Accepting request 404777 from home:oholecek:branches:multimedia:libs
- Remove webrtc-aarch64.patch, no longer needed - Adapt the rest of webrtc- patches to new arch naming - Remove unneeded explicit version dependency for automake - 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 - 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 - 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 - 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 OBS-URL: https://build.opensuse.org/request/show/404777 OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/webrtc-audio-processing?expand=0&rev=11
This commit is contained in:
parent
989d463d9f
commit
d47a474aa6
90
big_endian_support.patch
Normal file
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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}
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size_t WavReader::ReadSamples(size_t num_samples, int16_t* 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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// There could be metadata after the audio; ensure we don't read it.
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num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
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num_samples_remaining_);
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@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num
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RTC_CHECK(read == num_samples || feof(file_handle_));
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RTC_CHECK_LE(read, num_samples_remaining_);
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num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
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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 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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+ return std::string(reinterpret_cast<char*>(&x), 4);
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+}
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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
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24
big_endian_support_2.patch
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@ -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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#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
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@ -1,12 +0,0 @@
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--- src/typedefs.h
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+++ src/typedefs.h
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@@ -82,6 +82,9 @@
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#elif defined(__s390__)
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#define WEBRTC_BIG_ENDIAN
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#define WEBRTC_ARCH_32_BITS
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+#elif defined(__aarch64__)
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+#define WEBRTC_LITTLE_ENDIAN
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+#define WEBRTC_ARCH_64_BITS
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#else
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#error Please add support for your architecture in typedefs.h
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#endif
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@ -1,3 +0,0 @@
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version https://git-lfs.github.com/spec/v1
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oid sha256:ed4b52f9c2688b97628035a5565377d74704d7c04de4254a768df3342c7afedc
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size 392540
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3
webrtc-audio-processing-0.3.tar.xz
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3
webrtc-audio-processing-0.3.tar.xz
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@ -0,0 +1,3 @@
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version https://git-lfs.github.com/spec/v1
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oid sha256:756e291d4f557d88cd50c4fe3b8454ec238362d22cedb3e6173240d90f0a80fa
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size 688096
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@ -1,3 +1,80 @@
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-------------------------------------------------------------------
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Sat Jun 25 10:39:08 UTC 2016 - oholecek@suse.com
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- Remove webrtc-aarch64.patch, no longer needed
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- Adapt the rest of webrtc- patches to new arch naming
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-------------------------------------------------------------------
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Thu Jun 23 13:31:14 UTC 2016 - oholecek@suse.com
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- Remove unneeded explicit version dependency for automake
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-------------------------------------------------------------------
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Wed Jun 22 11:55:11 UTC 2016 - oholecek@suse.com
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- Update to 0.3
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* build: enforce linking with --no-undefined, add explicit -lpthread
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* build: Make sure files with SSE2 code are compiled with -msse2
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- Remove no-undefined.patch
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- Remove webrtc-audio-processing-0.2-x86_msse2.patch
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-------------------------------------------------------------------
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Mon Jun 20 13:02:06 UTC 2016 - oholecek@suse.com
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- Add no-undefined.patch patch
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https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6
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- Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
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- Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version
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- Adapt big_endian_support.patch to new version
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-------------------------------------------------------------------
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Mon May 30 09:00:51 UTC 2016 - oholecek@suse.com
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- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
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https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html
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- Add big_endian_support.patch
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https://bugs.freedesktop.org/show_bug.cgi?id=95738
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- New automake version dependency >= 1.5
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-------------------------------------------------------------------
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Thu May 26 21:19:28 UTC 2016 - oholecek@suse.com
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- Update to 0.2:
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Contains API breaking changes.
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Upstream changes include:
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* Rewritten AGC and voice activity detection
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* Intelligibility enhancer
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* Extended AEC filter
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* Beamformer
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* Transient suppressor
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* ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
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API changes:
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* We no longer include a top-level audio_processing.h. The webrtc tree format
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is used, so use webrtc/modules/audio_processing/include/audio_processing.h
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* The top-level module_common_types.h has also been moved to
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webrtc/modules/interface/module_common_types.h
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* C++11 support is now required while compiling client code
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* AudioProcessing::Create() does not take any arguments any more
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* AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
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* Stream parameters are now configured via StreamConfig and ProcessingConfig
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rather than set_sample_rate(), set_num_channels(), etc.
