mirror of
https://github.com/Swiftgram/Telegram-iOS.git
synced 2025-06-16 05:55:20 +00:00
Live muxer cleanup
This commit is contained in:
Isaac
parent
db1541b2c4
commit
33d4eef7a6
@ -19,8 +19,6 @@
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int *streams_list = NULL;
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int number_of_streams = 0;
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struct SwrContext *swr_ctx = NULL;
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in_filename = [path UTF8String];
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out_filename = [outPath UTF8String];
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@ -41,34 +39,6 @@
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goto end;
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}
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const AVCodec *aac_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
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if (!aac_codec) {
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fprintf(stderr, "Could not find AAC encoder\n");
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ret = AVERROR_UNKNOWN;
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goto end;
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}
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AVCodecContext *aac_codec_context = avcodec_alloc_context3(aac_codec);
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if (!aac_codec_context) {
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fprintf(stderr, "Could not allocate AAC codec context\n");
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ret = AVERROR_UNKNOWN;
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goto end;
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}
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const AVCodec *opus_decoder = avcodec_find_decoder(AV_CODEC_ID_AAC);
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if (!opus_decoder) {
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fprintf(stderr, "Could not find Opus decoder\n");
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ret = AVERROR_UNKNOWN;
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goto end;
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}
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AVCodecContext *opus_decoder_context = avcodec_alloc_context3(opus_decoder);
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if (!opus_decoder_context) {
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fprintf(stderr, "Could not allocate Opus decoder context\n");
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ret = AVERROR_UNKNOWN;
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goto end;
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}
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number_of_streams = input_format_context->nb_streams;
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streams_list = av_malloc_array(number_of_streams, sizeof(*streams_list));
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@ -76,15 +46,13 @@
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ret = AVERROR(ENOMEM);
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goto end;
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}
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bool hasAudio = false;
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for (i = 0; i < input_format_context->nb_streams; i++) {
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AVStream *out_stream;
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AVStream *in_stream = input_format_context->streams[i];
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AVCodecParameters *in_codecpar = in_stream->codecpar;
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if (/*in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO && */in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO) {
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if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO && in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO) {
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streams_list[i] = -1;
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continue;
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}
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@ -106,75 +74,24 @@
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out_stream->time_base = in_stream->time_base;
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out_stream->duration = in_stream->duration;
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} else if (in_codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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out_stream = avformat_new_stream(output_format_context, aac_codec);
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if (in_codecpar->codec_id != AV_CODEC_ID_AAC) {
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streams_list[i] = -1;
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continue;
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}
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out_stream = avformat_new_stream(output_format_context, NULL);
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if (!out_stream) {
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fprintf(stderr, "Failed allocating output stream\n");
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ret = AVERROR_UNKNOWN;
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goto end;
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}
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hasAudio = true;
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// Set the codec parameters for the AAC encoder
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aac_codec_context->sample_rate = in_codecpar->sample_rate;
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aac_codec_context->channel_layout = in_codecpar->channel_layout ? in_codecpar->channel_layout : AV_CH_LAYOUT_STEREO;
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aac_codec_context->channels = av_get_channel_layout_nb_channels(aac_codec_context->channel_layout);
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aac_codec_context->sample_fmt = aac_codec->sample_fmts ? aac_codec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP; // Use the first supported sample format
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aac_codec_context->bit_rate = 128000; // Set a default bitrate, you can adjust this as needed
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//aac_codec_context->time_base = (AVRational){1, 90000};
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ret = avcodec_open2(aac_codec_context, aac_codec, NULL);
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ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
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if (ret < 0) {
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fprintf(stderr, "Could not open AAC encoder\n");
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fprintf(stderr, "Failed to copy codec parameters\n");
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goto end;
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}
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ret = avcodec_parameters_from_context(out_stream->codecpar, aac_codec_context);
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if (ret < 0) {
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fprintf(stderr, "Failed initializing audio output stream\n");
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goto end;
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}
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out_stream->time_base = (AVRational){1, 90000};
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out_stream->duration = av_rescale_q(in_stream->duration, in_stream->time_base, out_stream->time_base);
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// Set up the Opus decoder context
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ret = avcodec_parameters_to_context(opus_decoder_context, in_codecpar);
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if (ret < 0) {
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fprintf(stderr, "Could not copy codec parameters to decoder context\n");
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goto end;
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}
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if (opus_decoder_context->channel_layout == 0) {
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opus_decoder_context->channel_layout = av_get_default_channel_layout(opus_decoder_context->channels);
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}
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ret = avcodec_open2(opus_decoder_context, opus_decoder, NULL);
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if (ret < 0) {
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fprintf(stderr, "Could not open Opus decoder\n");
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goto end;
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}
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// Reset the channel layout if it was unset before opening the codec
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if (opus_decoder_context->channel_layout == 0) {
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opus_decoder_context->channel_layout = av_get_default_channel_layout(opus_decoder_context->channels);
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}
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}
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}
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if (hasAudio) {
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// Set up the resampling context
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swr_ctx = swr_alloc_set_opts(NULL,
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aac_codec_context->channel_layout, aac_codec_context->sample_fmt, aac_codec_context->sample_rate,
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opus_decoder_context->channel_layout, opus_decoder_context->sample_fmt, opus_decoder_context->sample_rate,
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0, NULL);
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if (!swr_ctx) {
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fprintf(stderr, "Could not allocate resampler context\n");
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ret = AVERROR(ENOMEM);
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goto end;
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}
