Updated series of patches:

• 1/5: 0004_use-source-sample-format-in-ffmpeg-audio-resampler.patch (unchanged, file #34930)
• 2/5: 0005_dont-skip-resampling-8bit-audio.patch (unchanged, file #34931)
• 3/5: 0006_use-correct-8bit-pcm-ffmpeg-codec-v2.patch (file #34958)
• 4/5: 0007_use-original-sample-rate-for-ffmpeg-8bit-pcm-decoder-v2.patch (file #34959)
• 5/5: 0008-fix-ffmpeg-resample-factor-count-calculation-v2.patch (file #34960)

Note: This updated series also have been re-tested in the same configuration as comment 6.

Gnash: 0.8.11dev (patched against git 516a6ca 15-Sep-2015)
SDL: libsdl1.2debian 1.2.15-5 (debian)
libAV: libavcodec53 6:0.8.17-1 (debian), libavformat53 6:0.8.17-1 (debian), libavutil51 6:0.8.17-1 (debian)
CSound: libcsound64-5.2 1:5.17.11~dfsg-3 (debian)
System: Debian GNU/Linux 7.0 Wheezy i386

Does following comments look okay (for patch 3 and patch 5)?

3/5: 0006_use-correct-8bit-pcm-ffmpeg-codec.patch:

            case AUDIO_CODEC_UNCOMPRESSED:
            case AUDIO_CODEC_RAW:
                if (info.sampleSize == 2) {
                    // Flash's 16-bit UNCOMPRESSED and RAW audio comes as
                    // a signed PCM format. UNCOMPRESSED audio always use
                    // little-endian byte order. RAW audio uses encoder's
                    // native byte order; as majority of encoders run
                    // on little-endian machine, we use little-endian for it.
                    codec_id = AV_CODEC_ID_PCM_S16LE;
                } else {
                    // Flash's 8-bit UNCOMPRESSED and RAW audio comes as an
                    // unsigned PCM format. No difference between
                    // UNCOMPRESSED and RAW.
                    codec_id = AV_CODEC_ID_PCM_U8;
                }
                break;

Note: GStreamer backend have a similar problem in its code documentation;
the code itself correctly implements a 8-bit unsigned PCM codec,
but the comments still say 8-bit signed. I'll submit a fix for that as
a different patch entry.

5/5: 0008-fix-ffmpeg-resample-factor-count-calculation.patch:

    if (_resampler.init(_audioCodecCtx)) {
        // Resampling is needed.

        // Compute new size based on frame_size and
        // resampling configuration

        // Find out the needed sample rate scaling
        double resampleFactor = 44100.0/_audioCodecCtx->sample_rate;

        // Compute total number of input samples
        int inSamples = outSize;
        bool stereo = _audioCodecCtx->channels > 1 ? true : false;

        if (stereo) inSamples = inSamples >> 1;
        if (_audioCodecCtx->sample_fmt == AV_SAMPLE_FMT_S16 ||
            _audioCodecCtx->sample_fmt == AV_SAMPLE_FMT_S16P) {
            inSamples = inSamples >> 1;
        }

        // Compute total number of output samples
        int expectedMaxOutSamples = std::ceil(inSamples*resampleFactor);

        // Compute output buffer size (in bytes); by multiplying
        // output samples count with output sample format's frame size,
        // which is number of bytes per sample (2) times channels (2).
        int resampledFrameSize = expectedMaxOutSamples*2*2;

        // Allocate just the required amount of bytes
        std::uint8_t* resampledOutput = new std::uint8_t[resampledFrameSize];

For patch 1, 2, 4; they're just fixing coding oversight and removing hacks,
I didn't really have anything else to say about them.

If comments above are OK, I'll proceed to submit new version of patch 3, 4, 5.

Wow, lots of patches.
The signed vs. unsigned change really deserves some inline comment otherwise it'll be hard for anyone reading the code to understand why an unsigned codec is used for an advertised-as-signed one.

What about merging all patches in a single one and adding proper comments to explain what's being done ?

This is particularly needed as we have NO AUTOMATED TESTS to ensure a regression is not introduced.

`pcmplay-stereo-8bit.swf` from bug #45722 should now be playing properly
with `gnash --media=ffmpeg` once patch 5 is applied.

Gnash: 0.8.11dev (patched against git 516a6ca 15-Sep-2015)
SDL: libsdl1.2debian 1.2.15-5 (debian)
libAV: libavcodec53 6:0.8.17-1 (debian), libavformat53 6:0.8.17-1 (debian), libavutil51 6:0.8.17-1 (debian)
CSound: libcsound64-5.2 1:5.17.11~dfsg-3 (debian)
System: Debian GNU/Linux 7.0 Wheezy i386

Patch 4 of 5:
See the attached `0007_use-original-sample-rate-for-ffmpeg-8bit-pcm-decoder.patch`

Originally, FFmpeg backend divided sound sample rate by 2 if the source audio
is 8-bit `UNCOMPRESSED`/`RAW` PCM, for an unobvious reason.

This patch removes the divisor; allowing the resampler invoker to use
a correct sample rate for resampling factor calculation.

Note: `pcmplay-8bit.swf` from bug #45722 should now be playing properly
with `gnash --media=ffmpeg` once this patch is applied.

Gnash: 0.8.11dev (patched against git 516a6ca 15-Sep-2015)
SDL: libsdl1.2debian 1.2.15-5 (debian)
libAV: libavcodec53 6:0.8.17-1 (debian), libavformat53 6:0.8.17-1 (debian), libavutil51 6:0.8.17-1 (debian)
CSound: libcsound64-5.2 1:5.17.11~dfsg-3 (debian)
System: Debian GNU/Linux 7.0 Wheezy i386

(file #34933)

Patch 2 of 5:
See the attached `0005_dont-skip-resampling-8bit-audio.patch`

Originally, FFmpeg backend ignored non-16 bit audio when checking whether
it needed to call FFmpeg/libAV audio resampler, if user's
FFmpeg/libAV version didn't have both `swresample.h` and `avresample.h`.

This patch makes sure it always consider the possibility that source sample
format is not 16-bit (which is the case when original sound data is 8-bit PCM),
and invoke the resampler accordingly.

Gnash: 0.8.11dev (patched against git 516a6ca 15-Sep-2015)
SDL: libsdl1.2debian 1.2.15-5 (debian)
libAV: libavcodec53 6:0.8.17-1 (debian), libavformat53 6:0.8.17-1 (debian), libavutil51 6:0.8.17-1 (debian)
CSound: libcsound64-5.2 1:5.17.11~dfsg-3 (debian)
System: Debian GNU/Linux 7.0 Wheezy i386

(file #34931)