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https://github.com/FFmpeg/FFmpeg.git
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72f5a6d067
Adjust the code so that a working ffplay command is printed in the planar audio case.
342 lines
12 KiB
C
342 lines
12 KiB
C
/*
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* Copyright (c) 2012 Stefano Sabatini
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/**
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* @file
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* libavformat demuxing API use example.
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*
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* Show how to use the libavformat and libavcodec API to demux and
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* decode audio and video data.
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* @example doc/examples/demuxing.c
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*/
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#include <libavutil/imgutils.h>
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#include <libavutil/samplefmt.h>
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#include <libavutil/timestamp.h>
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#include <libavformat/avformat.h>
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static AVFormatContext *fmt_ctx = NULL;
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static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
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static AVStream *video_stream = NULL, *audio_stream = NULL;
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static const char *src_filename = NULL;
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static const char *video_dst_filename = NULL;
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static const char *audio_dst_filename = NULL;
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static FILE *video_dst_file = NULL;
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static FILE *audio_dst_file = NULL;
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static uint8_t *video_dst_data[4] = {NULL};
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static int video_dst_linesize[4];
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static int video_dst_bufsize;
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static int video_stream_idx = -1, audio_stream_idx = -1;
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static AVFrame *frame = NULL;
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static AVPacket pkt;
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static int video_frame_count = 0;
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static int audio_frame_count = 0;
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static int decode_packet(int *got_frame, int cached)
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{
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int ret = 0;
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int decoded = pkt.size;
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if (pkt.stream_index == video_stream_idx) {
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/* decode video frame */
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ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
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if (ret < 0) {
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fprintf(stderr, "Error decoding video frame\n");
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return ret;
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}
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if (*got_frame) {
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printf("video_frame%s n:%d coded_n:%d pts:%s\n",
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cached ? "(cached)" : "",
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video_frame_count++, frame->coded_picture_number,
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av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
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/* copy decoded frame to destination buffer:
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* this is required since rawvideo expects non aligned data */
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av_image_copy(video_dst_data, video_dst_linesize,
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(const uint8_t **)(frame->data), frame->linesize,
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video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
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/* write to rawvideo file */
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fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
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}
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} else if (pkt.stream_index == audio_stream_idx) {
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/* decode audio frame */
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ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
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if (ret < 0) {
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fprintf(stderr, "Error decoding audio frame\n");
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return ret;
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}
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/* Some audio decoders decode only part of the packet, and have to be
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* called again with the remainder of the packet data.
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* Sample: fate-suite/lossless-audio/luckynight-partial.shn
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* Also, some decoders might over-read the packet. */
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decoded = FFMIN(ret, pkt.size);
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if (*got_frame) {
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size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
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printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
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cached ? "(cached)" : "",
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audio_frame_count++, frame->nb_samples,
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av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
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/* Write the raw audio data samples of the first plane. This works
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* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
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* most audio decoders output planar audio, which uses a separate
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* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
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* In other words, this code will write only the first audio channel
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* in these cases.
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* You should use libswresample or libavfilter to convert the frame
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* to packed data. */
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fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
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}
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}
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return decoded;
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}
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static int open_codec_context(int *stream_idx,
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AVFormatContext *fmt_ctx, enum AVMediaType type)
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{
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int ret;
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AVStream *st;
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AVCodecContext *dec_ctx = NULL;
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AVCodec *dec = NULL;
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ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not find %s stream in input file '%s'\n",
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av_get_media_type_string(type), src_filename);
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return ret;
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} else {
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*stream_idx = ret;
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st = fmt_ctx->streams[*stream_idx];
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/* find decoder for the stream */
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dec_ctx = st->codec;
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dec = avcodec_find_decoder(dec_ctx->codec_id);
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if (!dec) {
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fprintf(stderr, "Failed to find %s codec\n",
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av_get_media_type_string(type));
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return ret;
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}
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if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
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fprintf(stderr, "Failed to open %s codec\n",
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av_get_media_type_string(type));
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return ret;
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}
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}
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return 0;
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}
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static int get_format_from_sample_fmt(const char **fmt,
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enum AVSampleFormat sample_fmt)
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{
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int i;
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struct sample_fmt_entry {
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enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
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} sample_fmt_entries[] = {
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{ AV_SAMPLE_FMT_U8, "u8", "u8" },
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{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
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{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
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{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
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{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
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};
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*fmt = NULL;
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for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
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struct sample_fmt_entry *entry = &sample_fmt_entries[i];
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if (sample_fmt == entry->sample_fmt) {
