ddbb8a2c24
Select "processing" rates based on the input and output sampling rates. Resample the input streams to those rates, and if necessary to the output rate. - Remove deprecated stream format APIs. - Remove deprecated device sample rate APIs. - Add a ChannelBuffer class to help manage deinterleaved channels. - Clean up the splitting filter state. - Add a unit test which verifies the output against known-working native format output. BUG=2894 R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
95 lines
3.0 KiB
C++
95 lines
3.0 KiB
C++
/*
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* Copyright (c) 2014 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/audio_processing/debug.pb.h"
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#include "webrtc/modules/audio_processing/common.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
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#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
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// Exits on failure; do not use in unit tests.
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static inline FILE* OpenFile(const std::string& filename, const char* mode) {
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FILE* file = fopen(filename.c_str(), mode);
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if (!file) {
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printf("Unable to open file %s\n", filename.c_str());
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exit(1);
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}
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return file;
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}
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static inline int SamplesFromRate(int rate) {
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return AudioProcessing::kChunkSizeMs * rate / 1000;
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}
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static inline void SetFrameSampleRate(AudioFrame* frame,
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int sample_rate_hz) {
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frame->sample_rate_hz_ = sample_rate_hz;
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frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
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sample_rate_hz / 1000;
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}
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template <typename T>
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void SetContainerFormat(int sample_rate_hz,
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int num_channels,
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AudioFrame* frame,
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scoped_ptr<ChannelBuffer<T> >* cb) {
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SetFrameSampleRate(frame, sample_rate_hz);
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frame->num_channels_ = num_channels;
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cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
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}
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static inline AudioProcessing::ChannelLayout LayoutFromChannels(
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int num_channels) {
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switch (num_channels) {
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case 1:
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return AudioProcessing::kMono;
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case 2:
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return AudioProcessing::kStereo;
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default:
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assert(false);
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return AudioProcessing::kMono;
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}
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}
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// Allocates new memory in the scoped_ptr to fit the raw message and returns the
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// number of bytes read.
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static inline size_t ReadMessageBytesFromFile(FILE* file,
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scoped_ptr<uint8_t[]>* bytes) {
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// The "wire format" for the size is little-endian. Assume we're running on
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// a little-endian machine.
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int32_t size = 0;
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if (fread(&size, sizeof(size), 1, file) != 1)
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return 0;
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if (size <= 0)
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return 0;
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bytes->reset(new uint8_t[size]);
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return fread(bytes->get(), sizeof((*bytes)[0]), size, file);
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}
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// Returns true on success, false on error or end-of-file.
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static inline bool ReadMessageFromFile(FILE* file,
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::google::protobuf::MessageLite* msg) {
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scoped_ptr<uint8_t[]> bytes;
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size_t size = ReadMessageBytesFromFile(file, &bytes);
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if (!size)
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return false;
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msg->Clear();
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return msg->ParseFromArray(bytes.get(), size);
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}
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} // namespace webrtc
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