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FFmpeg jni cpp (WIP)

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This commit is contained in:
shabinder 2021-09-02 11:12:30 +05:30
parent 40e9b0a80c
commit df6e969a56
19 changed files with 1079 additions and 84 deletions
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1
.gitignore vendored
View File

@ -13,3 +13,4 @@ Gemfile
Gemfile.lock
/maintenance-tasks/build/
/android/.cxx/Debug/5k2s1t1p/x86/
/ffmpeg/ffmpeg-kit-android-lib/.cxx/Debug/

4
.gitmodules vendored
View File

@ -1,6 +1,6 @@
[submodule "spotiflyer-ios"]
path = spotiflyer-ios
url = https://github.com/Shabinder/spotiflyer-ios
[submodule "ffmpeg-android-maker"]
path = ffmpeg-android-maker
[submodule "ffmpeg/ffmpeg-android-maker"]
path = ffmpeg/ffmpeg-android-maker
url = https://github.com/Shabinder/ffmpeg-android-maker

24
android/build.gradle.kts
View File

@ -57,20 +57,6 @@ android {
targetSdk = Versions.targetSdkVersion
versionCode = Versions.versionCode
versionName = Versions.versionName
ndk {
abiFilters.addAll(setOf("x86", "x86_64", "armeabi-v7a", "arm64-v8a"))
}
}
sourceSets {
named("main") {
jniLibs.srcDir("../ffmpeg-android-maker/output/lib")
}
}
externalNativeBuild {
cmake {
path("CMakeLists.txt")
}
}
buildTypes {
getByName("release") {
@ -103,16 +89,6 @@ android {
exclude(group = "androidx.compose.ui")
}
}
packagingOptions {
resources {
excludes.apply {
add("META-INF/*")
}
jniLibs.pickFirsts.apply {
add("**/*.so")
}
}
}
}
dependencies {
implementation(compose.material)

18
android/src/main/cpp/main.cpp
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@ -1,18 +0,0 @@
#include <stdio.h>
#include <stdlib.h>
#include <jni.h>
#include <string.h>
#include <android/log.h>
extern "C" {
#include <libavformat/avformat.h>
// #include <libavcodec/avcodec.h>
JNIEXPORT jint
JNICALL Java_com_shabinder_spotiflyer_ffmpeg_FFmpeg_testInit(JNIEnv *env, jclass c) {
__android_log_print(ANDROID_LOG_DEBUG, "FFmpeg", "%s", avcodec_configuration());
return (jint)
1;
}
}

