Add a wrapper around PushSincResampler and the old Resampler.
The old resampler is used whenever it supports the requested rates. Otherwise the sinc resampler is enabled. Integrated with output_mixer in order to test the change through output_mixer_unittest. The sinc resampler will not yet be used, since we don't feed VoE with any rates that trigger it. BUG=webrtc:1395 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1355004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
parent
5b7120c81b
commit
50b2efef6e
41
webrtc/common_audio/audio_util.cc
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41
webrtc/common_audio/audio_util.cc
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/*
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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/include/audio_util.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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void Deinterleave(const int16_t* interleaved, int samples_per_channel,
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int num_channels, int16_t** deinterleaved) {
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for (int i = 0; i < num_channels; i++) {
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int16_t* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; j++) {
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channel[j] = interleaved[interleaved_idx];
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interleaved_idx += num_channels;
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}
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}
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}
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void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
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int num_channels, int16_t* interleaved) {
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for (int i = 0; i < num_channels; ++i) {
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const int16_t* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; j++) {
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interleaved[interleaved_idx] = channel[j];
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interleaved_idx += num_channels;
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}
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}
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}
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} // namespace webrtc
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55
webrtc/common_audio/audio_util_unittest.cc
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55
webrtc/common_audio/audio_util_unittest.cc
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/*
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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 "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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void ExpectArraysEq(const int16_t* ref, const int16_t* test, int length) {
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for (int i = 0; i < length; ++i) {
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EXPECT_EQ(test[i], ref[i]);
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}
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}
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TEST(AudioUtilTest, InterleavingStereo) {
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const int16_t kInterleaved[] = {2, 3, 4, 9, 8, 27, 16, 81};
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const int kSamplesPerChannel = 4;
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const int kNumChannels = 2;
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const int kLength = kSamplesPerChannel * kNumChannels;
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int16_t left[kSamplesPerChannel], right[kSamplesPerChannel];
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int16_t* deinterleaved[] = {left, right};
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Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
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const int16_t kRefLeft[] = {2, 4, 8, 16};
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const int16_t kRefRight[] = {3, 9, 27, 81};
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ExpectArraysEq(left, kRefLeft, kSamplesPerChannel);
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ExpectArraysEq(right, kRefRight, kSamplesPerChannel);
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int16_t interleaved[kLength];
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Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
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ExpectArraysEq(interleaved, kInterleaved, kLength);
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}
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TEST(AudioUtilTest, InterleavingMonoIsIdentical) {
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const int16_t kInterleaved[] = {1, 2, 3, 4, 5};
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const int kSamplesPerChannel = 5;
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const int kNumChannels = 1;
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int16_t mono[kSamplesPerChannel];
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int16_t* deinterleaved[] = {mono};
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Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
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ExpectArraysEq(mono, kInterleaved, kSamplesPerChannel);
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int16_t interleaved[kSamplesPerChannel];
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Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
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ExpectArraysEq(interleaved, mono, kSamplesPerChannel);
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}
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} // namespace webrtc
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33
webrtc/common_audio/include/audio_util.h
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33
webrtc/common_audio/include/audio_util.h
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/*
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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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#ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Deinterleave audio from |interleaved| to the channel buffers pointed to
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// by |deinterleaved|. There must be sufficient space allocated in the
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// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
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// per buffer).
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void Deinterleave(const int16_t* interleaved, int samples_per_channel,
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int num_channels, int16_t** deinterleaved);
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// Interleave audio from the channel buffers pointed to by |deinterleaved| to
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// |interleaved|. There must be sufficient space allocated in |interleaved|
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// (|samples_per_channel| * |num_channels|).
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void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
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int num_channels, int16_t* interleaved);
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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61
webrtc/common_audio/resampler/include/push_resampler.h
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61
webrtc/common_audio/resampler/include/push_resampler.h
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/*
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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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#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Resampler;
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class PushSincResampler;
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// Wraps the old resampler and new arbitrary rate conversion resampler. The
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// old resampler will be used whenever it supports the requested rates, and
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// otherwise the sinc resampler will be enabled.
