blob: 38206959b27a104587d0b2fc269fc8330b3a52b6 [file] [log] [blame]
* 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.
#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,
int 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,
int source_len);
} // namespace voe
} // namespace webrtc