24
android/src/main/java/com/shabinder/spotiflyer/MainActivity.kt
View File

@ -54,10 +54,12 @@ import com.google.accompanist.insets.statusBarsPadding
import com.shabinder.common.core_components.ConnectionLiveData
import com.shabinder.common.core_components.analytics.AnalyticsManager
import com.shabinder.common.core_components.file_manager.FileManager
import com.shabinder.common.core_components.media_converter.AndroidMediaConverter
import com.shabinder.common.core_components.preference_manager.PreferenceManager
import com.shabinder.common.di.observeAsState
import com.shabinder.common.models.*
import com.shabinder.common.models.PlatformActions.Companion.SharedPreferencesKey
import com.shabinder.common.models.event.coroutines.success
import com.shabinder.common.providers.FetchPlatformQueryResult
import com.shabinder.common.root.SpotiFlyerRoot
import com.shabinder.common.root.callbacks.SpotiFlyerRootCallBacks
@ -65,7 +67,6 @@ import com.shabinder.common.translations.Strings
import com.shabinder.common.uikit.configurations.SpotiFlyerTheme
import com.shabinder.common.uikit.configurations.colorOffWhite
import com.shabinder.common.uikit.screens.SpotiFlyerRootContent
import com.shabinder.spotiflyer.ffmpeg.FFmpeg
import com.shabinder.spotiflyer.service.ForegroundService
import com.shabinder.spotiflyer.ui.AnalyticsDialog
import com.shabinder.spotiflyer.ui.NetworkDialog
@ -106,8 +107,15 @@ class MainActivity : ComponentActivity() {
// This app draws behind the system bars, so we want to handle fitting system windows
WindowCompat.setDecorFitsSystemWindows(window, false)
rootComponent = spotiFlyerRoot(defaultComponentContext())
lifecycleScope.launch {
Log.d("FFmpeg", "init")
FFmpeg.testInit()
AndroidMediaConverter().convertAudioFile("/storage/emulated/0/Music/SpotiFlyer/Playlists/Sing-along_Punjabi/Kya_Baat_Ay.mp3","/storage/emulated/0/Music/SpotiFlyer/Playlists/Sing-along_Punjabi/Kya_Baat_Ay.temp.mp3").fold({
Log.d("FFmpeg Success",it)
}){
it.printStackTrace()
}
}
/*FFmpeg.testInit()*/
setContent {
SpotiFlyerTheme {
Surface(contentColor = colorOffWhite) {
@ -246,7 +254,11 @@ class MainActivity : ComponentActivity() {
}
@Suppress("DEPRECATION")
override fun onRequestPermissionsResult(requestCode: Int, permissions: Array<out String>, grantResults: IntArray) {
override fun onRequestPermissionsResult(
requestCode: Int,
permissions: Array<out String>,
grantResults: IntArray
) {
super.onRequestPermissionsResult(requestCode, permissions, grantResults)
permissionGranted.value = checkPermissions()
}
@ -261,11 +273,13 @@ class MainActivity : ComponentActivity() {
override val fileManager: FileManager = this@MainActivity.fileManager
override val preferenceManager = this@MainActivity.preferenceManager
override val analyticsManager: AnalyticsManager = this@MainActivity.analyticsManager
override val downloadProgressFlow: MutableSharedFlow<HashMap<String, DownloadStatus>> = trackStatusFlow
override val downloadProgressFlow: MutableSharedFlow<HashMap<String, DownloadStatus>> =
trackStatusFlow
override val actions = object : Actions {
override val platformActions = object : PlatformActions {
override val imageCacheDir: String = applicationContext.cacheDir.absolutePath + File.separator
override val imageCacheDir: String =
applicationContext.cacheDir.absolutePath + File.separator
override val sharedPreferences = applicationContext.getSharedPreferences(
SharedPreferencesKey,
MODE_PRIVATE

9
android/src/main/java/com/shabinder/spotiflyer/ffmpeg/FFmpeg.kt
View File

@ -1,9 +0,0 @@
package com.shabinder.spotiflyer.ffmpeg
object FFmpeg {
external fun testInit(): Long
init {
System.loadLibrary("spotiflyer-converter")
}
}

3
common/core-components/build.gradle.kts
View File

@ -19,7 +19,8 @@ kotlin {
dependencies {
implementation(Extras.mp3agic)
implementation(Extras.Android.countly)
implementation("com.arthenica:ffmpeg-kit-audio:4.4.LTS")
implementation(project(":ffmpeg:ffmpeg-kit-android-lib"))
// implementation("com.arthenica:ffmpeg-kit-audio:4.4.LTS")
//api(files("$rootDir/libs/mobile-ffmpeg.aar"))
}
}

2
common/core-components/src/androidMain/kotlin/com/shabinder/common/core_components/analytics/AndroidAnalyticsManager.kt
View File

@ -24,7 +24,7 @@ internal class AndroidAnalyticsManager(private val mainActivity: Activity) : Ana
setIdMode(DeviceId.Type.OPEN_UDID)
setViewTracking(true)
enableCrashReporting()
setLoggingEnabled(true)
setLoggingEnabled(false)
setRecordAllThreadsWithCrash()
setRequiresConsent(true)
setShouldIgnoreAppCrawlers(true)

