Revert "gentoo auto-resync : 14:06:2023 - 15:13:45"

This reverts commit b9e8f3cc44aed3b6da71c7510c6287bf7bbbc66b.

1 files changed, 0 insertions, 580 deletions

diff --git a/media-libs/chromaprint/files/ffmpeg-5.patch b/media-libs/chromaprint/files/ffmpeg-5.patch
deleted file mode 100644
index a88ce7204f73..000000000000
--- a/media-libs/chromaprint/files/ffmpeg-5.patch
+++ /dev/null
@@ -1,580 +0,0 @@
-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 --
- 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/.github/workflows/build.yml b/.github/workflows/build.yml
-index 92761d9..baf67b7 100644
---- a/.github/workflows/build.yml
-+++ b/.github/workflows/build.yml
-@@ -6,7 +6,7 @@ on:
- 
- jobs:
-   test-linux:
--    runs-on: ubuntu-18.04
-+    runs-on: ubuntu-20.04
-     strategy:
-       matrix:
-         fft:
-@@ -50,7 +50,7 @@ jobs:
-         make check VERBOSE=1
- 
-   package-linux:
--    runs-on: ubuntu-18.04
-+    runs-on: ubuntu-20.04
-     strategy:
-       matrix:
-         arch:
-@@ -71,7 +71,7 @@ jobs:
-         path: artifacts/
- 
-   package-windows:
--    runs-on: ubuntu-18.04
-+    runs-on: ubuntu-20.04
-     strategy:
-       matrix:
-         arch:
-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()
-