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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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goto end;
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out_stream->time_base = in_stream->time_base;
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out_stream->duration = in_stream->duration;
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}
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}
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@ -209,91 +126,18 @@
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out_stream = output_format_context->streams[packet.stream_index];
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if (in_stream->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
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ret = avcodec_send_packet(opus_decoder_context, &packet);
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packet.pts = av_rescale_q_rnd(packet.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX);
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packet.dts = av_rescale_q_rnd(packet.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX);
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packet.pts += (int64_t)(offsetSeconds * out_stream->time_base.den);
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packet.dts += (int64_t)(offsetSeconds * out_stream->time_base.den);
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packet.duration = av_rescale_q(packet.duration, in_stream->time_base, out_stream->time_base);
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packet.pos = -1;
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ret = av_interleaved_write_frame(output_format_context, &packet);
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if (ret < 0) {
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fprintf(stderr, "Error sending packet to decoder\n");
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fprintf(stderr, "Error muxing packet\n");
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av_packet_unref(&packet);
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continue;
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}
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AVFrame *frame = av_frame_alloc();
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ret = avcodec_receive_frame(opus_decoder_context, frame);
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if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) {
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fprintf(stderr, "Error receiving frame from decoder\n");
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av_frame_free(&frame);
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av_packet_unref(&packet);
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continue;
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}
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if (ret >= 0) {
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frame->pts = frame->best_effort_timestamp;
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AVFrame *resampled_frame = av_frame_alloc();
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resampled_frame->channel_layout = aac_codec_context->channel_layout;
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resampled_frame->sample_rate = aac_codec_context->sample_rate;
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resampled_frame->format = aac_codec_context->sample_fmt;
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resampled_frame->nb_samples = aac_codec_context->frame_size;
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if ((ret = av_frame_get_buffer(resampled_frame, 0)) < 0) {
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fprintf(stderr, "Could not allocate resampled frame buffer\n");
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av_frame_free(&resampled_frame);
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av_frame_free(&frame);
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av_packet_unref(&packet);
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continue;
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}
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memset(resampled_frame->data[0], 0, resampled_frame->nb_samples * 2 * 2);
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//arc4random_buf(resampled_frame->data[0], resampled_frame->nb_samples * 2 * 2);
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//memset(frame->data[0], 0, frame->nb_samples * 2 * 2);
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if ((ret = swr_convert(swr_ctx,
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resampled_frame->data, resampled_frame->nb_samples,
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(const uint8_t **)frame->data, frame->nb_samples)) < 0) {
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fprintf(stderr, "Error while converting\n");
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av_frame_free(&resampled_frame);
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av_frame_free(&frame);
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av_packet_unref(&packet);
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continue;
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}
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resampled_frame->pts = av_rescale_q(frame->pts, opus_decoder_context->time_base, aac_codec_context->time_base);
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ret = avcodec_send_frame(aac_codec_context, resampled_frame);
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if (ret < 0) {
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fprintf(stderr, "Error sending frame to encoder\n");
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av_frame_free(&resampled_frame);
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av_frame_free(&frame);
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av_packet_unref(&packet);
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continue;
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}
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AVPacket out_packet;
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av_init_packet(&out_packet);
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out_packet.data = NULL;
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out_packet.size = 0;
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ret = avcodec_receive_packet(aac_codec_context, &out_packet);
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if (ret >= 0) {
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out_packet.pts = av_rescale_q_rnd(packet.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX);
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out_packet.dts = av_rescale_q_rnd(packet.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX);
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out_packet.pts += (int64_t)(offsetSeconds * out_stream->time_base.den);
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out_packet.dts += (int64_t)(offsetSeconds * out_stream->time_base.den);
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out_packet.duration = av_rescale_q(out_packet.duration, aac_codec_context->time_base, out_stream->time_base);
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out_packet.stream_index = packet.stream_index;
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ret = av_interleaved_write_frame(output_format_context, &out_packet);
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if (ret < 0) {
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fprintf(stderr, "Error muxing packet\n");
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av_packet_unref(&out_packet);
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av_frame_free(&resampled_frame);
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av_frame_free(&frame);
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av_packet_unref(&packet);
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break;
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}
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av_packet_unref(&out_packet);
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}
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av_frame_free(&resampled_frame);
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av_frame_free(&frame);
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break;
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}
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} else {
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packet.pts = av_rescale_q_rnd(packet.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX);
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@ -317,16 +161,17 @@
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av_write_trailer(output_format_context);
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end:
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avformat_close_input(&input_format_context);
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if (input_format_context) {
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avformat_close_input(&input_format_context);
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}
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if (output_format_context && !(output_format_context->oformat->flags & AVFMT_NOFILE)) {
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avio_closep(&output_format_context->pb);
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}
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avformat_free_context(output_format_context);
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avcodec_free_context(&aac_codec_context);
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avcodec_free_context(&opus_decoder_context);
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av_freep(&streams_list);
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if (swr_ctx) {
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swr_free(&swr_ctx);
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if (output_format_context) {
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avformat_free_context(output_format_context);
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}
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if (streams_list) {
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av_freep(&streams_list);
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}
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if (ret < 0 && ret != AVERROR_EOF) {
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fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
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