2
common/core-components/src/androidMain/kotlin/com/shabinder/common/core_components/file_manager/AndroidFileManager.kt
View File

@ -152,7 +152,7 @@ class AndroidFileManager(
SuspendableEvent.success(trackDetails.outputFilePath)
} catch (e: Throwable) {
e.printStackTrace()
if (songFile.exists()) songFile.delete()
//if (songFile.exists()) songFile.delete()
logger.e { "${songFile.absolutePath} could not be created" }
SuspendableEvent.error(e)
}

17
common/core-components/src/androidMain/kotlin/com/shabinder/common/core_components/media_converter/AndroidMediaConverter.kt
View File

@ -1,17 +1,9 @@
package com.shabinder.common.core_components.media_converter
import android.util.Log
import com.arthenica.ffmpegkit.FFmpegKit
import com.arthenica.ffmpegkit.ReturnCode
import com.shabinder.spotiflyer.ffmpeg.AndroidFFmpeg.runTranscode
import com.shabinder.common.models.AudioQuality
import com.shabinder.common.models.SpotiFlyerException
import org.koin.dsl.bind
import org.koin.dsl.module
import com.arthenica.ffmpegkit.FFprobeKit
import com.arthenica.ffmpegkit.MediaInformationSession
import kotlin.math.ceil
import kotlin.math.roundToInt
class AndroidMediaConverter : MediaConverter() {
@ -21,7 +13,10 @@ class AndroidMediaConverter : MediaConverter() {
audioQuality: AudioQuality,
progressCallbacks: (Long) -> Unit,
) = executeSafelyInPool {
val kbpsArg = if (audioQuality == AudioQuality.UNKNOWN) {
// 192 is Default
val audioBitrate = if (audioQuality == AudioQuality.UNKNOWN) 192 else audioQuality.kbps.toIntOrNull() ?: 192
runTranscode(inputFilePath,outputFilePath,audioBitrate).toString()
/*val kbpsArg = if (audioQuality == AudioQuality.UNKNOWN) {
val mediaInformation = FFprobeKit.getMediaInformation(inputFilePath)
val bitrate = ((mediaInformation.mediaInformation.bitrate).toFloat()/1000).roundToInt()
Log.d("MEDIA-INPUT Bit", bitrate.toString())
@ -41,7 +36,7 @@ class AndroidMediaConverter : MediaConverter() {
throw SpotiFlyerException.MP3ConversionFailed("FFmpeg Conversion Canceled for $inputFilePath")
}
else -> throw SpotiFlyerException.MP3ConversionFailed("FFmpeg Conversion Failed for $inputFilePath")
}
}*/
}
}

1
ffmpeg-android-maker

@ -1 +0,0 @@
Subproject commit b1dc4b643dc1c4015fc5f87075f9c135714def9e

27
android/CMakeLists.txt → ffmpeg/ffmpeg-kit-android-lib/CMakeLists.txt
View File

@ -9,7 +9,7 @@ set(
# List variable name
ffmpeg_libs_names
# Values in the list
avutil avformat avcodec avresample swresample
avutil avformat avcodec swresample avdevice avfilter swscale
)
foreach (ffmpeg_lib_name ${ffmpeg_libs_names})
@ -28,22 +28,29 @@ endforeach ()
add_library(
# Name for a library to build
spotiflyer-converter
spotiflyer-ffmpeg
# Type of a library
SHARED
# All cpp files to compile
src/main/cpp/main.cpp
# src/main/cpp/media_file_builder.cpp
# src/main/cpp/media_file_builder_jni.cpp
# src/main/cpp/frame_loader_context.cpp
# src/main/cpp/frame_loader_context_jni.cpp
# src/main/cpp/frame_extractor.cpp
# src/main/cpp/utils.cpp
# mobile-ffmpeg
src/main/cpp/doc_examples_transcode_aac.c
# ffmpeg-kit
# src/main/cpp/ffmpegkit.c
# src/main/cpp/ffprobekit.c
# src/main/cpp/ffmpegkit_exception.c
# src/main/cpp/fftools_cmdutils.c
# src/main/cpp/fftools_ffmpeg.c
# src/main/cpp/fftools_ffprobe.c
# src/main/cpp/fftools_ffmpeg_opt.c
# src/main/cpp/fftools_ffmpeg_hw.c
# src/main/cpp/fftools_ffmpeg_filter.c
# src/main/cpp/saf_wrapper.c
)
target_link_libraries(
# Library to link
spotiflyer-converter
spotiflyer-ffmpeg
# List of libraries to link against:
# Library for writing messages in LogCat
log

