From patchwork Sat Apr 11 15:06:13 2020 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Anton Khirnov X-Patchwork-Id: 18863 Return-Path: X-Original-To: patchwork@ffaux-bg.ffmpeg.org Delivered-To: patchwork@ffaux-bg.ffmpeg.org Received: from ffbox0-bg.mplayerhq.hu (ffbox0-bg.ffmpeg.org [79.124.17.100]) by ffaux.localdomain (Postfix) with ESMTP id 89F6D44BB3B for ; Sat, 11 Apr 2020 18:06:33 +0300 (EEST) Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 7573668B7A2; Sat, 11 Apr 2020 18:06:33 +0300 (EEST) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from mail.red.khirnov.net (red.khirnov.net [176.97.15.12]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 0E5BA68B769 for ; Sat, 11 Apr 2020 18:06:27 +0300 (EEST) Received: from localhost (localhost [IPv6:::1]) by mail.red.khirnov.net (Postfix) with ESMTP id D2E2C288550 for ; Sat, 11 Apr 2020 17:06:26 +0200 (CEST) Received: from mail.red.khirnov.net ([IPv6:::1]) by localhost (mail.red.khirnov.net [IPv6:::1]) (amavisd-new, port 10024) with ESMTP id 0fEhZWgPz67I for ; 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Sat, 11 Apr 2020 17:06:23 +0200 (CEST) Received: by libav.daenerys.khirnov.net (Postfix, from userid 1000) id 0B21220E0087; Sat, 11 Apr 2020 17:06:18 +0200 (CEST) From: Anton Khirnov To: ffmpeg-devel@ffmpeg.org Date: Sat, 11 Apr 2020 17:06:13 +0200 Message-Id: <20200411150614.24578-2-anton@khirnov.net> X-Mailer: git-send-email 2.25.1 In-Reply-To: <20200411150614.24578-1-anton@khirnov.net> References: <20200411150614.24578-1-anton@khirnov.net> MIME-Version: 1.0 Subject: [FFmpeg-devel] [PATCH 2/3] doc/examples/demuxing_decoding: convert to new decoding API X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" --- doc/examples/demuxing_decoding.c | 177 ++++++++++++++++--------------- 1 file changed, 91 insertions(+), 86 deletions(-) diff --git a/doc/examples/demuxing_decoding.c b/doc/examples/demuxing_decoding.c index 9bde927321..803e35d25c 100644 --- a/doc/examples/demuxing_decoding.c +++ b/doc/examples/demuxing_decoding.c @@ -55,87 +55,93 @@ static AVPacket pkt; static int video_frame_count = 0; static int audio_frame_count = 0; -static int decode_packet(int *got_frame, int cached) +static int output_video_frame(AVFrame *frame) +{ + if (frame->width != width || frame->height != height || + frame->format != pix_fmt) { + /* To handle this change, one could call av_image_alloc again and + * decode the following frames into another rawvideo file. */ + fprintf(stderr, "Error: Width, height and pixel format have to be " + "constant in a rawvideo file, but the width, height or " + "pixel format of the input video changed:\n" + "old: width = %d, height = %d, format = %s\n" + "new: width = %d, height = %d, format = %s\n", + width, height, av_get_pix_fmt_name(pix_fmt), + frame->width, frame->height, + av_get_pix_fmt_name(frame->format)); + return -1; + } + + printf("video_frame n:%d coded_n:%d\n", + video_frame_count++, frame->coded_picture_number); + + /* copy decoded frame to destination buffer: + * this is required since rawvideo expects non aligned data */ + av_image_copy(video_dst_data, video_dst_linesize, + (const uint8_t **)(frame->data), frame->linesize, + pix_fmt, width, height); + + /* write to rawvideo file */ + fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file); + return 0; +} + +static int output_audio_frame(AVFrame *frame) +{ + size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format); + printf("audio_frame n:%d nb_samples:%d pts:%s\n", + audio_frame_count++, frame->nb_samples, + av_ts2timestr(frame->pts, &audio_dec_ctx->time_base)); + + /* Write the raw audio data samples of the first plane. This works + * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However, + * most audio decoders output planar audio, which uses a separate + * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P). + * In other words, this code will write only the first audio channel + * in these cases. + * You should use libswresample or libavfilter to convert the frame + * to packed data. */ + fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file); + + return 0; +} + +static int decode_packet(AVCodecContext *dec, const AVPacket *pkt) { int ret = 0; - int decoded = pkt.size; - *got_frame = 0; + // submit the packet to the decoder + ret = avcodec_send_packet(dec, pkt); + if (ret < 0) { + fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret)); + return ret; + } - if (pkt.stream_index == video_stream_idx) { - /* decode