ffmpeg | branch: master | Clément Bœsch <u...@pkh.me> | Wed Mar 29 13:29:00 2017 +0200| [b785af48687fa839fbc25045d2201335753304b3] | committer: Clément Bœsch
Merge commit '40aaa8dadfd1c69ff4460d04750e1403b5535a6d' * commit '40aaa8dadfd1c69ff4460d04750e1403b5535a6d': examples/avcodec: split audio encoding into a separate example Merged-by: Clément Bœsch <u...@pkh.me> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b785af48687fa839fbc25045d2201335753304b3 --- configure | 2 + doc/Makefile | 1 + doc/examples/.gitignore | 1 + doc/examples/Makefile | 3 +- doc/examples/decoding_encoding.c | 192 -------------------------------- doc/examples/encode_audio.c | 235 +++++++++++++++++++++++++++++++++++++++ 6 files changed, 241 insertions(+), 193 deletions(-) diff --git a/configure b/configure index a84b126..bf1d2fe 100755 --- a/configure +++ b/configure @@ -1463,6 +1463,7 @@ EXAMPLE_LIST=" avio_reading_example decoding_encoding_example demuxing_decoding_example + encode_audio_example extract_mvs_example filter_audio_example filtering_audio_example @@ -3169,6 +3170,7 @@ avio_dir_cmd_deps="avformat avutil" avio_reading_deps="avformat avcodec avutil" decoding_encoding_example_deps="avcodec avformat avutil" demuxing_decoding_example_deps="avcodec avformat avutil" +encode_audio_example_deps="avcodec avutil" extract_mvs_example_deps="avcodec avformat avutil" filter_audio_example_deps="avfilter avutil" filtering_audio_example_deps="avfilter avcodec avformat avutil" diff --git a/doc/Makefile b/doc/Makefile index a180fbe..396d9e0 100644 --- a/doc/Makefile +++ b/doc/Makefile @@ -40,6 +40,7 @@ DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding +DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio diff --git a/doc/examples/.gitignore b/doc/examples/.gitignore index 430d3bf..1a6209b 100644 --- a/doc/examples/.gitignore +++ b/doc/examples/.gitignore @@ -2,6 +2,7 @@ /avio_reading /decoding_encoding /demuxing_decoding +/encode_audio /extract_mvs /filter_audio /filtering_audio diff --git a/doc/examples/Makefile b/doc/examples/Makefile index af38159..f76e5a2 100644 --- a/doc/examples/Makefile +++ b/doc/examples/Makefile @@ -15,6 +15,7 @@ EXAMPLES= avio_dir_cmd \ avio_reading \ decoding_encoding \ demuxing_decoding \ + encode_audio \ extract_mvs \ filtering_video \ filtering_audio \ @@ -31,7 +32,7 @@ OBJS=$(addsuffix .o,$(EXAMPLES)) # the following examples make explicit use of the math library avcodec: LDLIBS += -lm -decoding_encoding: LDLIBS += -lm +encode_audio: LDLIBS += -lm muxing: LDLIBS += -lm resampling_audio: LDLIBS += -lm diff --git a/doc/examples/decoding_encoding.c b/doc/examples/decoding_encoding.c index 1c5a78a..f863bdf 100644 --- a/doc/examples/decoding_encoding.c +++ b/doc/examples/decoding_encoding.c @@ -44,197 +44,6 @@ #define AUDIO_INBUF_SIZE 20480 #define AUDIO_REFILL_THRESH 4096 -/* check that a given sample format is supported by the encoder */ -static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) -{ - const enum AVSampleFormat *p = codec->sample_fmts; - - while (*p != AV_SAMPLE_FMT_NONE) { - if (*p == sample_fmt) - return 1; - p++; - } - return 0; -} - -/* just pick the highest supported samplerate */ -static int select_sample_rate(AVCodec *codec) -{ - const int *p; - int best_samplerate = 0; - - if (!codec->supported_samplerates) - return 44100; - - p = codec->supported_samplerates; - while (*p) { - best_samplerate = FFMAX(*p, best_samplerate); - p++; - } - return best_samplerate; -} - -/* select layout with the highest channel count */ -static int select_channel_layout(AVCodec *codec) -{ - const uint64_t *p; - uint64_t best_ch_layout = 0; - int best_nb_channels = 0; - - if (!codec->channel_layouts) - return AV_CH_LAYOUT_STEREO; - - p = codec->channel_layouts; - while (*p) { - int nb_channels = av_get_channel_layout_nb_channels(*p); - - if (nb_channels > best_nb_channels) { - best_ch_layout = *p; - best_nb_channels = nb_channels; - } - p++; - } - return best_ch_layout; -} - -/* - * Audio encoding example - */ -static void audio_encode_example(const char *filename) -{ - AVCodec *codec; - AVCodecContext *c= NULL; - AVFrame *frame; - AVPacket pkt; - int i, j, k, ret, got_output; - int buffer_size; - FILE *f; - uint16_t *samples; - float t, tincr; - - printf("Encode audio file %s\n", filename); - - /* find the MP2 encoder */ - codec = avcodec_find_encoder(AV_CODEC_ID_MP2); - if (!codec) { - fprintf(stderr, "Codec not found\n"); - exit(1); - } - - c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate audio codec context\n"); - exit(1); - } - - /* put sample parameters */ - c->bit_rate = 64000; - - /* check that the encoder supports s16 pcm input */ - c->sample_fmt = AV_SAMPLE_FMT_S16; - if (!check_sample_fmt(codec, c->sample_fmt)) { - fprintf(stderr, "Encoder does not support sample format %s", - av_get_sample_fmt_name(c->sample_fmt)); - exit(1); - } - - /* select