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https://gitee.com/openharmony/third_party_ffmpeg
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examples/muxing: add support to audio resampling
Allows to encode to output in case the destination sample format is different from AV_SAMPLE_FMT_S16.
This commit is contained in:
Stefano Sabatini
parent
561e05136f
commit
2aa2b4ac65
@ -34,9 +34,11 @@
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#include <string.h>
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#include <math.h>
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#include <libavutil/opt.h>
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#include <libavutil/mathematics.h>
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#include <libavformat/avformat.h>
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#include <libswscale/swscale.h>
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#include <libswresample/swresample.h>
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/* 5 seconds stream duration */
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#define STREAM_DURATION 200.0
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@ -46,13 +48,6 @@
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static int sws_flags = SWS_BICUBIC;
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/**************************************************************/
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/* audio output */
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static float t, tincr, tincr2;
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static int16_t *samples;
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static int audio_input_frame_size;
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/* Add an output stream. */
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static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
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enum AVCodecID codec_id)
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@ -78,7 +73,7 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
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switch ((*codec)->type) {
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case AVMEDIA_TYPE_AUDIO:
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c->sample_fmt = AV_SAMPLE_FMT_S16;
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c->sample_fmt = AV_SAMPLE_FMT_FLTP;
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c->bit_rate = 64000;
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c->sample_rate = 44100;
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c->channels = 2;
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@ -126,8 +121,17 @@ static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
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/* audio output */
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static float t, tincr, tincr2;
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static int16_t *samples;
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static int audio_input_frame_size;
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static uint8_t **src_samples_data;
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static int src_samples_linesize;
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static int src_nb_samples;
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static int max_dst_nb_samples;
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uint8_t **dst_samples_data;
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int dst_samples_linesize;
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int dst_samples_size;
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struct SwrContext *swr_ctx = NULL;
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static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
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{
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@ -149,17 +153,51 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
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/* increment frequency by 110 Hz per second */
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tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
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if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
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audio_input_frame_size = 10000;
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else
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audio_input_frame_size = c->frame_size;
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samples = av_malloc(audio_input_frame_size *
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av_get_bytes_per_sample(c->sample_fmt) *
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c->channels);
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if (!samples) {
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fprintf(stderr, "Could not allocate audio samples buffer\n");
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src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
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10000 : c->frame_size;
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ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
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src_nb_samples, c->sample_fmt, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate source samples\n");
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exit(1);
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}
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/* create resampler context */
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if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
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swr_ctx = swr_alloc();
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if (!swr_ctx) {
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fprintf(stderr, "Could not allocate resampler context\n");
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exit(1);
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}
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/* set options */
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av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
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av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
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av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
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/* initialize the resampling context */
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if ((ret = swr_init(swr_ctx)) < 0) {
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fprintf(stderr, "Failed to initialize the resampling context\n");
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exit(1);
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}
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}
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/* compute the number of converted samples: buffering is avoided
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* ensuring that the output buffer will contain at least all the
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* converted input samples */
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max_dst_nb_samples = src_nb_samples;
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ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
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max_dst_nb_samples, c->sample_fmt, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate destination samples\n");
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exit(1);
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}
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dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
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c->sample_fmt, 0);
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}
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/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
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@ -184,18 +222,45 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
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AVCodecContext *c;
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AVPacket pkt = { 0 }; // data and size must be 0;
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AVFrame *frame = avcodec_alloc_frame();
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int got_packet, ret;
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int got_packet, ret, dst_nb_samples;
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av_init_packet(&pkt);
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c = st->codec;
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get_audio_frame(samples, audio_input_frame_size, c->channels);
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frame->nb_samples = audio_input_frame_size;
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get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
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/* convert samples from native format to destination codec format, using the resampler */
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if (swr_ctx) {
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/* compute destination number of samples */
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dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
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c->sample_rate, c->sample_rate, AV_ROUND_UP);
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if (dst_nb_samples > max_dst_nb_samples) {
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av_free(dst_samples_data[0]);
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ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
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dst_nb_samples, c->sample_fmt, 0);
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if (ret < 0)
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exit(1);
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max_dst_nb_samples = dst_nb_samples;
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dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
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c->sample_fmt, 0);
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}
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/* convert to destination format */
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ret = swr_convert(swr_ctx,
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dst_samples_data, dst_nb_samples,
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(const uint8_t **)src_samples_data, src_nb_samples);
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if (ret < 0) {
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fprintf(stderr, "Error while converting\n");
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exit(1);
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}
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} else {
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dst_samples_data[0] = src_samples_data[0];
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dst_nb_samples = src_nb_samples;
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}
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frame->nb_samples = dst_nb_samples;
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avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
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(uint8_t *)samples,
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audio_input_frame_size *
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av_get_bytes_per_sample(c->sample_fmt) *
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c->channels, 1);
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dst_samples_data[0], dst_samples_size, 0);
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ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
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if (ret < 0) {
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@ -221,8 +286,8 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
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static void close_audio(AVFormatContext *oc, AVStream *st)
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{
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avcodec_close(st->codec);
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av_free(samples);
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av_free(src_samples_data[0]);
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av_free(dst_samples_data[0]);
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}
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/**************************************************************/
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