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Backport oficial ffmpeg 5 support

This commit is contained in:
Sérgio M. Basto 2022-11-22 22:47:38 +00:00
parent 946076e68f
commit e161501f1a
3 changed files with 570 additions and 77 deletions
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565
8ccad6937177b1b92e40ab8f4447ea27bac009a7.patch Normal file
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@ -0,0 +1,565 @@
From 8ccad6937177b1b92e40ab8f4447ea27bac009a7 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Luk=C3=A1=C5=A1=20Lalinsk=C3=BD?= <lalinsky@gmail.com>
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 ${AUDIO_PROCESSOR_LIB} for audio conversion")
else()
message(STATUS "Building without audio conversion support, please install FFmpeg with libswresample")
endif()
diff --git a/package/build.sh b/package/build.sh
index da631ae..b41d36e 100755
--- a/package/build.sh
+++ b/package/build.sh
@@ -7,8 +7,8 @@ set -eux
BASE_DIR=$(cd $(dirname $0)/.. && pwd)
-FFMPEG_VERSION=4.4.1
-FFMPEG_BUILD_TAG=v4.4.1-1
+FFMPEG_VERSION=5.1.2
+FFMPEG_BUILD_TAG=v${FFMPEG_VERSION}-1
TMP_BUILD_DIR=$BASE_DIR/$(mktemp -d build.XXXXXXXX)
trap 'rm -rf $TMP_BUILD_DIR' EXIT
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()

74
chromaprint-PR108-Port-to-ffmpeg-5.patch
View File

@ -1,74 +0,0 @@
From 6d938d70b1d52634f8b0d88cb29da87f8d5b35a2 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Bernhard=20Rosenkr=C3=A4nzer?= <bero@lindev.ch>
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 <bero@lindev.ch>
---
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 +284,21 @@ 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
+ ret = avcodec_receive_frame(m_codec_ctx, m_frame);
+ if (ret == 0)
+ m_got_frame = true;
+ if(ret == AVERROR(EAGAIN))
+ ret = 0;
+ if (ret == 0)
+ ret = avcodec_send_packet(m_codec_ctx, &m_packet);
+ if (ret == AVERROR(EAGAIN))
+ ret = 0;
+ if (ret >= 0)
+ ret = m_packet.size;
+#endif
if (ret < 0) {
if (m_decode_error) {
SetError("Error decoding audio frame", m_decode_error);

8
chromaprint.spec
View File

@ -6,15 +6,14 @@
Name: chromaprint
Version: 1.5.1
Release: 5%{?dist}
Release: 6%{?dist}
Summary: Library implementing the AcoustID fingerprinting
License: GPLv2+
URL: http://www.acoustid.org/chromaprint
Source: https://github.com/acoustid/%{name}/releases/download/v%{version}/%{name}-%{version}.tar.gz
# From: https://github.com/acoustid/chromaprint/pull/108
Patch: chromaprint-PR108-Port-to-ffmpeg-5.patch
Patch1: https://github.com/acoustid/chromaprint/commit/8ccad6937177b1b92e40ab8f4447ea27bac009a7.patch
BuildRequires: gcc
BuildRequires: gcc-c++
@ -107,6 +106,9 @@ rm -f %{buildroot}%{_libdir}/lib*.la
%endif
%changelog
* Tue Nov 22 2022 Sérgio Basto <sergio@serjux.com> - 1.5.1-6
- Backport oficial ffmpeg 5 support
* Mon Aug 29 2022 Neal Gompa <ngompa@fedoraproject.org> - 1.5.1-5
- Rebuild for ffmpeg 5.1 (#2121070)