63
ffmpeg/ffmpeg-kit-android-lib/build.gradle.kts Normal file
View File

@ -0,0 +1,63 @@
plugins {
id("com.android.library")
id("kotlin-android")
}
android {
//ndkVersion "22.0.7026061"
compileSdk = Versions.compileSdkVersion
buildToolsVersion = "30.0.3"
defaultConfig {
consumerProguardFile("proguard-rules.pro")
minSdk = Versions.minSdkVersion
targetSdk = Versions.targetSdkVersion
/*versionCode = Versions.versionCode
versionName = Versions.versionName*/
ndk {
abiFilters.addAll(setOf("x86", "x86_64", "armeabi-v7a", "arm64-v8a"))
}
}
sourceSets {
named("main") {
jniLibs.srcDir("../ffmpeg-android-maker/output/lib")
}
}
externalNativeBuild {
cmake {
path("CMakeLists.txt")
}
}
buildTypes {
getByName("release") {
isMinifyEnabled = false
proguardFiles(
getDefaultProguardFile("proguard-android.txt"),
"proguard-rules.pro"
)
}
}
compileOptions {
sourceCompatibility = JavaVersion.VERSION_1_8
targetCompatibility = JavaVersion.VERSION_1_8
}
packagingOptions {
resources {
excludes.apply {
add("META-INF/*")
}
jniLibs.pickFirsts.apply {
add("**/*.so")
}
}
}
}
dependencies { /**/ }

17
ffmpeg/ffmpeg-kit-android-lib/proguard-rules.pro vendored Normal file
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@ -0,0 +1,17 @@
# Add project specific ProGuard rules here.
# You can control the set of applied configuration files using the
# proguardFiles setting in build.gradle.kts.
#
# For more details, see
# http://developer.android.com/guide/developing/tools/proguard.html
-keep class com.arthenica.ffmpegkit.FFmpegKitConfig {
native <methods>;
void log(long, int, byte[]);
void statistics(long, int, float, float, long , int, double, double);
void closeParcelFileDescriptor(int);
}
-keep class com.arthenica.ffmpegkit.AbiDetect {
native <methods>;
}

4
ffmpeg/ffmpeg-kit-android-lib/src/main/AndroidManifest.xml Normal file
View File

@ -0,0 +1,4 @@
<manifest xmlns:android="http://schemas.android.com/apk/res/android"
package="com.shabinder.spotiflyer.ffmpeg">
</manifest>

2
ffmpeg/ffmpeg-kit-android-lib/src/main/cpp/.gitignore vendored Normal file
View File

@ -0,0 +1,2 @@
/android_lts_support.o
/libandroidltssupport.a

915
ffmpeg/ffmpeg-kit-android-lib/src/main/cpp/doc_examples_transcode_aac.c Normal file
View File

