ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
115 lines
4.2 KiB
C++
115 lines
4.2 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/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/system_wrappers/include/logging.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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int 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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int 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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LOG_FERR3(LS_ERROR, InitializeIfNeeded, sample_rate_hz,
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dst_frame->sample_rate_hz_, audio_ptr_num_channels);
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assert(false);
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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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LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_);
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assert(false);
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}
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dst_frame->samples_per_channel_ =
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static_cast<size_t>(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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int target_channel,
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const int16_t source[],
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int source_channel,
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size_t source_len) {
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assert(target_channel == 1 || target_channel == 2);
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assert(source_channel == 1 || 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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