[FFmpeg-devel,Libav-user] New libav API usage axamples

Submitted by Paolo Prete on March 27, 2017, 11:13 p.m.

Details

Message ID 647710073.8448733.1490656403410@mail.yahoo.com
State New
Headers show

Commit Message

Paolo Prete March 27, 2017, 11:13 p.m.
Il Lunedì 27 Marzo 2017 10:05, Carl Eugen Hoyos <ceffmpeg@gmail.com> ha scritto:


 

 2017-03-27 9:11 GMT+02:00 Gabor Alsecz <alseczg@gmail.com>:

> Thanks for the code snippet you have pasted here. Many of us
> struggling here because lack of samples even based on the latest API.
> I just can confirm to the Libav team we really would need new API
> examples and please try to figure out the way how Paolo can share
> his snippets commonly.

(Speaking for FFmpeg, not any forks)
If the snippets provide advantages over the code in doc/examples,
please send your patch (against current FFmpeg git head) made
with git format-patch to the FFmpeg development mailing list.
This is also where the feedback comes from.

Carl Eugen

Patch hide | download patch | download mbox

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

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..045df33
--- /dev/null
+++ b/doc/examples/encode_raw_audio_file_to_aac.c
@@ -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