Sync from SUSE:ALP:Source:Standard:1.0 webrtc-audio-processing revision 3359b25363aa46aa70faf001800fca6a
This commit is contained in:
commit
d20e6ece78
23
.gitattributes
vendored
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23
.gitattributes
vendored
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@ -0,0 +1,23 @@
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## Default LFS
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*.7z filter=lfs diff=lfs merge=lfs -text
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*.bsp filter=lfs diff=lfs merge=lfs -text
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*.bz2 filter=lfs diff=lfs merge=lfs -text
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*.gem filter=lfs diff=lfs merge=lfs -text
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*.gz filter=lfs diff=lfs merge=lfs -text
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*.jar filter=lfs diff=lfs merge=lfs -text
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*.lz filter=lfs diff=lfs merge=lfs -text
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*.lzma filter=lfs diff=lfs merge=lfs -text
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*.obscpio filter=lfs diff=lfs merge=lfs -text
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*.oxt filter=lfs diff=lfs merge=lfs -text
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*.pdf filter=lfs diff=lfs merge=lfs -text
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*.png filter=lfs diff=lfs merge=lfs -text
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*.rpm filter=lfs diff=lfs merge=lfs -text
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*.tbz filter=lfs diff=lfs merge=lfs -text
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*.tbz2 filter=lfs diff=lfs merge=lfs -text
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*.tgz filter=lfs diff=lfs merge=lfs -text
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*.ttf filter=lfs diff=lfs merge=lfs -text
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*.txz filter=lfs diff=lfs merge=lfs -text
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*.whl filter=lfs diff=lfs merge=lfs -text
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*.xz filter=lfs diff=lfs merge=lfs -text
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*.zip filter=lfs diff=lfs merge=lfs -text
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*.zst filter=lfs diff=lfs merge=lfs -text
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||||||
20
_service
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20
_service
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@ -0,0 +1,20 @@
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<?xml version="1.0"?>
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||||||
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<services>
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||||||
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<service name="obs_scm" mode="manual">
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||||||
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<param name="scm">git</param>
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||||||
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<param name="url">https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing.git</param>
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||||||
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<param name="revision">v1.3</param>
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||||||
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<param name="versionformat">1.3</param>
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||||||
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<!--
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||||||
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<param name="revision">master</param>
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||||||
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<param name="versionformat">@PARENT_TAG@+git%cd.%h</param>
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||||||
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-->
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||||||
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</service>
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||||||
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<service name="tar" mode="buildtime"/>
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||||||
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<service name="recompress" mode="buildtime">
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||||||
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<param name="file">*.tar</param>
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||||||
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<param name="compression">xz</param>
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</service>
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<service name="set_version" mode="manual" />
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</services>
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2
baselibs.conf
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2
baselibs.conf
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@ -0,0 +1,2 @@
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libwebrtc-audio-processing-1-3
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libwebrtc-audio-coding-1-3
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90
big_endian_support.patch
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90
big_endian_support.patch
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@ -0,0 +1,90 @@
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diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc
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--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than 2016-05-24 08:28:45.749940095 -0400
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+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc 2016-05-24 08:50:30.361020010 -0400
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@@ -64,9 +64,6 @@ WavReader::~WavReader() {
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size_t WavReader::ReadSamples(const size_t num_samples,
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int16_t* const samples) {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Need to convert samples to big-endian when reading from WAV file"
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-#endif
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size_t num_samples_left_to_read = num_samples;
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size_t next_chunk_start = 0;
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@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num
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num_samples_left_to_read -= num_samples_read;
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}
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+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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+ //convert to big-endian
