40ee3d07ed
Replace the two versions with a single DownConvertToCodecFormat. As mentioned in comments, this could be further consolidated with RemixAndResample but we should write a full audio converter class in that case. Along the way: - Fix the bug present in Channel::Demultiplex with mono input and a stereo codec. - Remove the 32 kHz max from the OnDataAvailable path. This avoids a 48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we get a straight pass-through to ACM. The 32 kHz conversion is still needed in the RecordedDataIsAvailable path until APM natively supports 48 kHz. - Merge resampler improvements from ACM1 to ACM2. This allows ACM to handle 44.1 kHz audio passed to VoE and was originally done here: https://webrtc-codereview.appspot.com/1590004 - Reuse the RemixAndResample unit tests for DownConvertToCodecFormat. - Remove unused functions from utility.cc. BUG=3155,3000,b/12867572 TESTED=voe_cmd_test using both the OnDataAvailable and RecordedDataIsAvailable paths, with a captured audio format of all combinations of {44.1,48} kHz and {1,2} channels, running through all codecs, and finally using both ACM1 and ACM2. R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11019005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
60 lines
2.1 KiB
C++
60 lines
2.1 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/typedefs.h"
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namespace webrtc {
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class AudioFrame;
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class PushResampler;
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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 |num_channels_| and |sample_rate_hz_| set to
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// the desired values. Updates |samples_per_channel_| accordingly.
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//
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// On failure, returns -1 and copies |src_frame| to |dst_frame|.
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void RemixAndResample(const AudioFrame& src_frame,
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PushResampler* 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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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* 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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