chromaprint: revbump for ffmpeg6
This commit is contained in:
parent
d6c20c3b3a
commit
2e42fac6b1
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@ -0,0 +1,569 @@
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From 8ccad6937177b1b92e40ab8f4447ea27bac009a7 Mon Sep 17 00:00:00 2001
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From: =?UTF-8?q?Luk=C3=A1=C5=A1=20Lalinsk=C3=BD?= <lalinsky@gmail.com>
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Date: Fri, 4 Nov 2022 21:47:38 +0100
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Subject: [PATCH] Use FFmpeg 5.x (#120)
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* Use FFmpeg 5.1.2 for CI builds
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* Build on Ubuntu 20.04
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* Upgrade code to FFmpeg 5.x APIs
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* Only set FFmpeg include dirs if building tools
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* No longer needed
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* Use ubuntu 20.04
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---
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.github/workflows/build.yml | 6 +-
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CMakeLists.txt | 16 --
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package/build.sh | 4 +-
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src/audio/ffmpeg_audio_processor.h | 2 -
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src/audio/ffmpeg_audio_processor_avresample.h | 72 -------
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src/audio/ffmpeg_audio_processor_swresample.h | 18 +-
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src/audio/ffmpeg_audio_reader.h | 197 +++++++++---------
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tests/CMakeLists.txt | 6 +
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8 files changed, 122 insertions(+), 199 deletions(-)
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delete mode 100644 src/audio/ffmpeg_audio_processor_avresample.h
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diff --git a/CMakeLists.txt b/CMakeLists.txt
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index f8d6a32..4da2405 100644
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--- a/CMakeLists.txt
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+++ b/CMakeLists.txt
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@@ -84,9 +84,6 @@ find_package(FFmpeg)
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if(FFMPEG_LIBRARIES)
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cmake_push_check_state(RESET)
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set(CMAKE_REQUIRED_LIBRARIES ${FFMPEG_LIBRARIES} ${CMAKE_THREAD_LIBS_INIT} -lm)
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- check_function_exists(av_packet_unref HAVE_AV_PACKET_UNREF)
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- check_function_exists(av_frame_alloc HAVE_AV_FRAME_ALLOC)
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- check_function_exists(av_frame_free HAVE_AV_FRAME_FREE)
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cmake_pop_check_state()
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endif()
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@@ -163,14 +160,11 @@ message(STATUS "Using ${FFT_LIB} for FFT calculations")
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if(NOT AUDIO_PROCESSOR_LIB)
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if(FFMPEG_LIBSWRESAMPLE_FOUND)
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set(AUDIO_PROCESSOR_LIB "swresample")
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- elseif(FFMPEG_LIBAVRESAMPLE_FOUND)
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- set(AUDIO_PROCESSOR_LIB "avresample")
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endif()
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endif()
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if(AUDIO_PROCESSOR_LIB STREQUAL "swresample")
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if(FFMPEG_LIBSWRESAMPLE_FOUND)
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- set(USE_AVRESAMPLE OFF)
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set(USE_SWRESAMPLE ON)
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set(AUDIO_PROCESSOR_LIBRARIES ${FFMPEG_LIBSWRESAMPLE_LIBRARIES})
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set(AUDIO_PROCESSOR_INCLUDE_DIRS ${FFMPEG_LIBSWRESAMPLE_INCLUDE_DIRS})
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@@ -178,16 +172,6 @@ if(AUDIO_PROCESSOR_LIB STREQUAL "swresample")
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message(FATAL_ERROR "Selected ${AUDIO_PROCESSOR_LIB} for audio processing, but the library is not found")
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endif()
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message(STATUS "Using ${AUDIO_PROCESSOR_LIB} for audio conversion")
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-elseif(AUDIO_PROCESSOR_LIB STREQUAL "avresample")
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- if(FFMPEG_LIBAVRESAMPLE_FOUND)
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- set(USE_AVRESAMPLE ON)
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- set(USE_SWRESAMPLE OFF)
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- set(AUDIO_PROCESSOR_LIBRARIES ${FFMPEG_LIBAVRESAMPLE_LIBRARIES})
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- set(AUDIO_PROCESSOR_INCLUDE_DIRS ${FFMPEG_LIBAVRESAMPLE_INCLUDE_DIRS})
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- else()
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- message(FATAL_ERROR "Selected ${AUDIO_PROCESSOR_LIB} for audio processing, but the library is not found")
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- endif()
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- message(STATUS "Using ${AUDIO_PROCESSOR_LIB} for audio conversion")
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else()
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message(STATUS "Building without audio conversion support, please install FFmpeg with libswresample")
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endif()
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diff --git a/package/build.sh b/package/build.sh
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index da631ae..b41d36e 100755
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--- a/package/build.sh
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+++ b/package/build.sh
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@@ -7,8 +7,8 @@ set -eux
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BASE_DIR=$(cd $(dirname $0)/.. && pwd)
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-FFMPEG_VERSION=4.4.1
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-FFMPEG_BUILD_TAG=v4.4.1-1
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+FFMPEG_VERSION=5.1.2
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+FFMPEG_BUILD_TAG=v${FFMPEG_VERSION}-1
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TMP_BUILD_DIR=$BASE_DIR/$(mktemp -d build.XXXXXXXX)
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trap 'rm -rf $TMP_BUILD_DIR' EXIT
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diff --git a/src/audio/ffmpeg_audio_processor.h b/src/audio/ffmpeg_audio_processor.h
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index 7628fc7..39f4f6d 100644
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--- a/src/audio/ffmpeg_audio_processor.h
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+++ b/src/audio/ffmpeg_audio_processor.h
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@@ -10,8 +10,6 @@
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#if defined(USE_SWRESAMPLE)