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+ for(size_t idx = 0; idx < num_samples; idx++) {
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+ samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
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+ }
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+#endif
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return num_samples - num_samples_left_to_read;
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}
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@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
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void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
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#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Need to convert samples to little-endian when writing to WAV file"
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-#endif
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+ int16_t * le_samples = new int16_t[num_samples];
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+ for(size_t idx = 0; idx < num_samples; idx++) {
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+ le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
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+ }
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+ const size_t written =
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+ fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_);
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+ delete []le_samples;
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+#else
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const size_t written =
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fwrite(samples, sizeof(*samples), num_samples, file_handle_);
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+#endif
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RTC_CHECK_EQ(num_samples, written);
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num_samples_ += static_cast<uint32_t>(written);
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RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
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diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc
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--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than 2016-05-24 08:50:52.591379263 -0400
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+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc 2016-05-24 08:52:08.552606848 -0400
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@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uin
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return std::string(reinterpret_cast<char*>(&x), 4);
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}
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#else
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-#error "Write be-to-le conversion functions"
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+static inline void WriteLE16(uint16_t* f, uint16_t x) {
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+ *f = ((x << 8) & 0xff00) | ( ( x >> 8) & 0x00ff);
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+}
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+
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+static inline void WriteLE32(uint32_t* f, uint32_t x) {
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+ *f = ( (x & 0x000000ff) << 24 )
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+ | ((x & 0x0000ff00) << 8)
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+ | ((x & 0x00ff0000) >> 8)
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+ | ((x & 0xff000000) >> 24 );
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+}
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+
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+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
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+ *f = (static_cast<uint32_t>(a) << 24 )
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+ | (static_cast<uint32_t>(b) << 16)
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+ | (static_cast<uint32_t>(c) << 8)
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+ | (static_cast<uint32_t>(d) );
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+}
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+
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+static inline uint16_t ReadLE16(uint16_t x) {
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+ return (( x & 0x00ff) << 8 )| ((x & 0xff00)>>8);
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|
+}
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+
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|
+static inline uint32_t ReadLE32(uint32_t x) {
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+ return ( (x & 0x000000ff) << 24 )
|
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+ | ( (x & 0x0000ff00) << 8 )
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+ | ( (x & 0x00ff0000) >> 8)
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+ | ( (x & 0xff000000) >> 24 );
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+}
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|
+
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||||||
|
+static inline std::string ReadFourCC(uint32_t x) {
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|
+ x = ReadLE32(x);
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||||||
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+ return std::string(reinterpret_cast<char*>(&x), 4);
|
||||||
|
+}
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||||||
|
#endif
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static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {
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24
big_endian_support_2.patch
Normal file
24
big_endian_support_2.patch
Normal file
@ -0,0 +1,24 @@
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diff -up webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef webrtc-audio-processing-0.2/webrtc/typedefs.h
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--- webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef 2016-05-12 09:08:53.885000410 -0500
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+++ webrtc-audio-processing-0.2/webrtc/typedefs.h 2016-05-12 09:12:38.006851953 -0500
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@@ -48,7 +48,19 @@
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#define WEBRTC_ARCH_32_BITS
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#define WEBRTC_ARCH_LITTLE_ENDIAN
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#else
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-#error Please add support for your architecture in typedefs.h
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+/* instead of failing, use typical unix defines... */
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+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
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+#define WEBRTC_ARCH_LITTLE_ENDIAN
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+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
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|
