Index: Makefile
===================================================================
--- Makefile (nonexistent)
+++ Makefile (revision 62)
@@ -0,0 +1,58 @@
+
+COMPONENT_TARGETS = $(HARDWARE_NOARCH)
+
+
+include ../../../../build-system/constants.mk
+
+
+url = $(DOWNLOAD_SERVER)/sources/packages/n/webrtc-audio-processing
+
+versions = 0.3.1
+pkgname = webrtc-audio-processing
+suffix = tar.xz
+
+tarballs = $(addsuffix .$(suffix), $(addprefix $(pkgname)-, $(versions)))
+sha1s = $(addsuffix .sha1sum, $(tarballs))
+
+patches = $(CURDIR)/patches/webrtc-audio-processing-0.3.1-cross.patch
+patches += $(CURDIR)/patches/webrtc-audio-processing-0.3.1-big-endian.patch
+
+.NOTPARALLEL: $(patches)
+
+
+BUILD_TARGETS = $(tarballs) $(sha1s) $(patches)
+
+
+include ../../../../build-system/core.mk
+
+
+.PHONY: download_clean
+
+
+$(tarballs):
+ @echo -e "\n======= Downloading source tarballs =======" ; \
+ for tarball in $(tarballs) ; do \
+ echo "$(url)/$$tarball" | xargs -n 1 -P 100 wget $(WGET_OPTIONS) - & \
+ done ; wait
+
+$(sha1s): $(tarballs)
+ @for sha in $@ ; do \
+ echo -e "\n======= Downloading '$$sha' signature =======\n" ; \
+ echo "$(url)/$$sha" | xargs -n 1 -P 100 wget $(WGET_OPTIONS) - & wait %1 ; \
+ touch $$sha ; \
+ echo -e "\n======= Check the '$$sha' sha1sum =======\n" ; \
+ sha1sum --check $$sha ; ret="$$?" ; \
+ if [ "$$ret" == "1" ]; then \
+ echo -e "\n======= ERROR: Bad '$$sha' sha1sum =======\n" ; \
+ exit 1 ; \
+ fi ; \
+ done
+
+$(patches): $(sha1s)
+ @echo -e "\n======= Create Patches =======\n" ; \
+ ( cd create-0.3.1-cross-patch ; ./create.patch.sh ) ; \
+ ( cd create-0.3.1-big-endian-patch ; ./create.patch.sh ) ; \
+ echo -e "\n"
+
+download_clean:
+ @rm -f $(tarballs) $(sha1s) $(patches)
Index: create-0.3.1-big-endian-patch/create.patch.sh
===================================================================
--- create-0.3.1-big-endian-patch/create.patch.sh (nonexistent)
+++ create-0.3.1-big-endian-patch/create.patch.sh (revision 62)
@@ -0,0 +1,15 @@
+#!/bin/sh
+
+VERSION=0.3.1
+
+tar --files-from=file.list -xJvf ../webrtc-audio-processing-$VERSION.tar.xz
+mv webrtc-audio-processing-$VERSION webrtc-audio-processing-$VERSION-orig
+
+cp -rf ./webrtc-audio-processing-$VERSION-new ./webrtc-audio-processing-$VERSION
+
+diff --unified -Nr webrtc-audio-processing-$VERSION-orig webrtc-audio-processing-$VERSION > webrtc-audio-processing-$VERSION-big-endian.patch
+
+mv webrtc-audio-processing-$VERSION-big-endian.patch ../patches
+
+rm -rf ./webrtc-audio-processing-$VERSION
+rm -rf ./webrtc-audio-processing-$VERSION-orig
Property changes on: create-0.3.1-big-endian-patch/create.patch.sh
___________________________________________________________________
Added: svn:executable
## -0,0 +1 ##
+*
\ No newline at end of property
Index: create-0.3.1-big-endian-patch/file.list
===================================================================
--- create-0.3.1-big-endian-patch/file.list (nonexistent)
+++ create-0.3.1-big-endian-patch/file.list (revision 62)
@@ -0,0 +1,2 @@
+webrtc-audio-processing-0.3.1/webrtc/common_audio/wav_file.cc
+webrtc-audio-processing-0.3.1/webrtc/common_audio/wav_header.cc
Index: create-0.3.1-big-endian-patch/webrtc-audio-processing-0.3.1-new/webrtc/common_audio/wav_file.cc
===================================================================
--- create-0.3.1-big-endian-patch/webrtc-audio-processing-0.3.1-new/webrtc/common_audio/wav_file.cc (nonexistent)
+++ create-0.3.1-big-endian-patch/webrtc-audio-processing-0.3.1-new/webrtc/common_audio/wav_file.cc (revision 62)
@@ -0,0 +1,192 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * 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.
