diff options
| -rw-r--r-- | intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp | 49 | ||||
| -rw-r--r-- | intern/ffmpeg/ffmpeg_compat.h | 17 | ||||
| -rw-r--r-- | source/blender/blenkernel/intern/writeffmpeg.c | 51 |
3 files changed, 112 insertions, 5 deletions
diff --git a/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp b/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp index 408a4e56f1c..3d830df83e8 100644 --- a/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp +++ b/intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp @@ -43,6 +43,7 @@ extern "C" { int AUD_FFMPEGReader::decode(AVPacket& packet, AUD_Buffer& buffer) { +#ifdef FFMPEG_HAVE_DECODE_AUDIO4 AVFrame* frame = NULL; int got_frame; int read_length; @@ -101,6 +102,54 @@ int AUD_FFMPEGReader::decode(AVPacket& packet, AUD_Buffer& buffer) av_free(frame); return buf_pos; +#else + // save packet parameters + uint8_t *audio_pkg_data = packet.data; + int audio_pkg_size = packet.size; + + int buf_size = buffer.getSize(); + int buf_pos = 0; + + int read_length, data_size; + + AVPacket tmp_pkt; + + av_init_packet(&tmp_pkt); + + // as long as there is still data in the package + while(audio_pkg_size > 0) + { + // resize buffer if needed + if(buf_size - buf_pos < AVCODEC_MAX_AUDIO_FRAME_SIZE) + { + buffer.resize(buf_size + AVCODEC_MAX_AUDIO_FRAME_SIZE, true); + buf_size += AVCODEC_MAX_AUDIO_FRAME_SIZE; + } + + // read samples from the packet + data_size = buf_size - buf_pos; + + tmp_pkt.data = audio_pkg_data; + tmp_pkt.size = audio_pkg_size; + + read_length = avcodec_decode_audio3( + m_codecCtx, + (int16_t*)(((data_t*)buffer.getBuffer()) + buf_pos), + &data_size, &tmp_pkt); + + // read error, next packet! + if(read_length < 0) + break; + + buf_pos += data_size; + + // move packet parameters + audio_pkg_data += read_length; + audio_pkg_size -= read_length; + } + + return buf_pos; +#endif } static const char* streaminfo_error = "AUD_FFMPEGReader: Stream info couldn't " diff --git a/intern/ffmpeg/ffmpeg_compat.h b/intern/ffmpeg/ffmpeg_compat.h index e072796abc4..0f97d1d42cc 100644 --- a/intern/ffmpeg/ffmpeg_compat.h +++ b/intern/ffmpeg/ffmpeg_compat.h @@ -76,9 +76,26 @@ #define FFMPEG_HAVE_CANON_H264_RESOLUTION_FIX #endif +#if ((LIBAVCODEC_VERSION_MAJOR > 53) || (LIBAVCODEC_VERSION_MAJOR >= 53) && (LIBAVCODEC_VERSION_MINOR >= 60)) +#define FFMPEG_HAVE_ENCODE_AUDIO2 +#endif + +#if ((LIBAVCODEC_VERSION_MAJOR > 53) || (LIBAVCODEC_VERSION_MAJOR >= 53) && (LIBAVCODEC_VERSION_MINOR >= 42)) +#define FFMPEG_HAVE_DECODE_AUDIO4 +#endif + #if ((LIBAVUTIL_VERSION_MAJOR > 51) || (LIBAVUTIL_VERSION_MAJOR == 51) && (LIBAVUTIL_VERSION_MINOR >= 32)) #define FFMPEG_FFV1_ALPHA_SUPPORTED #define FFMPEG_SAMPLE_FMT_S16P_SUPPORTED +#else + +static inline +int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt) +{ + /* no planar formats in FFmpeg < 0.9 */ + return 0; +} + #endif static inline diff --git a/source/blender/blenkernel/intern/writeffmpeg.c b/source/blender/blenkernel/intern/writeffmpeg.c index 404be7d6857..6f806903554 100644 --- a/source/blender/blenkernel/intern/writeffmpeg.c +++ b/source/blender/blenkernel/intern/writeffmpeg.c @@ -90,6 +90,10 @@ static int video_buffersize = 0; static uint8_t *audio_input_buffer = 0; static uint8_t *audio_deinterleave_buffer = 0; static int audio_input_samples = 0; +#ifndef FFMPEG_HAVE_ENCODE_AUDIO2 +static uint8_t *audio_output_buffer = 0; +static int audio_outbuf_size = 0; +#endif static double audio_time = 0.0f; static bool audio_deinterleave = false; static int audio_sample_size = 0; @@ -123,7 +127,7 @@ static int write_audio_frame(void) { AVCodecContext *c = NULL; AVPacket pkt; - AVFrame *frame; + AVFrame *frame = NULL; int got_output = 0; c = audio_stream->codec; @@ -132,6 +136,10 @@ static int write_audio_frame(void) pkt.size = 0; pkt.data = NULL; + AUD_readDevice(audio_mixdown_device, audio_input_buffer, audio_input_samples); + audio_time += (double) audio_input_samples / (double) c->sample_rate; + +#ifdef FFMPEG_HAVE_ENCODE_AUDIO2 frame = avcodec_alloc_frame(); frame->nb_samples = audio_input_samples; frame->format = c->sample_fmt; @@ -139,9 +147,6 @@ static int write_audio_frame(void) frame->channel_layout = c->channel_layout; #endif - AUD_readDevice(audio_mixdown_device, audio_input_buffer, audio_input_samples); - audio_time += (double) audio_input_samples / (double) c->sample_rate; - if (audio_deinterleave) { int channel, i; uint8_t *temp; @@ -166,6 +171,22 @@ static int write_audio_frame(void) return -1; } + if (!got_output) { + avcodec_free_frame(&frame); + return 0; + } +#else + pkt.size = avcodec_encode_audio(c, audio_output_buffer, audio_outbuf_size, (short *) audio_input_buffer); + + if (pkt.size < 0) { + // XXX error("Error writing audio packet"); + return -1; + } + + pkt.data = audio_output_buffer; + got_output = 1; +#endif + if (got_output) { if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE) { pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base, audio_stream->time_base); @@ -178,13 +199,16 @@ static int write_audio_frame(void) if (av_interleaved_write_frame(outfile, &pkt) != 0) { fprintf(stderr, "Error writing audio packet!\n"); + if (frame) + avcodec_free_frame(&frame); return -1; } av_free_packet(&pkt); } - avcodec_free_frame(&frame); + if (frame) + avcodec_free_frame(&frame); return 0; } @@ -718,6 +742,10 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex st->codec->time_base.num = 1; st->codec->time_base.den = st->codec->sample_rate; +#ifndef FFMPEG_HAVE_ENCODE_AUDIO2 + audio_outbuf_size = FF_MIN_BUFFER_SIZE; +#endif + if (c->frame_size == 0) // used to be if((c->codec_id >= CODEC_ID_PCM_S16LE) && (c->codec_id <= CODEC_ID_PCM_DVD)) // not sure if that is needed anymore, so let's try out if there are any @@ -725,6 +753,10 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex audio_input_samples = FF_MIN_BUFFER_SIZE * 8 / c->bits_per_coded_sample / c->channels; else { audio_input_samples = c->frame_size; +#ifndef FFMPEG_HAVE_ENCODE_AUDIO2 + if (c->frame_size * c->channels * sizeof(int16_t) * 4 > audio_outbuf_size) + audio_outbuf_size = c->frame_size * c->channels * sizeof(int16_t) * 4; +#endif } audio_deinterleave = av_sample_fmt_is_planar(c->sample_fmt); @@ -732,6 +764,9 @@ static AVStream *alloc_audio_stream(RenderData *rd, int codec_id, AVFormatContex audio_sample_size = av_get_bytes_per_sample(c->sample_fmt); audio_input_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * audio_sample_size); +#ifndef FFMPEG_HAVE_ENCODE_AUDIO2 + audio_output_buffer = (uint8_t *) av_malloc(audio_outbuf_size); +#endif if (audio_deinterleave) audio_deinterleave_buffer = (uint8_t *) av_malloc(audio_input_samples * c->channels * audio_sample_size); @@ -1182,6 +1217,12 @@ static void end_ffmpeg_impl(int is_autosplit) av_free(audio_input_buffer); audio_input_buffer = 0; } +#ifndef FFMPEG_HAVE_ENCODE_AUDIO2 + if (audio_output_buffer) { + av_free(audio_output_buffer); + audio_output_buffer = 0; + } +#endif if (audio_deinterleave_buffer) { av_free(audio_deinterleave_buffer); |