[packages/chromaprint] - upstream fixes for ffmpeg 5+, rel 2

baggins baggins at pld-linux.org
Sun Oct 15 18:32:00 CEST 2023


commit 64f232819541d947bfff9953a3cdfee08142a488
Author: Jan Rękorajski <baggins at pld-linux.org>
Date:   Sun Oct 15 18:31:33 2023 +0200

    - upstream fixes for ffmpeg 5+, rel 2

 chromaprint.spec           |   6 +-
 ffmpeg5-decode-retry.patch |  50 ++++
 ffmpeg5.patch              | 596 +++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 651 insertions(+), 1 deletion(-)
---
diff --git a/chromaprint.spec b/chromaprint.spec
index ca7d430..5015e31 100644
--- a/chromaprint.spec
+++ b/chromaprint.spec
@@ -10,12 +10,14 @@ Summary:	Library implementing the AcoustID fingerprinting
 Summary(pl.UTF-8):	Biblioteka implementująca odciski AcoustID
 Name:		chromaprint
 Version:	1.5.1
-Release:	1
+Release:	2
 License:	LGPL v2.1+
 Group:		Libraries
 #Source0Download: https://github.com/acoustid/chromaprint/releases
 Source0:	https://github.com/acoustid/chromaprint/releases/download/v%{version}/%{name}-%{version}.tar.gz
 # Source0-md5:	54e71f86bcf1d34989db639044ba9628
+Patch0:		ffmpeg5.patch
+Patch1:		ffmpeg5-decode-retry.patch
 URL:		https://acoustid.org/chromaprint
 BuildRequires:	cmake >= 3.3
 %{?with_ffmpeg:BuildRequires:	ffmpeg-devel >= 0.6}
@@ -82,6 +84,8 @@ tworzenia aplikacji wykorzystujących bibliotekę libchromaprint.
 
 %prep
 %setup -q
+%patch0 -p1
+%patch1 -p1
 
 %build
 install -d build
diff --git a/ffmpeg5-decode-retry.patch b/ffmpeg5-decode-retry.patch
new file mode 100644
index 0000000..ecfa7d0
--- /dev/null
+++ b/ffmpeg5-decode-retry.patch
@@ -0,0 +1,50 @@
+From 82781d02cd3063d071a501218297a90bde9a314f Mon Sep 17 00:00:00 2001
+From: Marshal Walker <CatmanIX at gmail.com>
+Date: Thu, 8 Dec 2022 11:53:58 -0500
+Subject: [PATCH] ffmpeg5 fix for issue #122
+
+tested on Arch Linux, needs testing on win/mac/etc (should be fine tho)
+---
+ src/audio/ffmpeg_audio_processor_swresample.h | 4 ++--
+ src/audio/ffmpeg_audio_reader.h               | 5 +++--
+ 2 files changed, 5 insertions(+), 4 deletions(-)
+
+diff --git a/src/audio/ffmpeg_audio_processor_swresample.h b/src/audio/ffmpeg_audio_processor_swresample.h
+index b1d4bea..e8fcb3f 100644
+--- a/src/audio/ffmpeg_audio_processor_swresample.h
++++ b/src/audio/ffmpeg_audio_processor_swresample.h
+@@ -29,7 +29,7 @@ class FFmpegAudioProcessor {
+ 	}
+ 
+ 	void SetInputChannelLayout(AVChannelLayout *channel_layout) {
+-		av_opt_set_int(m_swr_ctx, "in_channel_layout", channel_layout->u.mask, 0);
++		av_opt_set_chlayout(m_swr_ctx, "in_chlayout", channel_layout, 0);
+ 	}
+ 
+ 	void SetInputSampleFormat(AVSampleFormat sample_format) {
+@@ -41,7 +41,7 @@ class FFmpegAudioProcessor {
+ 	}
+ 
+ 	void SetOutputChannelLayout(AVChannelLayout *channel_layout) {
+-		av_opt_set_int(m_swr_ctx, "out_channel_layout", channel_layout->u.mask, 0);
++		av_opt_set_chlayout(m_swr_ctx, "out_chlayout", channel_layout, 0);
+ 	}
+ 
+ 	void SetOutputSampleFormat(AVSampleFormat sample_format) {
+diff --git a/src/audio/ffmpeg_audio_reader.h b/src/audio/ffmpeg_audio_reader.h
+index 1c6b346..35b2934 100644
+--- a/src/audio/ffmpeg_audio_reader.h
++++ b/src/audio/ffmpeg_audio_reader.h
+@@ -301,9 +301,10 @@ inline bool FFmpegAudioReader::Read(const int16_t **data, size_t *size) {
+ 				} else {
+ 					m_has_more_frames = false;
+ 				}
++			} else {
++				SetError("Error decoding the audio source", ret);
++				return false;
+ 			}
+-			SetError("Error decoding the audio source", ret);
+-			return false;
+ 		}
+ 
+ 		if (m_frame->nb_samples > 0) {
diff --git a/ffmpeg5.patch b/ffmpeg5.patch
new file mode 100644
index 0000000..9d12c43
--- /dev/null
+++ b/ffmpeg5.patch
@@ -0,0 +1,596 @@
+From 8ccad6937177b1b92e40ab8f4447ea27bac009a7 Mon Sep 17 00:00:00 2001
+From: =?UTF-8?q?Luk=C3=A1=C5=A1=20Lalinsk=C3=BD?= <lalinsky at 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/.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 ${AUDIO_PROCESSOR_LIB} for audio conversion")
+ else()
+ 	message(STATUS "Building without audio conversion support, please install FFmpeg with libswresample")
+ endif()
+diff --git a/package/build.sh b/package/build.sh
+index da631ae..b41d36e 100755
+--- a/package/build.sh
++++ b/package/build.sh
+@@ -7,8 +7,8 @@ set -eux
+ 
+ BASE_DIR=$(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()
+ 
================================================================

---- gitweb:

http://git.pld-linux.org/gitweb.cgi/packages/chromaprint.git/commitdiff/64f232819541d947bfff9953a3cdfee08142a488



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