+ */
+
+#include "webrtc/common_audio/wav_file.h"
+
+#include <algorithm>
+#include <cstdio>
+#include <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/common_audio/include/audio_util.h"
+#include "webrtc/common_audio/wav_header.h"
+
+namespace webrtc {
+
+// We write 16-bit PCM WAV files.
+static const WavFormat kWavFormat = kWavFormatPcm;
+static const int kBytesPerSample = 2;
+
+// Doesn't take ownership of the file handle and won't close it.
+class ReadableWavFile : public ReadableWav {
+ public:
+ explicit ReadableWavFile(FILE* file) : file_(file) {}
+ virtual size_t Read(void* buf, size_t num_bytes) {
+ return fread(buf, 1, num_bytes, file_);
+ }
+
+ private:
+ FILE* file_;
+};
+
+std::string WavFile::FormatAsString() const {
+ std::ostringstream s;
+ s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
+ << ", Duration: "
+ << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
+ return s.str();
+}
+
+WavReader::WavReader(const std::string& filename)
+ : file_handle_(fopen(filename.c_str(), "rb")) {
+ RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
+
+ ReadableWavFile readable(file_handle_);
+ WavFormat format;
+ int bytes_per_sample;
+ RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
+ &bytes_per_sample, &num_samples_));
+ num_samples_remaining_ = num_samples_;
+ RTC_CHECK_EQ(kWavFormat, format);
+ RTC_CHECK_EQ(kBytesPerSample, bytes_per_sample);
+}
+
+WavReader::~WavReader() {
+ Close();
+}
+
+size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
+ // There could be metadata after the audio; ensure we don't read it.
+ num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
+ num_samples_remaining_);
+ const size_t read =
+ fread(samples, sizeof(*samples), num_samples, file_handle_);
+ // If we didn't read what was requested, ensure we've reached the EOF.
+ RTC_CHECK(read == num_samples || feof(file_handle_));
+ RTC_CHECK_LE(read, num_samples_remaining_);
+ num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+ //convert to big-endian
+ for(size_t idx = 0; idx < num_samples; idx++) {
+ samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
+ }
+#endif
+ return read;
+}
+
+size_t WavReader::ReadSamples(size_t num_samples, float* samples) {
+ static const size_t kChunksize = 4096 / sizeof(uint16_t);
+ size_t read = 0;
+ for (size_t i = 0; i < num_samples; i += kChunksize) {
+ int16_t isamples[kChunksize];
+ size_t chunk = std::min(kChunksize, num_samples - i);
+ chunk = ReadSamples(chunk, isamples);
+ for (size_t j = 0; j < chunk; ++j)
+ samples[i + j] = isamples[j];
+ read += chunk;
+ }
+ return read;
+}
+
+void WavReader::Close() {
+ RTC_CHECK_EQ(0, fclose(file_handle_));
+ file_handle_ = NULL;
+}
+
+WavWriter::WavWriter(const std::string& filename, int sample_rate,
+ int num_channels)
+ : sample_rate_(sample_rate),
+ num_channels_(num_channels),
+ num_samples_(0),
+ file_handle_(fopen(filename.c_str(), "wb")) {
+ RTC_CHECK(file_handle_) << "Could not open wav file for writing.";
+ RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
+ kBytesPerSample, num_samples_));
+
+ // Write a blank placeholder header, since we need to know the total number
+ // of samples before we can fill in the real data.
