bool did_something = false;
if (video) {
- AVPacket packet;
- av_init_packet (&packet);
- packet.data = nullptr;
- packet.size = 0;
- if (decode_and_process_video_packet(&packet)) {
+ if (decode_and_process_video_packet(nullptr)) {
did_something = true;
}
}
for (auto i: ffmpeg_content()->ffmpeg_audio_streams()) {
- AVPacket packet;
- av_init_packet (&packet);
- packet.data = nullptr;
- packet.size = 0;
- auto result = decode_audio_packet (i, &packet);
- if (result.second) {
+ auto context = _codec_context[i->index(_format_context)];
+ int r = avcodec_send_packet (context, nullptr);
+ if (r < 0 && r != AVERROR_EOF) {
+ /* EOF can happen if we've already sent a flush packet */
+ throw DecodeError (N_("avcodec_send_packet"), N_("FFmpegDecoder::flush"), r);
+ }
+ r = avcodec_receive_frame (context, _frame);
+ if (r >= 0) {
process_audio_frame (i);
did_something = true;
}
* Only the first buffer will be used for non-planar data, otherwise there will be one per channel.
*/
shared_ptr<AudioBuffers>
-FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream) const
+FFmpegDecoder::deinterleave_audio (AVFrame* frame)
{
- DCPOMATIC_ASSERT (bytes_per_audio_sample (stream));
-
-DCPOMATIC_DISABLE_WARNINGS
- int const size = av_samples_get_buffer_size (
- 0, stream->stream(_format_context)->codec->channels, _frame->nb_samples, audio_sample_format (stream), 1
- );
-DCPOMATIC_ENABLE_WARNINGS
- DCPOMATIC_ASSERT (size >= 0);
+ auto format = static_cast<AVSampleFormat>(frame->format);
- /* XXX: can't we just use _frame->nb_samples directly here? */
/* XXX: can't we use swr_convert() to do the format conversion? */
- /* Deinterleave and convert to float */
-
- /* total_samples and frames will be rounded down here, so if there are stray samples at the end
- of the block that do not form a complete sample or frame they will be dropped.
- */
- int const total_samples = size / bytes_per_audio_sample (stream);
- int const channels = stream->channels();
- int const frames = total_samples / channels;
+ int const channels = frame->channels;
+ int const frames = frame->nb_samples;
+ int const total_samples = frames * channels;
auto audio = make_shared<AudioBuffers>(channels, frames);
auto data = audio->data();
- switch (audio_sample_format (stream)) {
+ switch (format) {
case AV_SAMPLE_FMT_U8:
{
- uint8_t* p = reinterpret_cast<uint8_t *> (_frame->data[0]);
+ auto p = reinterpret_cast<uint8_t *> (frame->data[0]);
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
case AV_SAMPLE_FMT_S16:
{
- int16_t* p = reinterpret_cast<int16_t *> (_frame->data[0]);
+ auto p = reinterpret_cast<int16_t *> (frame->data[0]);
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
case AV_SAMPLE_FMT_S16P:
{
- int16_t** p = reinterpret_cast<int16_t **> (_frame->data);
+ auto p = reinterpret_cast<int16_t **> (frame->data);
for (int i = 0; i < channels; ++i) {
for (int j = 0; j < frames; ++j) {
data[i][j] = static_cast<float>(p[i][j]) / (1 << 15);
case AV_SAMPLE_FMT_S32:
{
- int32_t* p = reinterpret_cast<int32_t *> (_frame->data[0]);
+ auto p = reinterpret_cast<int32_t *> (frame->data[0]);
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
case AV_SAMPLE_FMT_S32P:
{
- int32_t** p = reinterpret_cast<int32_t **> (_frame->data);
+ auto p = reinterpret_cast<int32_t **> (frame->data);
for (int i = 0; i < channels; ++i) {
for (int j = 0; j < frames; ++j) {
data[i][j] = static_cast<float>(p[i][j]) / 2147483648;
case AV_SAMPLE_FMT_FLT:
{
- float* p = reinterpret_cast<float*> (_frame->data[0]);
+ auto p = reinterpret_cast<float*> (frame->data[0]);
int sample = 0;
int channel = 0;
for (int i = 0; i < total_samples; ++i) {
case AV_SAMPLE_FMT_FLTP:
{
- float** p = reinterpret_cast<float**> (_frame->data);
- DCPOMATIC_ASSERT (_frame->channels <= channels);
- /* Sometimes there aren't as many channels in the _frame as in the stream */
- for (int i = 0; i < _frame->channels; ++i) {
+ auto p = reinterpret_cast<float**> (frame->data);
+ DCPOMATIC_ASSERT (frame->channels <= channels);
+ /* Sometimes there aren't as many channels in the frame as in the stream */
+ for (int i = 0; i < frame->channels; ++i) {
memcpy (data[i], p[i], frames * sizeof(float));
}
- for (int i = _frame->channels; i < channels; ++i) {
+ for (int i = frame->channels; i < channels; ++i) {
audio->make_silent (i);
}
}
break;
default:
- throw DecodeError (String::compose (_("Unrecognised audio sample format (%1)"), static_cast<int> (audio_sample_format (stream))));
+ throw DecodeError (String::compose(_("Unrecognised audio sample format (%1)"), static_cast<int>(format)));
