From patchwork Wed Mar 29 23:30:27 2017 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 8bit X-Patchwork-Submitter: Paolo Prete X-Patchwork-Id: 3182 Delivered-To: ffmpegpatchwork@gmail.com Received: by 10.103.44.195 with SMTP id s186csp1312317vss; Wed, 29 Mar 2017 16:37:34 -0700 (PDT) X-Received: by 10.223.160.239 with SMTP id n44mr2705627wrn.198.1490830654499; Wed, 29 Mar 2017 16:37:34 -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 z65si658568wrc.101.2017.03.29.16.37.33; Wed, 29 Mar 2017 16:37:34 -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 4B6B9689820; Thu, 30 Mar 2017 02:37:08 +0300 (EEST) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from sonic306-21.consmr.mail.ir2.yahoo.com (sonic306-21.consmr.mail.ir2.yahoo.com [77.238.176.207]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 1387268921C for ; Thu, 30 Mar 2017 02:37:01 +0300 (EEST) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=yahoo.it; s=s2048; t=1490830644; bh=IonehG0B1ahTw9Bl3WSbeccGKo36M+ePiQ3ZBqCbFRE=; h=Date:From:Reply-To:To:In-Reply-To:References:Subject:From:Subject; b=D6Xd3eZhDKO39rOHkOV8mF33RIl1IbQHTlRLuaAMKmbXRRVu4JwA9luKEzKmlczp7ckPlUOjRep0lIeQm+Dky2IBItExUOsgdb7PRiP8aK/CJ2Mo7sdT8yHEf1pHEvD2kQx7i/jMURa+6MHNdRSLrDGAHPKh7nOlCqpbdJsih08/vjd2I7ZH0uct3P0QJ3RSuMFFHCuzQbdOzLKBdasfe9hr0KTl/l/f2g9gtajqfJIkOonhlTOCZfSEgM6VBKvQ94w744AkMhymnQipCJbO2G22Kr8QITuT8uauVr5QhnfJt2TxTVn2quPsc0pSd1f3xkuyECFuCFcsKERyf2vXvQ== X-YMail-OSG: SCq_kdwVM1kmdxF0JliHR0a8zfguS34ODgvO3w8TLO0s2sASsILlqlr8mWrXCSf Wgr9h5Y04b_bDHG0G1AYaNmuN7DaUD.t9xzz3dYR10XsQ1KzM0.O165zXhz1uAi4IJ_mc3io__Dd NpX5XeCk12fQ2aq22vBzk3ypUBjpDrfk8WTkOe.a8GbO7NkOFv6bBY.N.a6YKnm.OxPa.yg7weJ6 ODHEwNsULg7BZtEG8CCFnCKcsiiEZzp0WJ_czXnN0gZmLArpVEGyF9xJwuWI.5_mO0NA7ilOJwCv Xh4.2F5Kxi4bXXYQ_3DgKWUxEiJ0ICB5at0pBKdnNPQIeZPs3ja_ypguih8HG.CNeS2LS40w7GdZ mtm2AQIgeQx765zpI7qsOkqHeK3To7sYyxnUKSHLJE5kM1coyc0WS7gzq09yGwC9vmKzy389vcKS met4sTDfPHrm0RWMkyUHCyj1lV_HB8ehXt91KqrXLbK.Iy7s.D1f8HfF3ubJyOcBbSwETBbF0Yb0 K8byknfDAkWxWeCyfjSCHyKH45aD8NA-- Received: from sonic.gate.mail.ne1.yahoo.com by sonic306.consmr.mail.ir2.yahoo.com with HTTP; Wed, 29 Mar 2017 23:37:24 +0000 Date: Wed, 29 Mar 2017 23:30:27 +0000 (UTC) From: Paolo Prete To: FFmpeg development discussions and patches Message-ID: <1579110658.11263984.1490830227554@mail.yahoo.com> In-Reply-To: <20170329225956.13930-1-p4olo_prete@yahoo.it> References: <1898638235.9769159.1490827955583@mail.yahoo.com> <20170329225956.13930-1-p4olo_prete@yahoo.it> MIME-Version: 1.0 X-Content-Filtered-By: Mailman/MimeDel 2.1.20 Subject: Re: [FFmpeg-devel] [PATCH] new API usage example (adts-aac encoding from raw audio file) 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 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" Sorry, the previous patch contains few typo errors. See the next one. Il Giovedì 30 Marzo 2017 1:00, Paolo Prete ha scritto: --- doc/examples/Makefile                      |  1 + doc/examples/encode_raw_audio_file_to_aac.c | 300 ++++++++++++++++++++++++++++ 2 files changed, 301 insertions(+) create mode 100644 doc/examples/encode_raw_audio_file_to_aac.c diff --git a/doc/examples/Makefile b/doc/examples/Makefile index af38159..81181c7 100644 --- a/doc/examples/Makefile +++ b/doc/examples/Makefile @@ -15,6 +15,7 @@ EXAMPLES=      avio_dir_cmd                      \                 avio_reading                      \                 decoding_encoding                  \                 demuxing_decoding                  \ +                encode_raw_audio_file_to_aac      \                 extract_mvs                        \                 filtering_video                    \                 filtering_audio                    \ 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..546e713 --- /dev/null +++ b/doc/examples/encode_raw_audio_file_to_aac.c @@ -0,0 +1,300 @@ +/* + * 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" + * 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. +    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 ((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; +    +}