+ static const uint8_t blank_header[kWavHeaderSize] = {0};
+ RTC_CHECK_EQ(1u, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
+}
+
+WavWriter::~WavWriter() {
+ Close();
+}
+
+void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+ int16_t * le_samples = new int16_t[num_samples];
+ for(size_t idx = 0; idx < num_samples; idx++) {
+ le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
+ }
+ const size_t written =
+ fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_);
+ delete []le_samples;
+#else
+ const size_t written =
+ fwrite(samples, sizeof(*samples), num_samples, file_handle_);
+#endif
+ RTC_CHECK_EQ(num_samples, written);
+ num_samples_ += static_cast<uint32_t>(written);
+ RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
+ num_samples_ >= written); // detect uint32_t overflow
+}
+
+void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
+ static const size_t kChunksize = 4096 / sizeof(uint16_t);
+ for (size_t i = 0; i < num_samples; i += kChunksize) {
+ int16_t isamples[kChunksize];
+ const size_t chunk = std::min(kChunksize, num_samples - i);
+ FloatS16ToS16(samples + i, chunk, isamples);
+ WriteSamples(isamples, chunk);
+ }
+}
+
+void WavWriter::Close() {
+ RTC_CHECK_EQ(0, fseek(file_handle_, 0, SEEK_SET));
+ uint8_t header[kWavHeaderSize];
+ WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat,
+ kBytesPerSample, num_samples_);
+ RTC_CHECK_EQ(1u, fwrite(header, kWavHeaderSize, 1, file_handle_));
+ RTC_CHECK_EQ(0, fclose(file_handle_));
+ file_handle_ = NULL;
+}
+
+} // namespace webrtc
+
+rtc_WavWriter* rtc_WavOpen(const char* filename,
+ int sample_rate,
+ int num_channels) {
+ return reinterpret_cast<rtc_WavWriter*>(
+ new webrtc::WavWriter(filename, sample_rate, num_channels));
+}
+
+void rtc_WavClose(rtc_WavWriter* wf) {
+ delete reinterpret_cast<webrtc::WavWriter*>(wf);
+}
+
+void rtc_WavWriteSamples(rtc_WavWriter* wf,
+ const float* samples,
+ size_t num_samples) {
+ reinterpret_cast<webrtc::WavWriter*>(wf)->WriteSamples(samples, num_samples);
+}
+
+int rtc_WavSampleRate(const rtc_WavWriter* wf) {
+ return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
+}
+
+int rtc_WavNumChannels(const rtc_WavWriter* wf) {
+ return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
+}
+
+uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
+ return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples();
+}
Index: create-0.3.1-big-endian-patch/webrtc-audio-processing-0.3.1-new/webrtc/common_audio/wav_header.cc
===================================================================
--- create-0.3.1-big-endian-patch/webrtc-audio-processing-0.3.1-new/webrtc/common_audio/wav_header.cc (nonexistent)
+++ create-0.3.1-big-endian-patch/webrtc-audio-processing-0.3.1-new/webrtc/common_audio/wav_header.cc (revision 62)
@@ -0,0 +1,274 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * 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.
+ */
+
+// Based on the WAV file format documentation at
+// https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ and
+// http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
+
+#include "webrtc/common_audio/wav_header.h"
+
+#include <algorithm>
+#include <cstring>
+#include <limits>
+#include <string>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/include/audio_util.h"
+
+namespace webrtc {
+namespace {
+
+struct ChunkHeader {
+ uint32_t ID;
+ uint32_t Size;
+};
+static_assert(sizeof(ChunkHeader) == 8, "ChunkHeader size");
+
+// We can't nest this definition in WavHeader, because VS2013 gives an error
+// on sizeof(WavHeader::fmt): "error C2070: 'unknown': illegal sizeof operand".