}
return audio;
AVSampleFormat
FFmpegDecoder::audio_sample_format (shared_ptr<FFmpegAudioStream> stream) const
{
-DCPOMATIC_DISABLE_WARNINGS
- return stream->stream (_format_context)->codec->sample_fmt;
-DCPOMATIC_ENABLE_WARNINGS
+ return static_cast<AVSampleFormat>(stream->stream(_format_context)->codecpar->format);
}
avcodec_flush_buffers (video_codec_context());
}
-DCPOMATIC_DISABLE_WARNINGS
for (auto i: ffmpeg_content()->ffmpeg_audio_streams()) {
- avcodec_flush_buffers (i->stream(_format_context)->codec);
+ avcodec_flush_buffers (_codec_context[i->index(_format_context)]);
}
-DCPOMATIC_ENABLE_WARNINGS
if (subtitle_codec_context ()) {
avcodec_flush_buffers (subtitle_codec_context ());
void
FFmpegDecoder::process_audio_frame (shared_ptr<FFmpegAudioStream> stream)
{
- auto data = deinterleave_audio (stream);
+ auto data = deinterleave_audio (_frame);
ContentTime ct;
if (_frame->pts == AV_NOPTS_VALUE) {
}
-pair<int, bool>
-FFmpegDecoder::decode_audio_packet (shared_ptr<FFmpegAudioStream> stream, AVPacket* packet)
-{
- int frame_finished;
- DCPOMATIC_DISABLE_WARNINGS
- int decode_result = avcodec_decode_audio4 (stream->stream(_format_context)->codec, _frame, &frame_finished, packet);
- DCPOMATIC_ENABLE_WARNINGS
- if (decode_result < 0) {
- /* avcodec_decode_audio4 can sometimes return an error even though it has decoded
- some valid data; for example dca_subframe_footer can return AVERROR_INVALIDDATA
- if it overreads the auxiliary data. ffplay carries on if frame_finished is true,
- even in the face of such an error, so I think we should too.
-
- Returning from the method here caused mantis #352.
- */
- LOG_WARNING ("avcodec_decode_audio4 failed (%1)", decode_result);
- }
- return make_pair(decode_result, frame_finished);
-}
-
-
void
FFmpegDecoder::decode_and_process_audio_packet (AVPacket* packet)
{
return;
}
- /* Audio packets can contain multiple frames, so we may have to call avcodec_decode_audio4
- several times. Make a simple copy so we can alter data and size.
- */
- AVPacket copy_packet = *packet;
+ auto context = _codec_context[stream->index(_format_context)];
- while (copy_packet.size > 0) {
- auto result = decode_audio_packet (stream, ©_packet);
- if (result.first < 0) {
- /* avcodec_decode_audio4 can sometimes return an error even though it has decoded
- some valid data; for example dca_subframe_footer can return AVERROR_INVALIDDATA
- if it overreads the auxiliary data. ffplay carries on if frame_finished is true,
- even in the face of such an error, so I think we should too.
-
- Returning from the method here caused mantis #352.
- */
- }
+ int r = avcodec_send_packet (context, packet);
+ if (r < 0) {
+ /* We could cope with AVERROR(EAGAIN) and re-send the packet but I think it should never happen.
+ * Likewise I think AVERROR_EOF should not happen.
+ */
+ throw DecodeError (N_("avcodec_send_packet"), N_("FFmpegDecoder::decode_and_process_audio_packet"), r);
+ }
- if (result.second) {
- process_audio_frame (stream);
+ while (r >= 0) {
+ r = avcodec_receive_frame (context, _frame);
+ if (r == AVERROR(EAGAIN)) {
+ /* More input is required */
+ return;
}
- if (result.first) {
- break;
- }
-
- copy_packet.data += result.first;
- copy_packet.size -= result.first;
+ /* We choose to be relaxed here about other errors; it seems that there may be valid
+ * data to decode even if an error occurred. #352 may be related (though this was
+ * when we were using an old version of the FFmpeg API).
+ */
+ process_audio_frame (stream);
}
}
{
DCPOMATIC_ASSERT (_video_stream);
- int frame_finished;
-DCPOMATIC_DISABLE_WARNINGS
- if (avcodec_decode_video2 (video_codec_context(), _frame, &frame_finished, packet) < 0 || !frame_finished) {
+ auto context = video_codec_context();
+
+ int r = avcodec_send_packet (context, packet);
+ if (r < 0 && !(r == AVERROR_EOF && !packet)) {
+ /* We could cope with AVERROR(EAGAIN) and re-send the packet but I think it should never happen.
+ * AVERROR_EOF can happen during flush if we've already sent a flush packet.
+ */
+ throw DecodeError (N_("avcodec_send_packet"), N_("FFmpegDecoder::decode_and_process_video_packet"), r);
+ }
+
+ r = avcodec_receive_frame (context, _frame);
+ if (r == AVERROR(EAGAIN) || r == AVERROR_EOF) {
+ /* More input is required, or no more frames are coming */
return false;
}
-DCPOMATIC_ENABLE_WARNINGS
+
+ /* We assume we'll only get one frame here, which I think is safe */
boost::mutex::scoped_lock lm (_filter_graphs_mutex);
projects / dcpomatic.git / blobdiff