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#include "audio/ffmpeg_audio_processor_swresample.h"
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-#elif defined(USE_AVRESAMPLE)
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-#include "audio/ffmpeg_audio_processor_avresample.h"
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#else
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#error "no audio processing library"
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#endif
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diff --git a/src/audio/ffmpeg_audio_processor_avresample.h b/src/audio/ffmpeg_audio_processor_avresample.h
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deleted file mode 100644
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index bd85f92..0000000
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--- a/src/audio/ffmpeg_audio_processor_avresample.h
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+++ /dev/null
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@@ -1,72 +0,0 @@
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-// Copyright (C) 2016 Lukas Lalinsky
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-// Distributed under the MIT license, see the LICENSE file for details.
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-
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-#ifndef CHROMAPRINT_AUDIO_FFMPEG_AUDIO_PROCESSOR_AVRESAMPLE_H_
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-#define CHROMAPRINT_AUDIO_FFMPEG_AUDIO_PROCESSOR_AVRESAMPLE_H_
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-
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-extern "C" {
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-#include <libavresample/avresample.h>
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-}
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-
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-namespace chromaprint {
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-
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-class FFmpegAudioProcessor {
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-public:
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- FFmpegAudioProcessor() {
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- m_resample_ctx = avresample_alloc_context();
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- }
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-
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- ~FFmpegAudioProcessor() {
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- avresample_free(&m_resample_ctx);
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- }
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-
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- void SetCompatibleMode() {
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- av_opt_set_int(m_resample_ctx, "filter_size", 16, 0);
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- av_opt_set_int(m_resample_ctx, "phase_shift", 8, 0);
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- av_opt_set_int(m_resample_ctx, "linear_interp", 1, 0);
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- av_opt_set_double(m_resample_ctx, "cutoff", 0.8, 0);
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- }
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-
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- void SetInputChannelLayout(int64_t channel_layout) {
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- av_opt_set_int(m_resample_ctx, "in_channel_layout", channel_layout, 0);
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- }
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-
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- void SetInputSampleFormat(AVSampleFormat sample_format) {
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- av_opt_set_int(m_resample_ctx, "in_sample_fmt", sample_format, 0);
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- }
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-
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- void SetInputSampleRate(int sample_rate) {
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- av_opt_set_int(m_resample_ctx, "in_sample_rate", sample_rate, 0);
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- }
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-
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- void SetOutputChannelLayout(int64_t channel_layout) {
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- av_opt_set_int(m_resample_ctx, "out_channel_layout", channel_layout, 0);
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- }
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-
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- void SetOutputSampleFormat(AVSampleFormat sample_format) {
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- av_opt_set_int(m_resample_ctx, "out_sample_fmt", sample_format, 0);
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- }
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-
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- void SetOutputSampleRate(int sample_rate) {
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- av_opt_set_int(m_resample_ctx, "out_sample_fmt", sample_rate, 0);
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- }
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-
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- int Init() {
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- return avresample_open(m_resample_ctx);
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- }
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-
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- int Convert(uint8_t **out, int out_count, const uint8_t **in, int in_count) {
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- return avresample_convert(m_resample_ctx, out, 0, out_count, (uint8_t **) in, 0, in_count);
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- }
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-
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- int Flush(uint8_t **out, int out_count) {
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- return avresample_read(m_resample_ctx, out, out_count);
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- }
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-
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-private:
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- AVAudioResampleContext *m_resample_ctx = nullptr;
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-};
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-
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-}; // namespace chromaprint
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-
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-#endif
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diff --git a/src/audio/ffmpeg_audio_processor_swresample.h b/src/audio/ffmpeg_audio_processor_swresample.h
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index b86266b..b1d4bea 100644
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--- a/src/audio/ffmpeg_audio_processor_swresample.h
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+++ b/src/audio/ffmpeg_audio_processor_swresample.h
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@@ -28,30 +28,28 @@ public:
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av_opt_set_double(m_swr_ctx, "cutoff", 0.8, 0);
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}
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- void SetInputChannelLayout(int64_t channel_layout) {
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- av_opt_set_int(m_swr_ctx, "icl", channel_layout, 0);
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- av_opt_set_int(m_swr_ctx, "ich", av_get_channel_layout_nb_channels(channel_layout), 0);