+struct FmtSubchunk {
+ ChunkHeader header;
+ uint16_t AudioFormat;
+ uint16_t NumChannels;
+ uint32_t SampleRate;
+ uint32_t ByteRate;
+ uint16_t BlockAlign;
+ uint16_t BitsPerSample;
+};
+static_assert(sizeof(FmtSubchunk) == 24, "FmtSubchunk size");
+const uint32_t kFmtSubchunkSize = sizeof(FmtSubchunk) - sizeof(ChunkHeader);
+
+struct WavHeader {
+ struct {
+ ChunkHeader header;
+ uint32_t Format;
+ } riff;
+ FmtSubchunk fmt;
+ struct {
+ ChunkHeader header;
+ } data;
+};
+static_assert(sizeof(WavHeader) == kWavHeaderSize, "no padding in header");
+
+} // namespace
+
+bool CheckWavParameters(int num_channels,
+ int sample_rate,
+ WavFormat format,
+ int bytes_per_sample,
+ uint32_t num_samples) {
+ // num_channels, sample_rate, and bytes_per_sample must be positive, must fit
+ // in their respective fields, and their product must fit in the 32-bit
+ // ByteRate field.
+ if (num_channels <= 0 || sample_rate <= 0 || bytes_per_sample <= 0)
+ return false;
+ if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max())
+ return false;
+ if (static_cast<uint64_t>(num_channels) >
+ std::numeric_limits<uint16_t>::max())
+ return false;
+ if (static_cast<uint64_t>(bytes_per_sample) * 8 >
+ std::numeric_limits<uint16_t>::max())
+ return false;
+ if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample >
+ std::numeric_limits<uint32_t>::max())
+ return false;
+
+ // format and bytes_per_sample must agree.
+ switch (format) {
+ case kWavFormatPcm:
+ // Other values may be OK, but for now we're conservative:
+ if (bytes_per_sample != 1 && bytes_per_sample != 2)
+ return false;
+ break;
+ case kWavFormatALaw:
+ case kWavFormatMuLaw:
+ if (bytes_per_sample != 1)
+ return false;
+ break;
+ default:
+ return false;
+ }
+
+ // The number of bytes in the file, not counting the first ChunkHeader, must
+ // be less than 2^32; otherwise, the ChunkSize field overflows.
+ const uint32_t max_samples =
+ (std::numeric_limits<uint32_t>::max()
+ - (kWavHeaderSize - sizeof(ChunkHeader))) /
+ bytes_per_sample;
+ if (num_samples > max_samples)
+ return false;
+
+ // Each channel must have the same number of samples.