@ -0,0 +1,915 @@
/*
* Copyright (c) 2013-2018 Andreas Unterweger
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
#include <stdio.h>
#include <jni.h>
#include <android/log.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
#include <jni.h>
/* The number of output channels */
#define OUTPUT_CHANNELS 2
/* The index of audio stream that will be transcoded */
static int audio_stream_idx = -1;
/**
* Open an input file and the required decoder.
* @param filename File to be opened
* @param[out] input_format_context Format context of opened file
* @param[out] input_codec_context Codec context of opened file
* @return Error code (0 if successful)
*/
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodecContext *avctx;
AVCodec *input_codec;
int error;
/* Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not open input file '%s' (error '%s')\n",
filename, av_err2str(error));
*input_format_context = NULL;
return error;
}
/* Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not open find stream info (error '%s')\n",
av_err2str(error));
avformat_close_input(input_format_context);
return error;
}
for (audio_stream_idx = 0; audio_stream_idx < (*input_format_context)->nb_streams; audio_stream_idx++) {
if ((*input_format_context)->streams[audio_stream_idx]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO)
break;
__android_log_print(ANDROID_LOG_INFO, "transcode_aac", "Skip non-audio input stream %d\n", audio_stream_idx);
}
/* Make sure that there is at least one audio stream in the input file. */
if (audio_stream_idx >= (*input_format_context)->nb_streams) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not find an audio (error '%s')\n",
av_err2str(error));
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/* Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[audio_stream_idx]->codecpar->codec_id))) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/* Allocate a new decoding context. */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate a decoding context\n");
avformat_close_input(input_format_context);
return AVERROR(ENOMEM);
}
/* Initialize the stream parameters with demuxer information. */
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[audio_stream_idx]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
avcodec_free_context(&avctx);
return error;
}
/* Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not open input codec (error '%s')\n",
av_err2str(error));
avcodec_free_context(&avctx);
avformat_close_input(input_format_context);
return error;
}
/* Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
}
/**
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
* @param filename File to be opened
* @param input_codec_context Codec context of input file
* @param[out] output_format_context Format context of output file
* @param[out] output_codec_context Codec context of output file
* @return Error code (0 if successful)
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context,
int audioBitrate
)
{
AVCodecContext *avctx = NULL;
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
int error;
/* Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not open output file '%s' (error '%s')\n",
filename, av_err2str(error));
return error;
}
/* Create a new format context for the output container format. */
if (!(*output_format_context = avformat_alloc_context())) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
/* Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
/* Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not find output file format\n");
goto cleanup;
}
if (!((*output_format_context)->url = av_strdup(filename))) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate url.\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/* Find the encoder to be used by its name. */
if (!(output_codec = avcodec_find_encoder((*output_format_context)->oformat->audio_codec))) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not find an encoder for %s(%d).\n",
(*output_format_context)->oformat->long_name,
(*output_format_context)->oformat->audio_codec);
goto cleanup;
}
/* Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
avctx = avcodec_alloc_context3(output_codec);
if (!avctx) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate an encoding context\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion. */
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = audioBitrate;
/* Allow the use of the experimental AAC encoder. */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
/* Some container formats (like MP4) require global headers to be present.
* Mark the encoder so that it behaves accordingly. */
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not open output codec (error '%s')\n",
av_err2str(error));
goto cleanup;
}
error = avcodec_parameters_from_context(stream->codecpar, avctx);
if (error < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not initialize stream parameters\n");
goto cleanup;
}
/* Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
cleanup:
avcodec_free_context(&avctx);
avio_closep(&(*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
}
/**
* Initialize one data packet for reading or writing.
* @param packet Packet to be initialized
*/
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
/* Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/**
* Initialize one audio frame for reading from the input file.
* @param[out] frame Frame to be initialized
* @return Error code (0 if successful)
*/
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate input frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param[out] resample_context Resample context for the required conversion
* @return Error code (0 if successful)
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext **resample_context)
{
int error;
/*