other audio parameters supported by the encoder */ - c->sample_rate = select_sample_rate(codec); - c->channel_layout = select_channel_layout(codec); - c->channels = av_get_channel_layout_nb_channels(c->channel_layout); - - /* open it */ - if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); - exit(1); - } - - f = fopen(filename, "wb"); - if (!f) { - fprintf(stderr, "Could not open %s\n", filename); - exit(1); - } - - /* frame containing input raw audio */ - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate audio frame\n"); - exit(1); - } - - frame->nb_samples = c->frame_size; - frame->format = c->sample_fmt; - frame->channel_layout = c->channel_layout; - - /* the codec gives us the frame size, in samples, - * we calculate the size of the samples buffer in bytes */ - buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, - c->sample_fmt, 0); - if (buffer_size < 0) { - fprintf(stderr, "Could not get sample buffer size\n"); - exit(1); - } - samples = av_malloc(buffer_size); - if (!samples) { - fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", - buffer_size); - exit(1); - } - /* setup the data pointers in the AVFrame */ - ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, - (const uint8_t*)samples, buffer_size, 0); - if (ret < 0) { - fprintf(stderr, "Could not setup audio frame\n"); - exit(1); - } - - /* encode a single tone sound */ - t = 0; - tincr = 2 * M_PI * 440.0 / c->sample_rate; - for (i = 0; i < 200; i++) { - av_init_packet(&pkt); - pkt.data = NULL; // packet data will be allocated by the encoder - pkt.size = 0; - - for (j = 0; j < c->frame_size; j++) { - samples[2*j] = (int)(sin(t) * 10000); - - for (k = 1; k < c->channels; k++) - samples[2*j + k] = samples[2*j]; - t += tincr; - } - /* encode the samples */ - ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding audio frame\n"); - exit(1); - } - if (got_output) { - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - - /* get the delayed frames */ - for (got_output = 1; got_output; i++) { - ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); - exit(1); - } - - if (got_output) { - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - fclose(f); - - av_freep(&samples); - av_frame_free(&frame); - avcodec_free_context(&c); -} - /* * Audio decoding. */ @@ -646,7 +455,6 @@ int main(int argc, char **argv) if (!strcmp(output_type, "h264")) { video_encode_example("test.h264", AV_CODEC_ID_H264); } else if (!strcmp(output_type, "mp2")) { - audio_encode_example("test.mp2"); audio_decode_example("test.pcm", "test.mp2"); } else if (!strcmp(output_type, "mpg")) { video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO); diff --git a/doc/examples/encode_audio.c b/doc/examples/encode_audio.c new file mode 100644 index 0000000..5932521 --- /dev/null +++ b/doc/examples/encode_audio.c @@ -0,0 +1,235 @@ +/* + * Copyright (c) 2001 Fabrice Bellard + * + * 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 + * audio encoding with libavcodec API example. + * + * @example encode_audio.c + */ + +#include <stdint.h> +#include <stdio.h> +#include <stdlib.h> + +#include "libavcodec/avcodec.h" + +#include "libavutil/channel_layout.h" +#include "libavutil/common.h" +#include "libavutil/frame.h" +#include "libavutil/samplefmt.h" + +/* check that a given sample format is supported by the encoder */ +static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) +{ + const enum AVSampleFormat *p = codec->sample_fmts; + + while (*p != AV_SAMPLE_FMT_NONE) { + if (*p == sample_fmt) + return 1; + p++; + } + return 0; +} + +/* just pick the highest supported samplerate */ +static int select_sample_rate(AVCodec *codec) +{ + const int *p; + int best_samplerate = 0; + + if (!codec->supported_samplerates) + return 44100; + + p = codec->supported_samplerates; + while (*p) { + best_samplerate = FFMAX(*p, best_samplerate); + p++; + } + return best_samplerate; +} + +/* select layout with the highest channel count */ +static int select_channel_layout(AVCodec *codec) +{ + const uint64_t *p; + uint64_t best_ch_layout = 0; + int best_nb_channels = 0; + + if (!codec->channel_layouts) + return AV_CH_LAYOUT_STEREO; + + p = codec->channel_layouts; + while (*p) { + int nb_channels = av_get_channel_layout_nb_channels(*p); + + if (nb_channels > best_nb_channels) { + best_ch_layout = *p; + best_nb_channels = nb_channels; + } + p++; + } + return best_ch_layout; +} + +int main(int argc, char **argv) +{ + const char *filename; + AVCodec *codec; + AVCodecContext *c= NULL; + AVFrame *frame; + AVPacket pkt; + int i, j, k, ret, got_output; + int buffer_size; + FILE *f; + uint16_t *samples; + float t, tincr; + + if (argc <= 1) { + fprintf(stderr, "Usage: %s <output file>\n", argv[0]); + return 0; + } + filename = argv[1]; + + /* register all the codecs */ + avcodec_register_all(); + + /* find