audio/webrtc-audio-processing: update to 1.0

Changes:	https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/compare/v0.3.1...v1.0
Reported by:	Repology

6 files changed, 278 insertions, 0 deletions

diff --git a/audio/webrtc-audio-processing0/files/patch-big-endian b/audio/webrtc-audio-processing0/files/patch-big-endian
new file mode 100644
index 000000000000..57daadce1870
--- /dev/null
+++ b/audio/webrtc-audio-processing0/files/patch-big-endian
@@ -0,0 +1,113 @@
+https://bugs.freedesktop.org/show_bug.cgi?id=95738
+
+--- webrtc/common_audio/wav_file.cc.orig	2018-07-23 14:02:57 UTC
++++ webrtc/common_audio/wav_file.cc
+@@ -64,9 +64,6 @@ WavReader::~WavReader() {
+ }
+ 
+ size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
+-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+-#error "Need to convert samples to big-endian when reading from WAV file"
+-#endif
+   // 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_);
+@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num_samples, int1
+   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;
+ }
+ 
+@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
+ 
+ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
+ #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+-#error "Need to convert samples to little-endian when writing to WAV file"
+-#endif
++  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() ||
+--- webrtc/common_audio/wav_header.cc.orig	2018-07-23 14:02:57 UTC
++++ webrtc/common_audio/wav_header.cc
+@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uint32_t x) {
+   return std::string(reinterpret_cast<char*>(&x), 4);
+ }
+ #else
+-#error "Write be-to-le conversion functions"
++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) {
+--- webrtc/typedefs.h.orig	2018-07-23 14:02:57 UTC
++++ webrtc/typedefs.h
+@@ -48,7 +48,19 @@
+ #define WEBRTC_ARCH_32_BITS
+ #define WEBRTC_ARCH_LITTLE_ENDIAN
+ #else
+-#error Please add support for your architecture in typedefs.h
++/* instead of failing, use typical unix defines... */
++#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
++#define WEBRTC_ARCH_LITTLE_ENDIAN
++#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
++#define WEBRTC_ARCH_BIG_ENDIAN
++#else
++#error __BYTE_ORDER__ is not defined
++#endif
++#if defined(__LP64__)
++#define WEBRTC_ARCH_64_BITS
++#else
++#define WEBRTC_ARCH_32_BITS
++#endif
+ #endif
+ 
+ #if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
diff --git a/audio/webrtc-audio-processing0/files/patch-configure.ac b/audio/webrtc-audio-processing0/files/patch-configure.ac
new file mode 100644
index 000000000000..b41e30d1a20a
--- /dev/null
+++ b/audio/webrtc-audio-processing0/files/patch-configure.ac
@@ -0,0 +1,18 @@
+- Add WEBRTC_BSD as it's closer to WEBRTC_LINUX than WEBRTC_MAC
+
+--- configure.ac.orig	2018-07-23 14:02:57 UTC
++++ configure.ac
+@@ -63,6 +63,13 @@ AS_CASE(["${host}"],
+          OS_LDFLAGS="-lrt -lpthread"
+          HAVE_POSIX=1
+         ],
++    [*-*dragonfly*|*-*bsd*],
++        [
++         OS_CFLAGS="-DWEBRTC_BSD -DWEBRTC_THREAD_RR"
++         PLATFORM_CFLAGS="-DWEBRTC_POSIX"
++         OS_LDFLAGS="-lpthread"
++         HAVE_POSIX=1
++        ],
+     [*-*darwin*],
+         [
+          OS_CFLAGS="-DWEBRTC_MAC -DWEBRTC_THREAD_RR -DWEBRTC_CLOCK_TYPE_REALTIME"
diff --git a/audio/webrtc-audio-processing0/files/patch-webrtc_base_checks.cc b/audio/webrtc-audio-processing0/files/patch-webrtc_base_checks.cc
new file mode 100644
index 000000000000..82ffcf49c7ec
--- /dev/null
+++ b/audio/webrtc-audio-processing0/files/patch-webrtc_base_checks.cc
@@ -0,0 +1,70 @@
+- Drop unnecessary dependency on libexecinfo for GCC build
+  https://chromium.googlesource.com/external/webrtc/+/7c4dedade158%5E!/
+
+--- webrtc/base/checks.cc.orig	2018-07-23 14:02:57 UTC
++++ webrtc/base/checks.cc
+@@ -11,16 +11,10 @@
+ // Most of this was borrowed (with minor modifications) from V8's and Chromium's
+ // src/base/logging.cc.