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* AudioFrame field names have changed
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* Use config API for newer audio processing options
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* Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
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when using the intelligibility enhancer
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* GainControl::set_analog_level_limits() is broken. The AGC implementation
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hard codes 0-255 as the volume range
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Other notes:
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* The new audio processing parameters are not all tested, and a few are not
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enabled upstream (in Chromium) either
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* The rewritten AGC appears to be less sensitive, and it might make sense to
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initialise the capture volume to something reasonable (33% or 50%, for
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example) to make sure there is sufficient energy in the stream to trigger
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the AGC mechanism
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- Adapted all 3 arch patches
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-------------------------------------------------------------------
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-------------------------------------------------------------------
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Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com
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Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com
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@ -2,7 +2,7 @@
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#
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#
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# spec file for package webrtc-audio-processing
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# spec file for package webrtc-audio-processing
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#
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#
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# Copyright (c) 2013 SUSE LINUX Products GmbH, Nuernberg, Germany.
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# Copyright (c) 2016 SUSE LINUX GmbH, Nuernberg, Germany.
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# Copyright (c) 2012 Pascal Bleser <pascal.bleser@opensuse.org>
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# Copyright (c) 2012 Pascal Bleser <pascal.bleser@opensuse.org>
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#
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#
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# All modifications and additions to the file contributed by third parties
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# All modifications and additions to the file contributed by third parties
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@ -18,18 +18,23 @@
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#
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#
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%define soname 1
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# Please submit bugfixes or comments via http://bugs.opensuse.org/
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# Please submit bugfixes or comments via http://bugs.opensuse.org/
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Name: webrtc-audio-processing
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Name: webrtc-audio-processing
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%define soname 0
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Version: 0.3
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Version: 0.1
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Release: 0
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Release: 0
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Summary: Real-Time Communication Library for Web Browsers
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Summary: Real-Time Communication Library for Web Browsers
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License: BSD-3-Clause
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License: BSD-3-Clause
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Group: System/Libraries
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Group: System/Libraries
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Source: http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.xz
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Url: http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
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Url: http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
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BuildRoot: %{_tmppath}/%{name}-%{version}-build
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Source: http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.xz
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# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
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Patch1: big_endian_support.patch
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# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
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Patch2: big_endian_support_2.patch
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# PATCH-FIX-OPENSUSE webrtc-(ppc64|s390x|aarch64).patch
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Patch100: webrtc-ppc64.patch
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Patch101: webrtc-s390x.patch
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BuildRequires: autoconf
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BuildRequires: autoconf
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BuildRequires: automake
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BuildRequires: automake
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BuildRequires: gcc-c++
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BuildRequires: gcc-c++
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@ -38,9 +43,7 @@ BuildRequires: libtool
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BuildRequires: make
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BuildRequires: make
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BuildRequires: pkgconfig
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BuildRequires: pkgconfig
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BuildRequires: xz
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BuildRequires: xz
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Patch0: webrtc-ppc64.patch
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BuildRoot: %{_tmppath}/%{name}-%{version}-build
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Patch1: webrtc-s390x.patch
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||||||
Patch2: webrtc-aarch64.patch
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%description
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%description
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WebRTC is an open source project that enables web browsers with Real-Time
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WebRTC is an open source project that enables web browsers with Real-Time
|
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@ -86,31 +89,29 @@ WebRTC implements the W3C's proposal for video conferencing on the web.
