1fddd6185d
This method is introduced to try to avoid inconsistent resetting of AudioFrame members to default/uninitialized values. Use it at the call points of DownConvertToCodecFormat(). Results in the following minor functional changes: - speech_activity_ is set to its uninitialized value. AFAICT, this member isn't used at all in the capture path. - timestamp_ is switched from -1 to 0. This member doesn't appear to be used either in the capture path, but left a TODO for wu to change the default value to better represent the uninitialized state. Bonus: Don't copy the frame on error in RemixAndResample(). An error indicates a logical fault (as pointed out by the asserts) that we should not attempt to recover from. BUG=3111 R=turaj@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21519007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6289 4adac7df-926f-26a2-2b94-8c16560cd09d
64 lines
2.4 KiB
C++
64 lines
2.4 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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/*
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* Contains functions often used by different parts of VoiceEngine.
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*/
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#ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_
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#define WEBRTC_VOICE_ENGINE_UTILITY_H_
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioFrame;
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namespace voe {
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// Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
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// Expects |dst_frame| to have its sample rate and channels members set to the
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// desired values. Updates the samples per channel member accordingly. No other
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// members will be changed.
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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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// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
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// specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
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// temporary space and must be of sufficient size to hold the downmixed source
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// audio (recommend using a size of kMaxMonoDataSizeSamples).
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//
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// |dst_af| will have its data and format members (sample rate, channels and
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// samples per channel) set appropriately. No other members will be changed.
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// TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
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// it shouldn't be needed.
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void DownConvertToCodecFormat(const int16_t* src_data,
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int samples_per_channel,
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int num_channels,
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int sample_rate_hz,
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int codec_num_channels,
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int codec_rate_hz,
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int16_t* mono_buffer,
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PushResampler<int16_t>* resampler,
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AudioFrame* dst_af);
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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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int source_len);
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} // namespace voe
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_UTILITY_H_
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