From: Paolo Prete <p4olo_prete-at-yahoo.it@ffmpeg.org>
To: FFmpeg Development Discussions and Patches <ffmpeg-devel@ffmpeg.org>
Subject: [FFmpeg-devel] [PATCH] New API usage example (reading, converting, encoding and muxing an audio file)
Date: Mon, 6 Jun 2022 17:40:42 +0000 (UTC)
Message-ID: <580594123.12475702.1654537242853@mail.yahoo.com> (raw)
In-Reply-To: <580594123.12475702.1654537242853.ref@mail.yahoo.com>
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Hello,
From what I see, the doc/examples files can be still improved. More precisely, their main() is often split into functions with short names that hide important (sometime essential) details of what they're doing and they worsen the readability, by forcing the user to jump from chunks to chunks of the code. See for example: muxing.c (and the open_audio/video(), write_audio/video_frame()...). In addition: they are hard to adapt to practical cases without doing big modifications.
Please consider to review/push the attached example: convert_encode_mux_audio.c .
It is deliberately written in a more "procedural" way, it should be much easier to read and it covers all the basic steps (reading from file, converting, encoding, muxing) of a common and practical pipeline made with ffMPEG.
Hope it helps.
P
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From 86db1c6dcd6f69028c9c25d03e6e19ef9fc2b331 Mon Sep 17 00:00:00 2001
From: paolo <paolopr976@gmail.com>
Date: Mon, 6 Jun 2022 15:34:05 +0200
Subject: [PATCH] Added a complete example - with clear, well readable and
sequential code - for doing basic operations on a raw audio file. It improves
the current state of the examples directory
---
configure | 2 +
doc/examples/Makefile | 47 ++--
doc/examples/Makefile.example | 1 +
doc/examples/convert_encode_mux_audio.c | 351 ++++++++++++++++++++++++
4 files changed, 378 insertions(+), 23 deletions(-)
create mode 100644 doc/examples/convert_encode_mux_audio.c
diff --git a/configure b/configure
index 5a167613a4..775750aff0 100755
--- a/configure
+++ b/configure
@@ -1725,6 +1725,7 @@ COMPONENT_LIST="
EXAMPLE_LIST="
avio_list_dir_example
avio_reading_example
+ convert_encode_mux_audio_example
decode_audio_example
decode_video_example
demuxing_decoding_example
@@ -3760,6 +3761,7 @@ yadif_videotoolbox_filter_deps="metal corevideo videotoolbox"
# examples
avio_list_dir_deps="avformat avutil"
avio_reading_deps="avformat avcodec avutil"
+convert_encode_mux_audio_example_deps="avcodec avformat avutil swresample"
decode_audio_example_deps="avcodec avutil"
decode_video_example_deps="avcodec avutil"
demuxing_decoding_example_deps="avcodec avformat avutil"
diff --git a/doc/examples/Makefile b/doc/examples/Makefile
index 81bfd34d5d..aed73e811c 100644
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -1,26 +1,27 @@
-EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
-EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
-EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
-EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
-EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
-EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video
-EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
-EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
-EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
-EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
-EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
-EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE) += hw_decode
-EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
-EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
-EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
-EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
-EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
-EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
-EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
-EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
-EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
-EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE) += vaapi_transcode
+EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
+EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
+EXAMPLES-$(CONFIG_CONVERT_ENCODE_MUX_AUDIO_EXAMPLE) += convert_encode_mux_audio
+EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
+EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
+EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
+EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video
+EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
+EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
+EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
+EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
+EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
+EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE) += hw_decode
+EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
+EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
+EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
+EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
+EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
+EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
+EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
+EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
+EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
+EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE) += vaapi_transcode
EXAMPLES := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
EXAMPLES_G := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
diff --git a/doc/examples/Makefile.example b/doc/examples/Makefile.example
index a232d97f98..f2438caeb8 100644
--- a/doc/examples/Makefile.example
+++ b/doc/examples/Makefile.example
@@ -13,6 +13,7 @@ LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_list_dir \
avio_reading \
+ convert_encode_mux_audio \
decode_audio \
decode_video \
demuxing_decoding \
diff --git a/doc/examples/convert_encode_mux_audio.c b/doc/examples/convert_encode_mux_audio.c
new file mode 100644
index 0000000000..abde611bf6
--- /dev/null
+++ b/doc/examples/convert_encode_mux_audio.c
@@ -0,0 +1,351 @@
+/*
+ * Copyright (c) 2019 Paolo Prete (paolopr976 at gmail.com)
+ *
+ * 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 converting, encoding and muxing raw audio files.