+ 
+-// Use the C++ version to provide __GLIBCXX__.
+ #include <cstdarg>
+ #include <cstdio>
+ #include <cstdlib>
+ 
+-#if defined(__GLIBCXX__) && !defined(__UCLIBC__)
+-#include <cxxabi.h>
+-#include <execinfo.h>
+-#endif
+-
+ #if defined(WEBRTC_ANDROID)
+ #define LOG_TAG "rtc"
+ #include <android/log.h>  // NOLINT
+@@ -51,39 +45,6 @@ void PrintError(const char* format, ...) {
+   va_end(args);
+ }
+ 
+-// TODO(ajm): This works on Mac (although the parsing fails) but I don't seem
+-// to get usable symbols on Linux. This is copied from V8. Chromium has a more
+-// advanced stace trace system; also more difficult to copy.
+-void DumpBacktrace() {
+-#if defined(__GLIBCXX__) && !defined(__UCLIBC__)
+-  void* trace[100];
+-  int size = backtrace(trace, sizeof(trace) / sizeof(*trace));
+-  char** symbols = backtrace_symbols(trace, size);
+-  PrintError("\n==== C stack trace ===============================\n\n");
+-  if (size == 0) {
+-    PrintError("(empty)\n");
+-  } else if (symbols == NULL) {
+-    PrintError("(no symbols)\n");
+-  } else {
+-    for (int i = 1; i < size; ++i) {
+-      char mangled[201];
+-      if (sscanf(symbols[i], "%*[^(]%*[(]%200[^)+]", mangled) == 1) {  // NOLINT
+-        PrintError("%2d: ", i);
+-        int status;
+-        size_t length;
+-        char* demangled = abi::__cxa_demangle(mangled, NULL, &length, &status);
+-        PrintError("%s\n", demangled != NULL ? demangled : mangled);
+-        free(demangled);
+-      } else {
+-        // If parsing failed, at least print the unparsed symbol.
+-        PrintError("%s\n", symbols[i]);
+-      }
+-    }
+-  }
+-  free(symbols);
+-#endif
+-}
+-
+ FatalMessage::FatalMessage(const char* file, int line) {
+   Init(file, line);
+ }
+@@ -99,7 +60,6 @@ NO_RETURN FatalMessage::~FatalMessage() {
+   fflush(stderr);
+   stream_ << std::endl << "#" << std::endl;
+   PrintError(stream_.str().c_str());
+-  DumpBacktrace();
+   fflush(stderr);
+   abort();
+ }
diff --git a/audio/webrtc-audio-processing0/files/patch-webrtc_base_platform__thread.cc b/audio/webrtc-audio-processing0/files/patch-webrtc_base_platform__thread.cc
new file mode 100644
index 000000000000..23840052102e
--- /dev/null
+++ b/audio/webrtc-audio-processing0/files/patch-webrtc_base_platform__thread.cc
@@ -0,0 +1,42 @@
+- Implement CurrentThreadId() using global thread ID
+- Implement SetCurrentThreadName()
+
+--- webrtc/base/platform_thread.cc.orig	2018-07-23 14:02:57 UTC
++++ webrtc/base/platform_thread.cc
+@@ -19,6 +19,12 @@
+ #include <sys/syscall.h>
+ #endif
+ 
++#if defined(__DragonFly__) || defined(__FreeBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
++#include <pthread_np.h>
++#elif defined(__NetBSD__) // WEBRTC_BSD
++#include <lwp.h>
++#endif
++
+ namespace rtc {
+ 
+ PlatformThreadId CurrentThreadId() {
+@@ -32,6 +38,12 @@ PlatformThreadId CurrentThreadId() {
+   ret =  syscall(__NR_gettid);
+ #elif defined(WEBRTC_ANDROID)
+   ret = gettid();
++#elif defined(__DragonFly__) || defined(__FreeBSD__) // WEBRTC_BSD
++  ret = pthread_getthreadid_np();
++#elif defined(__NetBSD__) // WEBRTC_BSD
++  ret = _lwp_self();
++#elif defined(__OpenBSD__) // WEBRTC_BSD
++  ret = getthrid();
+ #else
+   // Default implementation for nacl and solaris.