video frame */ - ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); + // get all the available frames from the decoder + while (ret >= 0) { + ret = avcodec_receive_frame(dec, frame); if (ret < 0) { - fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); - return ret; - } + // those two return values are special and mean there is no output + // frame available, but there were no errors during decoding + if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)) + return 0; - if (*got_frame) { - - if (frame->width != width || frame->height != height || - frame->format != pix_fmt) { - /* To handle this change, one could call av_image_alloc again and - * decode the following frames into another rawvideo file. */ - fprintf(stderr, "Error: Width, height and pixel format have to be " - "constant in a rawvideo file, but the width, height or " - "pixel format of the input video changed:\n" - "old: width = %d, height = %d, format = %s\n" - "new: width = %d, height = %d, format = %s\n", - width, height, av_get_pix_fmt_name(pix_fmt), - frame->width, frame->height, - av_get_pix_fmt_name(frame->format)); - return -1; - } - - printf("video_frame%s n:%d coded_n:%d\n", - cached ? "(cached)" : "", - video_frame_count++, frame->coded_picture_number); - - /* copy decoded frame to destination buffer: - * this is required since rawvideo expects non aligned data */ - av_image_copy(video_dst_data, video_dst_linesize, - (const uint8_t **)(frame->data), frame->linesize, - pix_fmt, width, height); - - /* write to rawvideo file */ - fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file); - } - } else if (pkt.stream_index == audio_stream_idx) { - /* decode audio frame */ - ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt); - if (ret < 0) { - fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret)); + fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret)); return ret; } - /* Some audio decoders decode only part of the packet, and have to be - * called again with the remainder of the packet data. - * Sample: fate-suite/lossless-audio/luckynight-partial.shn - * Also, some decoders might over-read the packet. */ - decoded = FFMIN(ret, pkt.size); - - if (*got_frame) { - size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format); - printf("audio_frame%s n:%d nb_samples:%d pts:%s\n", - cached ? "(cached)" : "", - audio_frame_count++, frame->nb_samples, - av_ts2timestr(frame->pts, &audio_dec_ctx->time_base)); - - /* Write the raw audio data samples of the first plane. This works - * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However, - * most audio decoders output planar audio, which uses a separate - * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P). - * In other words, this code will write only the first audio channel - * in these cases. - * You should use libswresample or libavfilter to convert the frame - * to packed data. */ - fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file); - } - } - if (*got_frame) + // write the frame data to output file + if (dec->codec->type == AVMEDIA_TYPE_VIDEO) + ret = output_video_frame(frame); + else + ret = output_audio_frame(frame); + av_frame_unref(frame); + if (ret < 0) + return ret; + } - return decoded; + return 0; } static int open_codec_context(int *stream_idx, @@ -221,7 +227,7 @@ static int get_format_from_sample_fmt(const char **fmt, int main (int argc, char **argv) { - int ret = 0, got_frame; + int ret = 0; if (argc != 4) { fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n" @@ -309,23 +315,22 @@ int main (int argc, char **argv) /* read frames from the file */ while (av_read_frame(fmt_ctx, &pkt) >= 0) { - AVPacket orig_pkt = pkt; - do { - ret = decode_packet(&got_frame, 0); - if (ret < 0) - break; - pkt.data += ret; - pkt.size -= ret; - } while (pkt.size > 0); - av_packet_unref(&orig_pkt); + // check if the packet belongs to a stream we are interested in, otherwise + // skip it + if (pkt.stream_index == video_stream_idx) + ret = decode_packet(video_dec_ctx, &pkt); + else if (pkt.stream_index == audio_stream_idx) + ret = decode_packet(audio_dec_ctx, &pkt); + av_packet_unref(&pkt); + if (ret < 0) + break; } - /* flush cached frames */ - pkt.data = NULL; - pkt.size = 0; - do { - decode_packet(&got_frame, 1); - } while (got_frame); + /* flush the decoders */ + if (video_dec_ctx) + decode_packet(video_dec_ctx, NULL); + if (audio_dec_ctx) + decode_packet(audio_dec_ctx, NULL); printf("Demuxing succeeded.\n");