+ * This example reads a raw audio input file, converts it to a sample format accepted by
+ * a given encoder, performs encoding and puts the encoded frames into a muxing container.
+ * The encoded stream is written to a file named "outfile".
+ * 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_muxed_data()) in order to show 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 48000 raw_audio_file.raw
+ *
+ * @example convert_encode_mux_audio.c
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavutil/timestamp.h>
+#include <libswresample/swresample.h>
+
+// Customizable parameters
+#define SAMPLE_RATE 48000
+#define ENCODER_ID AV_CODEC_ID_AAC
+#define ENCODER_BITRATE 64000
+#define INPUT_SAMPLE_FMT AV_SAMPLE_FMT_FLT
+#define OUTPUT_SAMPLE_FMT AV_SAMPLE_FMT_FLTP
+#define INPUT_CHANNELS 2
+#define OUTPUT_CHANNELS 2
+
+static int encoded_pkt_counter = 1;
+
+static int write_muxed_data(void *opaque, uint8_t *data, int size)
+{
+ FILE *encoded_audio_file = (FILE *)opaque;
+ fwrite(data, 1, size, encoded_audio_file); //(h)
+ return size;
+}
+
+static int mux_encoded_packet(AVPacket *encoded_audio_packet, AVFormatContext *out_container_ctx)
+{
+ int ret_val;
+ if ((ret_val = av_write_frame(out_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 packet %d, size=%d, pts_time=%s\n",
+ encoded_pkt_counter, encoded_audio_packet->size,
+ av_ts2timestr(encoded_audio_packet->pts, &out_container_ctx->streams[0]->time_base));
+ }
+ return ret_val;
+}
+
+static int check_if_samplerate_is_supported(const 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;
+ AVChannelLayout in_ch_layout;
+ AVChannelLayout out_ch_layout;
+ const AVCodec *audio_codec = NULL;
+ AVCodecContext *audio_encoder_ctx = NULL;
+ AVFrame *input_audio_frame = NULL, *converted_audio_frame = NULL;
+ SwrContext *audio_convert_context = NULL;
+ const AVOutputFormat *out_container = NULL;
+ AVFormatContext *out_container_ctx = NULL;
+ uint8_t *out_container_buffer = NULL;
+ size_t out_container_buffer_size = 4096;
+ AVIOContext *avio_ctx = NULL;
+ AVStream *out_stream = NULL;
+ AVPacket *encoded_audio_packet = NULL;
+ int ret_val = 0;
+ int audio_bytes_to_encode;
+
+ if (argc != 3) {
+ printf("Usage: %s <raw audio input file (INPUT_CHANNELS, INPUT_SAMPLE_FMT, SAMPLE_RATE)>, <container format>\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("outfile", "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;
+ }
+
+ /**
+ * Allocate the encoder's context and open the encoder
+ */
+ audio_codec = avcodec_find_encoder(ENCODER_ID);
+ if (!audio_codec) {
+ av_log(NULL, AV_LOG_ERROR, "Could not find encoder's 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;
+ }
+ av_channel_layout_default(&out_ch_layout, OUTPUT_CHANNELS);
+ audio_encoder_ctx->sample_fmt = OUTPUT_SAMPLE_FMT;
+ audio_encoder_ctx->bit_rate = ENCODER_BITRATE;
+ audio_encoder_ctx->sample_rate = SAMPLE_RATE;
+ audio_encoder_ctx->ch_layout = out_ch_layout;
+ 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;
+ }
+ av_channel_layout_default(&in_ch_layout, INPUT_CHANNELS);
+ input_audio_frame->nb_samples = audio_encoder_ctx->frame_size;
+ input_audio_frame->format = INPUT_SAMPLE_FMT;
+ input_audio_frame->sample_rate = SAMPLE_RATE;
+ input_audio_frame->ch_layout = in_ch_layout;
+ // 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 in the right format required by the 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())
+ */
+ swr_alloc_set_opts2(&audio_convert_context,
+ &out_ch_layout,
+ OUTPUT_SAMPLE_FMT,
+ SAMPLE_RATE,
+ &in_ch_layout,
+ 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->ch_layout = audio_encoder_ctx->ch_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 output container for the encoded frames
+ */
+ out_container = av_guess_format(argv[2], NULL, NULL);
+ if (!out_container) {
+ av_log(NULL, AV_LOG_ERROR, "Could not find output format\n");
+ ret_val = AVERROR_EXIT;
+ goto end;
+ }
+ if ((ret_val = avformat_alloc_output_context2(&out_container_ctx, out_container, "", NULL)) < 0) {
+ av_log(NULL, AV_LOG_ERROR, "Could not create output context (error '%s')\n", av_err2str(ret_val));
+ goto end;
+ }
+ if (!(out_container_buffer = av_malloc(out_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 muxer's container with a write callback (write_muxed_data()),
+ * so that muxed data will be accessed through this function and can be managed by the user.