+   ret = reinterpret_cast<pid_t>(pthread_self());
+@@ -76,6 +88,10 @@ void SetCurrentThreadName(const char* name) {
+   prctl(PR_SET_NAME, reinterpret_cast<unsigned long>(name));
+ #elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
+   pthread_setname_np(name);
++#elif defined(__DragonFly__) || defined(__FreeBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
++  pthread_set_name_np(pthread_self(), name);
++#elif defined(__NetBSD__) // WEBRTC_BSD
++  pthread_setname_np(pthread_self(), "%s", (void*)name);
+ #endif
+ }
+ 
diff --git a/audio/webrtc-audio-processing0/files/patch-webrtc_base_stringutils.h b/audio/webrtc-audio-processing0/files/patch-webrtc_base_stringutils.h
new file mode 100644
index 000000000000..c01ec8a2c14b
--- /dev/null
+++ b/audio/webrtc-audio-processing0/files/patch-webrtc_base_stringutils.h
@@ -0,0 +1,13 @@
+- BSD macro (in sys/param.h) is an archaic of the (University of California) past
+
+--- webrtc/base/stringutils.h.orig	2018-07-23 14:02:57 UTC
++++ webrtc/base/stringutils.h
+@@ -23,7 +23,7 @@
+ #endif  // WEBRTC_WIN 
+ 
+ #if defined(WEBRTC_POSIX)
+-#ifdef BSD
++#ifdef WEBRTC_BSD
+ #include <stdlib.h>
+ #else  // BSD
+ #include <alloca.h>
diff --git a/audio/webrtc-audio-processing0/files/patch-webrtc_system__wrappers_source_condition__variable.cc b/audio/webrtc-audio-processing0/files/patch-webrtc_system__wrappers_source_condition__variable.cc
new file mode 100644
index 000000000000..b1c7dd0e2d06
--- /dev/null
+++ b/audio/webrtc-audio-processing0/files/patch-webrtc_system__wrappers_source_condition__variable.cc
@@ -0,0 +1,22 @@
+- Match conditional in webrtc/system_wrappers/Makefile.am
+
+--- webrtc/system_wrappers/source/condition_variable.cc.orig	2018-07-23 14:02:57 UTC
++++ webrtc/system_wrappers/source/condition_variable.cc
+@@ -14,7 +14,7 @@
+ #include <windows.h>
+ #include "webrtc/system_wrappers/source/condition_variable_event_win.h"
+ #include "webrtc/system_wrappers/source/condition_variable_native_win.h"
+-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
++#elif defined(WEBRTC_POSIX)
+ #include <pthread.h>
+ #include "webrtc/system_wrappers/source/condition_variable_posix.h"
+ #endif
+@@ -31,7 +31,7 @@ ConditionVariableWrapper* ConditionVariableWrapper::Cr
+     ret_val = new ConditionVariableEventWin();
+   }
+   return ret_val;
+-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
++#elif defined(WEBRTC_POSIX)
+   return ConditionVariablePosix::Create();
+ #else
+   return NULL;