mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
doc/examples: convert to new channel layout-API
Signed-off-by: James Almer <jamrial@gmail.com>
This commit is contained in:
parent
50e9e11316
commit
f5ef91e020
@ -97,7 +97,7 @@ static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
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exit(1);
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}
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for (i = 0; i < frame->nb_samples; i++)
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for (ch = 0; ch < dec_ctx->channels; ch++)
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for (ch = 0; ch < dec_ctx->ch_layout.nb_channels; ch++)
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fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
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}
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}
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@ -215,7 +215,7 @@ int main(int argc, char **argv)
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sfmt = av_get_packed_sample_fmt(sfmt);
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}
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n_channels = c->channels;
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n_channels = c->ch_layout.nb_channels;
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if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
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goto end;
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@ -345,7 +345,7 @@ int main (int argc, char **argv)
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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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int n_channels = audio_dec_ctx->ch_layout.nb_channels;
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const char *fmt;
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if (av_sample_fmt_is_planar(sfmt)) {
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@ -70,26 +70,25 @@ static int select_sample_rate(const AVCodec *codec)
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}
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/* select layout with the highest channel count */
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static int select_channel_layout(const AVCodec *codec)
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static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
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{
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const uint64_t *p;
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uint64_t best_ch_layout = 0;
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const AVChannelLayout *p, *best_ch_layout;
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int best_nb_channels = 0;
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if (!codec->channel_layouts)
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return AV_CH_LAYOUT_STEREO;
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if (!codec->ch_layouts)
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return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
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p = codec->channel_layouts;
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while (*p) {
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int nb_channels = av_get_channel_layout_nb_channels(*p);
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p = codec->ch_layouts;
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while (p->nb_channels) {
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int nb_channels = p->nb_channels;
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if (nb_channels > best_nb_channels) {
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best_ch_layout = *p;
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best_ch_layout = p;
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best_nb_channels = nb_channels;
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}
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p++;
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}
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return best_ch_layout;
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return av_channel_layout_copy(dst, best_ch_layout);
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}
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static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
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@ -164,8 +163,9 @@ int main(int argc, char **argv)
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/* select other audio parameters supported by the encoder */
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c->sample_rate = select_sample_rate(codec);
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c->channel_layout = select_channel_layout(codec);
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c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
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ret = select_channel_layout(codec, &c->ch_layout);
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if (ret < 0)
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exit(1);
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/* open it */
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if (avcodec_open2(c, codec, NULL) < 0) {
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@ -195,7 +195,9 @@ int main(int argc, char **argv)
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frame->nb_samples = c->frame_size;
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frame->format = c->sample_fmt;
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frame->channel_layout = c->channel_layout;
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ret = av_channel_layout_copy(&frame->ch_layout, &c->ch_layout);
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if (ret < 0)
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exit(1);
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/* allocate the data buffers */
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ret = av_frame_get_buffer(frame, 0);
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@ -218,7 +220,7 @@ int main(int argc, char **argv)
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for (j = 0; j < c->frame_size; j++) {
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samples[2*j] = (int)(sin(t) * 10000);
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for (k = 1; k < c->channels; k++)
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for (k = 1; k < c->ch_layout.nb_channels; k++)
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samples[2*j + k] = samples[2*j];
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t += tincr;
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}
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@ -55,7 +55,7 @@
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#define INPUT_SAMPLERATE 48000
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#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
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#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
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#define INPUT_CHANNEL_LAYOUT (AVChannelLayout)AV_CHANNEL_LAYOUT_5POINT0
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#define VOLUME_VAL 0.90
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@ -100,7 +100,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
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}
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/* Set the filter options through the AVOptions API. */
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av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
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av_channel_layout_describe(&INPUT_CHANNEL_LAYOUT, ch_layout, sizeof(ch_layout));
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av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
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av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
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av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
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@ -154,9 +154,8 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
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/* A third way of passing the options is in a string of the form
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* key1=value1:key2=value2.... */
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snprintf(options_str, sizeof(options_str),
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"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
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av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
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(uint64_t)AV_CH_LAYOUT_STEREO);
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"sample_fmts=%s:sample_rates=%d:channel_layouts=stereo",
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av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100);
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err = avfilter_init_str(aformat_ctx, options_str);
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if (err < 0) {
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av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
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@ -215,7 +214,7 @@ static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
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static int process_output(struct AVMD5 *md5, AVFrame *frame)
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{
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int planar = av_sample_fmt_is_planar(frame->format);