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class PushResampler {
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public:
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PushResampler();
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virtual ~PushResampler();
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// Must be called whenever the parameters change. Free to be called at any
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// time as it is a no-op if parameters have not changed since the last call.
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int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
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int num_channels);
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// Returns the total number of samples provided in destination (e.g. 32 kHz,
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// 2 channel audio gives 640 samples).
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int Resample(const int16_t* src, int src_length, int16_t* dst,
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int dst_capacity);
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bool use_sinc_resampler() const { return use_sinc_resampler_; }
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private:
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int ResampleSinc(const int16_t* src, int src_length, int16_t* dst,
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int dst_capacity);
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scoped_ptr<Resampler> resampler_;
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scoped_ptr<PushSincResampler> sinc_resampler_;
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scoped_ptr<PushSincResampler> sinc_resampler_right_;
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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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bool use_sinc_resampler_;
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scoped_array<int16_t> src_left_;
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scoped_array<int16_t> src_right_;
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scoped_array<int16_t> dst_left_;
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scoped_array<int16_t> dst_right_;
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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133
webrtc/common_audio/resampler/push_resampler.cc
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133
webrtc/common_audio/resampler/push_resampler.cc
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@ -0,0 +1,133 @@
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/*
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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 <cstring>
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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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// Requires valid values at construction, so give it something arbitrary.
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: resampler_(new Resampler(48000, 48000, kResamplerSynchronous)),
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sinc_resampler_(NULL),
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sinc_resampler_right_(NULL),
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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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use_sinc_resampler_(false),
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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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}
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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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}
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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 ResamplerType resampler_type =
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num_channels == 1 ? kResamplerSynchronous : kResamplerSynchronousStereo;
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if (resampler_->Reset(src_sample_rate_hz, dst_sample_rate_hz,
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resampler_type) == 0) {
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// The resampler supports these rates.
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use_sinc_resampler_ = false;
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return 0;
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}
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use_sinc_resampler_ = true;
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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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}
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if (use_sinc_resampler_) {
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return ResampleSinc(src, src_length, dst, dst_capacity);
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}
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int resulting_length = 0;
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if (resampler_->Push(src, src_length, dst, dst_capacity,
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resulting_length) != 0) {
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return -1;
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}
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return resulting_length;
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}
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int PushResampler::ResampleSinc(const int16_t* src, int src_length,
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int16_t* dst, int dst_capacity) {
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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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107
webrtc/common_audio/resampler/push_resampler_unittest.cc
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107