+#define WEBRTC_ARCH_BIG_ENDIAN
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+#else
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+#error __BYTE_ORDER__ is not defined
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|
+#endif
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|
+#if defined(__LP64__)
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|
+#define WEBRTC_ARCH_64_BITS
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+#else
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+#define WEBRTC_ARCH_32_BITS
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+#endif
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|
#endif
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|
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||||||
|
#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
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60
fix-build.patch
Normal file
60
fix-build.patch
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
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|
===================================================================
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|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
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||||||
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@@ -39,6 +39,7 @@ float GetLevel(const VadLevelAnalyzer::R
|
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return vad_level.rms_dbfs;
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|
break;
|
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|
case LevelEstimatorType::kPeak:
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|
+ default:
|
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|
return vad_level.peak_dbfs;
|
||||||
|
break;
|
||||||
|
}
|
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|
Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc
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===================================================================
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|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/audio_processing_impl.cc
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/audio_processing_impl.cc
|
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@@ -112,6 +112,7 @@ GainControl::Mode Agc1ConfigModeToInterf
|
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|
case Agc1Config::kAdaptiveDigital:
|
||||||
|
return GainControl::kAdaptiveDigital;
|
||||||
|
case Agc1Config::kFixedDigital:
|
||||||
|
+ default:
|
||||||
|
return GainControl::kFixedDigital;
|
||||||
|
}
|
||||||
|
}
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|
@@ -1852,6 +1853,7 @@ void AudioProcessingImpl::InitializeNois
|
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|
return NsConfig::SuppressionLevel::k21dB;
|
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default:
|
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|
RTC_NOTREACHED();
|
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|
+ return NsConfig::SuppressionLevel::k21dB; // Just to keep the compiler happy
|
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|
}
|
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|
};
|
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|
|
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Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/include/audio_processing.cc
|
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===================================================================
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|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/include/audio_processing.cc
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+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/include/audio_processing.cc
|
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@@ -26,6 +26,7 @@ std::string NoiseSuppressionLevelToStrin
|
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case AudioProcessing::Config::NoiseSuppression::Level::kHigh:
|
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return "High";
|
||||||
|
case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
|
||||||
|
+ default:
|
||||||
|
return "VeryHigh";
|
||||||
|
}
|
||||||
|
}
|
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@@ -38,6 +39,7 @@ std::string GainController1ModeToString(
|
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case AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital:
|
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return "AdaptiveDigital";
|
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|
case AudioProcessing::Config::GainController1::Mode::kFixedDigital:
|
||||||
|
+ default:
|
||||||
|
return "FixedDigital";
|
||||||
|
}
|
||||||
|
}
|
||||||
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@@ -48,6 +50,7 @@ std::string GainController2LevelEstimato
|
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|
case AudioProcessing::Config::GainController2::LevelEstimator::kRms:
|
||||||
|
return "Rms";
|
||||||
|
case AudioProcessing::Config::GainController2::LevelEstimator::kPeak:
|
||||||
|
+ default:
|
||||||
|
return "Peak";
|
||||||
|
}
|
||||||
|
}
|
||||||
126
fix-i586.patch
Normal file
126
fix-i586.patch
Normal file
@ -0,0 +1,126 @@
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|
Index: webrtc-audio-processing-1.3/webrtc/third_party/pffft/src/pffft.c
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/webrtc/third_party/pffft/src/pffft.c
|
||||||
|
+++ webrtc-audio-processing-1.3/webrtc/third_party/pffft/src/pffft.c
|
||||||
|
@@ -131,7 +131,7 @@ inline v4sf ld_ps1(const float *p) { v4s
|
||||||
|
/*
|
||||||
|
SSE1 support macros
|
||||||
|
*/
|
||||||
|
-#elif !defined(PFFFT_SIMD_DISABLE) && (defined(__x86_64__) || defined(_M_X64) || defined(i386) || defined(__i386__) || defined(_M_IX86))
|
||||||
|
+#elif !defined(PFFFT_SIMD_DISABLE) && (defined(__x86_64__) || defined(_M_X64) || defined(i386) || defined(__i386__) || defined(_M_IX86)) && defined(__SSE2__)
|
||||||
|
|
||||||
|
#include <xmmintrin.h>
|
||||||
|
typedef __m128 v4sf;
|
||||||
|
Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
|
||||||
|
+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
|
||||||
|
@@ -88,6 +88,7 @@ void ComputeFrequencyResponse_Neon(
|
||||||
|
|
||||||
|
#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
// Computes and stores the frequency response of the filter.
|
||||||
|
+__attribute__((target("sse2")))
|
||||||
|
void ComputeFrequencyResponse_Sse2(
|
||||||
|
size_t num_partitions,
|
||||||
|
const std::vector<std::vector<FftData>>& H,
|
||||||
|
@@ -207,9 +208,10 @@ void AdaptPartitions_Neon(const RenderBu
|
||||||
|
} while (p < lim2);
|
||||||
|
}
|
||||||
|
#endif
|
||||||
|
-
|
||||||
|
+
|
||||||
|
#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
// Adapts the filter partitions. (SSE2 variant)
|
||||||
|
+__attribute__((target("sse2")))
|
||||||
|
void AdaptPartitions_Sse2(const RenderBuffer& render_buffer,
|
||||||
|
const FftData& G,
|
||||||
|
size_t num_partitions,
|
||||||
|
@@ -375,6 +377,7 @@ void ApplyFilter_Neon(const RenderBuffer
|
||||||
|
|
||||||
|
#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
// Produces the filter output (SSE2 variant).
