diff options
Diffstat (limited to 'src/plugins/previewopus_extractor.c')
| -rw-r--r-- | src/plugins/previewopus_extractor.c | 108 |
1 files changed, 67 insertions, 41 deletions
diff --git a/src/plugins/previewopus_extractor.c b/src/plugins/previewopus_extractor.c index 45ea420..f137f38 100644 --- a/src/plugins/previewopus_extractor.c +++ b/src/plugins/previewopus_extractor.c @@ -44,7 +44,7 @@ #include <libavformat/avformat.h> #include <libavcodec/avcodec.h> #include <libswscale/swscale.h> -#include <libavresample/avresample.h> +#include <libswresample/swresample.h> /** @@ -300,7 +300,7 @@ init_input_frame (AVFrame **frame) static int init_resampler (AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - AVAudioResampleContext **resample_context) + SwrContext **resample_context) { /** * Only initialize the resampler if it is necessary, i.e., @@ -312,7 +312,7 @@ init_resampler (AVCodecContext *input_codec_context, int error; /** Create a resampler context for the conversion. */ - if (! (*resample_context = avresample_alloc_context ())) + if (! (*resample_context = swr_alloc ())) { #if DEBUG fprintf (stderr, "Could not allocate resample context\n"); @@ -326,28 +326,35 @@ init_resampler (AVCodecContext *input_codec_context, * Default channel layouts based on the number of channels * are assumed for simplicity (they are sometimes not detected * properly by the demuxer and/or decoder). - */av_opt_set_int (*resample_context, "in_channel_layout", + */ + av_opt_set_int (*resample_context, + "in_channel_layout", av_get_default_channel_layout ( input_codec_context->channels), 0); - av_opt_set_int (*resample_context, "out_channel_layout", + av_opt_set_int (*resample_context, + "out_channel_layout", av_get_default_channel_layout ( output_codec_context->channels), 0); - av_opt_set_int (*resample_context, "in_sample_rate", + av_opt_set_int (*resample_context, + "in_sample_rate", input_codec_context->sample_rate, 0); - av_opt_set_int (*resample_context, "out_sample_rate", + av_opt_set_int (*resample_context, + "out_sample_rate", output_codec_context->sample_rate, 0); - av_opt_set_int (*resample_context, "in_sample_fmt", + av_opt_set_int (*resample_context, + "in_sample_fmt", input_codec_context->sample_fmt, 0); - av_opt_set_int (*resample_context, "out_sample_fmt", + av_opt_set_int (*resample_context, + "out_sample_fmt", output_codec_context->sample_fmt, 0); /** Open the resampler with the specified parameters. */ - if ((error = avresample_open (*resample_context)) < 0) + if ((error = swr_init (*resample_context)) < 0) { #if DEBUG fprintf (stderr, "Could not open resample context\n"); #endif - avresample_free (resample_context); + swr_free (resample_context); return error; } } @@ -461,7 +468,8 @@ decode_audio_frame (AVFrame *frame, * The number of audio samples to be allocated is specified in frame_size. */ static int -init_converted_samples (uint8_t ***converted_input_samples, int*out_linesize, +init_converted_samples (uint8_t ***converted_input_samples, + int*out_linesize, AVCodecContext *output_codec_context, int frame_size) { @@ -484,7 +492,8 @@ init_converted_samples (uint8_t ***converted_input_samples, int*out_linesize, * Allocate memory for the samples of all channels in one consecutive * block for convenience. */ - if ((error = av_samples_alloc (*converted_input_samples, out_linesize, + if ((error = av_samples_alloc (*converted_input_samples, + out_linesize, output_codec_context->channels, frame_size, output_codec_context->sample_fmt, 0)) < 0) @@ -509,19 +518,26 @@ init_converted_samples (uint8_t ***converted_input_samples, int*out_linesize, */ static int convert_samples (uint8_t **input_data, - uint8_t **converted_data, const int in_sample, const int - out_sample, const int out_linesize, - AVAudioResampleContext *resample_context) + uint8_t **converted_data, + int in_sample, + int out_sample, + int out_linesize, + SwrContext *resample_context) { int error; /** Convert the samples using the resampler. */ - if ((error = avresample_convert (resample_context, converted_data, - out_linesize, - out_sample, input_data, 0, in_sample)) < 0) + if ((error = swr_convert (resample_context, + converted_data, + out_linesize, + out_sample, + input_data, + 0, + in_sample)) < 0) { #if DEBUG - fprintf (stderr, "Could not convert input samples (error '%s')\n", + fprintf (stderr, + "Could not convert input samples (error '%s')\n", get_error_text (error)); #endif return error; @@ -532,11 +548,13 @@ convert_samples (uint8_t **input_data, * Perform a sanity check so that the number of converted samples is * not greater than the number of samples to be converted. * If the sample rates differ, this case has to be handled differently - */if (avresample_available (resample_context)) + */ + if (avresample_available (resample_context)) { #if DEBUG - fprintf (stderr, "%i Converted samples left over\n",avresample_available ( - resample_context)); + fprintf (stderr, + "%i Converted samples left over\n", + avresample_available (resample_context)); #endif } @@ -588,7 +606,7 @@ read_decode_convert_and_store (AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - AVAudioResampleContext *resampler_context, int + SwrContext *resampler_context, int audio_stream_index, int *finished) { @@ -609,22 +627,25 @@ read_decode_convert_and_store (AVAudioFifo *fifo, } /** Decode one frame worth of audio samples. */ - if (decode_audio_frame (input_frame, input_format_context, - input_codec_context, audio_stream_index, - &data_present, finished)) + if (decode_audio_frame (input_frame, + input_format_context, + input_codec_context, + audio_stream_index, + &data_present, + finished)) { #if DEBUG fprintf (stderr, "Failed at decode audio\n"); #endif goto cleanup; - } /** * If we are at the end of the file and there are no more samples * in the decoder which are delayed, we are actually finished. * This must not be treated as an error. - */if (*finished && ! data_present) + */ + if (*finished && ! data_present) { ret = 0; #if DEBUG @@ -636,7 +657,7 @@ read_decode_convert_and_store (AVAudioFifo *fifo, if (data_present) { int out_linesize; - // FIX ME: I'm losing samples, but can't get it to work. + // FIXME: I'm losing samples, but can't get it to work. int out_samples = avresample_available (resampler_context) + avresample_get_delay (resampler_context) + input_frame->nb_samples; @@ -645,7 +666,8 @@ read_decode_convert_and_store (AVAudioFifo *fifo, // fprintf(stderr, "Input nbsamples %i out_samples: %i \n",input_frame->nb_samples,out_samples); /** Initialize the temporary storage for the converted input samples. */ - if (init_converted_samples (&converted_input_samples, &out_linesize, + if (init_converted_samples (&converted_input_samples, + &out_linesize, output_codec_context, out_samples)) { @@ -659,20 +681,23 @@ read_decode_convert_and_store (AVAudioFifo *fifo, * Convert the input samples to the desired output sample format. * This requires a temporary storage provided by converted_input_samples. */ - if (convert_samples (input_frame->extended_data, converted_input_samples, - input_frame->nb_samples, out_samples, out_linesize, + if (convert_samples (input_frame->extended_data, + converted_input_samples, + input_frame->nb_samples, + out_samples, + out_linesize, resampler_context)) { - - #if DEBUG - fprintf (stderr, "Failed at convert_samples, input frame %i \n", + fprintf (stderr, + "Failed at convert_samples, input frame %i \n", input_frame->nb_samples); #endif goto cleanup; } /** Add the converted input samples to the FIFO buffer for later processing. */ - if (add_samples_to_fifo (fifo, converted_input_samples, + if (add_samples_to_fifo (fifo, + converted_input_samples, out_samples)) { #if DEBUG @@ -881,7 +906,7 @@ extract_audio (struct EXTRACTOR_ExtractContext *ec) AVDictionary *options; AVFrame *frame; AVCodecContext*output_codec_context = NULL; - AVAudioResampleContext *resample_context = NULL; + SwrContext *resample_context = NULL; AVAudioFifo *fifo = NULL; int audio_stream_index; @@ -1045,7 +1070,8 @@ extract_audio (struct EXTRACTOR_ExtractContext *ec) * Since the decoder's and the encoder's frame size may differ, we * need to FIFO buffer to store as many frames worth of input samples * that they make up at least one frame worth of output samples. - */while ((av_audio_fifo_size (fifo) < output_frame_size)) + */ + while ((av_audio_fifo_size (fifo) < output_frame_size)) { /** * Decode one frame worth of audio samples, convert it to the @@ -1143,8 +1169,8 @@ cleanup: av_audio_fifo_free (fifo); if (resample_context) { - avresample_close (resample_context); - avresample_free (&resample_context); + swr_close (resample_context); + swr_free (&resample_context); } if (output_codec_context) avcodec_close (output_codec_context); |