the MP2 encoder */ + codec = avcodec_find_encoder(AV_CODEC_ID_MP2); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } + + /* put sample parameters */ + c->bit_rate = 64000; + + /* check that the encoder supports s16 pcm input */ + c->sample_fmt = AV_SAMPLE_FMT_S16; + if (!check_sample_fmt(codec, c->sample_fmt)) { + fprintf(stderr, "Encoder does not support sample format %s", + av_get_sample_fmt_name(c->sample_fmt)); + exit(1); + } + + /* select other audio parameters supported by the encoder */ + c->sample_rate = select_sample_rate(codec); + c->channel_layout = select_channel_layout(codec); + c->channels = av_get_channel_layout_nb_channels(c->channel_layout); + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "wb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + /* frame containing input raw audio */ + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate audio frame\n"); + exit(1); + } + + frame->nb_samples = c->frame_size; + frame->format = c->sample_fmt; + frame->channel_layout = c->channel_layout; + + /* the codec gives us the frame size, in samples, + * we calculate the size of the samples buffer in bytes */ + buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, + c->sample_fmt, 0); + if (buffer_size < 0) { + fprintf(stderr, "Could not get sample buffer size\n"); + exit(1); + } + samples = av_malloc(buffer_size); + if (!samples) { + fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", + buffer_size); + exit(1); + } + /* setup the data pointers in the AVFrame */ + ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (const uint8_t*)samples, buffer_size, 0); + if (ret < 0) { + fprintf(stderr, "Could not setup audio frame\n"); + exit(1); + } + + /* encode a single tone sound */ + t = 0; + tincr = 2 * M_PI * 440.0 / c->sample_rate; + for (i = 0; i < 200; i++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + for (j = 0; j < c->frame_size; j++) { + samples[2*j] = (int)(sin(t) * 10000); + + for (k = 1; k < c->channels; k++) + samples[2*j + k] = samples[2*j]; + t += tincr; + } + /* encode the samples */ + ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding audio frame\n"); + exit(1); + } + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_packet_unref(&pkt); + } + } + + /* get the delayed frames */ + for (got_output = 1; got_output; i++) { + ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_packet_unref(&pkt); + } + } + fclose(f); + + av_freep(&samples); + av_frame_free(&frame); + avcodec_free_context(&c); +} ====================================================================== diff --cc configure index a84b126,28567bb..bf1d2fe --- a/configure +++ b/configure @@@ -1459,38 -1209,30 +1459,39 @@@ COMPONENT_LIST= " EXAMPLE_LIST=" - avcodec_example + avio_dir_cmd_example + avio_reading_example + decoding_encoding_example + demuxing_decoding_example + encode_audio_example + extract_mvs_example filter_audio_example + filtering_audio_example + filtering_video_example + http_multiclient_example metadata_example - output_example + muxing_example qsvdec_example + remuxing_example + resampling_audio_example + scaling_video_example transcode_aac_example + transcoding_example " - -HWACCEL_LIBRARY_NONFREE_LIST=" - cuda - libnpp -" -HWACCEL_LIBRARY_LIST=" - $HWACCEL_LIBRARY_NONFREE_LIST - d3d11va - dxva2 - libmfx - mmal - nvenc - omx - vaapi - vda - vdpau +EXTERNAL_AUTODETECT_LIBRARY_LIST=" + bzlib + iconv + libxcb + libxcb_shm + libxcb_shape + libxcb_xfixes + lzma + schannel + sdl + sdl2 + securetransport + xlib + zlib " EXTERNAL_LIBRARY_GPL_LIST=" @@@ -3165,23 -2435,13 +3166,24 @@@ zscale_filter_deps="libzimg scale_vaapi_filter_deps="vaapi VAProcPipelineParameterBuffer" # examples -avcodec_example_deps="avcodec avutil" +avio_dir_cmd_deps="avformat avutil" +avio_reading_deps="avformat avcodec avutil" +decoding_encoding_example_deps="avcodec avformat avutil" +demuxing_decoding_example_deps="avcodec avformat avutil" + encode_audio_example_deps="avcodec avutil" +extract_mvs_example_deps="avcodec avformat avutil" filter_audio_example_deps="avfilter avutil" +filtering_audio_example_deps="avfilter avcodec avformat avutil" +filtering_video_example_deps="avfilter avcodec avformat avutil" +http_multiclient_example_deps="avformat avutil fork" metadata_example_deps="avformat avutil" -output_example_deps="avcodec avformat avutil swscale" +muxing_example_deps="avcodec avformat avutil swscale" qsvdec_example_deps="avcodec avutil libmfx h264_qsv_decoder vaapi_x11" -transcode_aac_example_deps="avcodec avformat avresample" +remuxing_example_deps="avcodec avformat avutil" +resampling_audio_example_deps="avutil swresample" +scaling_video_example_deps="avutil swscale" +transcode_aac_example_deps="avcodec avformat swresample" +transcoding_example_deps="avfilter avcodec avformat avutil" # libraries, in linking order avcodec_deps="avutil" diff --cc doc/Makefile index a180fbe,738e601..396d9e0 --- a/doc/Makefile +++ b/doc/Makefile @@@ -27,42 -11,25 +27,43 @@@ HTMLPAGES = $(AVPROGS-yes:%=doc/%.htm doc/nut.html \ doc/platform.html \ -DOCS-$(CONFIG_POD2MAN) += $(MANPAGES) $(PODPAGES) -DOCS-$(CONFIG_TEXI2HTML) += $(HTMLPAGES) -DOCS = $(DOCS-yes) +TXTPAGES = doc/fate.txt \ -DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec -DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio -DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio -DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata -DOC_EXAMPLES-$(CONFIG_OUTPUT_EXAMPLE) += output -DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec -DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac -ALL_DOC_EXAMPLES = avcodec encode_audio filter_audio metadata output transcode_aac -DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(EXESUF)) -ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES:%=doc/examples/%$(EXESUF)) -PROGS += $(DOC_EXAMPLES) +DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES) +DOCS-$(CONFIG_PODPAGES) += $(PODPAGES) +DOCS-$(CONFIG_MANPAGES) += $(MANPAGES) +DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES) +DOCS = $(DOCS-yes) -all: $(DOCS) +DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd +DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading +DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding +DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding ++DOC_EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio +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)) +ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF)) +ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF)) +PROGS += $(DOC_EXAMPLES) + +all-$(CONFIG_DOC): doc + +doc: documentation apidoc: doc/doxy/html documentation: $(DOCS) diff --cc doc/examples/.gitignore index 430d3bf,3f60af7..1a6209b --- a/doc/examples/.gitignore +++ b/doc/examples/.gitignore @@@ -1,18 -1,5 +1,19 @@@ -/avcodec +/avio_dir_cmd +/avio_reading +/decoding_encoding +/demuxing_decoding ++/encode_audio +/extract_mvs /filter_audio +/filtering_audio +/filtering_video +/http_multiclient /metadata -/output +/muxing +/pc-uninstalled +/qsvdec +/remuxing +/resampling_audio +/scaling_video /transcode_aac +/transcoding diff --cc doc/examples/Makefile index af38159,0000000..f76e5a2 mode 100644,000000..100644 --- a/doc/examples/Makefile +++ b/doc/examples/Makefile @@@ -1,46 -1,0 +1,47 @@@ +# use pkg-config for getting CFLAGS and LDLIBS +FFMPEG_LIBS= libavdevice \ + libavformat \ + libavfilter \ + libavcodec \ + libswresample \ + libswscale \ + libavutil \ + +CFLAGS += -Wall -g +CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS) +LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS) + +EXAMPLES= avio_dir_cmd \ + avio_reading \ + decoding_encoding \ + demuxing_decoding \ ++ encode_audio \ + extract_mvs \ + filtering_video \ + filtering_audio \ + http_multiclient \ + metadata \ + muxing \ + remuxing \ + resampling_audio \ + scaling_video \ + transcode_aac \ + transcoding \ + +OBJS=$(addsuffix .o,$(EXAMPLES)) + +# the following examples make explicit use of the math library +avcodec: LDLIBS += -lm - decoding_encoding: LDLIBS += -lm ++encode_audio: LDLIBS += -lm +muxing: LDLIBS += -lm +resampling_audio: LDLIBS += -lm + +.phony: all clean-test clean + +all: $(OBJS) $(EXAMPLES) + +clean-test: + $(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg + +clean: clean-test + $(RM) $(EXAMPLES) $(OBJS) diff --cc doc/examples/decoding_encoding.c index 1c5a78a,0000000..f863bdf mode 100644,000000..100644 --- a/doc/examples/decoding_encoding.c +++ b/doc/examples/decoding_encoding.c @@@ -1,661 -1,0 +1,469 @@@ +/* + * Copyright (c) 2001 Fabrice Bellard + * + * 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 + * libavcodec API use example. + * + * @example decoding_encoding.c + * Note that libavcodec only handles codecs (MPEG, MPEG-4, etc...), + * not file formats (AVI, VOB, MP4, MOV, MKV, MXF, FLV, MPEG-TS, MPEG-PS, etc...). + * See library 'libavformat' for the format handling + */ + +#include <math.h> + +#include <libavutil/opt.h> +#include <libavcodec/avcodec.h> +#include <libavutil/channel_layout.h> +#include <libavutil/common.h> +#include <libavutil/imgutils.h> +#include <libavutil/mathematics.h> +#include <libavutil/samplefmt.h> + +#define INBUF_SIZE 4096 +#define AUDIO_INBUF_SIZE 20480 +#define AUDIO_REFILL_THRESH 4096 + - /* check that a given sample format is supported by the encoder */ - static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) - { - const enum AVSampleFormat *p = codec->sample_fmts; - - while (*p != AV_SAMPLE_FMT_NONE) { - if (*p == sample_fmt) - return 1; - p++; - } - return 0; - } - - /* just pick the highest supported samplerate */ - static int select_sample_rate(AVCodec *codec) - { - const int *p; - int