Compare commits
No commits in common. 'cs10' and 'c9' have entirely different histories.
@ -1,3 +1 @@
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SOURCES/abseil-cpp-20230125.1.tar.gz
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SOURCES/abseil-cpp_20230125.1-4_patch.zip
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SOURCES/webrtc-audio-processing-1.3.tar.xz
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SOURCES/webrtc-audio-processing-0.3.1.tar.xz
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@ -1,3 +1 @@
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609817274bf327339efc8488bb773d5f633ada62 SOURCES/abseil-cpp-20230125.1.tar.gz
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11b3e50fd05da4c77f270ae7167a44539ab34e59 SOURCES/abseil-cpp_20230125.1-4_patch.zip
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f7cd684db4429722e7f8d8ed6559094aaf4dfc18 SOURCES/webrtc-audio-processing-1.3.tar.xz
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7aa63a6bfe0e5056cfcf883b2de3496ab214ac13 SOURCES/webrtc-audio-processing-0.3.1.tar.xz
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@ -1,313 +0,0 @@
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--- a/webrtc/common_audio/wav_file.cc
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+++ b/webrtc/common_audio/wav_file.cc
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@@ -10,6 +10,7 @@
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#include "common_audio/wav_file.h"
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+#include <byteswap.h>
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#include <errno.h>
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#include <algorithm>
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@@ -34,6 +35,38 @@ bool FormatSupported(WavFormat format) {
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format == WavFormat::kWavFormatIeeeFloat;
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}
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+template <typename T>
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+void TranslateEndianness(T* destination, const T* source, size_t length) {
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+ static_assert(sizeof(T) == 2 || sizeof(T) == 4 || sizeof(T) == 8,
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+ "no converter, use integral types");
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+ if (sizeof(T) == 2) {
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+ const uint16_t* src = reinterpret_cast<const uint16_t*>(source);
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+ uint16_t* dst = reinterpret_cast<uint16_t*>(destination);
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+ for (size_t index = 0; index < length; index++) {
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+ dst[index] = bswap_16(src[index]);
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+ }
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+ }
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+ if (sizeof(T) == 4) {
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+ const uint32_t* src = reinterpret_cast<const uint32_t*>(source);
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+ uint32_t* dst = reinterpret_cast<uint32_t*>(destination);
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+ for (size_t index = 0; index < length; index++) {
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+ dst[index] = bswap_32(src[index]);
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+ }
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+ }
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+ if (sizeof(T) == 8) {
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+ const uint64_t* src = reinterpret_cast<const uint64_t*>(source);
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+ uint64_t* dst = reinterpret_cast<uint64_t*>(destination);
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+ for (size_t index = 0; index < length; index++) {
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+ dst[index] = bswap_64(src[index]);
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+ }
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+ }
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+}
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+
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+template <typename T>
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+void TranslateEndianness(T* buffer, size_t length) {
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+ TranslateEndianness(buffer, buffer, length);
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+}
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+
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// Doesn't take ownership of the file handle and won't close it.
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class WavHeaderFileReader : public WavHeaderReader {
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public:
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@@ -89,10 +122,6 @@ void WavReader::Reset() {
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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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-
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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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while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
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@@ -105,6 +134,9 @@ size_t WavReader::ReadSamples(const size_t num_samples,
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num_bytes_read = file_.Read(samples_to_convert.data(),
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chunk_size * sizeof(samples_to_convert[0]));
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num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(samples_to_convert.data(), num_samples_read);
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+#endif
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for (size_t j = 0; j < num_samples_read; ++j) {
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samples[next_chunk_start + j] = FloatToS16(samples_to_convert[j]);
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@@ -114,6 +146,10 @@ size_t WavReader::ReadSamples(const size_t num_samples,
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num_bytes_read = file_.Read(&samples[next_chunk_start],
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chunk_size * sizeof(samples[0]));
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num_samples_read = num_bytes_read / sizeof(samples[0]);
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+
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(&samples[next_chunk_start], num_samples_read);
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+#endif
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}
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RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0)
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<< "Corrupt file: file ended in the middle of a sample.";
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@@ -129,10 +165,6 @@ size_t WavReader::ReadSamples(const size_t num_samples,
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}
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size_t WavReader::ReadSamples(const size_t num_samples, float* 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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-
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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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while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
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@@ -145,6 +177,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
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num_bytes_read = file_.Read(samples_to_convert.data(),
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chunk_size * sizeof(samples_to_convert[0]));
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num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(samples_to_convert.data(), num_samples_read);
