From patchwork Fri Mar 31 13:11:11 2017 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Paolo Prete X-Patchwork-Id: 3218 Delivered-To: ffmpegpatchwork@gmail.com Received: by 10.103.44.195 with SMTP id s186csp2036154vss; Fri, 31 Mar 2017 06:11:46 -0700 (PDT) X-Received: by 10.28.98.135 with SMTP id w129mr3351789wmb.68.1490965906263; Fri, 31 Mar 2017 06:11:46 -0700 (PDT) Return-Path: Received: from ffbox0-bg.mplayerhq.hu (ffbox0-bg.ffmpeg.org. [79.124.17.100]) by mx.google.com with ESMTP id 71si3417847wmo.26.2017.03.31.06.11.45; Fri, 31 Mar 2017 06:11:46 -0700 (PDT) Received-SPF: pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) client-ip=79.124.17.100; Authentication-Results: mx.google.com; dkim=neutral (body hash did not verify) header.i=@yahoo.it; spf=pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) smtp.mailfrom=ffmpeg-devel-bounces@ffmpeg.org Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id BD97E689A3D; Fri, 31 Mar 2017 16:11:42 +0300 (EEST) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from nm32-vm8.bullet.mail.ir2.yahoo.com (nm32-vm8.bullet.mail.ir2.yahoo.com [212.82.97.104]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id 36659680C77 for ; Fri, 31 Mar 2017 16:11:35 +0300 (EEST) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=yahoo.it; s=s2048; t=1490965894; bh=xUqDoHfkebIlkcham3HesVvw2DcAR02mkXSCDfQSQLs=; h=From:To:Cc:Subject:Date:In-Reply-To:References:From:Subject; b=f6Ll9RUa8MKb7UBOkO31DhcJQS8rsOMqsKCxCk8JlzQw0lQvvOz4Yjy8cHo/rumkB0NBU9v5K7XAPQKIRQMnAQSDrm6z0UqZJuOfX3Ico8qSRpUbrtki7+bA3ic4rsNRrvGXGbuXKbJWKs4/ACJ0DU6RN5HjR1o+U94QXV1pLFfIBTu+G76iGeWtYEsc4bTRGmfruT63iKQuwe+dhlP1gFRiDUBG7XoxpCRAKr9TNKXgKSBa1lHrRskqep4Pn04GZSfft4AhsQ0igZ7e2IpMcjqhAztXrF84jcLqq3vnioIPZ27ZzR24RpwuqfcIdXQ4LAKNP0cTtj3IJf+dXemdfw== Received: from [212.82.98.62] by nm32.bullet.mail.ir2.yahoo.com with NNFMP; 31 Mar 2017 13:11:34 -0000 Received: from [46.228.39.76] by tm15.bullet.mail.ir2.yahoo.com with NNFMP; 31 Mar 2017 13:11:34 -0000 Received: from [127.0.0.1] by smtp113.mail.ir2.yahoo.com with NNFMP; 31 Mar 2017 13:11:34 -0000 X-Yahoo-Newman-Id: 856801.28269.bm@smtp113.mail.ir2.yahoo.com X-Yahoo-Newman-Property: ymail-3 X-YMail-OSG: zUyMB20VM1l.lDRJD1QabpuDSl_uDMtMJbAhgM3J5K147vS iJgMhF2NyPlXPXkvfmg2cYkxSRUf_h0NkIaXvee8WuSVPFMACotoNXj07a2m uVwXuAPzNe9csaQbOUo5RDIBauhwniIAQvk7vKO8mz4IMWUHIOVf_vSDVxJn _zBxOqGGdWxcO6nmeVdyblzaHqTlw2KLQB8NA7Gk7tFg4OIvvT_f7tTi_kky R_j0NRRv2gM3ldDmzCZt8s7L5fWTaFdTW62XNEUHu8bYk1Ob5wpXirhDRDPo XW4XXwKljvYa841EYq2jfdyOmJt8NdPTid0HBNTWQYCi8hKOkpYZClvfF4q8 MAkbrwg189aj_WuZANXVX3WX8hPzIdQncAdK4fDrramOXHwg0HDIg0FWA9CM S.Y2GnsO4TAG.8iePODd6ttKKFyUf_2TI5SDz0GrZDF8ES4_EejYVX.I1Lfp 784c8edKP1dNwJIg8GItE8nIypGJggft5YxleURUTi7PNkDqj2s0.QXoUloS wQAR2rLrNfnBxk4_7lYFRl9VFtZriW_bxG8ZSLR5osq1wvIZbz8UD2BO2_KY QMbkEGuw.3BlJfUw- X-Yahoo-SMTP: F3G2xp.swBB0A1wKHjDaeJZbPU5Pmnfl From: Paolo Prete To: ffmpeg-devel@ffmpeg.org Date: Fri, 31 Mar 2017 15:11:11 +0200 Message-Id: <20170331131111.25457-1-p4olo_prete@yahoo.it> X-Mailer: git-send-email 2.9.3 In-Reply-To: <20170331015957.GQ4714@nb4> References: <20170331015957.GQ4714@nb4> Subject: [FFmpeg-devel] [PATCH] New API usage example (encode_raw_audio_file_to_aac) X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Cc: Paolo Prete MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" --- configure | 2 + doc/Makefile | 41 ++-- doc/examples/.gitignore | 1 + doc/examples/Makefile | 1 + doc/examples/encode_raw_audio_file_to_aac.c | 338 ++++++++++++++++++++++++++++ 5 files changed, 363 insertions(+), 20 deletions(-) create mode 100644 doc/examples/encode_raw_audio_file_to_aac.c diff --git a/configure b/configure index 6d76cf7..1069f9f 100755 --- a/configure +++ b/configure @@ -1466,6 +1466,7 @@ EXAMPLE_LIST=" decode_video_example demuxing_decoding_example encode_audio_example + encode_raw_audio_file_to_aac_example