best_samplerate = 0; - - if (!codec->supported_samplerates) - return 44100; - - p = codec->supported_samplerates; - while (*p) { - best_samplerate = FFMAX(*p, best_samplerate); - p++; - } - return best_samplerate; - } - - /* select layout with the highest channel count */ - static int select_channel_layout(AVCodec *codec) - { - const uint64_t *p; - uint64_t best_ch_layout = 0; - int best_nb_channels = 0; - - if (!codec->channel_layouts) - return AV_CH_LAYOUT_STEREO; - - p = codec->channel_layouts; - while (*p) { - int nb_channels = av_get_channel_layout_nb_channels(*p); - - if (nb_channels > best_nb_channels) { - best_ch_layout = *p; - best_nb_channels = nb_channels; - } - p++; - } - return best_ch_layout; - } - - /* - * Audio encoding example - */ - static void audio_encode_example(const char *filename) - { - AVCodec *codec; - AVCodecContext *c= NULL; - AVFrame *frame; - AVPacket pkt; - int i, j, k, ret, got_output; - int buffer_size; - FILE *f; - uint16_t *samples; - float t, tincr; - - printf("Encode audio file %s\n", filename); - - /* find the MP2 encoder */ - codec = avcodec_find_encoder(AV_CODEC_ID_MP2); - if (!codec) { - fprintf(stderr, "Codec not found\n"); - exit(1); - } - - c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate audio codec context\n"); - exit(1); - } - - /* put sample parameters */ - c->bit_rate = 64000; - - /* check that the encoder supports s16 pcm input */ - c->sample_fmt = AV_SAMPLE_FMT_S16; - if (!check_sample_fmt(codec, c->sample_fmt)) { - fprintf(stderr, "Encoder does not support sample format %s", - av_get_sample_fmt_name(c->sample_fmt)); - exit(1); - } - - /* select other audio parameters supported by the encoder */ - c->sample_rate = select_sample_rate(codec); - c->channel_layout = select_channel_layout(codec); - c->channels = av_get_channel_layout_nb_channels(c->channel_layout); - - /* open it */ - if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); - exit(1); - } - - f = fopen(filename, "wb"); - if (!f) { - fprintf(stderr, "Could not open %s\n", filename); - exit(1); - } - - /* frame containing input raw audio */ - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate audio frame\n"); - exit(1); - } - - frame->nb_samples = c->frame_size; - frame->format = c->sample_fmt; - frame->channel_layout = c->channel_layout; - - /* the codec gives us the frame size, in samples, - * we calculate the size of the samples buffer in bytes */ - buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, - c->sample_fmt, 0); - if (buffer_size < 0) { - fprintf(stderr, "Could not get sample buffer size\n"); - exit(1); - } - samples = av_malloc(buffer_size); - if (!samples) { - fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", - buffer_size); - exit(1); - } - /* setup the data pointers in the AVFrame */ - ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, - (const uint8_t*)samples, buffer_size, 0); - if (ret < 0) { - fprintf(stderr, "Could not setup audio frame\n"); - exit(1); - } - - /* encode a single tone sound */ - t = 0; - tincr = 2 * M_PI * 440.0 / c->sample_rate; - for (i = 0; i < 200; i++) { - av_init_packet(&pkt); - pkt.data = NULL; // packet data will be allocated by the encoder - pkt.size = 0; - - for (j = 0; j < c->frame_size; j++) { - samples[2*j] = (int)(sin(t) * 10000); - - for (k = 1; k < c->channels; k++) - samples[2*j + k] = samples[2*j]; - t += tincr; - } - /* encode the samples */ - ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding audio frame\n"); - exit(1); - } - if (got_output) { - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - - /* get the delayed frames */ - for (got_output = 1; got_output; i++) { - ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); - exit(1); - } - - if (got_output) { - fwrite(pkt.data, 1, pkt.size, f); - av_packet_unref(&pkt); - } - } - fclose(f); - - av_freep(&samples); - av_frame_free(&frame); - avcodec_free_context(&c); - } - +/* + * Audio decoding. + */ +static void audio_decode_example(const char *outfilename, const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int len; + FILE *f, *outfile; + uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; + AVPacket avpkt; + AVFrame *decoded_frame = NULL; + + av_init_packet(&avpkt); + + printf("Decode audio file %s to %s\n", filename, outfilename); + + /* find the MPEG audio decoder */ + codec = avcodec_find_decoder(AV_CODEC_ID_MP2); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "rb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + outfile = fopen(outfilename, "wb"); + if (!outfile) { + av_free(c); + exit(1); + } + + /* decode until eof */ + avpkt.data = inbuf; + avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); + + while (avpkt.size > 0) { + int i, ch; + int got_frame = 0; + + if (!decoded_frame) { + if (!