From 3d7afbccf35186890b5b8151342cb17dfc9d38e4 Mon Sep 17 00:00:00 2001
From: Paolo Prete <p4olo_prete@yahoo.it>
Date: Mon, 27 Mar 2017 23:07:12 +0200
Subject: [PATCH] new API usage example
---
doc/examples/encode_raw_audio_file_to_aac.c | 326 ++++++++++++++++++++++++++++
1 file changed, 326 insertions(+)
create mode 100644 doc/examples/encode_raw_audio_file_to_aac.c
new file mode 100644
@@ -0,0 +1,326 @@
+/*
+ * 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 <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavutil/timestamp.h>
+#include <libswresample/swresample.h>
+
+
+#define ENCODER_BITRATE 64000
+#define SAMPLE_RATE 16000
+#define INPUT_SAMPLE_FMT AV_SAMPLE_FMT_FLT
+#define CHANNELS 2
+
+
+static char *const get_error_text(const int error)
+{
+ static char error_buffer[255];
+ av_strerror(error, error_buffer, sizeof(error_buffer));
+ return error_buffer;
+}
+
+
+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;
+}
+
+
+int main(int argc, char **argv)
+{
+
+
+ if (argc != 2) {
+ av_log(NULL, AV_LOG_ERROR, "Usage: %s <raw audio input file (CHANNELS, INPUT_SAMPLE_FMT, SAMPLE_RATE)>\n", argv[0]);
+ return 1;
+ }
+
+
+ int ret_val = 0;
+ int cleanup_step = 1;
+
+
+
+ FILE *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;
+ }
+
+ FILE *encoded_audio_file = fopen("out.aac", "wb");
+ if(!encoded_audio_file){
+ av_log(NULL, AV_LOG_ERROR, "Could not open output audio file\n");
+ ret_val = AVERROR_EXIT;
+ goto cleanup;
+ }
+ ++cleanup_step;
+
+
+
+ av_register_all();
+
+
+
+ //
+ // Allocate the encoder's context and open the encoder
+ //
+ AVCodec *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 cleanup;
+ }
+ AVCodecContext *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 cleanup;
+ }
+ ++cleanup_step;
+ audio_encoder_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+ audio_encoder_ctx->bit_rate = ENCODER_BITRATE;
+ audio_encoder_ctx->sample_rate = SAMPLE_RATE; // You can use any other sample rate provided by the input file on condition that it is supported by the codec (use AVCodec::supported_samplerates for listing supported sample rates)
+ 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", get_error_text(ret_val));
+ goto cleanup;
+ }
+ ++cleanup_step;
+
+
+ //
+ // Allocate an AVFrame which will be filled with the input file's data.
+ //
+ AVFrame *input_audio_frame;
+ if (!(input_audio_frame = av_frame_alloc())) {
+ av_log(NULL, AV_LOG_ERROR, "Could not allocate input frame\n");
+ ret_val = AVERROR(ENOMEM);
+ goto cleanup;
+ }
+ 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", get_error_text(ret_val));
+ ret_val = AVERROR(ENOMEM);
+ goto cleanup;
+ }
+
+
+
+ //
+ // 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())
+ //
+ SwrContext *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 cleanup;
+ }
+ ++cleanup_step;
+ AVFrame *converted_audio_frame;
+ if (!(converted_audio_frame = av_frame_alloc())) {
+ av_log(NULL, AV_LOG_ERROR, "Could not allocate resampled frame\n");
+ ret_val = AVERROR(ENOMEM);
+ goto cleanup;
+ }
+ ++cleanup_step;
+ 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", get_error_text(ret_val));
+ goto cleanup;
+ }
+
+
+
+ //
+ // Create the ADTS container for the encoded frames
+ //
+ AVOutputFormat *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 cleanup;
+ }
+ AVFormatContext *adts_container_ctx;
+ 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", get_error_text(ret_val));
+ goto cleanup;
+ }
+ ++cleanup_step;
+ size_t adts_container_buffer_size = 4096;
+ uint8_t *adts_container_buffer;
+ if(!(adts_container_buffer = (uint8_t* )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 cleanup;
+ }
+ ++cleanup_step;
+ // 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.