+ if (num_samples % num_channels != 0)
+ return false;
+
+ return true;
+}
+
+#ifdef WEBRTC_ARCH_LITTLE_ENDIAN
+static inline void WriteLE16(uint16_t* f, uint16_t x) { *f = x; }
+static inline void WriteLE32(uint32_t* f, uint32_t x) { *f = x; }
+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
+ *f = static_cast<uint32_t>(a)
+ | static_cast<uint32_t>(b) << 8
+ | static_cast<uint32_t>(c) << 16
+ | static_cast<uint32_t>(d) << 24;
+}
+
+static inline uint16_t ReadLE16(uint16_t x) { return x; }
+static inline uint32_t ReadLE32(uint32_t x) { return x; }
+static inline std::string ReadFourCC(uint32_t x) {
+ return std::string(reinterpret_cast<char*>(&x), 4);
+}
+#else
+static inline void WriteLE16(uint16_t* f, uint16_t x) {
+ *f = ((x << 8) & 0xff00) | ( ( x >> 8) & 0x00ff);
+}
+
+static inline void WriteLE32(uint32_t* f, uint32_t x) {
+ *f = ( (x & 0x000000ff) << 24 )
+ | ((x & 0x0000ff00) << 8)
+ | ((x & 0x00ff0000) >> 8)
+ | ((x & 0xff000000) >> 24 );
+}
+
+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
+ *f = (static_cast<uint32_t>(a) << 24 )
+ | (static_cast<uint32_t>(b) << 16)
+ | (static_cast<uint32_t>(c) << 8)
+ | (static_cast<uint32_t>(d) );
+}
+
+static inline uint16_t ReadLE16(uint16_t x) {
+ return (( x & 0x00ff) << 8 )| ((x & 0xff00)>>8);
+}
+
+static inline uint32_t ReadLE32(uint32_t x) {
+ return ( (x & 0x000000ff) << 24 )
+ | ( (x & 0x0000ff00) << 8 )
+ | ( (x & 0x00ff0000) >> 8)
+ | ( (x & 0xff000000) >> 24 );
+}
+
+static inline std::string ReadFourCC(uint32_t x) {
+ x = ReadLE32(x);
+ return std::string(reinterpret_cast<char*>(&x), 4);
+}
+#endif
+
+static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {
+ return bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader);
+}
+
+static inline uint32_t ByteRate(int num_channels, int sample_rate,
+ int bytes_per_sample) {
+ return static_cast<uint32_t>(num_channels) * sample_rate * bytes_per_sample;
+}
+
+static inline uint16_t BlockAlign(int num_channels, int bytes_per_sample) {
+ return num_channels * bytes_per_sample;
+}
+
+void WriteWavHeader(uint8_t* buf,
+ int num_channels,
+ int sample_rate,
+ WavFormat format,
+ int bytes_per_sample,
+ uint32_t num_samples) {
+ RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format,
+ bytes_per_sample, num_samples));
+
+ WavHeader header;
+ const uint32_t bytes_in_payload = bytes_per_sample * num_samples;
+
+ WriteFourCC(&header.riff.header.ID, 'R', 'I', 'F', 'F');
+ WriteLE32(&header.riff.header.Size, RiffChunkSize(bytes_in_payload));
+ WriteFourCC(&header.riff.Format, 'W', 'A', 'V', 'E');
+
+ WriteFourCC(&header.fmt.header.ID, 'f', 'm', 't', ' ');
+ WriteLE32(&header.fmt.header.Size, kFmtSubchunkSize);
+ WriteLE16(&header.fmt.AudioFormat, format);
+ WriteLE16(&header.fmt.NumChannels, num_channels);
+ WriteLE32(&header.fmt.SampleRate, sample_rate);
+ WriteLE32(&header.fmt.ByteRate, ByteRate(num_channels, sample_rate,
+ bytes_per_sample));
+ WriteLE16(&header.fmt.BlockAlign, BlockAlign(num_channels, bytes_per_sample));
+ WriteLE16(&header.fmt.BitsPerSample, 8 * bytes_per_sample);
+
+ WriteFourCC(&header.data.header.ID, 'd', 'a', 't', 'a');
+ WriteLE32(&header.data.header.Size, bytes_in_payload);
+
+ // Do an extra copy rather than writing everything to buf directly, since buf
+ // might not be correctly aligned.
+ memcpy(buf, &header, kWavHeaderSize);
+}
+
+bool ReadWavHeader(ReadableWav* readable,
+ int* num_channels,
+ int* sample_rate,
+ WavFormat* format,
+ int* bytes_per_sample,
+ uint32_t* num_samples) {
+ WavHeader header;
+ if (readable->Read(&header, kWavHeaderSize - sizeof(header.data)) !=
+ kWavHeaderSize - sizeof(header.data))
+ return false;
+
+ const uint32_t fmt_size = ReadLE32(header.fmt.header.Size);
+ if (fmt_size != kFmtSubchunkSize) {
+ // There is an optional two-byte extension field permitted to be present
+ // with PCM, but which must be zero.