|
||||||
|
+__attribute__((target("sse2")))
|
||||||
|
void ApplyFilter_Sse2(const RenderBuffer& render_buffer,
|
||||||
|
size_t num_partitions,
|
||||||
|
const std::vector<std::vector<FftData>>& H,
|
||||||
|
Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/matched_filter.cc
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/matched_filter.cc
|
||||||
|
+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/matched_filter.cc
|
||||||
|
@@ -143,7 +143,7 @@ void MatchedFilterCore_NEON(size_t x_sta
|
||||||
|
#endif
|
||||||
|
|
||||||
|
#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
-
|
||||||
|
+__attribute__((target("sse2")))
|
||||||
|
void MatchedFilterCore_SSE2(size_t x_start_index,
|
||||||
|
float x2_sum_threshold,
|
||||||
|
float smoothing,
|
||||||
|
Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/fft_data.h
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/fft_data.h
|
||||||
|
+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/fft_data.h
|
||||||
|
@@ -48,7 +48,7 @@ struct FftData {
|
||||||
|
rtc::ArrayView<float> power_spectrum) const {
|
||||||
|
RTC_DCHECK_EQ(kFftLengthBy2Plus1, power_spectrum.size());
|
||||||
|
switch (optimization) {
|
||||||
|
-#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
|
||||||
|
case Aec3Optimization::kSse2: {
|
||||||
|
constexpr int kNumFourBinBands = kFftLengthBy2 / 4;
|
||||||
|
constexpr int kLimit = kNumFourBinBands * 4;
|
||||||
|
Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/vector_math.h
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/vector_math.h
|
||||||
|
+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/vector_math.h
|
||||||
|
@@ -43,7 +43,7 @@ class VectorMath {
|
||||||
|
void SqrtAVX2(rtc::ArrayView<float> x);
|
||||||
|
void Sqrt(rtc::ArrayView<float> x) {
|
||||||
|
switch (optimization_) {
|
||||||
|
-#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
|
||||||
|
case Aec3Optimization::kSse2: {
|
||||||
|
const int x_size = static_cast<int>(x.size());
|
||||||
|
const int vector_limit = x_size >> 2;
|
||||||
|
@@ -123,7 +123,7 @@ class VectorMath {
|
||||||
|
RTC_DCHECK_EQ(z.size(), x.size());
|
||||||
|
RTC_DCHECK_EQ(z.size(), y.size());
|
||||||
|
switch (optimization_) {
|
||||||
|
-#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
|
||||||
|
case Aec3Optimization::kSse2: {
|
||||||
|
const int x_size = static_cast<int>(x.size());
|
||||||
|
const int vector_limit = x_size >> 2;
|
||||||
|
@@ -173,7 +173,7 @@ class VectorMath {
|
||||||
|
void Accumulate(rtc::ArrayView<const float> x, rtc::ArrayView<float> z) {
|
||||||
|
RTC_DCHECK_EQ(z.size(), x.size());
|
||||||
|
switch (optimization_) {
|
||||||
|
-#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__SSE2__)
|
||||||
|
case Aec3Optimization::kSse2: {
|
||||||
|
const int x_size = static_cast<int>(x.size());
|
||||||
|
const int vector_limit = x_size >> 2;
|
||||||
|
Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
|
||||||
|
+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
|
||||||
|
@@ -229,6 +229,7 @@ void ComputeFullyConnectedLayerOutput(
|
||||||
|
|
||||||
|
#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
// Fully connected layer SSE2 implementation.
|
||||||
|
+__attribute__((target("sse2")))
|
||||||
|
void ComputeFullyConnectedLayerOutputSse2(
|
||||||
|
size_t input_size,
|
||||||
|
size_t output_size,
|
||||||
|
Index: webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
|
||||||
|
+++ webrtc-audio-processing-1.3/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_erl.cc
|
||||||
|
@@ -57,6 +57,7 @@ void ErlComputer_NEON(
|
||||||
|
#if defined(WEBRTC_ARCH_X86_FAMILY)
|
||||||
|
// Computes and stores the echo return loss estimate of the filter, which is the
|
||||||
|
// sum of the partition frequency responses.
|
||||||
|
+__attribute__((target("sse2")))
|
||||||
|
void ErlComputer_SSE2(
|
||||||
|
const std::vector<std::array<float, kFftLengthBy2Plus1>>& H2,
|
||||||
|
rtc::ArrayView<float> erl) {
|
||||||
12
reduce-meson-dep.patch
Normal file
12
reduce-meson-dep.patch
Normal file
@ -0,0 +1,12 @@
|
|||||||
|
Index: webrtc-audio-processing-1.3/meson.build
|
||||||
|
===================================================================
|
||||||
|
--- webrtc-audio-processing-1.3.orig/meson.build
|
||||||
|
+++ webrtc-audio-processing-1.3/meson.build
|
||||||
|
@@ -1,6 +1,6 @@
|
||||||
|
project('webrtc-audio-processing', 'c', 'cpp',
|
||||||
|
version : '1.3',
|
||||||
|
- meson_version : '>= 0.63',
|
||||||
|
+ meson_version : '>= 0.59.4',
|
||||||
|
default_options : [ 'warning_level=1',
|
||||||
|
'buildtype=debugoptimized',
|
||||||
|
'c_std=c11',
|
||||||
BIN
webrtc-audio-processing-1.3.obscpio
(Stored with Git LFS)
Normal file
BIN
webrtc-audio-processing-1.3.obscpio
(Stored with Git LFS)
Normal file
Binary file not shown.