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int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
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int channels = frame->ch_layout.nb_channels;
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int planes = planar ? channels : 1;
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int bps = av_get_bytes_per_sample(frame->format);
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int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
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@ -248,7 +247,7 @@ static int get_input(AVFrame *frame, int frame_num)
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/* Set up the frame properties and allocate the buffer for the data. */
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frame->sample_rate = INPUT_SAMPLERATE;
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frame->format = INPUT_FORMAT;
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frame->channel_layout = INPUT_CHANNEL_LAYOUT;
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av_channel_layout_copy(&frame->ch_layout, &INPUT_CHANNEL_LAYOUT);
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frame->nb_samples = FRAME_SIZE;
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frame->pts = frame_num * FRAME_SIZE;
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@ -94,7 +94,6 @@ static int init_filters(const char *filters_descr)
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AVFilterInOut *outputs = avfilter_inout_alloc();
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AVFilterInOut *inputs = avfilter_inout_alloc();
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static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
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static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
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static const int out_sample_rates[] = { 8000, -1 };
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const AVFilterLink *outlink;
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AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
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@ -106,12 +105,13 @@ static int init_filters(const char *filters_descr)
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}
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/* buffer audio source: the decoded frames from the decoder will be inserted here. */
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if (!dec_ctx->channel_layout)
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dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
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snprintf(args, sizeof(args),
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"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
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if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
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av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
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ret = snprintf(args, sizeof(args),
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"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=",
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time_base.num, time_base.den, dec_ctx->sample_rate,
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av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
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av_get_sample_fmt_name(dec_ctx->sample_fmt));
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av_channel_layout_describe(&dec_ctx->ch_layout, args + ret, sizeof(args) - ret);
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ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
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args, NULL, filter_graph);
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if (ret < 0) {
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@ -134,7 +134,7 @@ static int init_filters(const char *filters_descr)
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goto end;
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}
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ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
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ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
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AV_OPT_SEARCH_CHILDREN);
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if (ret < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
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@ -185,7 +185,7 @@ static int init_filters(const char *filters_descr)
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/* Print summary of the sink buffer
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* Note: args buffer is reused to store channel layout string */
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outlink = buffersink_ctx->inputs[0];
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av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
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av_channel_layout_describe(&outlink->ch_layout, args, sizeof(args));
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av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
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(int)outlink->sample_rate,
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(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
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@ -200,7 +200,7 @@ end:
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static void print_frame(const AVFrame *frame)
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{
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const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
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const int n = frame->nb_samples * frame->ch_layout.nb_channels;
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const uint16_t *p = (uint16_t*)frame->data[0];
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const uint16_t *p_end = p + n;
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@ -170,16 +170,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
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c->sample_rate = 44100;
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}
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}
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c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
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c->channel_layout = AV_CH_LAYOUT_STEREO;
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if ((*codec)->channel_layouts) {
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c->channel_layout = (*codec)->channel_layouts[0];
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for (i = 0; (*codec)->channel_layouts[i]; i++) {
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if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
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c->channel_layout = AV_CH_LAYOUT_STEREO;
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}
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}
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c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
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av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
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ost->st->time_base = (AVRational){ 1, c->sample_rate };
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break;
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@ -224,7 +215,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
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/* audio output */
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static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
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uint64_t channel_layout,
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const AVChannelLayout *channel_layout,
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int sample_rate, int nb_samples)
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{
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AVFrame *frame = av_frame_alloc();
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@ -236,7 +227,7 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
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}
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frame->format = sample_fmt;
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frame->channel_layout = channel_layout;
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av_channel_layout_copy(&frame->ch_layout, channel_layout);
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frame->sample_rate = sample_rate;
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frame->nb_samples = nb_samples;
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@ -281,9 +272,9 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
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else
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nb_samples = c->frame_size;
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ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
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ost->frame = alloc_audio_frame(c->sample_fmt, &c->ch_layout,
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c->sample_rate, nb_samples);
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ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
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ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, &c->ch_layout,
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c->sample_rate, nb_samples);
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/* copy the stream parameters to the muxer */