+ */
+ if (!(avio_ctx = avio_alloc_context(out_container_buffer, out_container_buffer_size,
+ 1, encoded_audio_file, NULL,
+ &write_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
+ */
+ out_container_ctx->pb = avio_ctx;
+ if (!(out_stream = avformat_new_stream(out_container_ctx, NULL))) {
+ av_log(NULL, AV_LOG_ERROR, "Could not create new stream\n");
+ ret_val = AVERROR(ENOMEM);
+ goto end;
+ }
+ out_stream->id = out_container_ctx->nb_streams-1;
+
+ // Copy the encoder's parameters
+ avcodec_parameters_from_context(out_stream->codecpar, audio_encoder_ctx);
+ // Allocate the stream private data and write the stream header
+ if (avformat_write_header(out_container_ctx, NULL) < 0) {
+ av_log(NULL, AV_LOG_ERROR, "avformat_write_header() error\n");
+ ret_val = AVERROR_EXIT;
+ goto end;
+ }
+
+ /**
+ * (a) Fill the input frame's data buffer with input file data
+ * (b) Convert the input frame to output sample format
+ * (c) Assign a pts to the converted frame based on the number ot total samples read
+ * (d) Send the converted frame to the encoder
+ * (e) Get the encoded packet
+ * (f) Rescale the timestamps of the encoded packet to the muxer's timebase
+ * (g) Send the encoded packet, with rescaled timestamps, to the muxer
+ * (h) Muxed data is caught in write_muxed_data() callback (see above) and it is
+ * written to the output audio file
+ */
+ 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;
+ }
+
+ converted_audio_frame->pts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1); //(c)
+
+ if ((ret_val = avcodec_send_frame(audio_encoder_ctx, converted_audio_frame)) == 0) //(d)
+ ret_val = avcodec_receive_packet(audio_encoder_ctx, encoded_audio_packet); //(e)
+ else {
+ av_log(NULL, AV_LOG_ERROR,
+ "Error encoding frame (error '%s')\n", av_err2str(ret_val));
+ goto end;
+ }
+
+ if (ret_val == 0) {
+ encoded_audio_packet->dts = encoded_audio_packet->pts = converted_audio_frame->pts;
+ av_packet_rescale_ts(encoded_audio_packet, audio_encoder_ctx->time_base,
+ out_stream->time_base); //(f)
+ if ((ret_val = mux_encoded_packet(encoded_audio_packet, out_container_ctx)) < 0) //(g)
+ 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) {
+ encoded_audio_packet->dts = converted_audio_frame->nb_samples*(encoded_pkt_counter-1);
+ encoded_audio_packet->pts = encoded_audio_packet->dts;
+ av_packet_rescale_ts(encoded_audio_packet, audio_encoder_ctx->time_base,
+ out_stream->time_base);
+ if ((ret_val = mux_encoded_packet(encoded_audio_packet, out_container_ctx)) < 0)
+ goto end;
+ encoded_pkt_counter++;
+ }
+ } while (ret_val == 0);
+
+ av_write_trailer(out_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(out_container_ctx);
+ av_freep(&avio_ctx);
+ av_freep(&out_container_buffer);
+ av_packet_free(&encoded_audio_packet);
+
+ return ret_val;
+
+}
--
2.32.0
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next parent reply other threads:[~2022-06-06 17:36 UTC|newest]
Thread overview: 5+ messages / expand[flat|nested] mbox.gz Atom feed top
[not found] <580594123.12475702.1654537242853.ref@mail.yahoo.com>
2022-06-06 17:40 ` Paolo Prete [this message]
2022-06-07 9:42 ` Anton Khirnov
2022-06-07 10:59 ` Paolo Prete
2022-06-07 11:56 ` Anton Khirnov
2022-06-07 12:37 ` Paolo Prete
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