(decoded_frame = av_frame_alloc())) { + fprintf(stderr, "Could not allocate audio frame\n"); + exit(1); + } + } + + len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); + if (len < 0) { + fprintf(stderr, "Error while decoding\n"); + exit(1); + } + if (got_frame) { + /* if a frame has been decoded, output it */ + int data_size = av_get_bytes_per_sample(c->sample_fmt); + if (data_size < 0) { + /* This should not occur, checking just for paranoia */ + fprintf(stderr, "Failed to calculate data size\n"); + exit(1); + } + for (i=0; i<decoded_frame->nb_samples; i++) + for (ch=0; ch<c->channels; ch++) + fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile); + } + avpkt.size -= len; + avpkt.data += len; + avpkt.dts = + avpkt.pts = AV_NOPTS_VALUE; + if (avpkt.size < AUDIO_REFILL_THRESH) { + /* Refill the input buffer, to avoid trying to decode + * incomplete frames. Instead of this, one could also use + * a parser, or use a proper container format through + * libavformat. */ + memmove(inbuf, avpkt.data, avpkt.size); + avpkt.data = inbuf; + len = fread(avpkt.data + avpkt.size, 1, + AUDIO_INBUF_SIZE - avpkt.size, f); + if (len > 0) + avpkt.size += len; + } + } + + fclose(outfile); + fclose(f); + + avcodec_free_context(&c); + av_frame_free(&decoded_frame); +} + +/* + * Video encoding example + */ +static void video_encode_example(const char *filename, int codec_id) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int i, ret, x, y, got_output; + FILE *f; + AVFrame *frame; + AVPacket pkt; + uint8_t endcode[] = { 0, 0, 1, 0xb7 }; + + printf("Encode video file %s\n", filename); + + /* find the video encoder */ + codec = avcodec_find_encoder(codec_id); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } + + /* put sample parameters */ + c->bit_rate = 400000; + /* resolution must be a multiple of two */ + c->width = 352; + c->height = 288; + /* frames per second */ + c->time_base = (AVRational){1,25}; + /* emit one intra frame every ten frames + * check frame pict_type before passing frame + * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I + * then gop_size is ignored and the output of encoder + * will always be I frame irrespective to gop_size + */ + c->gop_size = 10; + c->max_b_frames = 1; + c->pix_fmt = AV_PIX_FMT_YUV420P; + + if (codec_id == AV_CODEC_ID_H264) + av_opt_set(c->priv_data, "preset", "slow", 0); + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "wb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + frame->format = c->pix_fmt; + frame->width = c->width; + frame->height = c->height; + + /* the image can be allocated by any means and av_image_alloc() is + * just the most convenient way if av_malloc() is to be used */ + ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height, + c->pix_fmt, 32); + if (ret < 0) { + fprintf(stderr, "Could not allocate raw picture buffer\n"); + exit(1); + } + + /* encode 1 second of video */ + for (i = 0; i < 25; i++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + fflush(stdout); + /* prepare a dummy image */ + /* Y */ + for (y = 0; y < c->height; y++) { + for (x = 0; x < c->width; x++) { + frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3; + } + } + + /* Cb and Cr */ + for (y = 0; y < c->height/2; y++) { + for (x = 0; x < c->width/2; x++) { + frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2; + frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5; + } + } + + frame->pts = i; + + /* encode the image */ + ret = avcodec_encode_video2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + printf("Write frame %3d (size=%5d)\n", i, pkt.size); + fwrite(pkt.data, 1, pkt.size, f); + av_packet_unref(&pkt); + } + } + + /* get the delayed frames */ + for (got_output = 1; got_output; i++) { + fflush(stdout); + + ret = avcodec_encode_video2(c, &pkt, NULL, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + printf("Write frame %3d (size=%5d)\n", i, pkt.size); + fwrite(pkt.data, 1, pkt.size, f); + av_packet_unref(&pkt); + } + } + + /* add sequence end code to have a real MPEG file */ + fwrite(endcode, 1, sizeof(endcode), f); + fclose(f); + + avcodec_free_context(&c); + av_freep(&frame->data[0]); + av_frame_free(&frame); + printf("\n"); +} + +/* + * Video decoding example + */ + +static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize, + char *filename) +{ + FILE *f; + int i; + + f = fopen(filename,"w"); + fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255); + for (i = 0; i < ysize; i++) + fwrite(buf + i * wrap, 1, xsize, f); + fclose(f); +} + +static int decode_write_frame(const char *outfilename, AVCodecContext *avctx, + AVFrame *frame, int *frame_count, AVPacket *pkt, int last) +{ + int len, got_frame; + char buf[1024]; + + len = avcodec_decode_video2(avctx, frame, &got_frame, pkt); + if (len < 0) { + fprintf(stderr, "Error while decoding frame %d\n", *frame_count); + return len; + } + if (got_frame) { + printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count); + fflush(stdout); + + /* the picture is allocated by the decoder, no need to free it */ + snprintf(buf, sizeof(buf), outfilename, *frame_count); + pgm_save(frame->data[0], frame->linesize[0], + frame->width, frame->height, buf); + (*frame_count)++; + } + if (pkt->data) { + pkt->size -= len; + pkt->data += len; + } + return 0; +} + +static void video_decode_example(const char *outfilename, const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int frame_count; + FILE *f; + AVFrame *frame; + uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; + AVPacket avpkt; + + av_init_packet(&avpkt); + + /* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */ + memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE); + + printf("Decode video file %s to %s\n", filename, outfilename); + + /* find the MPEG-1 video decoder */ + codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } + + if (codec->capabilities & AV_CODEC_CAP_TRUNCATED) + c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames + + /* For some codecs, such as msmpeg4 and mpeg4, width and height + MUST be initialized there because this information is not + available in the bitstream. */ + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "rb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + + frame_count = 0; + for (;;) { + avpkt.size = fread(inbuf, 1, INBUF_SIZE, f); + if (avpkt.size == 0) + break; + + /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio) + and this is the only method to use them because you cannot + know the compressed data size before analysing it. + + BUT some other codecs (msmpeg4, mpeg4) are inherently frame + based, so you must call them with all the data for one + frame exactly. You must also initialize 'width' and + 'height' before initializing them. */ + + /* NOTE2: some codecs allow the raw parameters (frame size, + sample rate) to be changed at any frame. We handle this, so + you should also take care of it */ + + /* here, we use a stream based decoder (mpeg1video), so we + feed decoder and see if it could decode a frame */ + avpkt.data = inbuf; + while (avpkt.size > 0) + if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0) + exit(1); + } + + /* Some codecs, such as MPEG, transmit the I- and P-frame with a + latency of one frame. You must do the following to have a + chance to get the last frame of the video. */ + avpkt.data = NULL; + avpkt.size = 0; + decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1); + + fclose(f); + + avcodec_free_context(&c); + av_frame_free(&frame); + printf("\n"); +} + +int main(int argc, char **argv) +{ + const char *output_type; + + /* register all the codecs */ + avcodec_register_all(); + + if (argc < 2) { + printf("usage: %s output_type\n" + "API example program to decode/encode a media stream with libavcodec.\n" + "This program generates a synthetic stream and encodes it to a file\n" + "named test.h264, test.mp2 or test.mpg depending on output_type.\n" + "The encoded stream is then decoded and written to a raw data output.\n" + "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n", + argv[0]); + return 1; + } + output_type = argv[1]; + + if (!strcmp(output_type, "h264")) { + video_encode_example("test.h264", AV_CODEC_ID_H264); + } else if (!strcmp(output_type, "mp2")) { - audio_encode_example("test.mp2"); + audio_decode_example("test.pcm", "test.mp2"); + } else if (!strcmp(output_type, "mpg")) { + video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO); + video_decode_example("test%02d.pgm", "test.mpg"); + } else { + fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n", + output_type); + return 1; + } + + return 0; +} diff --cc doc/examples/encode_audio.c index 0000000,cabe589..5932521 mode 000000,100644..100644 --- a/doc/examples/encode_audio.c +++ b/doc/examples/encode_audio.c @@@ -1,0 -1,211 +1,235 @@@ + /* - * copyright (c) 2001 Fabrice Bellard ++ * Copyright (c) 2001 Fabrice Bellard + * - * This file is part of Libav. ++ * 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: + * - * Libav is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. ++ * The above copyright notice and this permission notice shall be included in ++ * all copies or substantial portions of the Software. + * - * Libav is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA ++ * 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 + * audio encoding with libavcodec API example. + * + * @example encode_audio.c + */ + + #include <stdint.h> + #include <stdio.h> + #include <stdlib.h> + + #include "libavcodec/avcodec.h" + + #include "libavutil/channel_layout.h" + #include "libavutil/common.h" + #include "libavutil/frame.h" + #include "libavutil/samplefmt.h" + + /* check that a given sample format is supported by the encoder */ + static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) + { + const