Date: Saturday, November 5, 2022 @ 09:45:12 Author: arojas Revision: 460691
upgpkg: chromaprint 1.5.1-5: Use upstream ffmpeg 5 patch, build out of tree Added: chromaprint/trunk/8ccad693.patch Modified: chromaprint/trunk/PKGBUILD Deleted: chromaprint/trunk/ffmpeg5.patch ----------------+ 8ccad693.patch | 550 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ PKGBUILD | 21 -- ffmpeg5.patch | 76 ------- 3 files changed, 559 insertions(+), 88 deletions(-) Added: 8ccad693.patch =================================================================== --- 8ccad693.patch (rev 0) +++ 8ccad693.patch 2022-11-05 09:45:12 UTC (rev 460691) @@ -0,0 +1,550 @@ +From 8ccad6937177b1b92e40ab8f4447ea27bac009a7 Mon Sep 17 00:00:00 2001 +From: =?UTF-8?q?Luk=C3=A1=C5=A1=20Lalinsk=C3=BD?= <[email protected]> +Date: Fri, 4 Nov 2022 21:47:38 +0100 +Subject: [PATCH] Use FFmpeg 5.x (#120) + +* Use FFmpeg 5.1.2 for CI builds + +* Build on Ubuntu 20.04 + +* Upgrade code to FFmpeg 5.x APIs + +* Only set FFmpeg include dirs if building tools + +* No longer needed + +* Use ubuntu 20.04 +--- + .github/workflows/build.yml | 6 +- + CMakeLists.txt | 16 -- + package/build.sh | 4 +- + src/audio/ffmpeg_audio_processor.h | 2 - + src/audio/ffmpeg_audio_processor_avresample.h | 72 ------- + src/audio/ffmpeg_audio_processor_swresample.h | 18 +- + src/audio/ffmpeg_audio_reader.h | 197 +++++++++--------- + tests/CMakeLists.txt | 6 + + 8 files changed, 122 insertions(+), 199 deletions(-) + delete mode 100644 src/audio/ffmpeg_audio_processor_avresample.h + +diff --git a/CMakeLists.txt b/CMakeLists.txt +index f8d6a32..4da2405 100644 +--- a/CMakeLists.txt ++++ b/CMakeLists.txt +@@ -84,9 +84,6 @@ find_package(FFmpeg) + if(FFMPEG_LIBRARIES) + cmake_push_check_state(RESET) + set(CMAKE_REQUIRED_LIBRARIES ${FFMPEG_LIBRARIES} ${CMAKE_THREAD_LIBS_INIT} -lm) +- check_function_exists(av_packet_unref HAVE_AV_PACKET_UNREF) +- check_function_exists(av_frame_alloc HAVE_AV_FRAME_ALLOC) +- check_function_exists(av_frame_free HAVE_AV_FRAME_FREE) + cmake_pop_check_state() + endif() + +@@ -163,14 +160,11 @@ message(STATUS "Using ${FFT_LIB} for FFT calculations") + if(NOT AUDIO_PROCESSOR_LIB) + if(FFMPEG_LIBSWRESAMPLE_FOUND) + set(AUDIO_PROCESSOR_LIB "swresample") +- elseif(FFMPEG_LIBAVRESAMPLE_FOUND) +- set(AUDIO_PROCESSOR_LIB "avresample") + endif() + endif() + + if(AUDIO_PROCESSOR_LIB STREQUAL "swresample") + if(FFMPEG_LIBSWRESAMPLE_FOUND) +- set(USE_AVRESAMPLE OFF) + set(USE_SWRESAMPLE ON) + set(AUDIO_PROCESSOR_LIBRARIES ${FFMPEG_LIBSWRESAMPLE_LIBRARIES}) + set(AUDIO_PROCESSOR_INCLUDE_DIRS ${FFMPEG_LIBSWRESAMPLE_INCLUDE_DIRS}) +@@ -178,16 +172,6 @@ if(AUDIO_PROCESSOR_LIB STREQUAL "swresample") + message(FATAL_ERROR "Selected ${AUDIO_PROCESSOR_LIB} for audio processing, but the library is not found") + endif() + message(STATUS "Using ${AUDIO_PROCESSOR_LIB} for audio conversion") +-elseif(AUDIO_PROCESSOR_LIB STREQUAL "avresample") +- if(FFMPEG_LIBAVRESAMPLE_FOUND) +- set(USE_AVRESAMPLE ON) +- set(USE_SWRESAMPLE