+ int16_t ext_size;
+ if (kFmtSubchunkSize + sizeof(ext_size) != fmt_size)
+ return false;
+ if (readable->Read(&ext_size, sizeof(ext_size)) != sizeof(ext_size))
+ return false;
+ if (ext_size != 0)
+ return false;
+ }
+ if (readable->Read(&header.data, sizeof(header.data)) != sizeof(header.data))
+ return false;
+
+ // Parse needed fields.
+ *format = static_cast<WavFormat>(ReadLE16(header.fmt.AudioFormat));
+ *num_channels = ReadLE16(header.fmt.NumChannels);
+ *sample_rate = ReadLE32(header.fmt.SampleRate);
+ *bytes_per_sample = ReadLE16(header.fmt.BitsPerSample) / 8;
+ const uint32_t bytes_in_payload = ReadLE32(header.data.header.Size);
+ if (*bytes_per_sample <= 0)
+ return false;
+ *num_samples = bytes_in_payload / *bytes_per_sample;
+
+ // Sanity check remaining fields.
+ if (ReadFourCC(header.riff.header.ID) != "RIFF")
+ return false;
+ if (ReadFourCC(header.riff.Format) != "WAVE")
+ return false;
+ if (ReadFourCC(header.fmt.header.ID) != "fmt ")
+ return false;
+ if (ReadFourCC(header.data.header.ID) != "data")
+ return false;
+
+ if (ReadLE32(header.riff.header.Size) < RiffChunkSize(bytes_in_payload))
+ return false;
+ if (ReadLE32(header.fmt.ByteRate) !=
+ ByteRate(*num_channels, *sample_rate, *bytes_per_sample))
+ return false;
+ if (ReadLE16(header.fmt.BlockAlign) !=
+ BlockAlign(*num_channels, *bytes_per_sample))
+ return false;
+
+ return CheckWavParameters(*num_channels, *sample_rate, *format,
+ *bytes_per_sample, *num_samples);
+}
+
+
+} // namespace webrtc
Index: create-0.3.1-cross-patch/create.patch.sh
===================================================================
--- create-0.3.1-cross-patch/create.patch.sh (nonexistent)
+++ create-0.3.1-cross-patch/create.patch.sh (revision 62)
@@ -0,0 +1,15 @@
+#!/bin/sh
+
+VERSION=0.3.1
+
+tar --files-from=file.list -xJvf ../webrtc-audio-processing-$VERSION.tar.xz
+mv webrtc-audio-processing-$VERSION webrtc-audio-processing-$VERSION-orig
+
+cp -rf ./webrtc-audio-processing-$VERSION-new ./webrtc-audio-processing-$VERSION
+
+diff --unified -Nr webrtc-audio-processing-$VERSION-orig webrtc-audio-processing-$VERSION > webrtc-audio-processing-$VERSION-cross.patch
+
+mv webrtc-audio-processing-$VERSION-cross.patch ../patches
+
+rm -rf ./webrtc-audio-processing-$VERSION
+rm -rf ./webrtc-audio-processing-$VERSION-orig
Property changes on: create-0.3.1-cross-patch/create.patch.sh
___________________________________________________________________
Added: svn:executable
## -0,0 +1 ##
+*
\ No newline at end of property
Index: create-0.3.1-cross-patch/file.list
===================================================================
--- create-0.3.1-cross-patch/file.list (nonexistent)
+++ create-0.3.1-cross-patch/file.list (revision 62)
@@ -0,0 +1 @@
+webrtc-audio-processing-0.3.1/webrtc/typedefs.h
Index: create-0.3.1-cross-patch/webrtc-audio-processing-0.3.1-new/webrtc/typedefs.h
===================================================================
--- create-0.3.1-cross-patch/webrtc-audio-processing-0.3.1-new/webrtc/typedefs.h (nonexistent)
+++ create-0.3.1-cross-patch/webrtc-audio-processing-0.3.1-new/webrtc/typedefs.h (revision 62)
@@ -0,0 +1,123 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * 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.