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!*resample_context) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/*
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/* Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/**
* Initialize a FIFO buffer for the audio samples to be encoded.
* @param[out] fifo Sample buffer
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
/* Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* Write the header of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not write output file header (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
/**
* Decode one audio frame from the input file.
* @param frame Audio frame to be decoded
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param[out] data_present Indicates whether data has been decoded
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false, there
* is more data to be decoded, i.e., this
* function has to be called again.
* @return Error code (0 if successful)
*/
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/* Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/* If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
__android_log_print(ANDROID_LOG_WARN, "transcode_aac", "Could not read frame (error '%s')\n",
av_err2str(error));
return error;
}
}
if (error != AVERROR_EOF && input_packet.stream_index != audio_stream_idx) {
goto cleanup;
}
/* Send the audio frame stored in the temporary packet to the decoder.
* The input audio stream decoder is used to do this. */
if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not send packet for decoding (error '%s')\n",
av_err2str(error));
return error;
}
/* Receive one frame from the decoder. */
error = avcodec_receive_frame(input_codec_context, frame);
/* If the decoder asks for more data to be able to decode a frame,
* return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the end of the input file is reached, stop decoding. */
} else if (error == AVERROR_EOF) {
*finished = 1;
error = 0;
goto cleanup;
} else if (error < 0) {
__android_log_print(ANDROID_LOG_WARN, "transcode_aac", "Could not decode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return decoded data. */
} else {
*data_present = 1;
goto cleanup;
}
cleanup:
av_packet_unref(&input_packet);
return error;
}
/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
* @param[out] converted_input_samples Array of converted samples. The
* dimensions are reference, channel
* (for multi-channel audio), sample.
* @param output_codec_context Codec context of the output file
* @param frame_size Number of samples to be converted in
* each round
* @return Error code (0 if successful)
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac",
"Could not allocate converted input samples (error '%s')\n",
av_err2str(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
}
/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is
* specified by frame_size.
* @param input_data Samples to be decoded. The dimensions are
* channel (for multi-channel audio), sample.
* @param[out] converted_data Converted samples. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @param resample_context Resample context for the conversion
* @return Error code (0 if successful)
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
SwrContext *resample_context)
{
int error;
/* Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not convert input samples (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
/**
* Add converted input audio samples to the FIFO buffer for later processing.
* @param fifo Buffer to add the samples to
* @param converted_input_samples Samples to be added. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @return Error code (0 if successful)
*/
static int add_samples_to_fifo(AVAudioFifo *fifo,
uint8_t **converted_input_samples,
const int frame_size)
{
int error;
/* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples. */
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not reallocate FIFO\n");
return error;
}
/* Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
/**
* Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer.
* @param fifo Buffer used for temporary storage
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param resampler_context Resample context for the conversion
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false,
* there is more data to be decoded,
* i.e., this function has to be called
* again.
* @return Error code (0 if successful)
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext *resampler_context,
int *finished)
{
/* Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present = 0;
int ret = AVERROR_EXIT;
/* Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/* Decode one frame worth of audio samples. */
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error. */
if (*finished) {
ret = 0;
goto cleanup;
}
/* If there is decoded data, convert and store it. */
if (data_present) {
/* Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
/* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples. */
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
/* Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
* @param[out] frame Frame to be initialized
* @param output_codec_context Codec context of the output file
* @param frame_size Size of the frame
* @return Error code (0 if successful)
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/* Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
__android_log_print(ANDROID_LOG_WARN, "transcode_aac", "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity. */
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified. */
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
__android_log_print(ANDROID_LOG_WARN, "transcode_aac", "Could not allocate output frame samples (error '%s')\n",
av_err2str(error));
av_frame_free(frame);
