mirror of
https://github.com/tuxbox-neutrino/libstb-hal.git
synced 2025-08-29 00:11:16 +02:00
libeplayer3: move audio resampling to dedicated ipcm writer
This commit is contained in:
martii
@@ -940,6 +940,7 @@ static int Write(Context_t *context, AudioVideoOut_t *out)
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call.data = out->data;
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call.len = out->len;
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call.Pts = out->pts;
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call.packet = out->packet;
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if (writer->writeData)
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res = writer->writeData(&call);
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@@ -977,12 +978,14 @@ static int Write(Context_t *context, AudioVideoOut_t *out)
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call.data = out->data;
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call.len = out->len;
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call.Pts = out->pts;
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call.packet = out->packet;
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call.uNoOfChannels = out->uNoOfChannels;
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call.uSampleRate = out->uSampleRate;
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call.uBitsPerSample = out->uBitsPerSample;
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call.bLittleEndian = out->bLittleEndian;
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call.restart_audio_resampling = out->restart_audio_resampling;
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call.context = context;
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if (writer->writeData)
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res = writer->writeData(&call);
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@@ -289,34 +289,187 @@ static int writeData(WriterAVCallData_t *call)
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return size;
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}
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SwrContext *swr = NULL;
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AVFrame *decoded_frame = NULL;
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int out_sample_rate = 44100;
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int out_channels = 2;
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uint64_t out_channel_layout = AV_CH_LAYOUT_STEREO;
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int restart_audio_resampling = 0;
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static int resetIpcm()
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{
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if (swr)
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swr_free(&swr);
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if (decoded_frame)
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av_frame_free(&decoded_frame);
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return 0;
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}
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int64_t calcPts(AVFormatContext *, AVStream *, int64_t);
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static int writeDataIpcm(WriterAVCallData_t *call)
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{
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AVCodecContext *c = call->stream->codec;
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AVPacket *packet = call->packet;
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uint8_t *packet_data = packet->data;
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unsigned int packet_size = packet->size;
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if (call->restart_audio_resampling)
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call->restart_audio_resampling = 1;
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if (restart_audio_resampling) {
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restart_audio_resampling = 0;
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if (swr) {
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swr_free(&swr);
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swr = NULL;
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}
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if (decoded_frame) {
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av_frame_free(&decoded_frame);
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decoded_frame = NULL;
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}
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call->context->output->Command(call->context, OUTPUT_CLEAR, NULL);
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call->context->output->Command(call->context, OUTPUT_PLAY, NULL);
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AVCodec *codec = avcodec_find_decoder(c->codec_id);
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if (!codec || avcodec_open2(c, codec, NULL))
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fprintf(stderr, "%s %d: avcodec_open2 failed\n", __func__, __LINE__);
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}
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while (packet_size > 0) {
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int got_frame = 0;
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if (!decoded_frame) {
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if (!(decoded_frame = av_frame_alloc())) {
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fprintf(stderr, "out of memory\n");
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exit(1);
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}
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} else
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av_frame_unref(decoded_frame);
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int len = avcodec_decode_audio4(c, decoded_frame, &got_frame, packet);
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if (len < 0) {
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restart_audio_resampling = 1;
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break;
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}
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packet_data += len;
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packet_size -= len;
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if (!got_frame)
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continue;
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int e;
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if (!swr) {
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int rates[] = { 48000, 96000, 192000, 44100, 88200, 176400, 0 };
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int *rate = rates;
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int in_rate = c->sample_rate;
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while (*rate && ((*rate / in_rate) * in_rate != *rate) && (in_rate / *rate) * *rate != in_rate)
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rate++;
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out_sample_rate = *rate ? *rate : 44100;
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swr = swr_alloc();
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out_channels = c->channels;
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if (c->channel_layout == 0) {
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// FIXME -- need to guess, looks pretty much like a bug in the FFMPEG WMA decoder
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c->channel_layout = AV_CH_LAYOUT_STEREO;
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}
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out_channel_layout = c->channel_layout;
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// player2 won't play mono
