31628aae7e
Analogous to the recent libjingle change: http://cl/54929753-p10. This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather than scoped_array and scoped_ptr_malloc respectively. - Add Chromium's template-based COMPILE_ASSERT. We didn't have this previously in order to support the macro in C. Instead, move the existing macro to compile_assert_c.h. - Additionally copy the move.h and template_util.h depedencies and add the WARN_UNUSED_RESULT macro. - Leave scoped_array and scoped_ptr_malloc for now, but mark as deprecated. - Remove scoped_ptr foo(NULL) use. The default constructor handles it. - Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc. - Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove some repeated code. TESTED=trybots R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2449005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
102 lines
3.7 KiB
C++
102 lines
3.7 KiB
C++
/*
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* Copyright (c) 2013 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/common_audio/resampler/include/push_resampler.h"
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#include <string.h>
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
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namespace webrtc {
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PushResampler::PushResampler()
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: src_sample_rate_hz_(0),
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dst_sample_rate_hz_(0),
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num_channels_(0),
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src_left_(NULL),
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src_right_(NULL),
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dst_left_(NULL),
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dst_right_(NULL) {
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}
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PushResampler::~PushResampler() {
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}
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int PushResampler::InitializeIfNeeded(int src_sample_rate_hz,
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int dst_sample_rate_hz,
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int num_channels) {
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if (src_sample_rate_hz == src_sample_rate_hz_ &&
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dst_sample_rate_hz == dst_sample_rate_hz_ &&
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num_channels == num_channels_)
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// No-op if settings haven't changed.
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return 0;
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if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
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num_channels <= 0 || num_channels > 2)
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return -1;
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src_sample_rate_hz_ = src_sample_rate_hz;
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dst_sample_rate_hz_ = dst_sample_rate_hz;
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num_channels_ = num_channels;
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const int src_size_10ms_mono = src_sample_rate_hz / 100;
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const int dst_size_10ms_mono = dst_sample_rate_hz / 100;
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sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
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dst_size_10ms_mono));
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if (num_channels_ == 2) {
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src_left_.reset(new int16_t[src_size_10ms_mono]);
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src_right_.reset(new int16_t[src_size_10ms_mono]);
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dst_left_.reset(new int16_t[dst_size_10ms_mono]);
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dst_right_.reset(new int16_t[dst_size_10ms_mono]);
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sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
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dst_size_10ms_mono));
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}
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return 0;
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}
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int PushResampler::Resample(const int16_t* src, int src_length,
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int16_t* dst, int dst_capacity) {
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const int src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
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const int dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
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if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
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return -1;
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if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
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// The old resampler provides this memcpy facility in the case of matching
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// sample rates, so reproduce it here for the sinc resampler.
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memcpy(dst, src, src_length * sizeof(int16_t));
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return src_length;
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}
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if (num_channels_ == 2) {
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const int src_length_mono = src_length / num_channels_;
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const int dst_capacity_mono = dst_capacity / num_channels_;
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int16_t* deinterleaved[] = {src_left_.get(), src_right_.get()};
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Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
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int dst_length_mono =
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sinc_resampler_->Resample(src_left_.get(), src_length_mono,
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dst_left_.get(), dst_capacity_mono);
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sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
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dst_right_.get(), dst_capacity_mono);
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deinterleaved[0] = dst_left_.get();
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deinterleaved[1] = dst_right_.get();
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Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
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return dst_length_mono * num_channels_;
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} else {
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return sinc_resampler_->Resample(src, src_length, dst, dst_capacity);
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}
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}
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} // namespace webrtc
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