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+#endif
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for (size_t j = 0; j < num_samples_read; ++j) {
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samples[next_chunk_start + j] =
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@@ -155,6 +190,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
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num_bytes_read = file_.Read(&samples[next_chunk_start],
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chunk_size * sizeof(samples[0]));
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num_samples_read = num_bytes_read / sizeof(samples[0]);
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(&samples[next_chunk_start], num_samples_read);
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+#endif
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for (size_t j = 0; j < num_samples_read; ++j) {
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samples[next_chunk_start + j] =
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@@ -213,24 +251,32 @@ WavWriter::WavWriter(FileWrapper file,
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}
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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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-
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for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
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const size_t num_remaining_samples = num_samples - i;
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const size_t num_samples_to_write =
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std::min(kMaxChunksize, num_remaining_samples);
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if (format_ == WavFormat::kWavFormatPcm) {
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+#ifndef WEBRTC_ARCH_BIG_ENDIAN
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RTC_CHECK(
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file_.Write(&samples[i], num_samples_to_write * sizeof(samples[0])));
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+#else
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+ std::array<int16_t, kMaxChunksize> converted_samples;
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+ TranslateEndianness(converted_samples.data(), &samples[i],
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+ num_samples_to_write);
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+ RTC_CHECK(
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+ file_.Write(converted_samples.data(),
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+ num_samples_to_write * sizeof(converted_samples[0])));
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+#endif
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} else {
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RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat);
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std::array<float, kMaxChunksize> converted_samples;
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for (size_t j = 0; j < num_samples_to_write; ++j) {
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converted_samples[j] = S16ToFloat(samples[i + j]);
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}
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(converted_samples.data(), num_samples_to_write);
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+#endif
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RTC_CHECK(
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file_.Write(converted_samples.data(),
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num_samples_to_write * sizeof(converted_samples[0])));
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@@ -243,10 +289,6 @@ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
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}
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void WavWriter::WriteSamples(const float* 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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-
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for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
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const size_t num_remaining_samples = num_samples - i;
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const size_t num_samples_to_write =
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@@ -257,6 +299,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
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for (size_t j = 0; j < num_samples_to_write; ++j) {
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converted_samples[j] = FloatS16ToS16(samples[i + j]);
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}
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(converted_samples.data(), num_samples_to_write);
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+#endif
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RTC_CHECK(
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file_.Write(converted_samples.data(),
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num_samples_to_write * sizeof(converted_samples[0])));
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@@ -266,6 +311,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
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for (size_t j = 0; j < num_samples_to_write; ++j) {
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converted_samples[j] = FloatS16ToFloat(samples[i + j]);
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}
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+#ifdef WEBRTC_ARCH_BIG_ENDIAN
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+ TranslateEndianness(converted_samples.data(), num_samples_to_write);
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+#endif
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RTC_CHECK(
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file_.Write(converted_samples.data(),
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num_samples_to_write * sizeof(converted_samples[0])));
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--- a/webrtc/common_audio/wav_header.cc
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+++ b/webrtc/common_audio/wav_header.cc
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@@ -14,6 +14,8 @@
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#include "common_audio/wav_header.h"
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+#include <endian.h>
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+
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#include <cstring>
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#include <limits>
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#include <string>
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@@ -26,10 +28,6 @@
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namespace webrtc {
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namespace {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Code not working properly for big endian platforms."
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-#endif
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-
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#pragma pack(2)
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struct ChunkHeader {
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uint32_t ID;
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@@ -174,6 +172,8 @@ bool FindWaveChunk(ChunkHeader* chunk_header,
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if (readable->Read(chunk_header, sizeof(*chunk_header)) !=
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sizeof(*chunk_header))
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return false; // EOF.
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+ chunk_header->Size = le32toh(chunk_header->Size);
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+
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if (ReadFourCC(chunk_header->ID) == sought_chunk_id)
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return true; // Sought chunk found.
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// Ignore current chunk by skipping its payload.
