// // Created by xu fulong on 2022/7/12. // #include #ifdef __cplusplus extern "C" { #endif #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" #ifdef __cplusplus } #endif #include "ffmpeg_jni_define.h" #define ALOGE(Format, ...) LOGE("audio_resample", Format, ##__VA_ARGS__) /* Global timestamp for the audio frames. */ static int64_t pts = 0; static AVPacket input_packet; static AVPacket output_packet; static AVFrame *input_frame; static AVFrame *output_frame; /** * Open an input file and the required decoder. * */ static int open_input_file(const char *filename, AVFormatContext **input_format_context, AVCodecContext **input_codec_context) { int error; const AVCodec *input_codec; AVCodecContext *avctx = nullptr; AVStream *audio_stream = nullptr; if ((error = avformat_open_input(input_format_context, filename, nullptr, nullptr)) < 0) { ALOGE("Could not open input file:%s (error:%s)\n", filename, av_err2str(error)); *input_format_context = nullptr; return error; } if ((error = avformat_find_stream_info(*input_format_context, nullptr)) < 0) { ALOGE("Could not open find stream info (error:%s)\n", av_err2str(error)); goto cleanup; } for (int i = 0; i < (*input_format_context)->nb_streams; ++i) { if ((*input_format_context)->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { audio_stream = (*input_format_context)->streams[i]; } } if (!(input_codec = avcodec_find_decoder(audio_stream->codecpar->codec_id))) { ALOGE("Could not find input codec\n"); goto cleanup; } avctx = avcodec_alloc_context3(input_codec); if (!avctx) { ALOGE("Could not allocate a decoding context\n"); goto cleanup; } error = avcodec_parameters_to_context(avctx, audio_stream->codecpar); if (error < 0) { goto cleanup; } if ((error = avcodec_open2(avctx, input_codec, nullptr)) < 0) { ALOGE("Could not open input codec (error:%s)\n", av_err2str(error)); goto cleanup; } *input_codec_context = avctx; return 0; cleanup: if (avctx) avcodec_free_context(&avctx); avformat_close_input(input_format_context); return -1; } /** * Open an output file and the required encoder. * */ static int open_output_file(const char *filename, int sample_rate, AVCodecContext *input_codec_context, AVFormatContext **output_format_context, AVCodecContext **output_codec_context) { AVCodecContext *avctx = nullptr; AVIOContext *output_io_context = nullptr; AVStream *stream; const AVCodec *output_codec; int error; /* Open the output file to write to it. */ if ((error = avio_open(&output_io_context, filename, AVIO_FLAG_WRITE)) < 0) { ALOGE("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())) { ALOGE("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(nullptr, filename, nullptr))) { ALOGE("Could not find output file format\n"); goto cleanup; } if (!((*output_format_context)->url = av_strdup(filename))) { ALOGE("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(input_codec_context->codec_id))) { ALOGE( "Could not find encoder=%s\n", input_codec_context->codec->name); goto cleanup; } /* Create a new audio stream in the output file container. */ if (!(stream = avformat_new_stream(*output_format_context, nullptr))) { ALOGE("Could not create new stream\n"); error = AVERROR(ENOMEM); goto cleanup; } avctx = avcodec_alloc_context3(output_codec); if (!avctx) { ALOGE("Could not allocate an encoding context\n"); error = AVERROR(ENOMEM); goto cleanup; } /* Set the basic encoder parameters.*/ avctx->channels = input_codec_context->channels; avctx->channel_layout = av_get_default_channel_layout(input_codec_context->channels); avctx->sample_rate = sample_rate; avctx->sample_fmt = output_codec->sample_fmts[0]; /* 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 = 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, nullptr)) < 0) { ALOGE("Could not open output codec (error:%s)\n", av_err2str(error)); goto cleanup; } error = avcodec_parameters_from_context(stream->codecpar, avctx); if (error < 0) { ALOGE("Could not initialize stream parameters\n"); goto cleanup; } *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 = nullptr; return error < 0 ? error : AVERROR_EXIT; } /** * Initialize the audio resample based on the input and output codec settings. * */ static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext **resample_context) { *resample_context = swr_alloc_set_opts(nullptr, 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, nullptr); return swr_init(*resample_context); } /** * Decode one audio frame from the input file. * */ static int decode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, int *data_present, int *finished) { int error; if ((error = av_read_frame(input_format_context, &input_packet)) < 0) { if (error == AVERROR_EOF) *finished = 1; else { ALOGE("Could not read frame (error:%s)\n", av_err2str(error)); return error; } } if (input_format_context->streams[input_packet.stream_index]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) { error = 0; ALOGE("isn't audio packet, skip it..."); goto cleanup; } /* Send the audio frame stored in the temporary packet to the decoder.*/ if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) { ALOGE("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 (error == AVERROR(EAGAIN)) { error = 0; goto cleanup; } else if (error == AVERROR_EOF) { *finished = 1; error = 0; goto cleanup; } else if (error < 0) { ALOGE("Could not decode frame (error:%s)\n", av_err2str(error)); goto cleanup; } 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. * */ static int init_converted_samples(uint8_t ***converted_input_samples, AVCodecContext *output_codec_context, int frame_size) { int error; if (!