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@ -301,10 +292,10 @@ static void open_audio(AVFormatContext *oc, const AVCodec *codec,
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}
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/* set options */
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av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
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av_opt_set_chlayout (ost->swr_ctx, "in_chlayout", &c->ch_layout, 0);
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av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
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av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
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av_opt_set_chlayout (ost->swr_ctx, "out_chlayout", &c->ch_layout, 0);
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av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
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av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
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@ -330,7 +321,7 @@ static AVFrame *get_audio_frame(OutputStream *ost)
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for (j = 0; j <frame->nb_samples; j++) {
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v = (int)(sin(ost->t) * 10000);
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for (i = 0; i < ost->enc->channels; i++)
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for (i = 0; i < ost->enc->ch_layout.nb_channels; i++)
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*q++ = v;
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ost->t += ost->tincr;
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ost->tincr += ost->tincr2;
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@ -80,7 +80,7 @@ static void fill_samples(double *dst, int nb_samples, int nb_channels, int sampl
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int main(int argc, char **argv)
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{
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int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
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AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_LAYOUT_SURROUND;
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int src_rate = 48000, dst_rate = 44100;
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uint8_t **src_data = NULL, **dst_data = NULL;
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int src_nb_channels = 0, dst_nb_channels = 0;
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@ -92,6 +92,7 @@ int main(int argc, char **argv)
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int dst_bufsize;
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const char *fmt;
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struct SwrContext *swr_ctx;
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char buf[64];
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double t;
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int ret;
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@ -120,11 +121,11 @@ int main(int argc, char **argv)
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}
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/* set options */
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av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
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av_opt_set_chlayout(swr_ctx, "in_chlayout", &src_ch_layout, 0);
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av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
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av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
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av_opt_set_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 0);
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av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
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@ -136,7 +137,7 @@ int main(int argc, char **argv)
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/* allocate source and destination samples buffers */
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src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
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src_nb_channels = src_ch_layout.nb_channels;
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ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
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src_nb_samples, src_sample_fmt, 0);
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if (ret < 0) {
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@ -151,7 +152,7 @@ int main(int argc, char **argv)
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av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
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/* buffer is going to be directly written to a rawaudio file, no alignment */
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dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
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dst_nb_channels = dst_ch_layout.nb_channels;
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ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
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dst_nb_samples, dst_sample_fmt, 0);
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if (ret < 0) {
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@ -194,9 +195,10 @@ int main(int argc, char **argv)
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if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
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goto end;
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av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
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fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
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"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
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fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
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"ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
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fmt, buf, dst_nb_channels, dst_rate, dst_filename);
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end:
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fclose(dst_file);
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@ -200,8 +200,7 @@ static int open_output_file(const char *filename,
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/* Set the basic encoder parameters.
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* The input file's sample rate is used to avoid a sample rate conversion. */
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avctx->channels = OUTPUT_CHANNELS;
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avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
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av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
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avctx->sample_rate = input_codec_context->sample_rate;
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avctx->sample_fmt = output_codec->sample_fmts[0];
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avctx->bit_rate = OUTPUT_BIT_RATE;
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@ -290,21 +289,18 @@ static int init_resampler(AVCodecContext *input_codec_context,
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/*
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* Create a resampler context for the conversion.
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* Set the conversion parameters.
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* Default channel layouts based on the number of channels
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* are assumed for simplicity (they are sometimes not detected
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* properly by the demuxer and/or decoder).
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*/
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*resample_context = swr_alloc_set_opts(NULL,
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av_get_default_channel_layout(output_codec_context->channels),
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error = swr_alloc_set_opts2(resample_context,
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&output_codec_context->ch_layout,
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output_codec_context->sample_fmt,
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output_codec_context->sample_rate,
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av_get_default_channel_layout(input_codec_context->channels),
|
||||
&input_codec_context->ch_layout,
|
||||
input_codec_context->sample_fmt,
|
||||
input_codec_context->sample_rate,
|
||||
0, NULL);
|
||||
if (!*resample_context) {
|
||||
if (error < 0) {
|
||||
fprintf(stderr, "Could not allocate resample context\n");
|
||||
return AVERROR(ENOMEM);
|
||||
return error;
|
||||
}
|
||||
/*
|
||||
* Perform a sanity check so that the number of converted samples is
|
||||
@ -332,7 +328,7 @@ static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
|
||||
{
|
||||
/* Create the FIFO buffer based on the specified output sample format. */
|
||||
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
|
||||
output_codec_context->channels, 1))) {
|
||||
output_codec_context->ch_layout.nb_channels, 1))) {
|
||||
fprintf(stderr, "Could not allocate FIFO\n");
|
||||
return AVERROR(ENOMEM);
|
||||
}
|
||||
@ -450,7 +446,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
|
||||
* Each pointer will later point to the audio samples of the corresponding
|
||||
* channels (although it may be NULL for interleaved formats).