webrtc/common_audio/resampler/push_resampler_unittest.cc
Normal file
@ -0,0 +1,107 @@
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/*
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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
|
||||
* 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.
|
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*/
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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// Quality testing of PushResampler is handled through output_mixer_unittest.cc.
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namespace webrtc {
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typedef std::tr1::tuple<int, int, bool> PushResamplerTestData;
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class PushResamplerTest
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: public testing::TestWithParam<PushResamplerTestData> {
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public:
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PushResamplerTest()
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: input_rate_(std::tr1::get<0>(GetParam())),
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output_rate_(std::tr1::get<1>(GetParam())),
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use_sinc_resampler_(std::tr1::get<2>(GetParam())) {
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}
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virtual ~PushResamplerTest() {}
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protected:
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int input_rate_;
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int output_rate_;
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bool use_sinc_resampler_;
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};
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TEST_P(PushResamplerTest, SincResamplerOnlyUsedWhenNecessary) {
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PushResampler resampler;
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resampler.InitializeIfNeeded(input_rate_, output_rate_, 1);
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EXPECT_EQ(use_sinc_resampler_, resampler.use_sinc_resampler());
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}
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INSTANTIATE_TEST_CASE_P(
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PushResamplerTest, PushResamplerTest, testing::Values(
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// To 8 kHz
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std::tr1::make_tuple(8000, 8000, false),
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std::tr1::make_tuple(16000, 8000, false),
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std::tr1::make_tuple(32000, 8000, false),
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std::tr1::make_tuple(44100, 8000, true),
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std::tr1::make_tuple(48000, 8000, false),
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std::tr1::make_tuple(96000, 8000, false),
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std::tr1::make_tuple(192000, 8000, true),
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// To 16 kHz
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std::tr1::make_tuple(8000, 16000, false),
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std::tr1::make_tuple(16000, 16000, false),
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std::tr1::make_tuple(32000, 16000, false),
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std::tr1::make_tuple(44100, 16000, true),
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std::tr1::make_tuple(48000, 16000, false),
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std::tr1::make_tuple(96000, 16000, false),
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std::tr1::make_tuple(192000, 16000, false),
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// To 32 kHz
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std::tr1::make_tuple(8000, 32000, false),
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std::tr1::make_tuple(16000, 32000, false),
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std::tr1::make_tuple(32000, 32000, false),
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std::tr1::make_tuple(44100, 32000, true),
|
||||
std::tr1::make_tuple(48000, 32000, false),
|
||||
std::tr1::make_tuple(96000, 32000, false),
|
||||
std::tr1::make_tuple(192000, 32000, false),
|
||||
|
||||
// To 44.1kHz
|
||||
std::tr1::make_tuple(8000, 44100, true),
|
||||
std::tr1::make_tuple(16000, 44100, true),
|
||||
std::tr1::make_tuple(32000, 44100, true),
|
||||
std::tr1::make_tuple(44100, 44100, false),
|
||||
std::tr1::make_tuple(48000, 44100, true),
|
||||
std::tr1::make_tuple(96000, 44100, true),
|
||||
std::tr1::make_tuple(192000, 44100, true),
|
||||
|
||||
// To 48kHz
|
||||
std::tr1::make_tuple(8000, 48000, false),
|
||||
std::tr1::make_tuple(16000, 48000, false),
|
||||
std::tr1::make_tuple(32000, 48000, false),
|
||||
std::tr1::make_tuple(44100, 48000, true),
|
||||
std::tr1::make_tuple(48000, 48000, false),
|
||||
std::tr1::make_tuple(96000, 48000, false),
|
||||
std::tr1::make_tuple(192000, 48000, false),
|
||||
|
||||
// To 96kHz
|
||||
std::tr1::make_tuple(8000, 96000, false),
|
||||
std::tr1::make_tuple(16000, 96000, false),
|
||||
std::tr1::make_tuple(32000, 96000, false),
|
||||
std::tr1::make_tuple(44100, 96000, true),
|
||||
std::tr1::make_tuple(48000, 96000, false),
|
||||
std::tr1::make_tuple(96000, 96000, false),
|
||||
std::tr1::make_tuple(192000, 96000, false),
|
||||
|
||||
// To 192kHz
|
||||
std::tr1::make_tuple(8000, 192000, true),
|
||||
std::tr1::make_tuple(16000, 192000, false),
|
||||
std::tr1::make_tuple(32000, 192000, false),
|
||||
std::tr1::make_tuple(44100, 192000, true),
|
||||
std::tr1::make_tuple(48000, 192000, false),
|
||||
std::tr1::make_tuple(96000, 192000, false),
|
||||
std::tr1::make_tuple(192000, 192000, false)));