OFF) +- set(AUDIO_PROCESSOR_LIBRARIES ${FFMPEG_LIBAVRESAMPLE_LIBRARIES}) +- set(AUDIO_PROCESSOR_INCLUDE_DIRS ${FFMPEG_LIBAVRESAMPLE_INCLUDE_DIRS}) +- else() +- message(FATAL_ERROR "Selected ${AUDIO_PROCESSOR_LIB} for audio processing, but the library is not found") +- endif() +- message(STATUS "Using ${AUDIO_PROCESSOR_LIB} for audio conversion") + else() + message(STATUS "Building without audio conversion support, please install FFmpeg with libswresample") + endif() +diff --git a/src/audio/ffmpeg_audio_processor.h b/src/audio/ffmpeg_audio_processor.h +index 7628fc7..39f4f6d 100644 +--- a/src/audio/ffmpeg_audio_processor.h ++++ b/src/audio/ffmpeg_audio_processor.h +@@ -10,8 +10,6 @@ + + #if defined(USE_SWRESAMPLE) + #include "audio/ffmpeg_audio_processor_swresample.h" +-#elif defined(USE_AVRESAMPLE) +-#include "audio/ffmpeg_audio_processor_avresample.h" + #else + #error "no audio processing library" + #endif +diff --git a/src/audio/ffmpeg_audio_processor_avresample.h b/src/audio/ffmpeg_audio_processor_avresample.h +deleted file mode 100644 +index bd85f92..0000000 +--- a/src/audio/ffmpeg_audio_processor_avresample.h ++++ /dev/null +@@ -1,72 +0,0 @@ +-// Copyright (C) 2016 Lukas Lalinsky +-// Distributed under the MIT license, see the LICENSE file for details. +- +-#ifndef CHROMAPRINT_AUDIO_FFMPEG_AUDIO_PROCESSOR_AVRESAMPLE_H_ +-#define CHROMAPRINT_AUDIO_FFMPEG_AUDIO_PROCESSOR_AVRESAMPLE_H_ +- +-extern "C" { +-#include <libavresample/avresample.h> +-} +- +-namespace chromaprint { +- +-class FFmpegAudioProcessor { +-public: +- FFmpegAudioProcessor() { +- m_resample_ctx = avresample_alloc_context(); +- } +- +- ~FFmpegAudioProcessor() { +- avresample_free(&m_resample_ctx); +- } +- +- void SetCompatibleMode() { +- av_opt_set_int(m_resample_ctx, "filter_size", 16, 0); +- av_opt_set_int(m_resample_ctx, "phase_shift", 8, 0); +- av_opt_set_int(m_resample_ctx, "linear_interp", 1, 0); +- av_opt_set_double(m_resample_ctx, "cutoff", 0.8, 0); +- } +- +- void SetInputChannelLayout(int64_t channel_layout) { +- av_opt_set_int(m_resample_ctx, "in_channel_layout", channel_layout, 0); +- } +- +- void SetInputSampleFormat(AVSampleFormat sample_format) { +- av_opt_set_int(m_resample_ctx, "in_sample_fmt", sample_format, 0); +- } +- +- void SetInputSampleRate(int sample_rate) { +- av_opt_set_int(m_resample_ctx, "in_sample_rate", sample_rate, 0); +- } +- +- void SetOutputChannelLayout(int64_t channel_layout) { +- av_opt_set_int(m_resample_ctx, "out_channel_layout", channel_layout, 0); +- } +- +- void SetOutputSampleFormat(AVSampleFormat sample_format) { +- av_opt_set_int(m_resample_ctx, "out_sample_fmt", sample_format, 0); +- } +- +- void SetOutputSampleRate(int sample_rate) { +- av_opt_set_int(m_resample_ctx, "out_sample_fmt", sample_rate, 0); +- } +- +- int Init() { +- return avresample_open(m_resample_ctx); +- } +- +- int Convert(uint8_t **out, int out_count, const uint8_t **in, int in_count) { +- return avresample_convert(m_resample_ctx, out, 0, out_count, (uint8_t **) in, 