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+ void SetInputChannelLayout(AVChannelLayout *channel_layout) {
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+ av_opt_set_int(m_swr_ctx, "in_channel_layout", channel_layout->u.mask, 0);
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}
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void SetInputSampleFormat(AVSampleFormat sample_format) {
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- av_opt_set_int(m_swr_ctx, "isf", sample_format, 0);
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+ av_opt_set_sample_fmt(m_swr_ctx, "in_sample_fmt", sample_format, 0);
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}
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void SetInputSampleRate(int sample_rate) {
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- av_opt_set_int(m_swr_ctx, "isr", sample_rate, 0);
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+ av_opt_set_int(m_swr_ctx, "in_sample_rate", sample_rate, 0);
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}
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- void SetOutputChannelLayout(int64_t channel_layout) {
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- av_opt_set_int(m_swr_ctx, "ocl", channel_layout, 0);
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- av_opt_set_int(m_swr_ctx, "och", av_get_channel_layout_nb_channels(channel_layout), 0);
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+ void SetOutputChannelLayout(AVChannelLayout *channel_layout) {
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+ av_opt_set_int(m_swr_ctx, "out_channel_layout", channel_layout->u.mask, 0);
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}
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void SetOutputSampleFormat(AVSampleFormat sample_format) {
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- av_opt_set_int(m_swr_ctx, "osf", sample_format, 0);
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+ av_opt_set_sample_fmt(m_swr_ctx, "out_sample_fmt", sample_format, 0);
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}
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void SetOutputSampleRate(int sample_rate) {
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- av_opt_set_int(m_swr_ctx, "osr", sample_rate, 0);
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+ av_opt_set_int(m_swr_ctx, "out_sample_rate", sample_rate, 0);
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}
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int Init() {
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diff --git a/src/audio/ffmpeg_audio_reader.h b/src/audio/ffmpeg_audio_reader.h
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index 5550164..1c6b346 100644
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--- a/src/audio/ffmpeg_audio_reader.h
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+++ b/src/audio/ffmpeg_audio_reader.h
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@@ -62,7 +62,7 @@ public:
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bool Read(const int16_t **data, size_t *size);
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bool IsOpen() const { return m_opened; }
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- bool IsFinished() const { return m_finished && !m_got_frame; }
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+ bool IsFinished() const { return !m_has_more_packets && !m_has_more_frames; }
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std::string GetError() const { return m_error; }
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int GetErrorCode() const { return m_error_code; }
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@@ -74,20 +74,19 @@ private:
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uint8_t *m_convert_buffer[1] = { nullptr };
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int m_convert_buffer_nb_samples = 0;
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- AVInputFormat *m_input_fmt = nullptr;
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+ const AVInputFormat *m_input_fmt = nullptr;
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AVDictionary *m_input_opts = nullptr;
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AVFormatContext *m_format_ctx = nullptr;
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AVCodecContext *m_codec_ctx = nullptr;
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- AVFrame *m_frame = nullptr;
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int m_stream_index = -1;
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std::string m_error;
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int m_error_code = 0;
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- bool m_finished = false;
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bool m_opened = false;
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- int m_got_frame = 0;
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- AVPacket m_packet;
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- AVPacket m_packet0;
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+ bool m_has_more_packets = true;
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+ bool m_has_more_frames = true;
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+ AVPacket *m_packet = nullptr;
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+ AVFrame *m_frame = nullptr;
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int m_output_sample_rate = 0;
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int m_output_channels = 0;
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@@ -98,19 +97,12 @@ private:
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inline FFmpegAudioReader::FFmpegAudioReader() {
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av_log_set_level(AV_LOG_QUIET);
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-
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- av_init_packet(&m_packet);
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- m_packet.data = nullptr;
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- m_packet.size = 0;
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-
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- m_packet0 = m_packet;
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}
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inline FFmpegAudioReader::~FFmpegAudioReader() {
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Close();
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av_dict_free(&m_input_opts);
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av_freep(&m_convert_buffer[0]);
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- av_packet_unref(&m_packet0);
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}
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inline bool FFmpegAudioReader::SetInputFormat(const char *name) {
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@@ -135,11 +127,10 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) {
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Close();
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- av_init_packet(&m_packet);
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- m_packet.data = nullptr;
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- m_packet.size = 0;
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-
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- m_packet0 = m_packet;
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+ m_packet = av_packet_alloc();
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+ if (!m_packet) {
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+ return false;
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+ }
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ret = avformat_open_input(&m_format_ctx, file_name.c_str(), m_input_fmt, &m_input_opts);
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if (ret < 0) {