|
||||
|
||||
} // namespace webrtc
|
||||
@ -23,7 +23,13 @@
|
||||
],
|
||||
},
|
||||
'sources': [
|
||||
# TODO(ajm): Adding audio_util here for now. We should transition
|
||||
# to having a single common_audio target.
|
||||
'../audio_util.cc',
|
||||
'../include/audio_util.h',
|
||||
'include/push_resampler.h',
|
||||
'include/resampler.h',
|
||||
'push_resampler.cc',
|
||||
'push_sinc_resampler.cc',
|
||||
'push_sinc_resampler.h',
|
||||
'resampler.cc',
|
||||
@ -45,7 +51,9 @@
|
||||
'<(DEPTH)/testing/gtest.gyp:gtest',
|
||||
],
|
||||
'sources': [
|
||||
'../audio_util_unittest.cc',
|
||||
'resampler_unittest.cc',
|
||||
'push_resampler_unittest.cc',
|
||||
'push_sinc_resampler_unittest.cc',
|
||||
'sinc_resampler_unittest.cc',
|
||||
'sinusoidal_linear_chirp_source.cc',
|
||||
|
||||
@ -8,16 +8,16 @@
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "output_mixer.h"
|
||||
#include "webrtc/voice_engine/output_mixer.h"
|
||||
|
||||
#include "audio_processing.h"
|
||||
#include "audio_frame_operations.h"
|
||||
#include "critical_section_wrapper.h"
|
||||
#include "file_wrapper.h"
|
||||
#include "output_mixer_internal.h"
|
||||
#include "statistics.h"
|
||||
#include "trace.h"
|
||||
#include "voe_external_media.h"
|
||||
#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
||||
#include "webrtc/modules/utility/interface/audio_frame_operations.h"
|
||||
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
|
||||
#include "webrtc/system_wrappers/interface/file_wrapper.h"
|
||||
#include "webrtc/system_wrappers/interface/trace.h"
|
||||
#include "webrtc/voice_engine/include/voe_external_media.h"
|
||||
#include "webrtc/voice_engine/output_mixer_internal.h"
|
||||
#include "webrtc/voice_engine/statistics.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
@ -528,7 +528,7 @@ int OutputMixer::GetMixedAudio(int sample_rate_hz,
|
||||
frame->sample_rate_hz_ = sample_rate_hz;
|
||||
// TODO(andrew): Ideally the downmixing would occur much earlier, in
|
||||
// AudioCodingModule.
|
||||
return RemixAndResample(_audioFrame, &_resampler, frame);
|
||||
return RemixAndResample(_audioFrame, &resampler_, frame);
|
||||
}
|
||||
|
||||
int32_t
|
||||
@ -602,7 +602,7 @@ void OutputMixer::APMAnalyzeReverseStream() {
|
||||
AudioFrame frame;
|
||||
frame.num_channels_ = 1;
|
||||
frame.sample_rate_hz_ = _audioProcessingModulePtr->sample_rate_hz();
|
||||
if (RemixAndResample(_audioFrame, &_apmResampler, &frame) == -1)
|
||||
if (RemixAndResample(_audioFrame, &audioproc_resampler_, &frame) == -1)
|
||||
return;
|
||||
|
||||
if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) == -1) {
|
||||
|
||||
@ -11,14 +11,14 @@
|
||||
#ifndef WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H_
|
||||
#define WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H_
|
||||
|
||||
#include "audio_conference_mixer.h"
|
||||
#include "audio_conference_mixer_defines.h"
|
||||
#include "common_types.h"
|
||||
#include "dtmf_inband.h"
|
||||
#include "file_recorder.h"
|
||||
#include "level_indicator.h"
|
||||
#include "resampler.h"
|
||||
#include "voice_engine_defines.h"
|
||||
#include "webrtc/common_audio/resampler/include/push_resampler.h"
|
||||
#include "webrtc/common_types.h"
|
||||
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
|
||||
#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
|
||||
#include "webrtc/modules/utility/interface/file_recorder.h"
|
||||
#include "webrtc/voice_engine/dtmf_inband.h"
|
||||
#include "webrtc/voice_engine/level_indicator.h"
|
||||
#include "webrtc/voice_engine/voice_engine_defines.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
@ -133,8 +133,8 @@ private:
|
||||