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@@ -187,6 +187,13 @@ bool ReadFmtChunkData(FmtPcmSubchunk* fmt_subchunk, WavHeaderReader* readable) {
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if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtPcmSubchunkSize) !=
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kFmtPcmSubchunkSize)
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return false;
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+ fmt_subchunk->AudioFormat = le16toh(fmt_subchunk->AudioFormat);
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+ fmt_subchunk->NumChannels = le16toh(fmt_subchunk->NumChannels);
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+ fmt_subchunk->SampleRate = le32toh(fmt_subchunk->SampleRate);
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+ fmt_subchunk->ByteRate = le32toh(fmt_subchunk->ByteRate);
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+ fmt_subchunk->BlockAlign = le16toh(fmt_subchunk->BlockAlign);
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+ fmt_subchunk->BitsPerSample = le16toh(fmt_subchunk->BitsPerSample);
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+
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const uint32_t fmt_size = fmt_subchunk->header.Size;
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if (fmt_size != kFmtPcmSubchunkSize) {
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// There is an optional two-byte extension field permitted to be present
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@@ -214,19 +221,22 @@ void WritePcmWavHeader(size_t num_channels,
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auto header = rtc::MsanUninitialized<WavHeaderPcm>({});
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const size_t bytes_in_payload = bytes_per_sample * num_samples;
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- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F');
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- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size);
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- header.riff.Format = PackFourCC('W', 'A', 'V', 'E');
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- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' ');
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- header.fmt.header.Size = kFmtPcmSubchunkSize;
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- header.fmt.AudioFormat = MapWavFormatToHeaderField(WavFormat::kWavFormatPcm);
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- header.fmt.NumChannels = static_cast<uint16_t>(num_channels);
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- header.fmt.SampleRate = sample_rate;
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- header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample);
|
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- header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample);
|
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- header.fmt.BitsPerSample = static_cast<uint16_t>(8 * bytes_per_sample);
|
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- header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
|
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- header.data.header.Size = static_cast<uint32_t>(bytes_in_payload);
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+ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F'));
|
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+ header.riff.header.Size =
|
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+ htole32(RiffChunkSize(bytes_in_payload, *header_size));
|
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+ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E'));
|
||||
+ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' '));
|
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+ header.fmt.header.Size = htole32(kFmtPcmSubchunkSize);
|
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+ header.fmt.AudioFormat =
|
||||
+ htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatPcm));
|
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+ header.fmt.NumChannels = htole16(num_channels);
|
||||
+ header.fmt.SampleRate = htole32(sample_rate);
|
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+ header.fmt.ByteRate =
|
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+ htole32(ByteRate(num_channels, sample_rate, bytes_per_sample));
|
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+ header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample));
|
||||
+ header.fmt.BitsPerSample = htole16(8 * bytes_per_sample);
|
||||
+ header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a'));
|
||||
+ header.data.header.Size = htole32(bytes_in_payload);
|
||||
|
||||
// Do an extra copy rather than writing everything to buf directly, since buf
|
||||
// might not be correctly aligned.
|
||||
@@ -245,24 +255,26 @@ void WriteIeeeFloatWavHeader(size_t num_channels,
|
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auto header = rtc::MsanUninitialized<WavHeaderIeeeFloat>({});
|
||||
const size_t bytes_in_payload = bytes_per_sample * num_samples;
|
||||
|
||||
- header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F');
|
||||
- header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size);
|
||||
- header.riff.Format = PackFourCC('W', 'A', 'V', 'E');
|
||||
- header.fmt.header.ID = PackFourCC('f', 'm', 't', ' ');
|
||||
- header.fmt.header.Size = kFmtIeeeFloatSubchunkSize;
|
||||
+ header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F'));
|
||||
+ header.riff.header.Size =
|
||||
+ htole32(RiffChunkSize(bytes_in_payload, *header_size));
|
||||
+ header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E'));
|
||||
+ header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' '));
|
||||
+ header.fmt.header.Size = htole32(kFmtIeeeFloatSubchunkSize);
|
||||
header.fmt.AudioFormat =
|
||||
- MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat);
|
||||
- header.fmt.NumChannels = static_cast<uint16_t>(num_channels);
|
||||
- header.fmt.SampleRate = sample_rate;
|
||||
- header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample);
|
||||
- header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample);
|
||||
- header.fmt.BitsPerSample = static_cast<uint16_t>(8 * bytes_per_sample);
|
||||
- header.fmt.ExtensionSize = 0;
|
||||
- header.fact.header.ID = PackFourCC('f', 'a', 'c', 't');
|
||||
- header.fact.header.Size = 4;
|
||||
- header.fact.SampleLength = static_cast<uint32_t>(num_channels * num_samples);
|
||||
- header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
|
||||
- header.data.header.Size = static_cast<uint32_t>(bytes_in_payload);
|
||||
+ htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat));
|
||||
+ header.fmt.NumChannels = htole16(num_channels);
|
||||
+ header.fmt.SampleRate = htole32(sample_rate);
|
||||
+ header.fmt.ByteRate =
|
||||
+ htole32(ByteRate(num_channels, sample_rate, bytes_per_sample));
|
||||
+ header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample));
|
||||
+ header.fmt.BitsPerSample = htole16(8 * bytes_per_sample);
|
||||
+ header.fmt.ExtensionSize = htole16(0);
|
||||
+ header.fact.header.ID = htole32(PackFourCC('f', 'a', 'c', 't'));
|
||||
+ header.fact.header.Size = htole32(4);
|
||||
+ header.fact.SampleLength = htole32(num_channels * num_samples);
|
||||
+ header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a'));
|
||||
+ header.data.header.Size = htole32(bytes_in_payload);
|
||||
|
||||
// Do an extra copy rather than writing everything to buf directly, since buf
|
||||
// might not be correctly aligned.