+ */
+
+// This file contains platform-specific typedefs and defines.
+// Much of it is derived from Chromium's build/build_config.h.
+
+#ifndef WEBRTC_TYPEDEFS_H_
+#define WEBRTC_TYPEDEFS_H_
+
+// Processor architecture detection. For more info on what's defined, see:
+// http://msdn.microsoft.com/en-us/library/b0084kay.aspx
+// http://www.agner.org/optimize/calling_conventions.pdf
+// or with gcc, run: "echo | gcc -E -dM -"
+#if defined(_M_X64) || defined(__x86_64__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86_64
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+
+#elif defined(__riscv_xlen) && __riscv_xlen == 64
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__riscv_xlen) && __riscv_xlen == 32
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__powerpc64__) && __BIG_ENDIAN__ == 1
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_BIG_ENDIAN
+#elif defined(__powerpc64__) && __LITTLE_ENDIAN__ == 1
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__aarch64__)
+#define WEBRTC_ARCH_64_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(_M_IX86) || defined(__i386__)
+#define WEBRTC_ARCH_X86_FAMILY
+#define WEBRTC_ARCH_X86
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__ARMEL__)
+// TODO(ajm): We'd prefer to control platform defines here, but this is
+// currently provided by the Android makefiles. Commented to avoid duplicate
+// definition warnings.
+//#define WEBRTC_ARCH_ARM
+// TODO(ajm): Chromium uses the following two defines. Should we switch?
+//#define WEBRTC_ARCH_ARM_FAMILY
+//#define WEBRTC_ARCH_ARMEL
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__MIPSEL__)
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#elif defined(__pnacl__)
+#define WEBRTC_ARCH_32_BITS
+#define WEBRTC_ARCH_LITTLE_ENDIAN
+#else
+#error Please add support for your architecture in typedefs.h
+#endif
+
+#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
+#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
+#endif
+
+// TODO(zhongwei.yao): WEBRTC_CPU_DETECTION is only used in one place; we should
+// probably just remove it.
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)) || \
+ defined(WEBRTC_DETECT_NEON)
+#define WEBRTC_CPU_DETECTION
+#endif
+
+// TODO(pbos): Use webrtc/base/basictypes.h instead to include fixed-size ints.
+#include <stdint.h>
+
+// Annotate a function indicating the caller must examine the return value.
+// Use like:
+// int foo() WARN_UNUSED_RESULT;
+// TODO(ajm): Hack to avoid multiple definitions until the base/ of webrtc and
+// libjingle are merged.
+#if !defined(WARN_UNUSED_RESULT)
+#if defined(__GNUC__)
+#define WARN_UNUSED_RESULT __attribute__((warn_unused_result))
+#else
+#define WARN_UNUSED_RESULT
+#endif
+#endif // WARN_UNUSED_RESULT
+
+// Put after a variable that might not be used, to prevent compiler warnings:
+// int result ATTRIBUTE_UNUSED = DoSomething();
+// assert(result == 17);
+#ifndef ATTRIBUTE_UNUSED
+#if defined(__GNUC__) || defined(__clang__)
+#define ATTRIBUTE_UNUSED __attribute__((unused))
+#else
+#define ATTRIBUTE_UNUSED
+#endif
+#endif
+
+// Macro to be used for switch-case fallthrough (required for enabling
+// -Wimplicit-fallthrough warning on Clang).
+#ifndef FALLTHROUGH
+#if defined(__clang__)
+#define FALLTHROUGH() [[clang::fallthrough]]
+#else
+#define FALLTHROUGH() do { } while (0)
+#endif
+#endif
+
+// Annotate a function that will not return control flow to the caller.