CriticalSectionWrapper& _fileCritSect;
|
||||
AudioConferenceMixer& _mixerModule;
|
||||
AudioFrame _audioFrame;
|
||||
Resampler _resampler; // converts mixed audio to fit ADM format
|
||||
Resampler _apmResampler; // converts mixed audio to fit APM rate
|
||||
PushResampler resampler_; // converts mixed audio to fit ADM format
|
||||
PushResampler audioproc_resampler_; // converts mixed audio to fit APM rate
|
||||
AudioLevel _audioLevel; // measures audio level for the combined signal
|
||||
DtmfInband _dtmfGenerator;
|
||||
int _instanceId;
|
||||
|
||||
@ -8,18 +8,19 @@
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "output_mixer_internal.h"
|
||||
#include "webrtc/voice_engine/output_mixer_internal.h"
|
||||
|
||||
#include "audio_frame_operations.h"
|
||||
#include "common_audio/resampler/include/resampler.h"
|
||||
#include "module_common_types.h"
|
||||
#include "trace.h"
|
||||
#include "webrtc/common_audio/resampler/include/push_resampler.h"
|
||||
#include "webrtc/modules/interface/module_common_types.h"
|
||||
#include "webrtc/modules/utility/interface/audio_frame_operations.h"
|
||||
#include "webrtc/system_wrappers/interface/logging.h"
|
||||
#include "webrtc/system_wrappers/interface/trace.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace voe {
|
||||
|
||||
int RemixAndResample(const AudioFrame& src_frame,
|
||||
Resampler* resampler,
|
||||
PushResampler* resampler,
|
||||
AudioFrame* dst_frame) {
|
||||
const int16_t* audio_ptr = src_frame.data_;
|
||||
int audio_ptr_num_channels = src_frame.num_channels_;
|
||||
@ -34,30 +35,26 @@ int RemixAndResample(const AudioFrame& src_frame,
|
||||
audio_ptr_num_channels = 1;
|
||||
}
|
||||
|
||||
const ResamplerType resampler_type = audio_ptr_num_channels == 1 ?
|
||||
kResamplerSynchronous : kResamplerSynchronousStereo;
|
||||
if (resampler->ResetIfNeeded(src_frame.sample_rate_hz_,
|
||||
dst_frame->sample_rate_hz_,
|
||||
resampler_type) == -1) {
|
||||
if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
|
||||
dst_frame->sample_rate_hz_,
|
||||
audio_ptr_num_channels) == -1) {
|
||||
dst_frame->CopyFrom(src_frame);
|
||||
WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
|
||||
"%s ResetIfNeeded failed", __FUNCTION__);
|
||||
LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
|
||||
dst_frame->sample_rate_hz_, audio_ptr_num_channels);
|
||||
return -1;
|
||||
}
|
||||
|
||||
int out_length = 0;
|
||||
if (resampler->Push(audio_ptr,
|
||||
src_frame.samples_per_channel_* audio_ptr_num_channels,
|
||||
dst_frame->data_,
|
||||
AudioFrame::kMaxDataSizeSamples,
|
||||
out_length) == 0) {
|
||||
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
|
||||
} else {
|
||||
const int src_length = src_frame.samples_per_channel_ *
|
||||
audio_ptr_num_channels;
|
||||
int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
|
||||
AudioFrame::kMaxDataSizeSamples);
|
||||
if (out_length == -1) {
|
||||
dst_frame->CopyFrom(src_frame);
|
||||
WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
|
||||
"%s resampling failed", __FUNCTION__);
|
||||
LOG_FERR3(LS_ERROR, Resample, src_length, dst_frame->data_,
|
||||
AudioFrame::kMaxDataSizeSamples);
|
||||
return -1;
|
||||
}
|
||||
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
|
||||
|
||||
// Upmix after resampling.
|
||||
if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
|
||||
|
||||
@ -14,7 +14,7 @@
|
||||
namespace webrtc {
|
||||
|
||||
class AudioFrame;
|
||||
class Resampler;
|
||||
class PushResampler;
|
||||
|
||||
namespace voe {
|
||||
|
||||
@ -24,7 +24,7 @@ namespace voe {
|
||||
//
|
||||
// On failure, returns -1 and copies |src_frame| to |dst_frame|.