172
webrtc-audio-processing.changes
Normal file
172
webrtc-audio-processing.changes
Normal file
@ -0,0 +1,172 @@
|
|||||||
|
-------------------------------------------------------------------
|
||||||
|
Mon Oct 30 16:42:04 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
|
||||||
|
|
||||||
|
- ExcludeArch s390, s390x and ppc64 since big endian support is
|
||||||
|
not implemented.
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Wed Sep 20 09:49:19 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
|
||||||
|
|
||||||
|
- Remove the tar.xz file. Having the obscpio file is enough
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Wed Sep 20 09:38:21 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
|
||||||
|
|
||||||
|
- Use also dashes instead of underscores in the manual Requires
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Wed Sep 20 09:04:13 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
|
||||||
|
|
||||||
|
- Rename the generated library package names to add a dash between
|
||||||
|
the name and soname (libwebrtc*-1-3 instead of libwebrtc*1-3)
|
||||||
|
- Rename the generated packages to use dashes instead of underscores
|
||||||
|
- Change baselibs.conf accordingly
|
||||||
|
- Add patch to reduce the required meson version so the package
|
||||||
|
builds in Leap 15.4/15.5:
|
||||||
|
* reduce-meson-dep.patch
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Fri Sep 08 10:40:12 UTC 2023 - alarrosa@suse.com
|
||||||
|
|
||||||
|
- Update to version 1.3:
|
||||||
|
* build: Bump version to 1.3
|
||||||
|
* meson: Fix generation of pkgconfig files
|
||||||
|
* build: Bump version to 1.2
|
||||||
|
* meson: Update minimum version based on what abseil wrap needs
|
||||||
|
* build: Expose absl as a dependency of webrtc-audio-processing
|
||||||
|
* meson: Update to latest wrap, install required absl headers
|
||||||
|
* doc: Update tarball generation process
|
||||||
|
* file_utils.h: Fix build with gcc-13
|
||||||
|
* meson: Fixes for MSVC build
|
||||||
|
* meson: Ensure that abseil is built with c++17 too
|
||||||
|
* More changes not listed by upstream. Check
|
||||||
|
the following link to see them:
|
||||||
|
https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/commits/v1.3
|
||||||
|
- Add patch that fixes some compiler "control reaches end of
|
||||||
|
non-void function" errors:
|
||||||
|
* fix-build.patch
|
||||||
|
- Add patch that fixes i586 build:
|
||||||
|
* fix-i586.patch
|
||||||
|
- Disable patches until they're rebased to the current codebase:
|
||||||
|
* big_endian_support.patch
|
||||||
|
* big_endian_support_2.patch
|
||||||
|
- Rebased patches:
|
||||||
|
* webrtc-ppc64.patch
|
||||||
|
* webrtc-s390x.patch
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Mon Aug 17 15:30:03 UTC 2020 - Dirk Mueller <dmueller@suse.com>
|
||||||
|
|
||||||
|
- update to 0.3.1:
|
||||||
|
* doc: file invalid reference to pulseaudio mailing list
|
||||||
|
* various build system fixes
|
||||||
|
- spec-cleaner run
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Fri Aug 2 08:23:00 UTC 2019 - Martin Liška <mliska@suse.cz>
|
||||||
|
|
||||||
|
- Use FAT LTO objects in order to provide proper static library.
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu Jan 12 08:32:04 UTC 2017 - olaf@aepfle.de
|
||||||
|
|
||||||
|
- Add baselibs.conf for gstreamer-plugins-bad-32bit
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Sat Jun 25 10:39:08 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Remove webrtc-aarch64.patch, no longer needed
|
||||||
|
- Adapt the rest of webrtc- patches to new arch naming
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu Jun 23 13:31:14 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Remove unneeded explicit version dependency for automake
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Wed Jun 22 11:55:11 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Update to 0.3
|
||||||
|
* build: enforce linking with --no-undefined, add explicit -lpthread
|
||||||
|
* build: Make sure files with SSE2 code are compiled with -msse2
|
||||||
|
- Remove no-undefined.patch
|
||||||
|
- Remove webrtc-audio-processing-0.2-x86_msse2.patch
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Mon Jun 20 13:02:06 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Add no-undefined.patch patch
|
||||||
|
https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6
|
||||||
|
- Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
- Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version
|
||||||
|
- Adapt big_endian_support.patch to new version
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Mon May 30 09:00:51 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
|
||||||
|
https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html
|
||||||
|
- Add big_endian_support.patch
|
||||||
|
https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
- New automake version dependency >= 1.5
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu May 26 21:19:28 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Update to 0.2:
|
||||||
|
Contains API breaking changes.