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*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
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return 0;
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}
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}
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fprintf(stderr,
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"sample format %s is not supported as output format\n",
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av_get_sample_fmt_name(sample_fmt));
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return -1;
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}
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int main (int argc, char **argv)
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{
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int ret = 0, got_frame;
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if (argc != 4) {
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fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
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"API example program to show how to read frames from an input file.\n"
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"This program reads frames from a file, decodes them, and writes decoded\n"
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"video frames to a rawvideo file named video_output_file, and decoded\n"
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"audio frames to a rawaudio file named audio_output_file.\n"
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"\n", argv[0]);
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exit(1);
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}
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src_filename = argv[1];
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video_dst_filename = argv[2];
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audio_dst_filename = argv[3];
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/* register all formats and codecs */
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av_register_all();
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/* open input file, and allocate format context */
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if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
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fprintf(stderr, "Could not open source file %s\n", src_filename);
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exit(1);
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}
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/* retrieve stream information */
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if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
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fprintf(stderr, "Could not find stream information\n");
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exit(1);
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}
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if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
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video_stream = fmt_ctx->streams[video_stream_idx];
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video_dec_ctx = video_stream->codec;
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video_dst_file = fopen(video_dst_filename, "wb");
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if (!video_dst_file) {
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fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
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ret = 1;
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goto end;
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}
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/* allocate image where the decoded image will be put */
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ret = av_image_alloc(video_dst_data, video_dst_linesize,
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video_dec_ctx->width, video_dec_ctx->height,
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video_dec_ctx->pix_fmt, 1);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate raw video buffer\n");
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goto end;
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}
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video_dst_bufsize = ret;
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}
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if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
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audio_stream = fmt_ctx->streams[audio_stream_idx];
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audio_dec_ctx = audio_stream->codec;
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audio_dst_file = fopen(audio_dst_filename, "wb");
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if (!audio_dst_file) {
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fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
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ret = 1;
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goto end;
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}
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}
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/* dump input information to stderr */
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av_dump_format(fmt_ctx, 0, src_filename, 0);
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if (!audio_stream && !video_stream) {
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fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
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ret = 1;
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goto end;
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}
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frame = avcodec_alloc_frame();
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if (!frame) {
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fprintf(stderr, "Could not allocate frame\n");
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ret = AVERROR(ENOMEM);
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goto end;
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}
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/* initialize packet, set data to NULL, let the demuxer fill it */
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av_init_packet(&pkt);
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pkt.data = NULL;
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pkt.size = 0;
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if (video_stream)
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printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
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if (audio_stream)
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printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
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/* read frames from the file */
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while (av_read_frame(fmt_ctx, &pkt) >= 0) {
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AVPacket orig_pkt = pkt;
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do {
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ret = decode_packet(&got_frame, 0);
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if (ret < 0)
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break;
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pkt.data += ret;
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pkt.size -= ret;
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} while (pkt.size > 0);
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av_free_packet(&orig_pkt);
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}
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/* flush cached frames */
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pkt.data = NULL;
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pkt.size = 0;
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do {
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decode_packet(&got_frame, 1);
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} while (got_frame);
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printf("Demuxing succeeded.\n");
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if (video_stream) {
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printf("Play the output video file with the command:\n"
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"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
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av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
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video_dst_filename);
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}
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if (audio_stream) {
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enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
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int n_channels = audio_dec_ctx->channels;
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const char *fmt;
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if (av_sample_fmt_is_planar(sfmt)) {
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const char *packed = av_get_sample_fmt_name(sfmt);
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printf("Warning: the sample format the decoder produced is planar "
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"(%s). This example will output the first channel only.\n",
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packed ? packed : "?");
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sfmt = av_get_packed_sample_fmt(sfmt);
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n_channels = 1;
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}
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if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
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goto end;
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printf("Play the output audio file with the command:\n"
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"ffplay -f %s -ac %d -ar %d %s\n",
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fmt, n_channels, audio_dec_ctx->sample_rate,
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audio_dst_filename);
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}
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end:
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if (video_dec_ctx)
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avcodec_close(video_dec_ctx);
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if (audio_dec_ctx)
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avcodec_close(audio_dec_ctx);
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avformat_close_input(&fmt_ctx);
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if (video_dst_file)
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fclose(video_dst_file);
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if (audio_dst_file)
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fclose(audio_dst_file);
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av_free(frame);
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av_free(video_dst_data[0]);
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return ret < 0;
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}
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