|
||||
int RemixAndResample(const AudioFrame& src_frame,
|
||||
Resampler* resampler,
|
||||
PushResampler* resampler,
|
||||
AudioFrame* dst_frame);
|
||||
|
||||
} // namespace voe
|
||||
|
||||
@ -10,10 +10,9 @@
|
||||
|
||||
#include <math.h>
|
||||
|
||||
#include "gtest/gtest.h"
|
||||
|
||||
#include "output_mixer.h"
|
||||
#include "output_mixer_internal.h"
|
||||
#include "testing/gtest/include/gtest/gtest.h"
|
||||
#include "webrtc/voice_engine/output_mixer.h"
|
||||
#include "webrtc/voice_engine/output_mixer_internal.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace voe {
|
||||
@ -32,7 +31,7 @@ class OutputMixerTest : public ::testing::Test {
|
||||
void RunResampleTest(int src_channels, int src_sample_rate_hz,
|
||||
int dst_channels, int dst_sample_rate_hz);
|
||||
|
||||
Resampler resampler_;
|
||||
PushResampler resampler_;
|
||||
AudioFrame src_frame_;
|
||||
AudioFrame dst_frame_;
|
||||
AudioFrame golden_frame_;
|
||||
@ -42,6 +41,7 @@ class OutputMixerTest : public ::testing::Test {
|
||||
// used so non-integer values result in rounding error, but not an accumulating
|
||||
// error.
|
||||
void SetMonoFrame(AudioFrame* frame, float data, int sample_rate_hz) {
|
||||
memset(frame->data_, 0, sizeof(frame->data_));
|
||||
frame->num_channels_ = 1;
|
||||
frame->sample_rate_hz_ = sample_rate_hz;
|
||||
frame->samples_per_channel_ = sample_rate_hz / 100;
|
||||
@ -59,6 +59,7 @@ void SetMonoFrame(AudioFrame* frame, float data) {
|
||||
// each channel respectively.
|
||||
void SetStereoFrame(AudioFrame* frame, float left, float right,
|
||||
int sample_rate_hz) {
|
||||
memset(frame->data_, 0, sizeof(frame->data_));
|
||||
frame->num_channels_ = 2;
|
||||
frame->sample_rate_hz_ = sample_rate_hz;
|
||||
frame->samples_per_channel_ = sample_rate_hz / 100;
|
||||
@ -80,13 +81,14 @@ void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
|
||||
}
|
||||
|
||||
// Computes the best SNR based on the error between |ref_frame| and
|
||||
// |test_frame|. It allows for up to a 30 sample delay between the signals to
|
||||
// compensate for the resampling delay.
|
||||
float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
|
||||
// |test_frame|. It allows for up to a |max_delay| in samples between the
|
||||
// signals to compensate for the resampling delay.
|
||||
float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
|
||||
int max_delay) {
|
||||
VerifyParams(ref_frame, test_frame);
|
||||
float best_snr = 0;
|
||||
int best_delay = 0;
|
||||
for (int delay = 0; delay < 30; delay++) {
|
||||
for (int delay = 0; delay <= max_delay; delay++) {
|
||||
float mse = 0;
|
||||
float variance = 0;
|
||||
for (int i = 0; i < ref_frame.samples_per_channel_ *
|
||||
@ -120,14 +122,14 @@ void OutputMixerTest::RunResampleTest(int src_channels,
|
||||
int src_sample_rate_hz,
|
||||
int dst_channels,
|
||||
int dst_sample_rate_hz) {
|
||||
Resampler resampler; // Create a new one with every test.
|
||||
const int16_t kSrcLeft = 60; // Shouldn't overflow for any used sample rate.
|
||||
const int16_t kSrcRight = 30;
|
||||
const float kResamplingFactor = (1.0 * src_sample_rate_hz) /
|
||||
PushResampler resampler; // Create a new one with every test.
|
||||
const int16_t kSrcLeft = 30; // Shouldn't overflow for any used sample rate.
