[FFmpeg-devel] [PATCH] new API usage example (adts-aac encoding from raw audio file, added av_frame_make_writable() check)
Paolo Prete
p4olo_prete at yahoo.it
Thu Mar 30 03:20:18 EEST 2017
---
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
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 at 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;
+
+}
--
2.9.3
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