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@@ -153,26 +144,31 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) {
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return false;
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}
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- AVCodec *codec;
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+ const AVCodec *codec;
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ret = av_find_best_stream(m_format_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
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if (ret < 0) {
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SetError("Could not find any audio stream in the file", ret);
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return false;
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}
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m_stream_index = ret;
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+ auto stream = m_format_ctx->streams[m_stream_index];
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- m_codec_ctx = m_format_ctx->streams[m_stream_index]->codec;
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+ m_codec_ctx = avcodec_alloc_context3(codec);
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m_codec_ctx->request_sample_fmt = AV_SAMPLE_FMT_S16;
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+ ret = avcodec_parameters_to_context(m_codec_ctx, stream->codecpar);
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+ if (ret < 0) {
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+ SetError("Could not copy the stream parameters", ret);
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+ return false;
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+ }
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+
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ret = avcodec_open2(m_codec_ctx, codec, nullptr);
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if (ret < 0) {
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SetError("Could not open the codec", ret);
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return false;
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}
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- if (!m_codec_ctx->channel_layout) {
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- m_codec_ctx->channel_layout = av_get_default_channel_layout(m_codec_ctx->channels);
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- }
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+ av_dump_format(m_format_ctx, 0, "foo", 0);
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m_frame = av_frame_alloc();
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if (!m_frame) {
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@@ -183,19 +179,23 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) {
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m_output_sample_rate = m_codec_ctx->sample_rate;
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}
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- if (!m_output_channels) {
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- m_output_channels = m_codec_ctx->channels;
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+ AVChannelLayout output_channel_layout;
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+ if (m_output_channels) {
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+ av_channel_layout_default(&output_channel_layout, m_output_channels);
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+ } else {
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+ m_output_channels = m_codec_ctx->ch_layout.nb_channels;
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+ av_channel_layout_default(&output_channel_layout, m_output_channels);
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}
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- 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) {
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+ 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) {
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m_converter.reset(new FFmpegAudioProcessor());
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m_converter->SetCompatibleMode();
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m_converter->SetInputSampleFormat(m_codec_ctx->sample_fmt);
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m_converter->SetInputSampleRate(m_codec_ctx->sample_rate);
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- m_converter->SetInputChannelLayout(m_codec_ctx->channel_layout);
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+ m_converter->SetInputChannelLayout(&(m_codec_ctx->ch_layout));
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m_converter->SetOutputSampleFormat(AV_SAMPLE_FMT_S16);
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m_converter->SetOutputSampleRate(m_output_sample_rate);
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- m_converter->SetOutputChannelLayout(av_get_default_channel_layout(m_output_channels));
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+ m_converter->SetOutputChannelLayout(&output_channel_layout);
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auto ret = m_converter->Init();
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if (ret != 0) {
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SetError("Could not create an audio converter instance", ret);
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@@ -203,10 +203,11 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) {
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}
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}
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+ av_channel_layout_uninit(&output_channel_layout);
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+
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m_opened = true;
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- m_finished = false;
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- m_got_frame = 0;
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- m_nb_packets = 0;
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+ m_has_more_packets = true;
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+ m_has_more_frames = true;
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m_decode_error = 0;
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return true;
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@@ -214,6 +215,7 @@ inline bool FFmpegAudioReader::Open(const std::string &file_name) {
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inline void FFmpegAudioReader::Close() {
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av_frame_free(&m_frame);
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+ av_packet_free(&m_packet);
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m_stream_index = -1;
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@@ -252,91 +254,98 @@ inline bool FFmpegAudioReader::Read(const int16_t **data, size_t *size) {
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return false;
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}
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||||
|
||||
+ *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()
|
||||
|
||||
--
|
||||
2.38.1
|
||||
|
||||
|
|
@ -1,11 +1,11 @@
|
|||
# Template file for 'chromaprint'
|
||||
pkgname=chromaprint
|
||||
version=1.5.1
|
||||
revision=1
|
||||
revision=2
|
||||
build_style=cmake
|
||||
configure_args="-DBUILD_TOOLS=ON"
|
||||
make_check_target="check"
|
||||
makedepends="fftw-devel ffmpeg-devel"
|
||||
makedepends="fftw-devel ffmpeg6-devel"
|
||||
short_desc="Library that extracts fingerprints from any audio source"
|
||||
maintainer="Orphaned <orphan@voidlinux.org>"
|
||||
license="MIT, LGPL-2.1-only"
|
||||
|
|
Loading…
Reference in New Issue