+#if defined(_MSC_VER)
+#define NO_RETURN __declspec(noreturn)
+#elif defined(__GNUC__)
+#define NO_RETURN __attribute__((noreturn))
+#else
+#define NO_RETURN
+#endif
+
+#endif // WEBRTC_TYPEDEFS_H_
Index: patches/README
===================================================================
--- patches/README (nonexistent)
+++ patches/README (revision 62)
@@ -0,0 +1,6 @@
+
+/* begin *
+
+ TODO: Leave some comment here.
+
+ * end */
Index: patches
===================================================================
--- patches (nonexistent)
+++ patches (revision 62)
Property changes on: patches
___________________________________________________________________
Added: svn:ignore
## -0,0 +1,74 ##
+
+# install dir
+dist
+
+# Target build dirs
+.a1x-newlib
+.a2x-newlib
+.at91sam7s-newlib
+
+.build-machine
+
+.a1x-glibc
+.a2x-glibc
+.h3-glibc
+.h5-glibc
+.i586-glibc
+.i686-glibc
+.imx6-glibc
+.jz47xx-glibc
+.makefile
+.am335x-glibc
+.omap543x-glibc
+.p5600-glibc
+.power8-glibc
+.power8le-glibc
+.power9-glibc
+.power9le-glibc
+.m1000-glibc
+.riscv64-glibc
+.rk328x-glibc
+.rk33xx-glibc
+.rk339x-glibc
+.rk358x-glibc
+.s8xx-glibc
+.s9xx-glibc
+.x86_64-glibc
+
+# Hidden files (each file)
+.makefile
+.dist
+.rootfs
+
+# src & hw requires
+.src_requires
+.src_requires_depend
+.requires
+.requires_depend
+
+# Tarballs
+*.gz
+*.bz2
+*.lz
+*.xz
+*.tgz
+*.txz
+
+# Signatures
+*.asc
+*.sig
+*.sign
+*.sha1sum
+
+# Patches
+*.patch
+
+# Descriptions
+*.dsc
+*.txt
+
+# Default linux config files
+*.defconfig
+
+# backup copies
+*~
Index: .
===================================================================
--- . (nonexistent)
+++ . (revision 62)
Property changes on: .
___________________________________________________________________
Added: svn:ignore
## -0,0 +1,74 ##
+
+# install dir
+dist
+
+# Target build dirs
+.a1x-newlib
+.a2x-newlib
+.at91sam7s-newlib
+
+.build-machine
+
+.a1x-glibc
+.a2x-glibc
+.h3-glibc
+.h5-glibc
+.i586-glibc
+.i686-glibc
+.imx6-glibc
+.jz47xx-glibc
+.makefile
+.am335x-glibc
+.omap543x-glibc
+.p5600-glibc
+.power8-glibc
+.power8le-glibc
+.power9-glibc
+.power9le-glibc
+.m1000-glibc
+.riscv64-glibc
+.rk328x-glibc
+.rk33xx-glibc
+.rk339x-glibc
+.rk358x-glibc
+.s8xx-glibc
+.s9xx-glibc
+.x86_64-glibc
+
+# Hidden files (each file)
+.makefile
+.dist
+.rootfs
+
+# src & hw requires
+.src_requires
+.src_requires_depend
+.requires
+.requires_depend
+
+# Tarballs
+*.gz
+*.bz2
+*.lz
+*.xz
+*.tgz
+*.txz
+
+# Signatures
+*.asc
+*.sig
+*.sign
+*.sha1sum
+
+# Patches
+*.patch
+
+# Descriptions
+*.dsc
+*.txt
+
+# Default linux config files
+*.defconfig
+
+# backup copies
+*~
Radix cross Linux
The main Radix cross Linux repository contains the build scripts of packages, which have the most complete and common functionality for desktop machines
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1 Tag