Files
platform-external-webrtc/webrtc/voice_engine/utility.h
Peter Kasting dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00

64 lines
2.4 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.
*/
/*
* Contains functions often used by different parts of VoiceEngine.
*/
#ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_
#define WEBRTC_VOICE_ENGINE_UTILITY_H_
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
namespace voe {
// Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
// Expects |dst_frame| to have its sample rate and channels members set to the
// desired values. Updates the samples per channel member accordingly. No other
// members will be changed.
void RemixAndResample(const AudioFrame& src_frame,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
// specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
// temporary space and must be of sufficient size to hold the downmixed source
// audio (recommend using a size of kMaxMonoDataSizeSamples).
//
// |dst_af| will have its data and format members (sample rate, channels and
// samples per channel) set appropriately. No other members will be changed.
// TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
// it shouldn't be needed.
void DownConvertToCodecFormat(const int16_t* src_data,
size_t samples_per_channel,
int num_channels,
int sample_rate_hz,
int codec_num_channels,
int codec_rate_hz,
int16_t* mono_buffer,
PushResampler<int16_t>* resampler,
AudioFrame* dst_af);
void MixWithSat(int16_t target[],
int target_channel,
const int16_t source[],
int source_channel,
size_t source_len);
} // namespace voe
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_UTILITY_H_