[FFmpeg-devel] [PATCH] New API usage example (encode_raw_audio_file_to_aac)
Paolo Prete
p4olo_prete at yahoo.it
Fri Mar 31 16:11:11 EEST 2017
---
configure | 2 +
doc/Makefile | 41 ++--
doc/examples/.gitignore | 1 +
doc/examples/Makefile | 1 +
doc/examples/encode_raw_audio_file_to_aac.c | 338 ++++++++++++++++++++++++++++
5 files changed, 363 insertions(+), 20 deletions(-)
create mode 100644 doc/examples/encode_raw_audio_file_to_aac.c
diff --git a/configure b/configure
index 6d76cf7..1069f9f 100755
--- a/configure
+++ b/configure
@@ -1466,6 +1466,7 @@ EXAMPLE_LIST="
decode_video_example
demuxing_decoding_example
encode_audio_example
+ encode_raw_audio_file_to_aac_example
encode_video_example
extract_mvs_example
filter_audio_example
@@ -3175,6 +3176,7 @@ decode_audio_example_deps="avcodec avutil"
decode_video_example_deps="avcodec avutil"
demuxing_decoding_example_deps="avcodec avformat avutil"
encode_audio_example_deps="avcodec avutil"
+encode_raw_audio_file_to_aac_example_deps="avcodec avformat avutil swresample"
encode_video_example_deps="avcodec avutil"
extract_mvs_example_deps="avcodec avformat avutil"
filter_audio_example_deps="avfilter avutil"
diff --git a/doc/Makefile b/doc/Makefile
index c193fc3..a7c349b 100644
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -36,26 +36,27 @@ DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
-DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd
-DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
-DOC_EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
-DOC_EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
-DOC_EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video
-DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
-DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
-DOC_EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
-DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
-DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
-DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
-DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
-DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
-DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
-DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
-DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
+DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd
+DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
+DOC_EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
+DOC_EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
+DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
+DOC_EXAMPLES-$(CONFIG_ENCODE_RAW_AUDIO_FILE_TO_AAC_EXAMPLE) += encode_raw_audio_file_to_aac
+DOC_EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video
+DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
+DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
+DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
+DOC_EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
+DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
+DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
+DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
+DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
+DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
+DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
+DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
+DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
diff --git a/doc/examples/.gitignore b/doc/examples/.gitignore
index 6bd9dc1..7b25718 100644
--- a/doc/examples/.gitignore
+++ b/doc/examples/.gitignore
@@ -4,6 +4,7 @@
/decode_video
/demuxing_decoding
/encode_audio
+/encode_raw_audio_file_to_aac
/encode_video
/extract_mvs
/filter_audio
diff --git a/doc/examples/Makefile b/doc/examples/Makefile
index 2d0a306..24929e3 100644
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -17,6 +17,7 @@ EXAMPLES= avio_dir_cmd \
decode_video \
demuxing_decoding \
encode_audio \
+ encode_raw_audio_file_to_aac \
encode_video \
extract_mvs \
filtering_video \
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..5e6df5c
--- /dev/null
+++ b/doc/examples/encode_raw_audio_file_to_aac.c
@@ -0,0 +1,338 @@
+/*
+ * 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"
+ * It can be adapted, with few changes, to a custom raw audio source (i.e: a live one).
+ * It uses a custom I/O write callback (write_adts_muxed_data()) in order to show to the user
+ * how to access muxed packets written in memory, before they are written to the output file.
+ * 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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