enum AVSampleFormat *p = codec->sample_fmts; + + while (*p != AV_SAMPLE_FMT_NONE) { + if (*p == sample_fmt) + return 1; + p++; + } + return 0; + } + + /* just pick the highest supported samplerate */ + static int select_sample_rate(AVCodec *codec) + { + const int *p; + int best_samplerate = 0; + + if (!codec->supported_samplerates) + return 44100; + + p = codec->supported_samplerates; + while (*p) { + best_samplerate = FFMAX(*p, best_samplerate); + p++; + } + return best_samplerate; + } + + /* select layout with the highest channel count */ + static int select_channel_layout(AVCodec *codec) + { + const uint64_t *p; + uint64_t best_ch_layout = 0; + int best_nb_channels = 0; + + if (!codec->channel_layouts) + return AV_CH_LAYOUT_STEREO; + + p = codec->channel_layouts; + while (*p) { + int nb_channels = av_get_channel_layout_nb_channels(*p); + + if (nb_channels > best_nb_channels) { + best_ch_layout = *p; + best_nb_channels = nb_channels; + } + p++; + } + return best_ch_layout; + } + + int main(int argc, char **argv) + { + const char *filename; + AVCodec *codec; + AVCodecContext *c= NULL; + AVFrame *frame; + AVPacket pkt; + int i, j, k, ret, got_output; + int buffer_size; + FILE *f; + uint16_t *samples; + float t, tincr; + + if (argc <= 1) { + fprintf(stderr, "Usage: %s <output file>\n", argv[0]); + return 0; + } + filename = argv[1]; + + /* register all the codecs */ + avcodec_register_all(); + + /* find the MP2 encoder */ + codec = avcodec_find_encoder(AV_CODEC_ID_MP2); + if (!codec) { - fprintf(stderr, "codec not found\n"); ++ fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); ++ if (!c) { ++ fprintf(stderr, "Could not allocate audio codec context\n"); ++ exit(1); ++ } + + /* put sample parameters */ + c->bit_rate = 64000; + + /* check that the encoder supports s16 pcm input */ + c->sample_fmt = AV_SAMPLE_FMT_S16; + if (!check_sample_fmt(codec, c->sample_fmt)) { - fprintf(stderr, "encoder does not support %s", ++ fprintf(stderr, "Encoder does not support sample format %s", + av_get_sample_fmt_name(c->sample_fmt)); + exit(1); + } + + /* select other audio parameters supported by the encoder */ + c->sample_rate = select_sample_rate(codec); + c->channel_layout = select_channel_layout(codec); + c->channels = av_get_channel_layout_nb_channels(c->channel_layout); + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "could not open codec\n"); ++ fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "wb"); + if (!f) { - fprintf(stderr, "could not open %s\n", filename); ++ fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + /* frame containing input raw audio */ + frame = av_frame_alloc(); + if (!frame) { - fprintf(stderr, "could not allocate audio frame\n"); ++ fprintf(stderr, "Could not allocate audio frame\n"); + exit(1); + } + + frame->nb_samples = c->frame_size; + frame->format = c->sample_fmt; + frame->channel_layout = c->channel_layout; + + /* the codec gives us the frame size, in samples, + * we calculate the size of the samples buffer in bytes */ + buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, + c->sample_fmt, 0); ++ if (buffer_size < 0) { ++ fprintf(stderr, "Could not get sample buffer size\n"); ++ exit(1); ++ } + samples = av_malloc(buffer_size); + if (!samples) { - fprintf(stderr, "could not allocate %d bytes for samples buffer\n", ++ fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", + buffer_size); + exit(1); + } + /* setup the data pointers in the AVFrame */ + ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (const uint8_t*)samples, buffer_size, 0); + if (ret < 0) { - fprintf(stderr, "could not setup audio frame\n"); ++ fprintf(stderr, "Could not setup audio frame\n"); + exit(1); + } + + /* encode a single tone sound */ + t = 0; + tincr = 2 * M_PI * 440.0 / c->sample_rate; - for(i=0;i<200;i++) { ++ for (i = 0; i < 200; i++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + for (j = 0; j < c->frame_size; j++) { + samples[2*j] = (int)(sin(t) * 10000); + + for (k = 1; k < c->channels; k++) + samples[2*j + k] = samples[2*j]; + t += tincr; + } + /* encode the samples */ + ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); + if (ret < 0) { - fprintf(stderr, "error encoding audio frame\n"); ++ fprintf(stderr, "Error encoding audio frame\n"); + exit(1); + } + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_packet_unref(&pkt); + } + } ++ ++ /* get the delayed frames */ ++ for (got_output = 1; got_output; i++) { ++ ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); ++ if (ret < 0) { ++ fprintf(stderr, "Error encoding frame\n"); ++ exit(1); ++ } ++ ++ if (got_output) { ++ fwrite(pkt.data, 1, pkt.size, f); ++ av_packet_unref(&pkt); ++ } ++ } + fclose(f); + + av_freep(&samples); + av_frame_free(&frame); + avcodec_free_context(&c); + } _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog
