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dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
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// Upmix after resampling.
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if (num_channels == 1 && dst_frame->num_channels_ == 2) {
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// The audio in dst_frame really is mono at this point; MonoToStereo will
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// set this back to stereo.
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dst_frame->num_channels_ = 1;
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AudioFrameOperations::MonoToStereo(dst_frame);
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}
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}
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void MixWithSat(int16_t target[],
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size_t target_channel,
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const int16_t source[],
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size_t source_channel,
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size_t source_len) {
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RTC_DCHECK_GE(target_channel, 1u);
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RTC_DCHECK_LE(target_channel, 2u);
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RTC_DCHECK_GE(source_channel, 1u);
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RTC_DCHECK_LE(source_channel, 2u);
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if (target_channel == 2 && source_channel == 1) {
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// Convert source from mono to stereo.
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int32_t left = 0;
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int32_t right = 0;
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for (size_t i = 0; i < source_len; ++i) {
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left = source[i] + target[i * 2];
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right = source[i] + target[i * 2 + 1];
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target[i * 2] = WebRtcSpl_SatW32ToW16(left);
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target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right);
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}
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} else if (target_channel == 1 && source_channel == 2) {
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// Convert source from stereo to mono.
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int32_t temp = 0;
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for (size_t i = 0; i < source_len / 2; ++i) {
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temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
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target[i] = WebRtcSpl_SatW32ToW16(temp);
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}
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} else {
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int32_t temp = 0;
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for (size_t i = 0; i < source_len; ++i) {
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temp = source[i] + target[i];
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target[i] = WebRtcSpl_SatW32ToW16(temp);
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}
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}
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}
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} // namespace voe
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} // namespace webrtc
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