36b1a5fcec
const int16_t* data() const; int16_t* mutable_data(); - data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames. - mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_. These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation. This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later. BUG=webrtc:7343 TBR=henrika Review-Url: https://codereview.webrtc.org/2750783004 Cr-Commit-Position: refs/heads/master@{#18543}
129 lines
5.0 KiB
C++
129 lines
5.0 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/audio/utility/audio_frame_operations.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/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 downsmixed_audio[AudioFrame::kMaxDataSizeSamples];
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// Downmix before resampling.
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if (num_channels > dst_frame->num_channels_) {
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RTC_DCHECK(num_channels == 2 || num_channels == 4)
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<< "num_channels: " << num_channels;
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RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
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<< "dst_frame->num_channels_: " << dst_frame->num_channels_;
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AudioFrameOperations::DownmixChannels(
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src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
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downsmixed_audio);
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audio_ptr = downsmixed_audio;
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audio_ptr_num_channels = dst_frame->num_channels_;
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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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// TODO(yujo): for muted input frames, don't resample. Either 1) allow
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// resampler to return output length without doing the resample, so we know
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// how much to zero here; or 2) make resampler accept a hint that the input is
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// zeroed.
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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,
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dst_frame->mutable_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->mutable_data() = " << dst_frame->mutable_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, 1);
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RTC_DCHECK_LE(target_channel, 2);
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RTC_DCHECK_GE(source_channel, 1);
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RTC_DCHECK_LE(source_channel, 2);
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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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