diff options
Diffstat (limited to 'codec.c')
| -rw-r--r-- | codec.c | 112 |
1 files changed, 100 insertions, 12 deletions
@@ -1,7 +1,7 @@ /// /// @file codec.c @brief Codec functions /// -/// Copyright (c) 2009 - 2013 by Johns. All Rights Reserved. +/// Copyright (c) 2009 - 2014 by Johns. All Rights Reserved. /// /// Contributor(s): /// @@ -38,8 +38,12 @@ #define USE_AC3_DRIFT_CORRECTION /// use ffmpeg libswresample API (autodected, Makefile) #define noUSE_SWRESAMPLE + /// use libav libavresample API (autodected, Makefile) +#define noUSE_AVRESAMPLE #include <stdio.h> +#include <stdlib.h> +#include <string.h> #include <unistd.h> #ifdef __FreeBSD__ #include <sys/endian.h> @@ -56,6 +60,7 @@ #include <alsa/iatomic.h> #include <libavcodec/avcodec.h> +#include <libavutil/mem.h> // support old ffmpeg versions <1.0 #if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,18,102) #define AVCodecID CodecID @@ -70,6 +75,10 @@ #ifdef USE_SWRESAMPLE #include <libswresample/swresample.h> #endif +#ifdef USE_AVRESAMPLE +#include <libavresample/avresample.h> +#include <libavutil/opt.h> +#endif #ifndef __USE_GNU #define __USE_GNU @@ -201,7 +210,9 @@ static int Codec_get_buffer(AVCodecContext * video_ctx, AVFrame * frame) //Debug(3, "codec: use surface %#010x\n", surface); +#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(52,48,101) frame->type = FF_BUFFER_TYPE_USER; +#endif #if LIBAVCODEC_VERSION_INT <= AV_VERSION_INT(53,46,0) frame->age = 256 * 256 * 256 * 64; #endif @@ -228,7 +239,9 @@ static int Codec_get_buffer(AVCodecContext * video_ctx, AVFrame * frame) //Debug(3, "codec: use surface %#010x\n", surface); +#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(52,48,101) frame->type = FF_BUFFER_TYPE_USER; +#endif #if LIBAVCODEC_VERSION_INT <= AV_VERSION_INT(53,46,0) frame->age = 256 * 256 * 256 * 64; #endif @@ -664,15 +677,18 @@ struct _audio_decoder_ int HwSampleRate; ///< hw sample rate int HwChannels; ///< hw channels -#ifndef USE_SWRESAMPLE - ReSampleContext *ReSample; ///< audio resampling context +#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE) + ReSampleContext *ReSample; ///< old resampling context #endif #ifdef USE_SWRESAMPLE #if LIBSWRESAMPLE_VERSION_INT < AV_VERSION_INT(0, 15, 100) - struct SwrContext *Resample; ///< audio software resample context + struct SwrContext *Resample; ///< ffmpeg software resample context #else - SwrContext *Resample; ///< audio software resample context + SwrContext *Resample; ///< ffmpeg software resample context +#endif #endif +#ifdef USE_AVRESAMPLE + AVAudioResampleContext *Resample; ///< libav software resample context #endif uint16_t Spdif[24576 / 2]; ///< SPDIF output buffer @@ -687,7 +703,7 @@ struct _audio_decoder_ int DriftCorr; ///< audio drift correction value int DriftFrac; ///< audio drift fraction for ac3 -#ifndef USE_SWRESAMPLE +#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE) struct AVResampleContext *AvResample; ///< second audio resample context #define MAX_CHANNELS 8 ///< max number of channels supported int16_t *Buffer[MAX_CHANNELS]; ///< deinterleave sample buffers @@ -785,7 +801,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name, AV_CH_LAYOUT_STEREO_DOWNMIX; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,61,100) - // this has no effect + // this has no effect (with ffmpeg and libav) // audio_decoder->AudioCtx->request_sample_fmt = AV_SAMPLE_FMT_S16; #endif pthread_mutex_lock(&CodecLockMutex); @@ -834,7 +850,7 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name, void CodecAudioClose(AudioDecoder * audio_decoder) { // FIXME: output any buffered data -#ifndef USE_SWRESAMPLE +#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE) if (audio_decoder->AvResample) { int ch; @@ -860,6 +876,11 @@ void CodecAudioClose(AudioDecoder * audio_decoder) swr_free(&audio_decoder->Resample); } #endif +#ifdef USE_AVRESAMPLE + if (audio_decoder->Resample) { + avresample_free(&audio_decoder->Resample); + } +#endif if (audio_decoder->AudioCtx) { pthread_mutex_lock(&CodecLockMutex); avcodec_close(audio_decoder->AudioCtx); @@ -1155,7 +1176,7 @@ static int CodecAudioPassthroughHelper(AudioDecoder * audio_decoder, return 0; } -#ifndef USE_SWRESAMPLE +#if !defined(USE_SWRESAMPLE) && !defined(USE_AVRESAMPLE) /** ** Set/update audio pts clock. @@ -1557,7 +1578,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) #endif -#ifdef USE_SWRESAMPLE +#if defined(USE_SWRESAMPLE) || defined(USE_AVRESAMPLE) /** ** Set/update audio pts clock. @@ -1646,6 +1667,7 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts) } } +#ifdef USE_SWRESAMPLE if (audio_decoder->Resample && audio_decoder->DriftCorr) { int distance; @@ -1660,6 +1682,18 @@ static void CodecAudioSetClock(AudioDecoder * audio_decoder, int64_t pts) Debug(3, "codec/audio: swr_set_compensation failed\n"); } } +#endif +#ifdef USE_AVRESAMPLE + if (audio_decoder->Resample && audio_decoder->DriftCorr) { + int distance; + + distance = (pts_diff * audio_decoder->HwSampleRate) / (900 * 1000); + if (avresample_set_compensation(audio_decoder->Resample, + audio_decoder->DriftCorr / 10, distance)) { + Debug(3, "codec/audio: swr_set_compensation failed\n"); + } + } +#endif if (1) { static int c; @@ -1699,6 +1733,7 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder) } #endif +#ifdef USE_SWRESAMPLE audio_decoder->Resample = swr_alloc_set_opts(audio_decoder->Resample, audio_ctx->channel_layout, AV_SAMPLE_FMT_S16, audio_decoder->HwSampleRate, @@ -1709,6 +1744,33 @@ static void CodecAudioUpdateFormat(AudioDecoder * audio_decoder) } else { Error(_("codec/audio: can't setup resample\n")); } +#endif +#ifdef USE_AVRESAMPLE + if (!(audio_decoder->Resample = avresample_alloc_context())) { + Error(_("codec/audio: can't setup resample\n")); + return; + } + + av_opt_set_int(audio_decoder->Resample, "in_channel_layout", + audio_ctx->channel_layout, 0); + av_opt_set_int(audio_decoder->Resample, "in_sample_fmt", + audio_ctx->sample_fmt, 0); + av_opt_set_int(audio_decoder->Resample, "in_sample_rate", + audio_ctx->sample_rate, 0); + av_opt_set_int(audio_decoder->Resample, "out_channel_layout", + audio_ctx->channel_layout, 0); + av_opt_set_int(audio_decoder->Resample, "out_sample_fmt", + AV_SAMPLE_FMT_S16, 0); + av_opt_set_int(audio_decoder->Resample, "out_sample_rate", + audio_decoder->HwSampleRate, 0); + + if (avresample_open(audio_decoder->Resample)) { + avresample_free(&audio_decoder->Resample); + audio_decoder->Resample = NULL; + Error(_("codec/audio: can't open resample\n")); + return; + } +#endif } /** @@ -1731,7 +1793,9 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) audio_ctx = audio_decoder->AudioCtx; // FIXME: don't need to decode pass-through codecs - frame.data[0] = NULL; + // libav needs memset, frame.data[0] = NULL; + memset(&frame, 0, sizeof(frame)); + got_frame = 0; n = avcodec_decode_audio4(audio_ctx, &frame, &got_frame, (AVPacket *) avpkt); if (n != avpkt->size) { @@ -1785,7 +1849,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) "codec/audio: channels %d samples %d plane %d data %d\n", audio_ctx->channels, frame.nb_samples, plane_sz, data_sz); } - +#ifdef USE_SWRESAMPLE if (audio_decoder->Resample) { uint8_t outbuf[8192 * 2 * 8]; uint8_t *out[1]; @@ -1804,6 +1868,30 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt) } return; } +#endif + +#ifdef USE_AVRESAMPLE + if (audio_decoder->Resample) { + uint8_t outbuf[8192 * 2 * 8]; + uint8_t *out[1]; + + out[0] = outbuf; + n = avresample_convert(audio_decoder->Resample, out, 0, + sizeof(outbuf) / (2 * audio_decoder->HwChannels), + (uint8_t **) frame.extended_data, 0, frame.nb_samples); + // FIXME: set out_linesize, in_linesize correct + if (n > 0) { + if (!(audio_decoder->Passthrough & CodecPCM)) { + CodecReorderAudioFrame((int16_t *) outbuf, + n * 2 * audio_decoder->HwChannels, + audio_decoder->HwChannels); + } + AudioEnqueue(outbuf, n * 2 * audio_decoder->HwChannels); + } + return; + } +#endif + #ifdef DEBUG // should be never reached fprintf(stderr, "oops\n"); |