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|
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||||||
%prep
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%prep
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%setup -q -T -c "%{name}-%{version}"
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%setup -q -T -c "%{name}-%{version}"
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xz --decompress --stdout "%{SOURCE0}" | %__tar xf - --strip-components=1
|
xz --decompress --stdout "%{SOURCE0}" | tar xf - --strip-components=1
|
||||||
%__sed -i 's/\r$//' AUTHORS
|
sed -i 's/\r$//' AUTHORS
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%patch0 -p1
|
%patch1 -p1
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||||||
%patch1
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%patch2 -p1
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%patch2
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%patch100
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%patch101
|
||||||
|
|
||||||
%build
|
%build
|
||||||
%configure
|
%configure
|
||||||
%__make %{?_smp_mflags} V=1
|
make %{?_smp_mflags} V=1
|
||||||
|
|
||||||
%install
|
%install
|
||||||
%makeinstall
|
%makeinstall
|
||||||
|
|
||||||
%__rm -f "%{buildroot}%{_libdir}"/*.la
|
rm -f "%{buildroot}%{_libdir}"/*.la
|
||||||
|
|
||||||
%post -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
|
%post -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
|
||||||
|
|
||||||
%postun -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
|
%postun -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
|
||||||
|
|
||||||
%clean
|
|
||||||
%{?buildroot:%__rm -rf "%{buildroot}"}
|
|
||||||
|
|
||||||
%files -n libwebrtc_audio_processing%{soname}
|
%files -n libwebrtc_audio_processing%{soname}
|
||||||
%defattr(-,root,root)
|
%defattr(-,root,root)
|
||||||
%doc AUTHORS COPYING NEWS PATENTS README
|
%doc AUTHORS COPYING NEWS README.md UPDATING.md
|
||||||
%{_libdir}/libwebrtc_audio_processing.so.%{soname}
|
%{_libdir}/libwebrtc_audio_processing.so.%{soname}
|
||||||
%{_libdir}/libwebrtc_audio_processing.so.%{soname}.*.*
|
%{_libdir}/libwebrtc_audio_processing.so.%{soname}.*.*
|
||||||
|
|
||||||
|
@ -1,17 +1,17 @@
|
|||||||
Index: webrtc-audio-processing-0.1/src/typedefs.h
|
Index: webrtc/typedefs.h
|
||||||
===================================================================
|
===================================================================
|
||||||
--- webrtc-audio-processing-0.1.orig/src/typedefs.h
|
--- webrtc/typedefs.h.org
|
||||||
+++ webrtc-audio-processing-0.1/src/typedefs.h
|
+++ webrtc/typedefs.h
|
||||||
@@ -76,6 +76,12 @@
|
@@ -47,6 +47,12 @@
|
||||||
//#define WEBRTC_ARCH_ARMEL
|
#elif defined(__pnacl__)
|
||||||
#define WEBRTC_ARCH_32_BITS
|
#define WEBRTC_ARCH_32_BITS
|
||||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
+#elif defined(__powerpc64__)
|
+#elif defined(__powerpc64__)
|
||||||
+#define WEBRTC_BIG_ENDIAN
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
+#define WEBRTC_ARCH_64_BITS
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
+#elif defined(__powerpc__)
|
+#elif defined(__powerpc__)
|
||||||
+#define WEBRTC_BIG_ENDIAN
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
+#define WEBRTC_ARCH_32_BITS
|
+#define WEBRTC_ARCH_32_BITS
|
||||||
#else
|
#else
|
||||||
#error Please add support for your architecture in typedefs.h
|
/* instead of failing, use typical unix defines... */
|
||||||
#endif
|
#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
||||||
|
@ -1,15 +1,15 @@
|
|||||||
--- src/typedefs.h
|
--- webrtc/typedefs.h
|
||||||
+++ src/typedefs.h
|
+++ webrtc/typedefs.h
|
||||||
@@ -82,6 +82,12 @@
|
@@ -53,6 +53,12 @@
|
||||||
#elif defined(__powerpc__)
|
#elif defined(__powerpc__)
|
||||||
#define WEBRTC_BIG_ENDIAN
|
#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
#define WEBRTC_ARCH_32_BITS
|
#define WEBRTC_ARCH_32_BITS
|
||||||
+#elif defined(__s390x__)
|
+#elif defined(__s390x__)
|
||||||
+#define WEBRTC_BIG_ENDIAN
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
+#define WEBRTC_ARCH_64_BITS
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
+#elif defined(__s390__)
|
+#elif defined(__s390__)
|
||||||
+#define WEBRTC_BIG_ENDIAN
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
+#define WEBRTC_ARCH_32_BITS
|
+#define WEBRTC_ARCH_32_BITS
|
||||||
#else
|
#else
|
||||||
#error Please add support for your architecture in typedefs.h
|
/* instead of failing, use typical unix defines... */
|
||||||
#endif
|
#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
||||||
|
Loading…
Reference in New Issue
Block a user