(*converted_input_samples = (uint8_t **) calloc(output_codec_context->channels, sizeof(**converted_input_samples)))) { ALOGE("Could not allocate converted input sample pointers\n"); return AVERROR(ENOMEM); } if ((error = av_samples_alloc(*converted_input_samples, nullptr, output_codec_context->channels, frame_size, output_codec_context->sample_fmt, 0)) < 0) { ALOGE("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; } /** * Read one audio frame from the input file, decode, convert and store * it in the FIFO buffer. * */ static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, SwrContext *resample_context, int *finished) { uint8_t **converted_dst_samples = nullptr; int data_present = 0; int ret = AVERROR_EXIT; /* 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 (*finished) { ret = 0; goto cleanup; } /* If there is decoded data, convert and store it. */ if (data_present) { int dst_nb_samples = (int) av_rescale_rnd(input_frame->nb_samples, output_codec_context->sample_rate, input_codec_context->sample_rate, AV_ROUND_UP); if (init_converted_samples(&converted_dst_samples, output_codec_context, dst_nb_samples)) goto cleanup; ret = swr_convert(resample_context, converted_dst_samples, dst_nb_samples, (const uint8_t**)input_frame->extended_data, input_frame->nb_samples); if (ret < 0) { ALOGE("Could not convert input samples (error:%s)\n", av_err2str(ret)); goto cleanup; } if (av_audio_fifo_write(fifo, (void **)converted_dst_samples, dst_nb_samples) < dst_nb_samples) goto cleanup; } ret = 0; cleanup: if (converted_dst_samples) { av_freep(&converted_dst_samples[0]); free(converted_dst_samples); } return ret; } static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context) { *frame = av_frame_alloc(); (*frame)->nb_samples = output_codec_context->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; int ret = av_frame_get_buffer(*frame, 0); if (ret < 0) { ALOGE("Could not allocate output frame samples (error:%s)\n", av_err2str(ret)); } return ret; } /** * Encode one frame worth of audio to the output file. * */ static int encode_audio_frame(AVFrame *frame, AVFormatContext *output_format_context, AVCodecContext *output_codec_context, int *data_present) { int error; /* Set a timestamp based on the sample rate for the container. */ if (frame) { frame->pts = pts; pts += frame->nb_samples; } /* Send frame stored in the temporary packet to the encoder.*/ error = avcodec_send_frame(output_codec_context, frame); if (error == AVERROR_EOF) { error = 0; goto cleanup; } else if (error < 0) { ALOGE("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 (error == AVERROR(EAGAIN)) { error = 0; goto cleanup; } else if (error == AVERROR_EOF) { error = 0; goto cleanup; } else if (error < 0) { ALOGE("Could not encode frame (error:%s)\n", av_err2str(error)); goto cleanup; } 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) { ALOGE("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. * */ static int load_encode_and_write(AVAudioFifo *fifo, AVFormatContext *output_format_context, AVCodecContext *output_codec_context) { int data_written; const int frame_size = FFMIN(av_audio_fifo_size(fifo), output_codec_context->frame_size); output_frame->nb_samples = frame_size; if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) { ALOGE("Could not read data from FIFO\n"); return AVERROR_EXIT; } if (encode_audio_frame(output_frame, output_format_context, output_codec_context, &data_written)) { return AVERROR_EXIT; } return 0; } int resampling(const char *src_file, const char *dst_file, int sampleRate) { int ret = AVERROR_EXIT; AVAudioFifo *fifo = nullptr; SwrContext *resample_context = nullptr; AVCodecContext *input_codec_context = nullptr; AVCodecContext *output_codec_context = nullptr; AVFormatContext *input_format_context = nullptr; AVFormatContext *output_format_context = nullptr; /* Open the input file for reading. */ if (open_input_file(src_file, &input_format_context, &input_codec_context)) goto cleanup; /* Open the output file for writing. */ if (open_output_file(dst_file, sampleRate, input_codec_context, &output_format_context, &output_codec_context)) goto cleanup; /* Initialize the re-sampler 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. */ fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt, output_codec_context->channels, 1024 * 10); input_frame = av_frame_alloc(); if (init_output_frame(&output_frame, output_codec_context)) goto cleanup; /* Write the header of the output file container. */ if ((ret = avformat_write_header(output_format_context, nullptr)) < 0) { ALOGE("write header error=%s", av_err2str(ret)); } while (1) { const int output_frame_size = output_codec_context->frame_size; int finished = 0; while (av_audio_fifo_size(fifo) < output_frame_size) { /* Decode one frame, convert 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 (finished) break; } /* If we have enough samples for the encoder, we encode them.*/ while (av_audio_fifo_size(fifo) >= output_frame_size || (finished && av_audio_fifo_size(fifo) > 0)) if (load_encode_and_write(fifo, output_format_context, output_codec_context)) goto cleanup; /* encode all the remaining samples. */ if (finished) { int data_written; do { data_written = 0; if (encode_audio_frame(nullptr, output_format_context, output_codec_context, &data_written)) goto cleanup; } while (data_written); break; } } /* Write the trailer of the output file container. */ if (av_write_trailer(output_format_context)) { ALOGE("write trailer error..."); } 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); if (output_frame) av_frame_free(&output_frame); if (input_frame) av_frame_free(&input_frame); return ret; } extern "C" VIDEO_PLAYER_FUNC(int, audioResample, jstring srcFile, jstring dstFile, int sampleRate) { const char *src_file = env->GetStringUTFChars(srcFile, JNI_FALSE); const char *dst_file = env->GetStringUTFChars(dstFile, JNI_FALSE); int ret = resampling(src_file, dst_file, sampleRate); env->ReleaseStringUTFChars(dstFile, dst_file); env->ReleaseStringUTFChars(srcFile, src_file); return ret; }