0, in_count); +- } +- +- int Flush(uint8_t **out, int out_count) { +- return avresample_read(m_resample_ctx, out, out_count); +- } +- +-private: +- AVAudioResampleContext *m_resample_ctx = nullptr; +-}; +- +-}; // namespace chromaprint +- +-#endif +diff --git a/src/audio/ffmpeg_audio_processor_swresample.h b/src/audio/ffmpeg_audio_processor_swresample.h +index b86266b..b1d4bea 100644 +--- a/src/audio/ffmpeg_audio_processor_swresample.h ++++ b/src/audio/ffmpeg_audio_processor_swresample.h +@@ -28,30 +28,28 @@ class FFmpegAudioProcessor { + av_opt_set_double(m_swr_ctx, "cutoff", 0.8, 0); + } + +- void SetInputChannelLayout(int64_t channel_layout) { +- av_opt_set_int(m_swr_ctx, "icl", channel_layout, 0); +- av_opt_set_int(m_swr_ctx, "ich", av_get_channel_layout_nb_channels(channel_layout), 0); ++ void SetInputChannelLayout(AVChannelLayout *channel_layout) { ++ av_opt_set_int(m_swr_ctx, "in_channel_layout", channel_layout->u.mask, 0); + } + + void SetInputSampleFormat(AVSampleFormat sample_format) { +- av_opt_set_int(m_swr_ctx, "isf", sample_format, 0); ++ av_opt_set_sample_fmt(m_swr_ctx, "in_sample_fmt", sample_format, 0); + } + + void SetInputSampleRate(int sample_rate) { +- av_opt_set_int(m_swr_ctx, "isr", sample_rate, 0); ++ av_opt_set_int(m_swr_ctx, "in_sample_rate", sample_rate, 0); + } + +- void SetOutputChannelLayout(int64_t channel_layout) { +- av_opt_set_int(m_swr_ctx, "ocl", channel_layout, 0); +- av_opt_set_int(m_swr_ctx, "och", av_get_channel_layout_nb_channels(channel_layout), 0); ++ void SetOutputChannelLayout(AVChannelLayout *channel_layout) { ++ av_opt_set_int(m_swr_ctx, "out_channel_layout", channel_layout->u.mask, 0); + } + + void SetOutputSampleFormat(AVSampleFormat sample_format) { +- av_opt_set_int(m_swr_ctx, "osf", sample_format, 0); ++ av_opt_set_sample_fmt(m_swr_ctx, "out_sample_fmt", sample_format, 0); + } + + void SetOutputSampleRate(int sample_rate) { +- av_opt_set_int(m_swr_ctx, "osr", sample_rate, 0); ++ av_opt_set_int(m_swr_ctx, "out_sample_rate", sample_rate, 0); + } + + int Init() { +diff --git a/src/audio/ffmpeg_audio_reader.h b/src/audio/ffmpeg_audio_reader.h +index 5550164..1c6b346 100644 +--- a/src/audio/ffmpeg_audio_reader.h ++++ b/src/audio/ffmpeg_audio_reader.h +@@ -62,7 +62,7 @@ class FFmpegAudioReader { + bool Read(const int16_t **data, size_t *size); + + bool IsOpen() const { return m_opened; } +- bool IsFinished() const { return m_finished && !m_got_frame; } ++ bool IsFinished() const { return !m_has_more_packets && !m_has_more_frames; } + + std::string GetError() const { return m_error; } + int GetErrorCode() const { return m_error_code; } +@@ -74,20 +74,19 @@ class FFmpegAudioReader { + uint8_t *m_convert_buffer[1] = { nullptr }; + int m_convert_buffer_nb_samples = 0; + +- AVInputFormat *m_input_fmt = nullptr; ++ const AVInputFormat *m_input_fmt = nullptr; + AVDictionary *m_input_opts = nullptr; + + AVFormatContext *m_format_ctx = nullptr; + AVCodecContext *m_codec_ctx = nullptr; +- AVFrame *m_frame = nullptr; + int m_stream_index = -1; + std::string m_error; + int m_error_code = 0; +- bool m_finished = false; + bool m_opened = false; +- int m_got_frame = 0; +- AVPacket m_packet; +- AVPacket m_packet0; ++ bool m_has_more_packets = true; ++ bool m_has_more_frames = true; ++ AVPacket *m_packet = nullptr; ++ AVFrame *m_frame = nullptr; + + int m_output_sample_rate = 0; + int m_output_channels = 0; +@@ -98,19 +97,12 @@ class FFmpegAudioReader { + + inline FFmpegAudioReader::FFmpegAudioReader() { + av_log_set_level(AV_LOG_QUIET); +- +- av_init_packet(&m_packet); +- m_packet.data = nullptr; +- m_packet.size = 0; +- +- m_packet0 = m_packet; + } + + inline FFmpegAudioReader::~FFmpegAudioReader() { + Close(); + av_dict_free(&m_input_opts); + av_freep(&m_convert_buffer[0]); +- av_packet_unref(&m_packet0); + } + + inline bool FFmpegAudioReader::SetInputFormat(const char *name) { +@@ -135,11 +127,10 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) { + + Close(); + +- av_init_packet(&m_packet); +- m_packet.data = nullptr; +- m_packet.size = 0; +- +- m_packet0 = m_packet; ++ m_packet = av_packet_alloc(); ++ if (!m_packet) { ++ return false; ++ } + + ret = avformat_open_input(&m_format_ctx, file_name.c_str(), m_input_fmt, &m_input_opts); + if (ret < 0) { +@@ -153,26 +144,31 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) { + return false; + } + +- AVCodec *codec; ++ const AVCodec *codec; + ret = av_find_best_stream(m_format_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0); + if (ret < 0) { + SetError("Could not find any audio stream in the file", ret); + return false; + } + m_stream_index = ret; ++ auto stream = m_format_ctx->streams[m_stream_index]; + +- m_codec_ctx = m_format_ctx->streams[m_stream_index]->codec; ++ m_codec_ctx = avcodec_alloc_context3(codec); + m_codec_ctx->request_sample_fmt = AV_SAMPLE_FMT_S16; + ++ ret = avcodec_parameters_to_context(m_codec_ctx, stream->codecpar); ++ if (ret < 0) { ++ SetError("Could not copy the stream parameters", ret); ++ return false; ++ } ++ + ret = avcodec_open2(m_codec_ctx, codec, nullptr); + if (ret < 0) { + SetError("Could not open the codec", ret); + return false; + } + +- if (!m_codec_ctx->channel_layout) { +- m_codec_ctx->channel_layout = av_get_default_channel_layout(m_codec_ctx->channels); +- } ++ av_dump_format(m_format_ctx, 0, "foo", 0); + + m_frame = av_frame_alloc(); + if (!m_frame) { +@@ -183,19 +179,23 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) { + m_output_sample_rate = m_codec_ctx->sample_rate; + } + +- if (!m_output_channels) { +- m_output_channels = m_codec_ctx->channels; ++ AVChannelLayout output_channel_layout; ++ if (m_output_channels) { ++ av_channel_layout_default(&output_channel_layout, m_output_channels); ++ } else { ++ m_output_channels = m_codec_ctx->ch_layout.nb_channels; ++ av_channel_layout_default(&output_channel_layout, m_output_channels); + } + +- if (m_codec_ctx->sample_fmt != AV_SAMPLE_FMT_S16 || m_codec_ctx->channels != m_output_channels || m_codec_ctx->sample_rate != m_output_sample_rate) { ++ if (m_codec_ctx->sample_fmt != AV_SAMPLE_FMT_S16 || m_codec_ctx->ch_layout.nb_channels != m_output_channels || m_codec_ctx->sample_rate != m_output_sample_rate) { + m_converter.reset(new FFmpegAudioProcessor()); + m_converter->SetCompatibleMode(); + m_converter->SetInputSampleFormat(m_codec_ctx->sample_fmt); + m_converter->SetInputSampleRate(m_codec_ctx->sample_rate); +- m_converter->SetInputChannelLayout(m_codec_ctx->channel_layout); ++ m_converter->SetInputChannelLayout(&(m_codec_ctx->ch_layout)); + m_converter->SetOutputSampleFormat(AV_SAMPLE_FMT_S16); + m_converter->SetOutputSampleRate(m_output_sample_rate); +- m_converter->SetOutputChannelLayout(av_get_default_channel_layout(m_output_channels)); ++ m_converter->SetOutputChannelLayout(&output_channel_layout); + auto ret = m_converter->Init(); + if (ret != 0) { + SetError("Could not create an audio converter instance", ret); +@@ -203,10 +203,11 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) { + } + } + ++ av_channel_layout_uninit(&output_channel_layout); ++ + m_opened = true; +- m_finished = false; +- m_got_frame = 0; +- m_nb_packets = 0; ++ m_has_more_packets = true; ++ m_has_more_frames = true; + m_decode_error = 0; + + return true; +@@ -214,6 +215,7 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) { + + inline void FFmpegAudioReader::Close() { + av_frame_free(&m_frame); ++ av_packet_free(&m_packet); + + m_stream_index = -1; + +@@ -252,91 +254,98 @@ inline bool FFmpegAudioReader::Read(const int16_t **data, size_t *size) { + return false; + } + ++ *data = nullptr; ++ *size = 0; ++ + int ret; ++ bool needs_packet = false; + while (true) { +- while (m_packet.size <= 0) { +- av_packet_unref(&m_packet0); +- av_init_packet(&m_packet); +- m_packet.data = nullptr; +- m_packet.size = 0; +- ret = av_read_frame(m_format_ctx, &m_packet); ++ while (needs_packet && m_packet->size == 0) { ++ ret = av_read_frame(m_format_ctx, m_packet); + if (ret < 0) { + if (ret == AVERROR_EOF) { +- m_finished = true; ++ needs_packet = false; ++ m_has_more_packets = false; + break; +- } else { ++ } ++ SetError("Error reading from the audio source", ret); ++ return false; ++ } ++ if (m_packet->stream_index == m_stream_index) { ++ needs_packet = false; ++ } else { ++ av_packet_unref(m_packet); ++ } ++ } ++ ++ if (m_packet->size != 0) { ++ ret = avcodec_send_packet(m_codec_ctx, m_packet); ++ if (ret < 0) { ++ if (ret != AVERROR(EAGAIN)) { + SetError("Error reading from the audio source", ret); + return false; + } +- } +- m_packet0 = m_packet; +- if (m_packet.stream_index != m_stream_index) { +- m_packet.data = nullptr; +- m_packet.size = 0; + } else { +- m_nb_packets++; ++ av_packet_unref(m_packet); + } + } + +- ret = avcodec_decode_audio4(m_codec_ctx, m_frame, &m_got_frame, &m_packet); ++ ret = avcodec_receive_frame(m_codec_ctx, m_frame); + if (ret < 0) { +- if (m_decode_error) { +- SetError("Error decoding audio frame", m_decode_error); +- return false; ++ if (ret == AVERROR_EOF) { ++ m_has_more_frames = false; ++ } else if (ret == AVERROR(EAGAIN)) { ++ if (m_has_more_packets) { ++ needs_packet = true; ++ continue; ++ } else { ++ m_has_more_frames = false; ++ } + } +- m_decode_error = ret; +- m_packet.data = nullptr; +- m_packet.size = 0; +- continue; ++ SetError("Error decoding the audio source", ret); ++ return false; + } + +- break; +- } +- +- m_decode_error = 0; +- +- const int decoded = std::min(ret, m_packet.size); +- m_packet.data += decoded; +- m_packet.size -= decoded; +- +- if (m_got_frame) { +- if (m_converter) { +- if (m_frame->nb_samples > m_convert_buffer_nb_samples) { +- int linsize; +- av_freep(&m_convert_buffer[0]); +- m_convert_buffer_nb_samples = std::max(1024 * 8, m_frame->nb_samples); +- ret = av_samples_alloc(m_convert_buffer, &linsize, m_codec_ctx->channels, m_convert_buffer_nb_samples, AV_SAMPLE_FMT_S16, 1); +- if (ret < 0) { +- SetError("Couldn't allocate audio converter buffer", ret); ++ if (m_frame->nb_samples > 0) { ++ if (m_converter) { ++ if (m_frame->nb_samples > m_convert_buffer_nb_samples) { ++ int linsize; ++ av_freep(&m_convert_buffer[0]); ++ m_convert_buffer_nb_samples = std::max(1024 * 8, m_frame->nb_samples); ++ ret = av_samples_alloc(m_convert_buffer, &linsize, m_codec_ctx->ch_layout.nb_channels, m_convert_buffer_nb_samples, AV_SAMPLE_FMT_S16, 1); ++ if (ret < 0) { ++ SetError("Couldn't allocate audio converter buffer", ret); ++ return false; ++ } ++ } ++ auto nb_samples = m_converter->Convert(m_convert_buffer, m_convert_buffer_nb_samples, (const uint8_t **) m_frame->data, m_frame->nb_samples); ++ if (nb_samples < 0) { ++ SetError("Couldn't convert audio", ret); + return false; + } +- } +- auto nb_samples = m_converter->Convert(m_convert_buffer, m_convert_buffer_nb_samples, (const uint8_t **) m_frame->data, m_frame->nb_samples); +- if (nb_samples < 0) { +- SetError("Couldn't convert audio", ret); +- return false; +- } +- *data = (const int16_t *) m_convert_buffer[0]; +- *size = nb_samples; +- } else { +- *data = (const int16_t *) m_frame->data[0]; +- *size = m_frame->nb_samples; +- } +- } else { +- if (m_finished && m_converter) { +- auto nb_samples = m_converter->Flush(m_convert_buffer, m_convert_buffer_nb_samples); +- if (nb_samples < 0) { +- SetError("Couldn't convert audio", ret); +- return false; +- } else if (nb_samples > 0) { +- m_got_frame = 1; + *data = (const int16_t *) m_convert_buffer[0]; + *size = nb_samples; ++ } else { ++ *data = (const int16_t *) m_frame->data[0]; ++ *size = m_frame->nb_samples; ++ } ++ } else { ++ if (m_converter) { ++ if (IsFinished()) { ++ auto nb_samples = m_converter->Flush(m_convert_buffer, m_convert_buffer_nb_samples); ++ if (nb_samples < 0) { ++ SetError("Couldn't convert audio", ret); ++ return false; ++ } else if (nb_samples > 0) { ++ *data = (const int16_t *) m_convert_buffer[0]; ++ *size = nb_samples; ++ } ++ } + } + } +- } + +- return true; ++ return true; ++ } + } + + inline void FFmpegAudioReader::SetError(const char *message, int errnum) { +diff --git a/tests/CMakeLists.txt b/tests/CMakeLists.txt +index a2b517b..123e643 100644 +--- a/tests/CMakeLists.txt ++++ b/tests/CMakeLists.txt +@@ -38,6 +38,12 @@ set(SRCS + + if(BUILD_TOOLS) + set(SRCS ${SRCS} ../src/audio/ffmpeg_audio_reader_test.cpp) ++ include_directories( ++ ${FFMPEG_LIBAVFORMAT_INCLUDE_DIRS} ++ ${FFMPEG_LIBAVCODEC_INCLUDE_DIRS} ++ ${FFMPEG_LIBAVUTIL_INCLUDE_DIRS} ++ ${AUDIO_PROCESSOR_INCLUDE_DIRS} ++ ) + link_libraries(fpcalc_libs) + endif() + Modified: PKGBUILD =================================================================== --- PKGBUILD 2022-11-05 08:46:40 UTC (rev 460690) +++ PKGBUILD 2022-11-05 09:45:12 UTC (rev 460691) @@ -4,7 +4,7 @@ pkgname=chromaprint pkgver=1.5.1 -pkgrel=4 +pkgrel=5 pkgdesc="Library for extracting fingerprints from any audio source" url="https://acoustid.org/chromaprint"; arch=('x86_64') @@ -14,18 +14,17 @@ provides=('libchromaprint.so') # upstream signs with DSA key: https://github.com/acoustid/chromaprint/issues/81 source=(https://github.com/acoustid/${pkgname}/archive/v$pkgver/$pkgname-$pkgver.tar.gz - chromaprint-ffmpeg5.patch::https://patch-diff.githubusercontent.com/raw/acoustid/chromaprint/pull/114.patch) + 8ccad693.patch) sha512sums=('ea16e4d2b879c15b1d9b9ec93878da8b893f1834c70942663e1d2d106c2e0a661094fe2dd3bae7a6c2a1f9d5d8fab5e0b0ba493561090cf57b2228606fad1e66' - 'b487c7b61dd67a2a84d51b8ead3aee58f678f1179da33b39071093af4d6fb0bdc536f8c872b0edaeefc81d4c6c9b0f4a634ab8df623b75e03b629ebd88c897b9') + '9eb62b6b165fb2a48fe6399025a51161347cb56b8b7d28819a90be38425d145d2997aecb203dbadf40bcb469b0757ad8f836dca7dbb1db95f923a427f3c2d9ee') b2sums=('9f7f030e97d3114cf679df298d313ea826c0fb05e7e7d8a10090d0a27ed0811b380b81b29fce973e0493826c478964367396311fd0484619cb2fc4c2d8e0d4c0' - '6f386ff4690e5286df7609f2211eb4edea07508eff087622aacbee81d8ca2147378a244886fb6d0951c1dca627dc022f661f68c36d155f17a8ac69e8a9287138') + '085dd1c64347a76cf912d327d8782fafd392777aae9dd565b1e84860e12a1262d9b25f908c54b96147abf74c82824529c6e100f5943ddff725358bb7aa19da47') prepare() { - patch -d $pkgname-$pkgver -p1 < chromaprint-ffmpeg5.patch + patch -d $pkgname-$pkgver -p1 < 8ccad693.patch # Port to ffmpeg 5 } build() { - cd "${pkgname}-${pkgver}" cmake -DCMAKE_INSTALL_PREFIX=/usr \ -DCMAKE_BUILD_TYPE='None' \ -DBUILD_TESTS=ON \ @@ -33,19 +32,17 @@ -DGTEST_SOURCE_DIR=/usr/src/googletest/ \ -Wno-dev \ -B build \ - -S . + -S $pkgname-$pkgver make VERBOSE=1 -C build } check() { - cd "${pkgname}-${pkgver}" make VERBOSE=1 -C build check } package() { depends+=('libavcodec.so' 'libavformat.so' 'libavutil.so' 'libswresample.so') - cd "${pkgname}-${pkgver}" - make VERBOSE=1 DESTDIR="${pkgdir}" install -C build - install -vDm 644 LICENSE.md -t "${pkgdir}/usr/share/licenses/${pkgname}" - install -vDm 644 {NEWS.txt,README.md} -t "${pkgdir}/usr/share/doc/${pkgname}" + make VERBOSE=1 DESTDIR="$pkgdir" install -C build + install -vDm 644 $pkgname-$pkgver/LICENSE.md -t "$pkgdir/usr/share/licenses/$pkgname" + install -vDm 644 $pkgname-$pkgver/{NEWS.txt,README.md} -t "$pkgdir/usr/share/doc/$pkgname" } Deleted: ffmpeg5.patch =================================================================== --- ffmpeg5.patch 2022-11-05 08:46:40 UTC (rev 460690) +++ ffmpeg5.patch 2022-11-05 09:45:12 UTC (rev 460691) @@ -1,76 +0,0 @@ -From 6d938d70b1d52634f8b0d88cb29da87f8d5b35a2 Mon Sep 17 00:00:00 2001 -From: =?UTF-8?q?Bernhard=20Rosenkr=C3=A4nzer?