encode_video_example extract_mvs_example filter_audio_example @@ -3175,6 +3176,7 @@ decode_audio_example_deps="avcodec avutil" decode_video_example_deps="avcodec avutil" demuxing_decoding_example_deps="avcodec avformat avutil" encode_audio_example_deps="avcodec avutil" +encode_raw_audio_file_to_aac_example_deps="avcodec avformat avutil swresample" encode_video_example_deps="avcodec avutil" extract_mvs_example_deps="avcodec avformat avutil" filter_audio_example_deps="avfilter avutil" diff --git a/doc/Makefile b/doc/Makefile index c193fc3..a7c349b 100644 --- a/doc/Makefile +++ b/doc/Makefile @@ -36,26 +36,27 @@ DOCS-$(CONFIG_MANPAGES) += $(MANPAGES) DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES) DOCS = $(DOCS-yes) -DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd -DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading -DOC_EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio -DOC_EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video -DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding -DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio -DOC_EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video -DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs -DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio -DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio -DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video -DOC_EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient -DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata -DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing -DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec -DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing -DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio -DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video -DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac -DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding +DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd +DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading +DOC_EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio +DOC_EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video +DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding +DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio +DOC_EXAMPLES-$(CONFIG_ENCODE_RAW_AUDIO_FILE_TO_AAC_EXAMPLE) += encode_raw_audio_file_to_aac +DOC_EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video +DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs +DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio +DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio +DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video +DOC_EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient +DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata +DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing +DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec +DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing +DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio +DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video +DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac +DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes) DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF)) diff --git a/doc/examples/.gitignore b/doc/examples/.gitignore index 6bd9dc1..7b25718 100644 --- a/doc/examples/.gitignore +++ b/doc/examples/.gitignore @@ -4,6 +4,7 @@ /decode_video /demuxing_decoding /encode_audio +/encode_raw_audio_file_to_aac /encode_video /extract_mvs /filter_audio diff --git a/doc/examples/Makefile b/doc/examples/Makefile index 2d0a306..24929e3 100644 --- a/doc/examples/Makefile +++ b/doc/examples/Makefile @@ -17,6 +17,7 @@ EXAMPLES= avio_dir_cmd \ decode_video \ demuxing_decoding \ encode_audio \ + encode_raw_audio_file_to_aac \ encode_video \ extract_mvs \ filtering_video \ diff --git a/doc/examples/encode_raw_audio_file_to_aac.c b/doc/examples/encode_raw_audio_file_to_aac.c new file mode 100644 index 0000000..5e6df5c --- /dev/null +++ b/doc/examples/encode_raw_audio_file_to_aac.c @@ -0,0 +1,338 @@ +/* + * Copyright (c) 2017 Paolo Prete (p4olo_prete@yahoo.it) + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * API example for adts-aac encoding raw audio files. + * This example reads a raw audio input file, converts it to float-planar format, performs + * aac encoding and puts the encoded frames into an ADTS container. + * The encoded stream is written to a file named "out.aac" + * It can be adapted, with few changes, to a custom raw audio source (i.e: a live one). + * It uses a custom I/O write callback (write_adts_muxed_data()) in order to show to the user + * how to access muxed packets written in memory, before they are written to the output file. + * The raw input audio file can be created with: + * + * ffmpeg -i some_audio_file -f f32le -acodec pcm_f32le -ac 2 -ar 16000 raw_audio_file.raw + * + * @example encode_raw_audio_file_to_aac.c + */ + +#include +#include +#include +#include + +#define ENCODER_BITRATE 64000 +#define SAMPLE_RATE 16000 +#define INPUT_SAMPLE_FMT AV_SAMPLE_FMT_FLT +#define CHANNELS 2 + +static int encoded_pkt_counter = 1; + +static int write_adts_muxed_data(void *opaque, uint8_t *adts_data, int size) +{ + FILE *encoded_audio_file = (FILE *)opaque; + fwrite(adts_data, 1, size, encoded_audio_file); //(f) + return size; +} + +static int mux_aac_packet_to_adts (AVPacket *encoded_audio_packet, AVFormatContext *adts_container_ctx) +{ + int ret_val; + if ((ret_val = av_write_frame(adts_container_ctx, encoded_audio_packet)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Error calling av_write_frame() (error '%s')\n", av_err2str(ret_val)); + } + else { + av_log(NULL, AV_LOG_INFO, "Encoded AAC packet %d, size=%d, pts_time=%s\n", + encoded_pkt_counter, encoded_audio_packet->size, + av_ts2timestr(encoded_audio_packet->pts, &adts_container_ctx->streams[0]->time_base)); + } + return ret_val; +} + +static int check_if_samplerate_is_supported(AVCodec *audio_codec, int samplerate) +{ + const int *samplerates_list = audio_codec->supported_samplerates; + while (*samplerates_list) { + if (*samplerates_list == samplerate) + return 0; + samplerates_list++; + } + return 1; +} + +int main(int argc, char **argv) +{ + FILE *input_audio_file = NULL, *encoded_audio_file = NULL; + AVCodec *audio_codec = NULL; + AVCodecContext *audio_encoder_ctx = NULL; + AVFrame *input_audio_frame = NULL, *converted_audio_frame = NULL; + SwrContext *audio_convert_context = NULL; + AVOutputFormat *adts_container = NULL; + AVFormatContext *adts_container_ctx = NULL; + uint8_t *adts_container_buffer = NULL; + size_t adts_container_buffer_size = 4096; + AVIOContext *adts_avio_ctx = NULL; + AVStream *adts_stream = NULL; + AVPacket *encoded_audio_packet = NULL; + int ret_val = 0; + int audio_bytes_to_encode; + int64_t curr_pts; + + if (argc != 2) { + printf("Usage: %s \n", argv[0]); + return 1; + } + + input_audio_file = fopen(argv[1], "rb"); + if (!input_audio_file) { + av_log(NULL, AV_LOG_ERROR, "Could not open input audio file\n"); + return AVERROR_EXIT; + } + + encoded_audio_file = fopen("out.aac", "wb"); + if (!encoded_audio_file) { + av_log(NULL, AV_LOG_ERROR, "Could not open output audio file\n"); + fclose(input_audio_file); + return AVERROR_EXIT; + } + + av_register_all(); + + /** + * Allocate the encoder's context and open the encoder + */ + audio_codec = avcodec_find_encoder(AV_CODEC_ID_AAC); + if (!audio_codec) { + av_log(NULL, AV_LOG_ERROR, "Could not find aac codec\n"); + ret_val = AVERROR_EXIT; + goto end; + } + if ((ret_val = check_if_samplerate_is_supported(audio_codec, SAMPLE_RATE)) != 0) { + av_log(NULL, AV_LOG_ERROR, "Audio codec doesn't support input samplerate %d\n", SAMPLE_RATE); + goto end; + } + audio_encoder_ctx = avcodec_alloc_context3(audio_codec); + if (!audio_codec) { + av_log(NULL, AV_LOG_ERROR, "Could not allocate the encoding context\n"); + ret_val = AVERROR_EXIT; + goto end; + } + audio_encoder_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP; + audio_encoder_ctx->bit_rate = ENCODER_BITRATE; + audio_encoder_ctx->sample_rate = SAMPLE_RATE; + audio_encoder_ctx->channels = CHANNELS; + audio_encoder_ctx->channel_layout = av_get_default_channel_layout(CHANNELS); + audio_encoder_ctx->time_base = (AVRational){1, SAMPLE_RATE}; + audio_encoder_ctx->codec_type = AVMEDIA_TYPE_AUDIO ; + if ((ret_val = avcodec_open2(audio_encoder_ctx, audio_codec, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Could not open input codec (error '%s')\n", av_err2str(ret_val)); + goto end; + } + + /** + * Allocate an AVFrame which will be filled with the input file's data. + */ + if (!(input_audio_frame = av_frame_alloc())) { + av_log(NULL, AV_LOG_ERROR, "Could not allocate input frame\n"); + ret_val = AVERROR(ENOMEM); + goto end; + } + input_audio_frame->nb_samples = audio_encoder_ctx->frame_size; + input_audio_frame->format = INPUT_SAMPLE_FMT; + input_audio_frame->channels = CHANNELS; + input_audio_frame->sample_rate = SAMPLE_RATE; + input_audio_frame->channel_layout = av_get_default_channel_layout(CHANNELS); + // Allocate the frame's data buffer + if ((ret_val = av_frame_get_buffer(input_audio_frame, 0)) < 0) { + av_log(NULL, AV_LOG_ERROR, + "Could not allocate container for input frame samples (error '%s')\n", av_err2str(ret_val)); + ret_val = AVERROR(ENOMEM); + goto end; + } + + /** + * Input data must be converted to float-planar format, which is the format required by the AAC encoder. + * We allocate a SwrContext and an AVFrame (which will contain the converted samples) for this task. + * The AVFrame will feed the encoding function (avcodec_send_frame()) + */ + audio_convert_context = swr_alloc_set_opts(NULL, + av_get_default_channel_layout(CHANNELS), + AV_SAMPLE_FMT_FLTP, + SAMPLE_RATE, + av_get_default_channel_layout(CHANNELS), + INPUT_SAMPLE_FMT, + SAMPLE_RATE, + 0, + NULL); + if (!audio_convert_context) { + av_log(NULL, AV_LOG_ERROR, "Could not allocate resample context\n"); + ret_val = AVERROR(ENOMEM); + goto end; + } + if (!(converted_audio_frame = av_frame_alloc())) { + av_log(NULL, AV_LOG_ERROR, "Could not allocate resampled frame\n"); + ret_val = AVERROR(ENOMEM); + goto end; + } + converted_audio_frame->nb_samples = audio_encoder_ctx->frame_size; + converted_audio_frame->format = audio_encoder_ctx->sample_fmt; + converted_audio_frame->channels = audio_encoder_ctx->channels; + converted_audio_frame->channel_layout = audio_encoder_ctx->channel_layout; + converted_audio_frame->sample_rate = SAMPLE_RATE; + if ((ret_val = av_frame_get_buffer(converted_audio_frame, 0)) < 0) { + av_log(NULL, AV_LOG_ERROR, + "Could not allocate a buffer for resampled frame samples (error '%s')\n", av_err2str(ret_val)); + goto end; + } + + /** + * Create the ADTS container for the encoded frames + */ + adts_container = av_guess_format("adts", NULL, NULL); + if (!adts_container) { + av_log(NULL, AV_LOG_ERROR, "Could not find adts output format\n"); + ret_val = AVERROR_EXIT; + goto end; + } + if ((ret_val = avformat_alloc_output_context2(&adts_container_ctx, adts_container, "", NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Could not create output context (error '%s')\n", av_err2str(ret_val)); + goto end; + } + if (!(adts_container_buffer = av_malloc(adts_container_buffer_size))) { + av_log(NULL, AV_LOG_ERROR, "Could not allocate a buffer for the I/O output context\n"); + ret_val = AVERROR(ENOMEM); + goto end; + } + /** + * Create an I/O context for the adts container with a write callback (write_adts_muxed_data()), + * so that muxed data will be accessed through this function and can be managed by the user. + */ + if (!(adts_avio_ctx = avio_alloc_context(adts_container_buffer, adts_container_buffer_size, + 1, encoded_audio_file, NULL, + &write_adts_muxed_data, NULL))) { + av_log(NULL, AV_LOG_ERROR, "Could not create I/O output context\n"); + ret_val = AVERROR_EXIT; + goto end; + } + // Link the container's context to the previous I/O context + adts_container_ctx->pb = adts_avio_ctx; + if (!(adts_stream = avformat_new_stream(adts_container_ctx, NULL))) { + av_log(NULL, AV_LOG_ERROR, "Could not create new stream\n"); + ret_val = AVERROR(ENOMEM); + goto end; + } + adts_stream->id = adts_container_ctx->nb_streams-1; + // Copy the encoder's parameters + avcodec_parameters_from_context(adts_stream->codecpar, audio_encoder_ctx); + // Allocate the stream private data and write the stream header + if (avformat_write_header(adts_container_ctx, NULL) < 0) { + av_log(NULL, AV_LOG_ERROR, "avformat_write_header() error\n"); + ret_val = AVERROR_EXIT; + goto end; + } + + /** + * Fill the input frame's data buffer with input file data (a), + * Convert the input frame to float-planar format (b), + * Send the converted frame to the encoder (c), + * Get the encoded packet (d), + * Send the encoded packet to the adts muxer (e). + * Muxed data is caught in write_adts_muxed_data() callback and it is written + * to the output audio file ( (f) : see above) + */ + encoded_audio_packet = av_packet_alloc(); + while (1) { + + audio_bytes_to_encode = fread(input_audio_frame->data[0], 1, + input_audio_frame->linesize[0], input_audio_file); //(a) + if (audio_bytes_to_encode != input_audio_frame->linesize[0]) { + break; + } + else { + if (av_frame_make_writable(converted_audio_frame) < 0) { + av_log(NULL, AV_LOG_ERROR, "av_frame_make_writable() error\n"); + ret_val = AVERROR_EXIT; + goto end; + } + + if ((ret_val = swr_convert_frame(audio_convert_context, + converted_audio_frame, + (const AVFrame *)input_audio_frame)) != 0) { //(b) + av_log(NULL, AV_LOG_ERROR, + "Error resampling input audio frame (error '%s')\n", av_err2str(ret_val)); + goto end; + } + + if ((ret_val = avcodec_send_frame(audio_encoder_ctx, converted_audio_frame)) == 0) //(c) + ret_val = avcodec_receive_packet(audio_encoder_ctx, encoded_audio_packet); //(d) + else { + av_log(NULL, AV_LOG_ERROR, + "Error encoding frame (error '%s')\n", av_err2str(ret_val)); + goto end; + } + + if (ret_val == 0) { + curr_pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1); + encoded_audio_packet->pts = encoded_audio_packet->dts = curr_pts; + if ((ret_val = mux_aac_packet_to_adts(encoded_audio_packet, adts_container_ctx)) < 0) //(e) + goto end; + encoded_pkt_counter++; + } + else if (ret_val != AVERROR(EAGAIN)) { + av_log(NULL, AV_LOG_ERROR, + "Error receiving encoded packet (error '%s')\n", av_err2str(ret_val)); + goto end; + } + } + } + // Flush cached packets + if ((ret_val = avcodec_send_frame(audio_encoder_ctx, NULL)) == 0) + do { + ret_val = avcodec_receive_packet(audio_encoder_ctx, encoded_audio_packet); + if (ret_val == 0) { + curr_pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1); + encoded_audio_packet->pts = encoded_audio_packet->dts = curr_pts; + if ((ret_val = mux_aac_packet_to_adts(encoded_audio_packet, adts_container_ctx)) < 0) + goto end; + encoded_pkt_counter++; + } + } while (ret_val == 0); + + av_write_trailer(adts_container_ctx); + +end: + + fclose(input_audio_file); + fclose(encoded_audio_file); + avcodec_free_context(&audio_encoder_ctx); + av_frame_free(&input_audio_frame); + swr_free(&audio_convert_context); + av_frame_free(&converted_audio_frame); + avformat_free_context(adts_container_ctx); + av_freep(&adts_avio_ctx); + av_freep(&adts_container_buffer); + av_packet_free(&encoded_audio_packet); + + return ret_val; + +}