|
||||
@@ -391,6 +403,7 @@ bool ReadWavHeader(WavHeaderReader* readable,
|
||||
return false;
|
||||
if (ReadFourCC(header.riff.Format) != "WAVE")
|
||||
return false;
|
||||
+ header.riff.header.Size = le32toh(header.riff.header.Size);
|
||||
|
||||
// Find "fmt " and "data" chunks. While the official Wave file specification
|
||||
// does not put requirements on the chunks order, it is uncommon to find the
|
||||
@ -1,86 +0,0 @@
|
||||
--- webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h 2023-09-05 10:19:47.000000000 -0500
|
||||
+++ webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h 2024-02-12 10:04:12.114812565 -0600
|
||||
@@ -15,8 +15,9 @@
|
||||
#define RTC_BASE_SYSTEM_ARCH_H_
|
||||
|
||||
// Processor architecture detection. For more info on what's defined, see:
|
||||
-// http://msdn.microsoft.com/en-us/library/b0084kay.aspx
|
||||
-// http://www.agner.org/optimize/calling_conventions.pdf
|
||||
+// https://docs.microsoft.com/en-us/cpp/preprocessor/predefined-macros
|
||||
+// https://www.agner.org/optimize/calling_conventions.pdf
|
||||
+// https://sourceforge.net/p/predef/wiki/Architectures/
|
||||
// or with gcc, run: "echo | gcc -E -dM -"
|
||||
#if defined(_M_X64) || defined(__x86_64__)
|
||||
#define WEBRTC_ARCH_X86_FAMILY
|
||||
@@ -27,29 +28,50 @@
|
||||
#define WEBRTC_ARCH_ARM_FAMILY
|
||||
#define WEBRTC_ARCH_64_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#elif defined(__riscv) || defined(__riscv__)
|
||||
-#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#if __riscv_xlen == 64
|
||||
-#define WEBRTC_ARCH_64_BITS
|
||||
-#else
|
||||
-#define WEBRTC_ARCH_32_BITS
|
||||
-#endif
|
||||
#elif defined(_M_IX86) || defined(__i386__)
|
||||
#define WEBRTC_ARCH_X86_FAMILY
|
||||
#define WEBRTC_ARCH_X86
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#elif defined(__ARMEL__)
|
||||
+#elif defined(_M_ARM) || defined(__ARMEL__)
|
||||
#define WEBRTC_ARCH_ARM_FAMILY
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#elif defined(__MIPSEL__)
|
||||
+#elif defined(__MIPSEL__) || defined(__MIPSEB__)
|
||||
#define WEBRTC_ARCH_MIPS_FAMILY
|
||||
#if defined(__LP64__)
|
||||
#define WEBRTC_ARCH_64_BITS
|
||||
#else
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#endif
|
||||
+#if defined(__MIPSEL__)
|
||||
+#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
+#else
|
||||
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
+#endif
|
||||
+#elif defined(__PPC__)
|
||||
+#if defined(__PPC64__)
|
||||
+#define WEBRTC_ARCH_64_BITS
|
||||
+#else
|
||||
+#define WEBRTC_ARCH_32_BITS
|
||||
+#endif
|
||||
+#if defined(__LITTLE_ENDIAN__)
|
||||
+#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
+#else
|
||||
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
+#endif
|
||||
+#elif defined(__sparc) || defined(__sparc__)
|
||||
+#if __SIZEOF_LONG__ == 8
|
||||
+#define WEBRTC_ARCH_64_BITS
|
||||
+#else
|
||||
+#define WEBRTC_ARCH_32_BITS
|
||||
+#endif
|
||||
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
+#elif defined(__riscv) && __riscv_xlen == 64
|
||||
+#define WEBRTC_ARCH_64_BITS
|
||||
+#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
+#elif defined(__riscv) && __riscv_xlen == 32
|
||||
+#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
#elif defined(__pnacl__)
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
--- webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h~ 2024-02-12 10:14:11.277835532 -0600
|
||||
+++ webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h 2024-02-12 10:25:11.558554823 -0600
|
||||
@@ -79,6 +79,9 @@
|
||||
#elif defined(__EMSCRIPTEN__)
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
+#elif defined(__s390x__)
|
||||
+#define WEBRTC_ARCH_64_BITS
|
||||
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
#else
|
||||
#error Please add support for your architecture in rtc_base/system/arch.h
|
||||
#endif
|
||||
@ -1,17 +0,0 @@
|
||||
diff -ru webrtc-audio-processing-1.3.old/subprojects/abseil-cpp.wrap webrtc-audio-processing-1.3/subprojects/abseil-cpp.wrap
|
||||