= <[email protected]> -Date: Mon, 17 Jan 2022 04:41:33 +0100 -Subject: [PATCH] Port to ffmpeg 5.0 -MIME-Version: 1.0 -Content-Type: text/plain; charset=UTF-8 -Content-Transfer-Encoding: 8bit - -Replace removed functionality like accessing the codec context -from an AVStream and avcodec_decode_audio4() - -Signed-off-by: Bernhard Rosenkränzer <[email protected]> ---- - src/audio/ffmpeg_audio_reader.h | 24 ++++++++++++++++++++++-- - 1 file changed, 22 insertions(+), 2 deletions(-) - -diff --git a/src/audio/ffmpeg_audio_reader.h b/src/audio/ffmpeg_audio_reader.h -index 5550164..a3b8de7 100644 ---- a/src/audio/ffmpeg_audio_reader.h -+++ b/src/audio/ffmpeg_audio_reader.h -@@ -74,7 +74,7 @@ class FFmpegAudioReader { - uint8_t *m_convert_buffer[1] = { nullptr }; - int m_convert_buffer_nb_samples = 0; - -- AVInputFormat *m_input_fmt = nullptr; -+ const AVInputFormat *m_input_fmt = nullptr; - AVDictionary *m_input_opts = nullptr; - - AVFormatContext *m_format_ctx = nullptr; -@@ -153,7 +153,7 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) { - return false; - } - -- AVCodec *codec; -+ const AVCodec *codec; - ret = av_find_best_stream(m_format_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0); - if (ret < 0) { - SetError("Could not find any audio stream in the file", ret); -@@ -161,7 +161,13 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) { - } - m_stream_index = ret; - -+#if LIBAVCODEC_VERSION_MAJOR >= 59 -+ const AVCodec *streamcodec = avcodec_find_decoder(m_format_ctx->streams[m_stream_index]->codecpar->codec_id); -+ m_codec_ctx = avcodec_alloc_context3(streamcodec); -+ avcodec_parameters_to_context(m_codec_ctx, m_format_ctx->streams[m_stream_index]->codecpar); -+#else - m_codec_ctx = m_format_ctx->streams[m_stream_index]->codec; -+#endif - m_codec_ctx->request_sample_fmt = AV_SAMPLE_FMT_S16; - - ret = avcodec_open2(m_codec_ctx, codec, nullptr); -@@ -278,7 +284,23 @@ inline bool FFmpegAudioReader::Read(const int16_t **data, size_t *size) { - } - } - -+#if LIBAVCODEC_VERSION_MAJOR < 59 - ret = avcodec_decode_audio4(m_codec_ctx, m_frame, &m_got_frame, &m_packet); -+#else -+ m_got_frame = 0; -+ ret = avcodec_send_packet(m_codec_ctx, &m_packet); -+ if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) -+ ret = 0; -+ if(ret >= 0) { -+ ret = avcodec_receive_frame(m_codec_ctx, m_frame); -+ if (ret == 0) { -+ m_got_frame = 1; -+ ret = m_packet.size; -+ } -+ } -+ if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) -+ ret = 0; -+#endif - if (ret < 0) { - if (m_decode_error) { - SetError("Error decoding audio frame", m_decode_error);