|
||||||
|
|
||||||
|
Upstream changes include:
|
||||||
|
* Rewritten AGC and voice activity detection
|
||||||
|
* Intelligibility enhancer
|
||||||
|
* Extended AEC filter
|
||||||
|
* Beamformer
|
||||||
|
* Transient suppressor
|
||||||
|
* ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
|
||||||
|
|
||||||
|
API changes:
|
||||||
|
* We no longer include a top-level audio_processing.h. The webrtc tree format
|
||||||
|
is used, so use webrtc/modules/audio_processing/include/audio_processing.h
|
||||||
|
* The top-level module_common_types.h has also been moved to
|
||||||
|
webrtc/modules/interface/module_common_types.h
|
||||||
|
* C++11 support is now required while compiling client code
|
||||||
|
* AudioProcessing::Create() does not take any arguments any more
|
||||||
|
* AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
|
||||||
|
* Stream parameters are now configured via StreamConfig and ProcessingConfig
|
||||||
|
rather than set_sample_rate(), set_num_channels(), etc.
|
||||||
|
* AudioFrame field names have changed
|
||||||
|
* Use config API for newer audio processing options
|
||||||
|
* Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
|
||||||
|
when using the intelligibility enhancer
|
||||||
|
* GainControl::set_analog_level_limits() is broken. The AGC implementation
|
||||||
|
hard codes 0-255 as the volume range
|
||||||
|
|
||||||
|
Other notes:
|
||||||
|
* The new audio processing parameters are not all tested, and a few are not
|
||||||
|
enabled upstream (in Chromium) either
|
||||||
|
* The rewritten AGC appears to be less sensitive, and it might make sense to
|
||||||
|
initialise the capture volume to something reasonable (33% or 50%, for
|
||||||
|
example) to make sure there is sufficient energy in the stream to trigger
|
||||||
|
the AGC mechanism
|
||||||
|
- Adapted all 3 arch patches
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com
|
||||||
|
|
||||||
|
- Add patch webrtc-aarch64.patch from algraf to add aarch64 support
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Wed Dec 19 10:39:23 CET 2012 - ro@suse.de
|
||||||
|
|
||||||
|
- add s390 and s390x to known platforms
|
||||||
|
by adding webrtc-s390x.patch
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Tue Jul 3 15:00:06 UTC 2012 - dvaleev@suse.com
|
||||||
|
|
||||||
|
- add ppc64 to known platforms
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Tue May 15 10:40:38 CET 2012 - pascal.bleser@opensuse.org
|
||||||
|
|
||||||
|
- initial version (0.1)
