diff --git a/app/src/main/cpp/audio_resample.cpp b/app/src/main/cpp/audio_resample.cpp index 4ae066a..da330df 100644 --- a/app/src/main/cpp/audio_resample.cpp +++ b/app/src/main/cpp/audio_resample.cpp @@ -41,14 +41,12 @@ static int64_t pts = 0; */ static int open_input_file(const char *filename, AVFormatContext **input_format_context, - AVCodecContext **input_codec_context) -{ + AVCodecContext **input_codec_context) { int error; - AVCodec *input_codec; + const AVCodec *input_codec; AVCodecContext *avctx = nullptr; AVStream *audio_stream = nullptr; - /* Open the input file to read from it. */ 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)); @@ -56,13 +54,11 @@ static int open_input_file(const char *filename, return error; } - /* Get information on the input file (number of streams etc.). */ 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; } - /* Find a decoder for the audio stream. */ 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]; @@ -73,7 +69,6 @@ static int open_input_file(const char *filename, goto cleanup; } - /* Allocate a new decoding context. */ avctx = avcodec_alloc_context3(input_codec); if (!avctx) { ALOGE("Could not allocate a decoding context\n"); @@ -85,13 +80,10 @@ static int open_input_file(const char *filename, goto cleanup; } - /* Open the decoder for the audio stream to use it later. */ if ((error = avcodec_open2(avctx, input_codec, nullptr)) < 0) { ALOGE("Could not open input codec (error:%s)\n", av_err2str(error)); goto cleanup; } - - /* Save the decoder context for easier access later. */ *input_codec_context = avctx; return 0; @@ -105,20 +97,17 @@ cleanup: /** * 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. * */ static int open_output_file(const char *filename, int sample_rate, AVCodecContext *input_codec_context, AVFormatContext **output_format_context, - AVCodecContext **output_codec_context) -{ + AVCodecContext **output_codec_context) { AVCodecContext *avctx = nullptr; AVIOContext *output_io_context = nullptr; - AVStream *stream = nullptr; - AVCodec *output_codec = nullptr; + AVStream *stream; + const AVCodec *output_codec; int error; /* Open the output file to write to it. */ @@ -170,8 +159,7 @@ static int open_output_file(const char *filename, goto cleanup; } - /* Set the basic encoder parameters. - * The input file's sample rate is used to avoid a sample rate conversion. */ + /* Set the basic encoder parameters.*/ avctx->channels = OUTPUT_CHANNELS; avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS); avctx->sample_rate = sample_rate; @@ -217,8 +205,7 @@ cleanup: * Initialize one data packet for reading or writing. * @param packet Packet to be initialized */ -static void init_packet(AVPacket *packet) -{ +static void init_packet(AVPacket *packet) { av_init_packet(packet); packet->data = nullptr; packet->size = 0; @@ -228,8 +215,7 @@ static void init_packet(AVPacket *packet) * Initialize one audio frame for reading from the input file. * */ -static int init_input_frame(AVFrame **frame) -{ +static int init_input_frame(AVFrame **frame) { if (!(*frame = av_frame_alloc())) { ALOGE("Could not allocate input frame\n"); return AVERROR(ENOMEM); @@ -243,17 +229,8 @@ static int init_input_frame(AVFrame **frame) */ static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - SwrContext **resample_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(nullptr, av_get_default_channel_layout(output_codec_context->channels), output_codec_context->sample_fmt, @@ -280,8 +257,7 @@ static int init_resampler(AVCodecContext *input_codec_context, * Initialize a FIFO buffer for the audio samples to be encoded. * */ -static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context) -{ +static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context) { if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt, output_codec_context->channels, 1))) { ALOGE("Could not allocate FIFO\n"); @@ -294,8 +270,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context) * Write the header of the output file container. * */ -static int write_output_file_header(AVFormatContext *output_format_context) -{ +static int write_output_file_header(AVFormatContext *output_format_context) { int error; if ((error = avformat_write_header(output_format_context, nullptr)) < 0) { ALOGE("Could not write output file header (error:%s)\n", av_err2str(error)); @@ -311,14 +286,12 @@ static int write_output_file_header(AVFormatContext *output_format_context) static int decode_audio_frame(AVFrame *frame, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, - int *data_present, int *finished) -{ + int *data_present, int *finished) { int error; AVPacket input_packet; init_packet(&input_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 { @@ -334,8 +307,7 @@ static int decode_audio_frame(AVFrame *frame, goto cleanup; } - /* Send the audio frame stored in the temporary packet to the decoder. - * The input audio stream decoder is used to do this. */ + /* 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; @@ -343,12 +315,9 @@ static int decode_audio_frame(AVFrame *frame, /* 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; @@ -368,27 +337,18 @@ cleanup: /** * 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. * */ static int init_converted_samples(uint8_t ***converted_input_samples, AVCodecContext *output_codec_context, - int frame_size) -{ + 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 - */ 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); } - /* Allocate memory for the samples of all channels in one consecutive - * block for convenience. */ if ((error = av_samples_alloc(*converted_input_samples, nullptr, output_codec_context->channels, frame_size, @@ -403,14 +363,12 @@ static int init_converted_samples(uint8_t ***converted_input_samples, /** * 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. + * The size of which is specified by frame_size. * */ static int convert_samples(const uint8_t **input_data, const int input_size, uint8_t **converted_data, const int output_size, - SwrContext *resample_context) -{ + SwrContext *resample_context) { int error; if ((error = swr_convert(resample_context, converted_data, output_size, input_data, input_size)) < 0) { @@ -427,12 +385,9 @@ static int convert_samples(const uint8_t **input_data, const int input_size, */ static int add_samples_to_fifo(AVAudioFifo *fifo, uint8_t **converted_input_samples, - const int frame_size) -{ + 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. */ + /* 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) { ALOGE("Could not reallocate FIFO\n"); return error; @@ -457,11 +412,8 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, 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. */ + int *finished) { AVFrame *input_frame = nullptr; - /* Temporary storage for the converted input samples. */ uint8_t **converted_dst_samples = nullptr; int data_present = 0; int ret = AVERROR_EXIT; @@ -473,9 +425,6 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, 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; @@ -485,19 +434,15 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, 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); - /* Initialize the temporary storage for the converted input samples. */ if (init_converted_samples(&converted_dst_samples, output_codec_context, dst_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, input_frame->nb_samples, converted_dst_samples, dst_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_dst_samples, dst_nb_samples)) goto cleanup; @@ -516,33 +461,23 @@ cleanup: /** * Initialize one input frame for writing to the output file. - * The frame will be exactly frame_size samples large. * */ static int init_output_frame(AVFrame **frame, AVCodecContext *output_codec_context, - int frame_size) -{ + int frame_size) { int error; - /* Create a new frame to store the audio samples. */ if (!(*frame = av_frame_alloc())) { ALOGE("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) { ALOGE("Could not allocate output frame samples (error:%s)\n", av_err2str(error)); av_frame_free(frame); @@ -559,8 +494,7 @@ static int init_output_frame(AVFrame **frame, static int encode_audio_frame(AVFrame *frame, AVFormatContext *output_format_context, AVCodecContext *output_codec_context, - int *data_present) -{ + int *data_present) { int error; AVPacket output_packet; init_packet(&output_packet); @@ -571,10 +505,8 @@ static int encode_audio_frame(AVFrame *frame, 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. */ + /* Send frame stored in the temporary packet to the encoder.*/ 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; @@ -585,12 +517,9 @@ static int encode_audio_frame(AVFrame *frame, /* 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; @@ -620,30 +549,21 @@ cleanup: */ 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. */ + AVCodecContext *output_codec_context) { + int data_written; 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) { ALOGE("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); @@ -657,8 +577,7 @@ static int load_encode_and_write(AVAudioFifo *fifo, * Write the trailer of the output file container. * */ -static int write_output_file_trailer(AVFormatContext *output_format_context) -{ +static int write_output_file_trailer(AVFormatContext *output_format_context) { int error; if ((error = av_write_trailer(output_format_context)) < 0) { ALOGE("Could not write output file trailer (error:%s)\n", av_err2str(error)); @@ -667,8 +586,7 @@ static int write_output_file_trailer(AVFormatContext *output_format_context) return 0; } -int resampling(const char *src_file, const char *dst_file, int sampleRate) -{ +int resampling(const char *src_file, const char *dst_file, int sampleRate) { int ret = AVERROR_EXIT; AVAudioFifo *fifo = nullptr; SwrContext *resample_context = nullptr; @@ -694,46 +612,31 @@ int resampling(const char *src_file, const char *dst_file, int sampleRate) 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; - /* 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. */ + /* 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 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. */ + /* 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)) - /* 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. */ + /* encode all the remaining samples. */ if (finished) { int data_written; - /* Flush the encoder as it may have delayed frames. */ do { data_written = 0; if (encode_audio_frame(nullptr, output_format_context,