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
2016-06-25 18:50:49 +02:00
|
|
|
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
|
|
|
|
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than 2016-05-24 08:28:45.749940095 -0400
|
|
|
|
+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc 2016-05-24 08:50:30.361020010 -0400
|
|
|
|
@@ -64,9 +64,6 @@ WavReader::~WavReader() {
|
|
|
|
|
2023-09-18 14:05:03 +02:00
|
|
|
size_t WavReader::ReadSamples(const size_t num_samples,
|
|
|
|
int16_t* const samples) {
|
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
2016-06-25 18:50:49 +02:00
|
|
|
-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
|
|
|
|
-#error "Need to convert samples to big-endian when reading from WAV file"
|
|
|
|
-#endif
|
2023-09-18 14:05:03 +02:00
|
|
|
|
|
|
|
size_t num_samples_left_to_read = num_samples;
|
|
|
|
size_t next_chunk_start = 0;
|
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
2016-06-25 18:50:49 +02:00
|
|
|
@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num
|
2023-09-18 14:05:03 +02:00
|
|
|
num_samples_left_to_read -= num_samples_read;
|
|
|
|
}
|
|
|
|
|
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
2016-06-25 18:50:49 +02:00
|
|
|
+#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
|
2023-09-18 14:05:03 +02:00
|
|
|
return num_samples - num_samples_left_to_read;
|
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
2016-06-25 18:50:49 +02:00
|
|
|
}
|
|
|
|
|
|
|
|
@@ -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() ||
|
|
|
|
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
|
|
|
|
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than 2016-05-24 08:50:52.591379263 -0400
|
|
|
|
+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc 2016-05-24 08:52:08.552606848 -0400
|
|
|
|
@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uin
|
|
|
|
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) {
|