work on audio encoding
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@ -206,6 +206,13 @@ Debug(2, "Have audio_output_context");
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av_strerror(ret, error_buffer, sizeof(error_buffer));
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Fatal( "could not open codec (%d) (%s)\n", ret, error_buffer );
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} else {
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/** Create the FIFO buffer based on the specified output sample format. */
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if (!(fifo = av_audio_fifo_alloc(audio_output_context->sample_fmt,
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audio_output_context->channels, 1))) {
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Error("Could not allocate FIFO\n");
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return;
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}
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output_frame_size = audio_output_context->frame_size;
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Debug(2, "Success opening AAC codec");
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}
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av_dict_free(&opts);
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@ -514,60 +521,101 @@ if ( 0 ) {
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zm_av_unref_packet(&opkt);
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return 0;
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}
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if ( data_present ) {
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/** Create a new frame to store the audio samples. */
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if (!(output_frame = av_frame_alloc())) {
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uint8_t **converted_input_samples = NULL;
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/**
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* Allocate as many pointers as there are audio channels.
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* Each pointer will later point to the audio samples of the corresponding
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* channels (although it may be NULL for interleaved formats).
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*/
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if (!(converted_input_samples = calloc( audio_output_context->channels, sizeof(*converted_input_samples)))) {
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Error( "Could not allocate converted input sample pointers\n");
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return 0;
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}
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/**
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* Allocate memory for the samples of all channels in one consecutive
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* block for convenience.
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*/
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if ((ret = av_samples_alloc(converted_input_samples, NULL,
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audio_output_context->channels,
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frame_size,
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audio_output_context->sample_fmt, 0)) < 0) {
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Error( "Could not allocate converted input samples (error '%s')\n",
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av_make_error_string(ret).c_str() );
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av_freep(&(converted_input_samples)[0]);
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free(*converted_input_samples);
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return 0;
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}
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if ((ret = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
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Error( "Could not reallocate FIFO\n");
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return 0;
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}
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/** Store the new samples in the FIFO buffer. */
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if (av_audio_fifo_write(fifo, (void **)converted_input_samples, frame_size) < frame_size) {
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Error( "Could not write data to FIFO\n");
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return 0;
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}
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/** Create a new frame to store the audio samples. */
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if (!(output_frame = av_frame_alloc())) {
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Error("Could not allocate output frame");
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av_frame_free(&input_frame);
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zm_av_unref_packet(&opkt);
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return 0;
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} else {
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Debug(2, "Got output frame alloc");
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}
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/**
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* Set the frame's parameters, especially its size and format.
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* av_frame_get_buffer needs this to allocate memory for the
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* audio samples of the frame.
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* Default channel layouts based on the number of channels
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* are assumed for simplicity.
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*/
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output_frame->nb_samples = audio_stream->codec->frame_size;
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output_frame->channel_layout = audio_output_context->channel_layout;
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output_frame->channels = audio_output_context->channels;
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output_frame->format = audio_output_context->sample_fmt;
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output_frame->sample_rate = audio_output_context->sample_rate;
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/**
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* Allocate the samples of the created frame. This call will make
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* sure that the audio frame can hold as many samples as specified.
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*/
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Debug(2, "getting buffer");
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if (( ret = av_frame_get_buffer( output_frame, 0)) < 0) {
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} else {
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Debug(2, "Got output frame alloc");
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}
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/**
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* Set the frame's parameters, especially its size and format.
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* av_frame_get_buffer needs this to allocate memory for the
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* audio samples of the frame.
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* Default channel layouts based on the number of channels
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* are assumed for simplicity.
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*/
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output_frame->nb_samples = audio_stream->codec->frame_size;
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output_frame->channel_layout = audio_output_context->channel_layout;
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output_frame->channels = audio_output_context->channels;
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output_frame->format = audio_output_context->sample_fmt;
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output_frame->sample_rate = audio_output_context->sample_rate;
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/**
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* Allocate the samples of the created frame. This call will make
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* sure that the audio frame can hold as many samples as specified.
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*/
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Debug(2, "getting buffer");
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if (( ret = av_frame_get_buffer( output_frame, 0)) < 0) {
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Error( "Couldnt allocate output frame buffer samples (error '%s')",
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av_make_error_string(ret).c_str() );
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av_make_error_string(ret).c_str() );
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Error("Frame: samples(%d) layout (%d) format(%d) rate(%d)", output_frame->nb_samples,
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output_frame->channel_layout, output_frame->format , output_frame->sample_rate
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);
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output_frame->channel_layout, output_frame->format , output_frame->sample_rate
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);
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av_frame_free(&input_frame);
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av_frame_free(&output_frame);
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zm_av_unref_packet(&opkt);
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return 0;
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}
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}
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/** Set a timestamp based on the sample rate for the container. */
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if (output_frame) {
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/** Set a timestamp based on the sample rate for the container. */
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if (output_frame) {
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output_frame->pts = opkt.pts;
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}
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/**
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* Encode the audio frame and store it in the temporary packet.
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* The output audio stream encoder is used to do this.
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*/
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if (( ret = avcodec_encode_audio2( audio_output_context, &opkt,
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input_frame, &data_present )) < 0) {
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}
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/**
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* Encode the audio frame and store it in the temporary packet.
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* The output audio stream encoder is used to do this.
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*/
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if (( ret = avcodec_encode_audio2( audio_output_context, &opkt,
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input_frame, &data_present )) < 0) {
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Error( "Could not encode frame (error '%s')",
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av_make_error_string(ret).c_str());
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av_make_error_string(ret).c_str());
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zm_av_unref_packet(&opkt);
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return 0;
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}
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}
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} else {
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Debug(2, "Not data present" );
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} // end if data_present
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}
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} else {
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opkt.data = ipkt->data;
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@ -2,6 +2,7 @@
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#define ZM_VIDEOSTORE_H
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#include "zm_ffmpeg.h"
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#include "libavutil/audio_fifo.h"
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#if HAVE_LIBAVCODEC
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@ -19,6 +20,8 @@ private:
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AVCodec *audio_output_codec;
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AVCodecContext *audio_output_context;
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int data_present;
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AVAudioFifo *fifo;
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int output_frame_size;
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const char *filename;
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const char *format;
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