|
||||||
|
|
||||||
4
webrtc-audio-processing.obsinfo
Normal file
4
webrtc-audio-processing.obsinfo
Normal file
@ -0,0 +1,4 @@
|
|||||||
|
name: webrtc-audio-processing
|
||||||
|
version: 1.3
|
||||||
|
mtime: 1693927187
|
||||||
|
commit: 8e258a1933d405073c9e6465628a69ac7d2a1f13
|
||||||
190
webrtc-audio-processing.spec
Normal file
190
webrtc-audio-processing.spec
Normal file
@ -0,0 +1,190 @@
|
|||||||
|
# vim: set sw=4 ts=4 et nu:
|
||||||
|
#
|
||||||
|
# spec file for package webrtc-audio-processing
|
||||||
|
#
|
||||||
|
# Copyright (c) 2023 SUSE LLC
|
||||||
|
# Copyright (c) 2012 Pascal Bleser <pascal.bleser@opensuse.org>
|
||||||
|
#
|
||||||
|
# All modifications and additions to the file contributed by third parties
|
||||||
|
# remain the property of their copyright owners, unless otherwise agreed
|
||||||
|
# upon. The license for this file, and modifications and additions to the
|
||||||
|
# file, is the same license as for the pristine package itself (unless the
|
||||||
|
# license for the pristine package is not an Open Source License, in which
|
||||||
|
# case the license is the MIT License). An "Open Source License" is a
|
||||||
|
# license that conforms to the Open Source Definition (Version 1.9)
|
||||||
|
# published by the Open Source Initiative.
|
||||||
|
|
||||||
|
# Please submit bugfixes or comments via https://bugs.opensuse.org/
|
||||||
|
#
|
||||||
|
|
||||||
|
|
||||||
|
%define pkg_soname 1-3
|
||||||
|
%define soname 3
|
||||||
|
# Please submit bugfixes or comments via http://bugs.opensuse.org/
|
||||||
|
Name: webrtc-audio-processing
|
||||||
|
Version: 1.3
|
||||||
|
Release: 0
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
License: BSD-3-Clause
|
||||||
|
Group: System/Libraries
|
||||||
|
URL: https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
|
||||||
|
Source: webrtc-audio-processing-%{version}.tar.xz
|
||||||
|
Source1: baselibs.conf
|
||||||
|
# PATCH-FIX-UPSTREAM fix-build.patch alarrosa@suse.com -- Fix a number of "control reaches end of non-void function" errors
|
||||||
|
Patch0: fix-build.patch
|
||||||
|
# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
Patch1: big_endian_support.patch
|
||||||
|
# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
Patch2: big_endian_support_2.patch
|
||||||
|
Patch3: fix-i586.patch
|
||||||
|
# PATCH-FIX-OPENSUSE webrtc-(ppc64|s390x|aarch64).patch
|
||||||
|
Patch100: webrtc-ppc64.patch
|
||||||
|
Patch101: webrtc-s390x.patch
|
||||||
|
# PATCH-FIX-OPENSUSE reduce-meson-dep.patch
|
||||||
|
Patch102: reduce-meson-dep.patch
|
||||||
|
BuildRequires: cmake
|
||||||
|
BuildRequires: gcc-c++
|
||||||
|
BuildRequires: glibc-devel
|
||||||
|
BuildRequires: libtool
|
||||||
|
BuildRequires: make
|
||||||
|
BuildRequires: meson >= 0.59.4
|
||||||
|
BuildRequires: pkgconfig
|
||||||
|
BuildRequires: xz
|
||||||
|
BuildRequires: cmake(absl)
|
||||||
|
ExcludeArch: s390 s390x ppc64
|
||||||
|
|
||||||
|
%description
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
%package -n libwebrtc-audio-processing-%{pkg_soname}
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: System/Libraries
|
||||||
|
|
||||||
|
%description -n libwebrtc-audio-processing-%{pkg_soname}
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
%package -n libwebrtc-audio-processing-devel
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: Development/Libraries/C and C++
|
||||||
|
Requires: libwebrtc-audio-processing-%{pkg_soname} = %{version}
|
||||||
|
|
||||||
|
%description -n libwebrtc-audio-processing-devel
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
%package -n libwebrtc-audio-processing-devel-static
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: Development/Libraries/C and C++
|
||||||
|
Requires: libwebrtc-audio-processing-devel = %{version}
|
||||||
|
|
||||||
|
%description -n libwebrtc-audio-processing-devel-static
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
%package -n libwebrtc-audio-coding-%{pkg_soname}
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: System/Libraries
|
||||||
|
|
||||||
|
%description -n libwebrtc-audio-coding-%{pkg_soname}
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
%package -n libwebrtc-audio-coding-devel
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: Development/Libraries/C and C++
|
||||||
|
Requires: libwebrtc-audio-coding-%{pkg_soname} = %{version}
|
||||||
|
|
||||||
|
%description -n libwebrtc-audio-coding-devel
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