|
||||
*/
|
||||
if (!(*converted_input_samples = calloc(output_codec_context->channels,
|
||||
if (!(*converted_input_samples = calloc(output_codec_context->ch_layout.nb_channels,
|
||||
sizeof(**converted_input_samples)))) {
|
||||
fprintf(stderr, "Could not allocate converted input sample pointers\n");
|
||||
return AVERROR(ENOMEM);
|
||||
@ -459,7 +455,7 @@ static int init_converted_samples(uint8_t ***converted_input_samples,
|
||||
/* Allocate memory for the samples of all channels in one consecutive
|
||||
* block for convenience. */
|
||||
if ((error = av_samples_alloc(*converted_input_samples, NULL,
|
||||
output_codec_context->channels,
|
||||
output_codec_context->ch_layout.nb_channels,
|
||||
frame_size,
|
||||
output_codec_context->sample_fmt, 0)) < 0) {
|
||||
fprintf(stderr,
|
||||
@ -633,7 +629,7 @@ static int init_output_frame(AVFrame **frame,
|
||||
* Default channel layouts based on the number of channels
|
||||
* are assumed for simplicity. */
|
||||
(*frame)->nb_samples = frame_size;
|
||||
(*frame)->channel_layout = output_codec_context->channel_layout;
|
||||
av_channel_layout_copy(&(*frame)->ch_layout, &output_codec_context->ch_layout);
|
||||
(*frame)->format = output_codec_context->sample_fmt;
|
||||
(*frame)->sample_rate = output_codec_context->sample_rate;
|
||||
|
||||
|
@ -175,8 +175,9 @@ static int open_output_file(const char *filename)
|
||||
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
|
||||
} else {
|
||||
enc_ctx->sample_rate = dec_ctx->sample_rate;
|
||||
enc_ctx->channel_layout = dec_ctx->channel_layout;
|
||||
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
|
||||
ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
/* take first format from list of supported formats */
|
||||
enc_ctx->sample_fmt = encoder->sample_fmts[0];
|
||||
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
|
||||
@ -289,6 +290,7 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
|
||||
goto end;
|
||||
}
|
||||
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
|
||||
char buf[64];
|
||||
buffersrc = avfilter_get_by_name("abuffer");
|
||||
buffersink = avfilter_get_by_name("abuffersink");
|
||||
if (!buffersrc || !buffersink) {
|
||||
@ -297,14 +299,14 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
|
||||
goto end;
|
||||
}
|
||||
|
||||
if (!dec_ctx->channel_layout)
|
||||
dec_ctx->channel_layout =
|
||||
av_get_default_channel_layout(dec_ctx->channels);
|
||||
if (dec_ctx->ch_layout.order == AV_CHANNEL_ORDER_UNSPEC)
|
||||
av_channel_layout_default(&dec_ctx->ch_layout, dec_ctx->ch_layout.nb_channels);
|
||||
av_channel_layout_describe(&dec_ctx->ch_layout, buf, sizeof(buf));
|
||||
snprintf(args, sizeof(args),
|
||||
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
|
||||
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=%s",
|
||||
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
|
||||
av_get_sample_fmt_name(dec_ctx->sample_fmt),
|
||||
dec_ctx->channel_layout);
|
||||
buf);
|
||||
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
|
||||
args, NULL, filter_graph);
|
||||
if (ret < 0) {
|
||||
@ -327,9 +329,9 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
|
||||
goto end;
|
||||
}
|
||||
|
||||
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
|
||||
(uint8_t*)&enc_ctx->channel_layout,
|
||||
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
|
||||
av_channel_layout_describe(&enc_ctx->ch_layout, buf, sizeof(buf));
|
||||
ret = av_opt_set(buffersink_ctx, "ch_layouts",
|
||||
buf, AV_OPT_SEARCH_CHILDREN);
|
||||
if (ret < 0) {
|
||||
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
|
||||
goto end;
|
||||
|
Loading…
Reference in New Issue
Block a user