+ AVIOContext *adts_avio_ctx;
+ 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 cleanup;
+ }
+ ++cleanup_step;
+ // Link the container's context to the previous I/O context
+ adts_container_ctx->pb = adts_avio_ctx;
+ AVStream *adts_stream;
+ 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 cleanup;
+ }
+ 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 cleanup;
+ }
+ ++cleanup_step;
+
+
+
+ //
+ // 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)
+ //
+ AVPacket encoded_audio_packet;
+ av_init_packet(&encoded_audio_packet);
+ int encoded_pkt_counter = 1;
+ while(1) {
+ int audio_bytes_to_encode = fread(input_audio_frame->data[0], 1, input_audio_frame->linesize[0], input_audio_file); //(a)
+ swr_convert_frame(audio_convert_context, converted_audio_frame, (const AVFrame *)input_audio_frame); //(b)
+ if(audio_bytes_to_encode != input_audio_frame->linesize[0]){
+ break;
+ }
+ else {
+ // Do encode
+ ret_val = avcodec_send_frame(audio_encoder_ctx, converted_audio_frame); //(c)
+ if(ret_val == 0)
+ 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", get_error_text(ret_val));
+ goto cleanup;
+ }
+
+ if(ret_val == 0){
+ int64_t pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1);
+ encoded_audio_packet.pts = encoded_audio_packet.dts = pts;
+ if((ret_val == av_write_frame(adts_container_ctx, &encoded_audio_packet)) < 0){ //(e)
+ av_log(NULL, AV_LOG_ERROR, "Error calling av_write_frame() (error '%s')\n", get_error_text(ret_val));
+ goto cleanup;
+ }
+ 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, &audio_encoder_ctx->time_base));
+ ++encoded_pkt_counter;
+ }
+ }
+ }
+ }
+ // Flush delayed packets
+ int still_pkts_to_flush = 1;
+ int delayed_pkt_counter = 1;
+ while(still_pkts_to_flush){
+ int ret = avcodec_send_frame(audio_encoder_ctx, NULL);
+ if(ret != 0)
+ still_pkts_to_flush = 0;
+ ret = avcodec_receive_packet(audio_encoder_ctx, &encoded_audio_packet);
+ if(ret == 0){
+ int64_t pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1);
+ encoded_audio_packet.pts = encoded_audio_packet.dts = pts;
+ av_write_frame(adts_container_ctx, &encoded_audio_packet);
+ av_log(NULL, AV_LOG_INFO, "Flushed encoded AAC delayed packet %d, size=%d, pts_time=%s\n", delayed_pkt_counter, encoded_audio_packet.size, av_ts2timestr(encoded_audio_packet.pts, &audio_encoder_ctx->time_base));
+ ++delayed_pkt_counter;
+ ++encoded_pkt_counter;
+ }
+ }
+
+
+ av_write_trailer(adts_container_ctx);
+
+
+
+
+cleanup:
+
+
+ if(cleanup_step > 0)
+ fclose(input_audio_file);
+ if(cleanup_step > 1)
+ fclose(encoded_audio_file);
+ if(cleanup_step > 2)
+ avcodec_free_context(&audio_encoder_ctx);
+ if(cleanup_step > 3)
+ av_frame_free(&input_audio_frame);
+ if(cleanup_step > 4)
+ swr_free(&audio_convert_context);
+ if(cleanup_step > 5)
+ av_frame_free(&converted_audio_frame);
+ if(cleanup_step > 6)
+ avformat_free_context(adts_container_ctx);
+ if(cleanup_step > 7)
+ av_free(adts_container_buffer);
+ if(cleanup_step > 8)
+ av_free(adts_avio_ctx);
+ if(cleanup_step > 9)
+ av_packet_unref(&encoded_audio_packet);
+
+
+ return ret_val;
+
+}
+
+
+
--
2.9.3