--- webrtc-audio-processing-1.3.old/subprojects/abseil-cpp.wrap 2023-09-05 17:19:47.000000000 +0200
|
||||
+++ webrtc-audio-processing-1.3/subprojects/abseil-cpp.wrap 2024-07-12 17:45:02.377170755 +0200
|
||||
@@ -1,13 +1,7 @@
|
||||
[wrap-file]
|
||||
directory = abseil-cpp-20230125.1
|
||||
-source_url = https://github.com/abseil/abseil-cpp/archive/20230125.1.tar.gz
|
||||
source_filename = abseil-cpp-20230125.1.tar.gz
|
||||
-source_hash = 81311c17599b3712069ded20cca09a62ab0bf2a89dfa16993786c8782b7ed145
|
||||
patch_filename = abseil-cpp_20230125.1-4_patch.zip
|
||||
-patch_url = https://wrapdb.mesonbuild.com/v2/abseil-cpp_20230125.1-4/get_patch
|
||||
-patch_hash = 112ee72052049d930396c2778fc1c6e184137905dd75d60a97dcfc386426610d
|
||||
-source_fallback_url = https://github.com/mesonbuild/wrapdb/releases/download/abseil-cpp_20230125.1-4/abseil-cpp-20230125.1.tar.gz
|
||||
-wrapdb_version = 20230125.1-4
|
||||
|
||||
[provide]
|
||||
absl_base = absl_base_dep
|
||||
@ -0,0 +1,90 @@
|
||||
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() {
|
||||
}
|
||||
|
||||
size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
|
||||
-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#error "Need to convert samples to big-endian when reading from WAV file"
|
||||
-#endif
|
||||
// There could be metadata after the audio; ensure we don't read it.
|
||||
num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
|
||||
num_samples_remaining_);
|
||||
@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num
|
||||
RTC_CHECK(read == num_samples || feof(file_handle_));
|
||||
RTC_CHECK_LE(read, num_samples_remaining_);
|
||||
num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
|
||||
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
+ //convert to big-endian
|
||||
+ for(size_t idx = 0; idx < num_samples; idx++) {
|
||||
+ samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
|
||||
+ }
|
||||
+#endif
|
||||
return read;
|
||||
}
|
||||
|
||||
@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
|
||||
|
||||
void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
|
||||
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
-#error "Need to convert samples to little-endian when writing to WAV file"
|
||||
-#endif
|
||||
+ int16_t * le_samples = new int16_t[num_samples];
|
||||
+ for(size_t idx = 0; idx < num_samples; idx++) {
|
||||
+ le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
|
||||
+ }
|
||||
+ const size_t written =
|
||||
+ fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_);
|
||||
+ delete []le_samples;
|
||||
+#else
|
||||
const size_t written =
|
||||
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
|
||||
+#endif
|
||||
RTC_CHECK_EQ(num_samples, written);
|
||||
num_samples_ += static_cast<uint32_t>(written);
|
||||
RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
|
||||
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) {
|
||||
@ -0,0 +1,24 @@
|
||||
diff -up webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef webrtc-audio-processing-0.2/webrtc/typedefs.h
|
||||
--- webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef 2016-05-12 09:08:53.885000410 -0500
|
||||
+++ webrtc-audio-processing-0.2/webrtc/typedefs.h 2016-05-12 09:12:38.006851953 -0500
|
||||
@@ -48,7 +48,19 @@
|
||||
#define WEBRTC_ARCH_32_BITS
|
||||
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
#else
|
||||
-#error Please add support for your architecture in typedefs.h
|
||||
+/* instead of failing, use typical unix defines... */
|
||||
+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
||||
+#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||
+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
|
||||
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||
+#else
|
||||
+#error __BYTE_ORDER__ is not defined
|
||||
+#endif
|
||||
+#if defined(__LP64__)
|
||||
+#define WEBRTC_ARCH_64_BITS
|
||||
+#else
|
||||
+#define WEBRTC_ARCH_32_BITS
|
||||
+#endif
|
||||
#endif
|
||||
|
||||
#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
|
||||
Loading…
Reference in new issue