[FFmpeg-devel] new API usage example (adts-aac encoding from raw audio file, added av_frame_make_writable() check)

Submitted by Paolo Prete on March 30, 2017, 12:20 a.m.

Details

Message ID 20170330002018.27917-1-p4olo_prete@yahoo.it
State New
Headers show

Commit Message

Paolo Prete March 30, 2017, 12:20 a.m.
---
 doc/examples/Makefile                       |   1 +
 doc/examples/encode_raw_audio_file_to_aac.c | 307 ++++++++++++++++++++++++++++
 2 files changed, 308 insertions(+)
 create mode 100644 doc/examples/encode_raw_audio_file_to_aac.c

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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..140aff6
--- /dev/null
+++ b/doc/examples/encode_raw_audio_file_to_aac.c
@@ -0,0 +1,307 @@ 
+/*
+ * 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 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 <raw audio input file (CHANNELS, INPUT_SAMPLE_FMT, SAMPLE_RATE)>\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;
+    
+}