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if (out_channel_layout == AV_CH_LAYOUT_MONO) {
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out_channel_layout = AV_CH_LAYOUT_STEREO;
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out_channels = 2;
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}
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av_opt_set_int(swr, "in_channel_layout", c->channel_layout, 0);
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av_opt_set_int(swr, "out_channel_layout", out_channel_layout, 0);
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av_opt_set_int(swr, "in_sample_rate", c->sample_rate, 0);
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av_opt_set_int(swr, "out_sample_rate", out_sample_rate, 0);
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av_opt_set_int(swr, "in_sample_fmt", c->sample_fmt, 0);
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av_opt_set_int(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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e = swr_init(swr);
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if (e < 0) {
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fprintf(stderr,
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"swr_init: %d (icl=%d ocl=%d isr=%d osr=%d isf=%d osf=%d\n",
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-e, (int) c->channel_layout,
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(int) out_channel_layout, c->sample_rate, out_sample_rate, c->sample_fmt, AV_SAMPLE_FMT_S16);
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swr_free(&swr);
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swr = NULL;
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}
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}
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uint8_t *output = NULL;
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int in_samples = decoded_frame->nb_samples;
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int out_samples = av_rescale_rnd(swr_get_delay(swr, c->sample_rate) + in_samples, out_sample_rate, c->sample_rate, AV_ROUND_UP);
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e = av_samples_alloc(&output, NULL, out_channels, out_samples, AV_SAMPLE_FMT_S16, 1);
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if (e < 0) {
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fprintf(stderr, "av_samples_alloc: %d\n", -e);
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continue;
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}
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// FIXME. PTS calculation is probably broken.
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int64_t pts;
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int64_t next_in_pts = av_rescale(av_frame_get_best_effort_timestamp(decoded_frame),
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call->stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate,
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call->stream->time_base.den);
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int64_t next_out_pts = av_rescale(swr_next_pts(swr, next_in_pts),
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call->stream->time_base.den,
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call->stream->time_base.num * (int64_t) out_sample_rate * c->sample_rate);
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*(call->context->currentAudioPtsP) = /* audioTrack->pts = */ pts = calcPts(call->avfc, call->stream, next_out_pts);
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out_samples = swr_convert(swr, &output, out_samples, (const uint8_t **)
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&decoded_frame->data[0], in_samples);
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WriterAVCallData_t pcmOut;
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pcmOut.fd = call->fd;
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pcmOut.uSampleRate = out_sample_rate;
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pcmOut.uNoOfChannels = av_get_channel_layout_nb_channels(out_channel_layout);
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pcmOut.uBitsPerSample = 16;
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pcmOut.bLittleEndian = 1;
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pcmOut.data = output;
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pcmOut.len = out_samples * sizeof(short) * out_channels;
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pcmOut.Pts = pts; // FIXME videoTrack ? pts : 0;
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pcmOut.stream = call->stream;
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pcmOut.avfc = call->avfc;
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pcmOut.packet = NULL;
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writeData(&pcmOut);
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av_freep(&output);
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}
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return packet->size;
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}
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/* ***************************** */
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/* Writer Definition */
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/* ***************************** */
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static WriterCaps_t caps_pcm = {
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"pcm",
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eAudio,
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"A_PCM",
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AUDIO_ENCODING_LPCMA
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"pcm",
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eAudio,
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"A_PCM",
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AUDIO_ENCODING_LPCMA
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};
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struct Writer_s WriterAudioPCM = {
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&reset,
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&writeData,
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NULL,
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&caps_pcm
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&reset,
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&writeData,
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NULL,
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&caps_pcm
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};
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static WriterCaps_t caps_ipcm = {
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"ipcm",
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eAudio,
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"A_IPCM",
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AUDIO_ENCODING_LPCMA
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"ipcm",
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eAudio,
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"A_IPCM",
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AUDIO_ENCODING_LPCMA
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};
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struct Writer_s WriterAudioIPCM = {
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&reset,
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&writeData,
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NULL,
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&caps_ipcm
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&resetIpcm,
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&writeDataIpcm,
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NULL,
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&caps_ipcm
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};
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