|
||||
const int16_t kSrcRight = 15;
|
||||
const float resampling_factor = (1.0 * src_sample_rate_hz) /
|
||||
dst_sample_rate_hz;
|
||||
const float kDstLeft = kResamplingFactor * kSrcLeft;
|
||||
const float kDstRight = kResamplingFactor * kSrcRight;
|
||||
const float kDstMono = (kDstLeft + kDstRight) / 2;
|
||||
const float dst_left = resampling_factor * kSrcLeft;
|
||||
const float dst_right = resampling_factor * kSrcRight;
|
||||
const float dst_mono = (dst_left + dst_right) / 2;
|
||||
if (src_channels == 1)
|
||||
SetMonoFrame(&src_frame_, kSrcLeft, src_sample_rate_hz);
|
||||
else
|
||||
@ -136,27 +138,27 @@ void OutputMixerTest::RunResampleTest(int src_channels,
|
||||
if (dst_channels == 1) {
|
||||
SetMonoFrame(&dst_frame_, 0, dst_sample_rate_hz);
|
||||
if (src_channels == 1)
|
||||
SetMonoFrame(&golden_frame_, kDstLeft, dst_sample_rate_hz);
|
||||
SetMonoFrame(&golden_frame_, dst_left, dst_sample_rate_hz);
|
||||
else
|
||||
SetMonoFrame(&golden_frame_, kDstMono, dst_sample_rate_hz);
|
||||
SetMonoFrame(&golden_frame_, dst_mono, dst_sample_rate_hz);
|
||||
} else {
|
||||
SetStereoFrame(&dst_frame_, 0, 0, dst_sample_rate_hz);
|
||||
if (src_channels == 1)
|
||||
SetStereoFrame(&golden_frame_, kDstLeft, kDstLeft, dst_sample_rate_hz);
|
||||
SetStereoFrame(&golden_frame_, dst_left, dst_left, dst_sample_rate_hz);
|
||||
else
|
||||
SetStereoFrame(&golden_frame_, kDstLeft, kDstRight, dst_sample_rate_hz);
|
||||
SetStereoFrame(&golden_frame_, dst_left, dst_right, dst_sample_rate_hz);
|
||||
}
|
||||
|
||||
// The sinc resampler has a known delay, which we compute here. Multiplying by
|
||||
// two gives us a crude maximum for any resampling, as the old resampler
|
||||
// typically (but not always) has lower delay.
|
||||
static const int kInputKernelDelaySamples = 16;
|
||||
const int max_delay = static_cast<double>(dst_sample_rate_hz)
|
||||
/ src_sample_rate_hz * kInputKernelDelaySamples * dst_channels * 2;
|
||||
printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later.
|
||||
src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
|
||||
EXPECT_EQ(0, RemixAndResample(src_frame_, &resampler, &dst_frame_));
|
||||
EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_), 40.0f);
|
||||
}
|
||||
|
||||
TEST_F(OutputMixerTest, RemixAndResampleFailsWithBadSampleRate) {
|
||||
SetMonoFrame(&dst_frame_, 10, 44100);
|
||||
EXPECT_EQ(-1, RemixAndResample(src_frame_, &resampler_, &dst_frame_));
|
||||
VerifyFramesAreEqual(src_frame_, dst_frame_);
|
||||
EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 39.0f);
|
||||
}
|
||||
|
||||
TEST_F(OutputMixerTest, RemixAndResampleCopyFrameSucceeds) {
|
||||
@ -190,10 +192,9 @@ TEST_F(OutputMixerTest, RemixAndResampleMixingOnlySucceeds) {
|
||||
}
|
||||
|
||||
TEST_F(OutputMixerTest, RemixAndResampleSucceeds) {
|
||||
// We don't attempt to be exhaustive here, but just get good coverage. Some
|
||||
// combinations of rates will not be resampled, and some give an odd
|
||||
// resampling factor which makes it more difficult to evaluate.
|
||||
const int kSampleRates[] = {16000, 32000, 48000};
|
||||
// TODO(ajm): convert this to the parameterized TEST_P style used in
|
||||
// sinc_resampler_unittest.cc. We can then easily add tighter SNR thresholds.
|
||||
const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000, 96000};
|
||||
const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
|
||||
const int kChannels[] = {1, 2};
|
||||
const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
|
||||
|
||||
Loading…
Reference in New Issue
Block a user