return error;
}
return 0;
}
/* Global timestamp for the audio frames. */
static int64_t pts = 0;
/**
* Encode one frame worth of audio to the output file.
* @param frame Samples to be encoded
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @param[out] data_present Indicates whether data has been
* encoded
* @return Error code (0 if successful)
*/
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/* Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
/* Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
/* Send the audio frame stored in the temporary packet to the encoder.
* The output audio stream encoder is used to do this. */
error = avcodec_send_frame(output_codec_context, frame);
/* The encoder signals that it has nothing more to encode. */
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
__android_log_print(ANDROID_LOG_WARN, "transcode_aac", "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
return error;
}
/* Receive one encoded frame from the encoder. */
error = avcodec_receive_packet(output_codec_context, &output_packet);
/* If the encoder asks for more data to be able to provide an
* encoded frame, return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the last frame has been encoded, stop encoding. */
} else if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not encode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return encoded data. */
} else {
*data_present = 1;
}
/* Write one audio frame from the temporary packet to the output file. */
if (*data_present &&
(error = av_write_frame(output_format_context, &output_packet)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not write frame (error '%s')\n",
av_err2str(error));
goto cleanup;
}
cleanup:
av_packet_unref(&output_packet);
return error;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
* @param fifo Buffer used for temporary storage
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
/* Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size. */
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
/* Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
/* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily. */
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
__android_log_print(ANDROID_LOG_WARN, "transcode_aac", "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
/* Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
av_frame_free(&output_frame);
return 0;
}
/**
* Write the trailer of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
if ((error = av_write_trailer(output_format_context)) < 0) {
__android_log_print(ANDROID_LOG_ERROR, "transcode_aac", "Could not write output file trailer (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
JNIEXPORT jint JNICALL Java_com_shabinder_spotiflyer_ffmpeg_AndroidFFmpeg_runTranscode(
JNIEnv *env, jobject c,
jstring inFilename,
jstring outFilename,
jint audioBitrate
) {
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
const char *in_filename = (*env)->GetStringUTFChars(env, inFilename, 0);
const char *out_filename = (*env)->GetStringUTFChars(env, outFilename, 0);
__android_log_print(ANDROID_LOG_INFO, "transcode_aac", "Bitrate:%d :: %s -> %s\n", audioBitrate, in_filename, out_filename);
/* Open the input file for reading. */
if (open_input_file(in_filename, &input_format_context,
&input_codec_context))
goto cleanup;
__android_log_print(ANDROID_LOG_INFO, "transcode_aac", "Input format: %s.\n",
input_format_context->iformat->long_name);
/* Open the output file for writing. */
if (open_output_file(out_filename, input_codec_context,
&output_format_context, &output_codec_context, (audioBitrate*1000)))
goto cleanup;
__android_log_print(ANDROID_LOG_INFO, "transcode_aac", "Output format: %s.\n",
output_format_context->oformat->long_name);
/* Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/* Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
/* Write the header of the output file container. */
if (write_output_file_header(output_format_context))
goto cleanup;
/* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither. */
while (1) {
/* Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples. */
while (av_audio_fifo_size(fifo) < output_frame_size) {
/* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer. */
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file. */
if (finished)
break;
}
/* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder. */
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish. */
if (finished) {
int data_written;
/* Flush the encoder as it may have delayed frames. */
do {
data_written = 0;
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/* Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_free_context(&output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_free_context(&input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}

27
ffmpeg/ffmpeg-kit-android-lib/src/main/java/com/shabinder/spotiflyer/ffmpeg/AndroidFFmpeg.kt Normal file
View File

@ -0,0 +1,27 @@
package com.shabinder.spotiflyer.ffmpeg
import android.util.Log
object AndroidFFmpeg {
/**
*
* Run transcode_aac from doc/examples.
*
* @return zero if transcoding was successful
*/
@JvmStatic
external fun runTranscode(inFilename: String?, outFilename: String?, audioBitrate: Int): Int
init {
Log.i("FFmpeg", "Loading mobile-ffmpeg.")
System.loadLibrary("avutil")
System.loadLibrary("swscale")
System.loadLibrary("swresample")
System.loadLibrary("avcodec")
System.loadLibrary("avformat")
System.loadLibrary("avfilter")
System.loadLibrary("avdevice")
//System.loadLibrary("avresample")
System.loadLibrary("spotiflyer-ffmpeg")
}
}

1
settings.gradle.kts
View File

@ -27,6 +27,7 @@ include(
":common:providers",
":common:core-components",
":common:dependency-injection",
":ffmpeg:ffmpeg-kit-android-lib",
":android",
":desktop",
":web-app",