%package -n libwebrtc-audio-coding-devel-static
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: Development/Libraries/C and C++
|
||||||
|
Requires: libwebrtc-audio-coding-devel = %{version}
|
||||||
|
|
||||||
|
%description -n libwebrtc-audio-coding-devel-static
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
%prep
|
||||||
|
%autosetup -p1 -N
|
||||||
|
sed -i 's/\r$//' AUTHORS
|
||||||
|
%patch0 -p1
|
||||||
|
#%%patch1 -p1
|
||||||
|
#%%patch2 -p1
|
||||||
|
%patch3 -p1
|
||||||
|
%patch100 -p1
|
||||||
|
%patch101 -p1
|
||||||
|
%patch102 -p1
|
||||||
|
|
||||||
|
%build
|
||||||
|
%global _lto_cflags %{_lto_cflags} -ffat-lto-objects
|
||||||
|
%meson \
|
||||||
|
-Dc_std=gnu11 \
|
||||||
|
-Dcpp_std=gnu++17 \
|
||||||
|
-Ddefault_library=both \
|
||||||
|
-Dc_args="${CFLAGS} ${LDFLAGS}" \
|
||||||
|
-Dcpp_args="${CXXFLAGS} ${LDFLAGS}" \
|
||||||
|
%{nil}
|
||||||
|
%meson_build
|
||||||
|
|
||||||
|
%install
|
||||||
|
%meson_install
|
||||||
|
|
||||||
|
find %{buildroot} -type f -name "*.la" -delete -print
|
||||||
|
|
||||||
|
%post -n libwebrtc-audio-processing-%{pkg_soname} -p /sbin/ldconfig
|
||||||
|
%postun -n libwebrtc-audio-processing-%{pkg_soname} -p /sbin/ldconfig
|
||||||
|
%post -n libwebrtc-audio-coding-%{pkg_soname} -p /sbin/ldconfig
|
||||||
|
%postun -n libwebrtc-audio-coding-%{pkg_soname} -p /sbin/ldconfig
|
||||||
|
|
||||||
|
%files -n libwebrtc-audio-processing-%{pkg_soname}
|
||||||
|
%license COPYING
|
||||||
|
%doc AUTHORS NEWS README.md UPDATING.md
|
||||||
|
%{_libdir}/libwebrtc-audio-processing-1.so.%{soname}*
|
||||||
|
|
||||||
|
%files -n libwebrtc-audio-processing-devel
|
||||||
|
%{_includedir}/webrtc-audio-processing-1
|
||||||
|
%{_libdir}/libwebrtc-audio-processing-1.so
|
||||||
|
%{_libdir}/pkgconfig/webrtc-audio-processing-1.pc
|
||||||
|
|
||||||
|
%files -n libwebrtc-audio-processing-devel-static
|
||||||
|
%{_libdir}/libwebrtc-audio-processing-1.a
|
||||||
|
|
||||||
|
%files -n libwebrtc-audio-coding-%{pkg_soname}
|
||||||
|
%license COPYING
|
||||||
|
%doc AUTHORS NEWS README.md UPDATING.md
|
||||||
|
%{_libdir}/libwebrtc-audio-coding-1.so.%{soname}*
|
||||||
|
|
||||||
|
%files -n libwebrtc-audio-coding-devel
|
||||||
|
%{_libdir}/libwebrtc-audio-coding-1.so
|
||||||
|
%{_libdir}/pkgconfig/webrtc-audio-coding-1.pc
|
||||||
|
|
||||||
|
%files -n libwebrtc-audio-coding-devel-static
|
||||||
|
%{_libdir}/libwebrtc-audio-coding-1.a
|
||||||
|
|
||||||
|
%changelog
|
||||||
26
webrtc-ppc64.patch
Normal file
26
webrtc-ppc64.patch
Normal file
@ -0,0 +1,26 @@
|
|||||||
|
Index: webrtc/typedefs.h
|
||||||
|
===================================================================
|
||||||
|
--- a/webrtc/rtc_base/system/arch.h.orig
|
||||||
|
+++ b/webrtc/rtc_base/system/arch.h
|
||||||
|
@@ -57,6 +57,15 @@
|
||||||
|
# #elif defined(__pnacl__)
|
||||||
|
# #define WEBRTC_ARCH_32_BITS
|
||||||
|
# #define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
|
#elif defined(__EMSCRIPTEN__)
|
||||||
|
#define WEBRTC_ARCH_32_BITS
|
||||||
|
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
|
+#elif defined(__powerpc64__) && defined(__LITTLE_ENDIAN__)
|
||||||
|
+#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
|
+#elif defined(__powerpc64__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
|
+#elif defined(__powerpc__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_32_BITS
|
||||||
|
#else
|
||||||
|
#error Please add support for your architecture in rtc_base/system/arch.h
|
||||||
|
#endif
|
||||||
|
# #else
|
||||||
|
# /* instead of failing, use typical unix defines... */
|
||||||
|
# #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
||||||
18
webrtc-s390x.patch
Normal file
18
webrtc-s390x.patch
Normal file
@ -0,0 +1,18 @@
|
|||||||
|
--- a/webrtc/rtc_base/system/arch.h.orig
|
||||||
|
+++ b/webrtc/rtc_base/system/arch.h
|
||||||
|
@@ -63,6 +63,12 @@
|
||||||
|
#elif defined(__powerpc__)
|
||||||
|
#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
#define WEBRTC_ARCH_32_BITS
|
||||||
|
+#elif defined(__s390x__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
|
+#elif defined(__s390__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_32_BITS
|
||||||
|
#else
|
||||||
|
#error Please add support for your architecture in rtc_base/system/arch.h
|
||||||
|
#endif
|
||||||
|
# #else
|
||||||
|
# /* instead of failing, use typical unix defines... */
|
||||||
|
# #if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
||||||
Loading…
Reference in New Issue
Block a user